diff --git a/girs/GES-1.0.gir b/girs/GES-1.0.gir
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+
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+
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+
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+
+
+
+ The Assets in the GStreamer Editing Services represent the resources
+that can be used. You can create assets for any type that implements the #GESExtractable
+interface, for example #GESClips, #GESFormatter, and #GESTrackElement do implement it.
+This means that assets will represent for example a #GESUriClips, #GESBaseEffect etc,
+and then you can extract objects of those types with the appropriate parameters from the asset
+using the #ges_asset_extract method:
+
+|[
+GESAsset *effect_asset;
+GESEffect *effect;
+
+// You create an asset for an effect
+effect_asset = ges_asset_request (GES_TYPE_EFFECT, "agingtv", NULL);
+
+// And now you can extract an instance of GESEffect from that asset
+effect = GES_EFFECT (ges_asset_extract (effect_asset));
+
+]|
+
+In that example, the advantages of having a #GESAsset are that you can know what effects
+you are working with and let your user know about the avalaible ones, you can add metadata
+to the #GESAsset through the #GESMetaContainer interface and you have a model for your
+custom effects. Note that #GESAsset management is making easier thanks to the #GESProject class.
+
+Each asset is represented by a pair of @extractable_type and @id (string). Actually the @extractable_type
+is the type that implements the #GESExtractable interface, that means that for example for a #GESUriClip,
+the type that implements the #GESExtractable interface is #GESClip.
+The identifier represents different things depending on the @extractable_type and you should check
+the documentation of each type to know what the ID of #GESAsset actually represents for that type. By default,
+we only have one #GESAsset per type, and the @id is the name of the type, but this behaviour is overriden
+to be more useful. For example, for GESTransitionClips, the ID is the vtype of the transition
+you will extract from it (ie crossfade, box-wipe-rc etc..) For #GESEffect the ID is the
+@bin-description property of the extracted objects (ie the gst-launch style description of the bin that
+will be used).
+
+Each and every #GESAsset is cached into GES, and you can query those with the #ges_list_assets function.
+Also the system will automatically register #GESAssets for #GESFormatters and #GESTransitionClips
+and standard effects (actually not implemented yet) and you can simply query those calling:
+|[
+ GList *formatter_assets, *tmp;
+
+ // List all the transitions
+ formatter_assets = ges_list_assets (GES_TYPE_FORMATTER);
+
+ // Print some infos about the formatter GESAsset
+ for (tmp = formatter_assets; tmp; tmp = tmp->next) {
+ g_print ("Name of the formatter: %s, file extension it produces: %s",
+ ges_meta_container_get_string (GES_META_CONTAINER (tmp->data), GES_META_FORMATTER_NAME),
+ ges_meta_container_get_string (GES_META_CONTAINER (tmp->data), GES_META_FORMATTER_EXTENSION));
+ }
+
+ g_list_free (transition_assets);
+
+]|
+
+You can request the creation of #GESAssets using either #ges_asset_request_async or
+#ges_asset_request_async. All the #GESAssets are cached and thus any asset that has already
+been created can be requested again without overhead.
+
+
+
+
+ Sets an asset from the internal cache as needing reload. An asset needs reload
+in the case where, for example, we were missing a GstPlugin to use it and that
+plugin has been installed, or, that particular asset content as changed
+meanwhile (in the case of the usage of proxies).
+
+Once an asset has been set as "needs reload", requesting that asset again
+will lead to it being re discovered, and reloaded as if it was not in the
+cache before.
+
+ %TRUE if the asset was in the cache and could be set as needing reload,
+%FALSE otherwise.
+
+
+
+
+ The #GType of the object that can be extracted from the
+ asset to be reloaded.
+
+
+
+ The identifier of the asset to mark as needing reload
+
+
+
+
+
+ Create a #GESAsset in the most simple cases, you should look at the @extractable_type
+documentation to see if that constructor can be called for this particular type
+
+As it is recommanded not to instanciate assets for GESUriClip synchronously,
+it will not work with this method, but you can instead use the specific
+#ges_uri_clip_asset_request_sync method if you really want to.
+
+ A reference to the wanted #GESAsset or %NULL
+
+
+
+
+ The #GType of the object that can be extracted from the new asset.
+
+
+
+ The Identifier or %NULL
+
+
+
+
+
+ Request a new #GESAsset asyncronously, @callback will be called when the materail is
+ready to be used or if an error occured.
+
+Example of request of a GESAsset async:
+|[
+// The request callback
+static void
+asset_loaded_cb (GESAsset * source, GAsyncResult * res, gpointer user_data)
+{
+ GESAsset *asset;
+ GError *error = NULL;
+
+ asset = ges_asset_request_finish (res, &error);
+ if (asset) {
+ g_print ("The file: %s is usable as a FileSource",
+ ges_asset_get_id (asset));
+ } else {
+ g_print ("The file: %s is *not* usable as a FileSource because: %s",
+ ges_asset_get_id (source), error->message);
+ }
+
+ gst_object_unref (mfs);
+}
+
+// The request:
+ges_asset_request_async (GES_TYPE_URI_CLIP, some_uri, NULL,
+ (GAsyncReadyCallback) asset_loaded_cb, user_data);
+]|
+
+
+
+
+
+ The #GType of the object that can be extracted from the
+ new asset. The class must implement the #GESExtractable interface.
+
+
+
+ The Identifier of the asset we want to create. This identifier depends of the extractable,
+type you want. By default it is the name of the class itself (or %NULL), but for example for a
+GESEffect, it will be the pipeline description, for a GESUriClip it
+will be the name of the file, etc... You should refer to the documentation of the #GESExtractable
+type you want to create a #GESAsset for.
+
+
+
+ optional %GCancellable object, %NULL to ignore.
+
+
+
+ a #GAsyncReadyCallback to call when the initialization is finished,
+Note that the @source of the callback will be the #GESAsset, but you need to
+make sure that the asset is properly loaded using the #ges_asset_request_finish
+method. This asset can not be used as is.
+
+
+
+ The user data to pass when @callback is called
+
+
+
+
+
+ Finalize the request of an async #GESAsset
+
+ The #GESAsset previously requested
+
+
+
+
+ The #GAsyncResult from which to get the newly created #GESAsset
+
+
+
+
+
+ Extracts a new #GObject from @asset. The type of the object is
+defined by the extractable-type of @asset, you can check what
+type will be extracted from @asset using
+#ges_asset_get_extractable_type
+
+ A newly created #GESExtractable
+
+
+
+
+ The #GESAsset to get extract an object from
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
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+
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+
+
+
+
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+
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+
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+
+
+
+
+
+
+
+
+ Extracts a new #GObject from @asset. The type of the object is
+defined by the extractable-type of @asset, you can check what
+type will be extracted from @asset using
+#ges_asset_get_extractable_type
+
+ A newly created #GESExtractable
+
+
+
+
+ The #GESAsset to get extract an object from
+
+
+
+
+
+
+ The #GError of the asset or %NULL if
+the asset was loaded without issue
+
+
+
+
+ The asset to retrieve the error from
+
+
+
+
+
+ Gets the type of object that can be extracted from @self
+
+ the type of object that can be extracted from @self
+
+
+
+
+ The #GESAsset
+
+
+
+
+
+ Gets the ID of a #GESAsset
+
+ The ID of @self
+
+
+
+
+ The #GESAsset to get ID from
+
+
+
+
+
+
+ The proxy in use for @asset
+
+
+
+
+ The #GESAsset to get currenlty used proxy
+
+
+
+
+
+
+ The #GESAsset that is proxied by @proxy
+
+
+
+
+ The #GESAsset from which to get the the asset it proxies.
+
+
+
+
+
+
+ The list of proxies @asset has. Note that the default asset to be
+used is always the first in that list.
+
+
+
+
+
+
+ The #GESAsset to get proxies from
+
+
+
+
+
+ A proxying asset is an asset that can substitue the real @asset. For example if you
+have a full HD #GESUriClipAsset you might want to set a lower resolution (HD version
+of the same file) as proxy. Note that when an asset is proxied, calling
+#ges_asset_request will actually return the proxy asset.
+
+ %TRUE if @proxy has been set on @asset, %FALSE otherwise.
+
+
+
+
+ The #GESAsset to set proxy on
+
+
+
+ The #GESAsset that should be used as default proxy for @asset or
+%NULL if you want to use the currently set proxy. Note that an asset can proxy one and only
+one other asset.
+
+
+
+
+
+ Removes @proxy from the list of known proxies for @asset.
+If @proxy was the current proxy for @asset, stop using it.
+
+ %TRUE if @proxy was a known proxy for @asset, %FALSE otherwise.
+
+
+
+
+ The #GESAsset to stop proxying with @proxy
+
+
+
+ The #GESAsset to stop considering as a proxy for @asset
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ A newly created #GESExtractable
+
+
+
+
+ The #GESAsset to get extract an object from
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
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+
+
+
+
+
+
+
+
+ ## Children Properties
+
+You can use the following children properties through the
+#ges_track_element_set_child_property and alike set of methods:
+
+<informaltable frame="none">
+<tgroup cols="3">
+<colspec colname="properties_type" colwidth="150px"/>
+<colspec colname="properties_name" colwidth="200px"/>
+<colspec colname="properties_flags" colwidth="400px"/>
+<tbody>
+<row>
+ <entry role="property_type"><link linkend="gdouble"><type>double</type></link></entry>
+ <entry role="property_name"><link linkend="GESAudioSource--volume">volume</link></entry>
+ <entry>volume factor, 1.0=100%.</entry>
+</row>
+<row>
+ <entry role="property_type"><link linkend="gboolean"><type>gboolean</type></link></entry>
+ <entry role="property_name"><link linkend="GESAudioSource--mute">mute</link></entry>
+ <entry>mute channel.</entry>
+</row>
+</tbody>
+</tgroup>
+</informaltable>
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
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+
+
+
+
+
+
+
+
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+
+
+
+
+
+
+
+
+
+
+
+
+
+ Outputs a test audio stream using audiotestsrc. The default property values
+output silence. Useful for testing pipelines, or to fill gaps in an audio
+track.
+
+
+
+ Get the current frequency of @self.
+
+ The current frequency of @self.
+
+
+
+
+ a #GESAudioTestSource
+
+
+
+
+
+ Get the current volume of @self.
+
+ The current volume of @self
+
+
+
+
+ a #GESAudioTestSource
+
+
+
+
+
+ Lets you set the frequency applied on the track element
+
+
+
+
+
+ a #GESAudioTestSource
+
+
+
+ The frequency you want to apply on @self
+
+
+
+
+
+ Sets the volume of the test audio signal.
+
+
+
+
+
+ a #GESAudioTestSource
+
+
+
+ The volume you want to apply on @self
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Sane default properties to specify and fixate the output stream are
+set as restriction-caps.
+It is advised, to modify these properties, to use
+#ges_track_update_restriction_caps, setting them directly is
+possible through #ges_track_set_restriction_caps, but not specifying
+one of them can lead to negotiation issues, only use that function
+if you actually know what you're doing :)
+
+The default properties are:
+- format: S32LE
+- channels: 2
+- rate: 44100
+- layout: interleaved
+
+
+
+ Creates a new #GESAudioTrack of type #GES_TRACK_TYPE_AUDIO and with generic
+raw audio caps ("audio/x-raw");
+
+ A new #GESTrack
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Creates a new #GESAudioTransition.
+
+ The newly created #GESAudioTransition.
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ The location of the file/resource to use.
+
+
+
+
+
+
+
+
+
+
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+
+
+
+
+
+
+
+
+
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+
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+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ parent class
+
+
+
+
+
+
+
+
+
+ The effect will be applied on the sources that have lower priorities
+(higher number) between the inpoint and the end of it.
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
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+
+
+
+
+
+
+
+ A #GESClip is a 'natural' object which controls one or more
+#GESTrackElement(s) in one or more #GESTrack(s).
+
+Keeps a reference to the #GESTrackElement(s) it created and
+sets/updates their properties.
+
+
+
+
+ the #GESTrackElement to be used, or %NULL if it can't provide one
+for the given @track.
+
+
+
+
+ a #GESClip
+
+
+
+ a #GESTrackType
+
+
+
+
+
+
+ %TRUE on success %FALSE on failure.
+
+
+
+
+
+
+ a #GESClip
+
+
+
+ a #GESTrackType
+
+
+
+
+
+ Extracts a #GESTrackElement from @asset and adds it to the @clip.
+Should only be called in order to add operations to a #GESClip,
+ni other cases TrackElement are added automatically when adding the
+#GESClip/#GESAsset to a layer.
+
+Takes a reference on @track_element.
+
+ Created #GESTrackElement or NULL
+if an error happened
+
+
+
+
+ a #GESClip
+
+
+
+ a #GESAsset with #GES_TYPE_TRACK_ELEMENT as extractable_type
+
+
+
+
+
+ Finds the #GESTrackElement controlled by @clip that is used in @track. You
+may optionally specify a GType to further narrow search criteria.
+
+Note: If many objects match, then the one with the highest priority will be
+returned.
+
+ The #GESTrackElement used by @track,
+else %NULL. Unref after usage
+
+
+
+
+ a #GESClip
+
+
+
+ a #GESTrack or NULL
+
+
+
+ a #GType indicating the type of track element you are looking
+for or %G_TYPE_NONE if you do not care about the track type.
+
+
+
+
+
+ Finds all the #GESTrackElement controlled by @clip that is used in @track. You
+may optionally specify a GType to further narrow search criteria.
+
+ a #GList of the
+#GESTrackElement contained in @clip.
+The refcount of the objects will be increased. The user will have to
+unref each #GESTrackElement and free the #GList.
+
+
+
+
+
+
+ a #GESClip
+
+
+
+ a #GESTrack or NULL
+
+
+
+ a #GESTrackType indicating the type of tracks in which elements
+should be searched.
+
+
+
+ a #GType indicating the type of track element you are looking
+for or %G_TYPE_NONE if you do not care about the track type.
+
+
+
+
+
+ Get the #GESLayer to which this clip belongs.
+
+ The #GESLayer where this @clip is being
+used, or %NULL if it is not used on any layer. The caller should unref it
+usage.
+
+
+
+
+ a #GESClip
+
+
+
+
+
+ Get the formats supported by @clip.
+
+ The formats supported by @clip.
+
+
+
+
+ the #GESClip
+
+
+
+
+
+ Gets the index position of an effect.
+
+ The top index of the effect, -1 if something went wrong.
+
+
+
+
+ The origin #GESClip
+
+
+
+ The #GESBaseEffect we want to get the top index from
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Get effects applied on @clip
+
+ a #GList of the
+#GESBaseEffect that are applied on @clip order by ascendant priorities.
+The refcount of the objects will be increased. The user will have to
+unref each #GESBaseEffect and free the #GList.
+
+
+
+
+
+
+ The origin #GESClip
+
+
+
+
+
+ Moves @clip to @layer. If @clip is not in any layer, it adds it to
+@layer, else, it removes it from its current layer, and adds it to @layer.
+
+ %TRUE if @clip could be moved %FALSE otherwize
+
+
+
+
+ a #GESClip
+
+
+
+ the new #GESLayer
+
+
+
+
+
+ Sets the formats supported by the file.
+
+
+
+
+
+ the #GESClip to set supported formats on
+
+
+
+ the #GESTrackType defining formats supported by @clip
+
+
+
+
+
+ This is a convenience method that lets you set the index of a top effect.
+
+ %TRUE if @effect was successfuly moved, %FALSE otherwise.
+
+
+
+
+ The origin #GESClip
+
+
+
+ The #GESBaseEffect to move
+
+
+
+ the new index at which to move the @effect inside this
+#GESClip
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ The function modifies @clip, and creates another #GESClip so we have two
+clips at the end, splitted at the time specified by @position, as a position
+in the timeline (not in the clip to be split). For example, if
+ges_clip_split is called on a 4-second clip playing from 0:01.00 until
+0:05.00, with a split position of 0:02.00, this will result in one clip of 1
+second and one clip of 3 seconds, not in two clips of 2 seconds.
+
+The newly created clip will be added to the same layer as @clip is in. This
+implies that @clip must be in a #GESLayer for the operation to be possible.
+
+This method supports clips playing at a different tempo than one second per
+second. For example, splitting a clip with a #GESEffect 'pitch tempo=1.5'
+four seconds after it starts, will set the inpoint of the new clip to six
+seconds after that of the clip to split. For this, the rate-changing
+property must be registered using @ges_effect_class_register_rate_property;
+for the 'pitch' plugin, this is already done.
+
+ The newly created #GESClip resulting
+from the splitting or %NULL if the clip can't be split.
+
+
+
+
+ the #GESClip to split
+
+
+
+ a #GstClockTime representing the timeline position at which to split
+
+
+
+
+
+ The GESLayer where this clip is being used. If you want to connect to its
+notify signal you should connect to it with g_signal_connect_after as the
+signal emission can be stop in the first fase.
+
+
+
+ The formats supported by the clip.
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ The #GESUriClipAsset is a special #GESAsset specilized in #GESClip.
+it is mostly used to get information about the #GESTrackType-s the objects extracted
+from it can potentialy create #GESTrackElement for.
+
+
+
+
+ Gets track types for which objects extracted from @self can create #GESTrackElement
+
+ The track types on which @self will create TrackElement when added to
+a layer
+
+
+
+
+ a #GESClipAsset
+
+
+
+
+
+ Sets track types for which objects extracted from @self can create #GESTrackElement
+
+
+
+
+
+ a #GESClipAsset
+
+
+
+ The track types supported by the GESClipAsset
+
+
+
+
+
+ The formats supported by the asset.
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Subclasses can override the @create_track_element.
+
+
+
+
+ method to create a single #GESTrackElement for a given #GESTrack.
+
+
+
+ method to create multiple #GESTrackElements for a
+#GESTrack.
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ The #GESContainer base class.
+
+
+
+ Groups the #GESContainer-s provided in @containers. It creates a subclass
+of #GESContainer, depending on the containers provided in @containers.
+Basically, if all the containers in @containers should be contained in a same
+clip (all the #GESTrackElement they contain have the exact same
+start/inpoint/duration and are in the same layer), it will create a #GESClip
+otherwise a #GESGroup will be created
+
+ The #GESContainer (subclass) resulting of the
+grouping
+
+
+
+
+ The
+#GESContainer to group, they must all be in a same #GESTimeline
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Edit @container in the different exisiting #GESEditMode modes. In the case of
+slide, and roll, you need to specify a #GESEdge
+
+ %TRUE if the container as been edited properly, %FALSE if an error
+occured
+
+
+
+
+ the #GESClip to edit
+
+
+
+ The layers you want the edit to
+ happen in, %NULL means that the edition is done in all the
+ #GESLayers contained in the current timeline.
+
+
+
+
+
+ The priority of the layer @container should land in.
+ If the layer you're trying to move the container to doesn't exist, it will
+ be created automatically. -1 means no move.
+
+
+
+ The #GESEditMode in which the editition will happen.
+
+
+
+ The #GESEdge the edit should happen on.
+
+
+
+ The position at which to edit @container (in nanosecond)
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Ungroups the #GESTimelineElement contained in this GESContainer,
+creating new #GESContainer containing those #GESTimelineElement
+apropriately.
+
+ The list of
+#GESContainer resulting from the ungrouping operation
+The user is responsible for unreffing the contained objects
+and freeing the list.
+
+
+
+
+
+
+ The #GESContainer to ungroup
+
+
+
+ Wether to recursively ungroup @container
+
+
+
+
+
+ Add the #GESTimelineElement to the container.
+
+ %TRUE on success, %FALSE on failure.
+
+
+
+
+ a #GESContainer
+
+
+
+ the #GESTimelineElement
+
+
+
+
+
+ Edit @container in the different exisiting #GESEditMode modes. In the case of
+slide, and roll, you need to specify a #GESEdge
+
+ %TRUE if the container as been edited properly, %FALSE if an error
+occured
+
+
+
+
+ the #GESClip to edit
+
+
+
+ The layers you want the edit to
+ happen in, %NULL means that the edition is done in all the
+ #GESLayers contained in the current timeline.
+
+
+
+
+
+ The priority of the layer @container should land in.
+ If the layer you're trying to move the container to doesn't exist, it will
+ be created automatically. -1 means no move.
+
+
+
+ The #GESEditMode in which the editition will happen.
+
+
+
+ The #GESEdge the edit should happen on.
+
+
+
+ The position at which to edit @container (in nanosecond)
+
+
+
+
+
+ Get the list of #GESTimelineElement contained in @container
+The user is responsible for unreffing the contained objects
+and freeing the list.
+
+ The list of
+timeline element contained in @container.
+
+
+
+
+
+
+ a #GESContainer
+
+
+
+ Whether to recursively get children in @container
+
+
+
+
+
+ Release the @child from the control of @container.
+
+ %TRUE if the @child was properly released, else %FALSE.
+
+
+
+
+ a #GESContainer
+
+
+
+ the #GESTimelineElement to release
+
+
+
+
+
+ Ungroups the #GESTimelineElement contained in this GESContainer,
+creating new #GESContainer containing those #GESTimelineElement
+apropriately.
+
+ The list of
+#GESContainer resulting from the ungrouping operation
+The user is responsible for unreffing the contained objects
+and freeing the list.
+
+
+
+
+
+
+ The #GESContainer to ungroup
+
+
+
+ Wether to recursively ungroup @container
+
+
+
+
+
+ The span of priorities which this container occupies.
+
+
+
+
+
+
+ A list of TimelineElement
+controlled by this Container. NOTE: Do not modify.
+
+
+
+
+
+ The span of priorities this container occupies
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Will be emitted after a child was added to @container.
+Usually you should connect with #g_signal_connect_after
+as in the first emission stage, the signal emission might
+get stopped internally.
+
+
+
+
+
+ the #GESTimelineElement that was added.
+
+
+
+
+
+ Will be emitted after a child was removed from @container.
+
+
+
+
+
+ the #GESTimelineElement that was removed.
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ The list of
+#GESContainer resulting from the ungrouping operation
+The user is responsible for unreffing the contained objects
+and freeing the list.
+
+
+
+
+
+
+ The #GESContainer to ungroup
+
+
+
+ Wether to recursively ungroup @container
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ %TRUE if the container as been edited properly, %FALSE if an error
+occured
+
+
+
+
+ the #GESClip to edit
+
+
+
+ The layers you want the edit to
+ happen in, %NULL means that the edition is done in all the
+ #GESLayers contained in the current timeline.
+
+
+
+
+
+ The priority of the layer @container should land in.
+ If the layer you're trying to move the container to doesn't exist, it will
+ be created automatically. -1 means no move.
+
+
+
+ The #GESEditMode in which the editition will happen.
+
+
+
+ The #GESEdge the edit should happen on.
+
+
+
+ The position at which to edit @container (in nanosecond)
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ A function that will be called to create the #GstElement that will be used
+as a source to fill the gaps in @track.
+
+ A #GstElement (must be a source) that will be used to
+fill the gaps (periods of time in @track that containes no source).
+
+
+
+
+ the #GESTrack
+
+
+
+
+
+ Creates the 'primary' track element for this @clip.
+
+Subclasses should implement this method if they only provide a
+single #GESTrackElement per track.
+
+If the subclass needs to create more than one #GESTrackElement for a
+given track, then it should implement the 'create_track_elements'
+method instead.
+
+The implementer of this function shall return the proper #GESTrackElement
+that should be controlled by @clip for the given @track.
+
+The returned #GESTrackElement will be automatically added to the list
+of objects controlled by the #GESClip.
+
+ the #GESTrackElement to be used, or %NULL if it can't provide one
+for the given @track.
+
+
+
+
+ a #GESClip
+
+
+
+ a #GESTrackType
+
+
+
+
+
+ Create all track elements this clip handles for this type of track.
+
+Subclasses should implement this method if they potentially need to
+return more than one #GESTrackElement(s) for a given #GESTrack.
+
+ %TRUE on success %FALSE on failure.
+
+
+
+
+
+
+ a #GESClip
+
+
+
+ a #GESTrackType
+
+
+
+
+
+ The edges of an object contain in a #GESTimeline or #GESTrack
+
+ Represents the start of an object.
+
+
+ Represents the end of an object.
+
+
+ Represent the fact we are not workin with any edge of an
+ object.
+
+
+
+ You can also find more explanation about the behaviour of those modes at:
+<ulink url="http://pitivi.org/manual/trimming.html"> trim, ripple and roll</ulink>
+and <ulink url="http://pitivi.org/manual/usingclips.html">clip management</ulink>.
+
+ The object is edited the normal way (default).
+
+
+ The objects are edited in ripple mode.
+ The Ripple mode allows you to modify the beginning/end of a clip
+ and move the neighbours accordingly. This will change the overall
+ timeline duration. In the case of ripple end, the duration of the
+ clip being rippled can't be superior to its max_duration - inpoint
+ otherwise the action won't be executed.
+
+
+ The object is edited in roll mode.
+ The Roll mode allows you to modify the position of an editing point
+ between two clips without modifying the inpoint of the first clip
+ nor the out-point of the second clip. This will not change the
+ overall timeline duration.
+
+
+ The object is edited in trim mode.
+ The Trim mode allows you to modify the in-point/duration of a clip
+ without modifying its position in the timeline.
+
+
+ The object is edited in slide mode.
+ The Slide mode allows you to modify the position of a clip in a
+ timeline without modifying its duration or its in-point, but will
+ modify the duration of the previous clip and in-point of the
+ following clip so does not modify the overall timeline duration.
+ (not implemented yet)
+
+
+
+
+
+
+ Creates a new #GESEffect from the description of the bin. It should be
+possible to determine the type of the effect through the element
+'klass' metadata of the GstElements that will be created.
+In that corner case, you should use:
+#ges_asset_request (GES_TYPE_EFFECT, "audio your ! bin ! description", NULL);
+and extract that asset to be in full control.
+
+ a newly created #GESEffect, or %NULL if something went
+wrong.
+
+
+
+
+ The gst-launch like bin description of the effect
+
+
+
+
+
+ The description of the effect bin with a gst-launch-style
+pipeline description.
+
+Example: "videobalance saturation=1.5 hue=+0.5"
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ parent class
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Register an element that can change the rate at which media is playing. The
+property type must be float or double, and must be a factor of the rate,
+i.e. a value of 2.0 must mean that the media plays twice as fast. For
+example, this is true for the properties 'rate' and 'tempo' of the element
+'pitch', which is already registered by default. By registering the element,
+timeline duration can be correctly converted into media duration, allowing
+the right segment seeks to be sent to the sources.
+
+A reference to the GESEffectClass can be obtained as follows:
+ GES_EFFECT_CLASS (g_type_class_ref (GES_TYPE_EFFECT));
+
+ whether the rate property was succesfully registered. When this
+method returns false, a warning is emitted with more information.
+
+
+
+
+ Instance of the GESEffectClass
+
+
+
+ Name of the GstElement that changes the rate
+
+
+
+ Name of the property that changes the rate
+
+
+
+
+
+
+ The effect will be applied on the sources that have lower priorities
+(higher number) between the inpoint and the end of it.
+
+
+
+ Creates a new #GESEffectClip from the description of the bin.
+
+ a newly created #GESEffectClip, or
+%NULL if something went wrong.
+
+
+
+
+ The gst-launch like bin description of the effect
+
+
+
+ The gst-launch like bin description of the effect
+
+
+
+
+
+ The description of the audio track of the effect bin with a gst-launch-style
+pipeline description. This should be used for test purposes.
+
+Example: "audiopanorama panorama=1.0"
+
+
+
+ The description of the video track of the effect bin with a gst-launch-style
+pipeline description. This should be used for test purposes.
+
+Example: "videobalance saturation=1.5 hue=+0.5"
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ The ID passed is malformed
+
+
+ An error happened while loading the asset
+
+
+ The formatted files was malformed
+
+
+
+ FIXME: Long description needed
+
+
+
+ The #id of the associated #GESAsset, free with #g_free
+
+
+
+
+ The #GESExtractable
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Method for getting an asset from a #GESExtractable
+
+ The #GESAsset or %NULL if none has
+been set
+
+
+
+
+ The #GESExtractable from which to retrieve a #GESAsset
+
+
+
+
+
+
+ The #id of the associated #GESAsset, free with #g_free
+
+
+
+
+ The #GESExtractable
+
+
+
+
+
+ Method to set the asset which instantiated the specified object
+
+ %TRUE if @asset could be set %FALSE otherwize
+
+
+
+
+ Target object
+
+
+
+ The #GESAsset to set
+
+
+
+
+
+
+
+ The ID to use for the asset or %NULL if @id is not valid
+
+
+
+
+ The #GType to check @id for:
+
+
+
+ The id to check
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ The #id of the associated #GESAsset, free with #g_free
+
+
+
+
+ The #GESExtractable
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ A function that will be called when the GNonLin object of a corresponding
+track element needs to be filled.
+
+The implementer of this function shall add the proper #GstElement to @nleobj
+using gst_bin_add().
+
+ TRUE if the implementer succesfully filled the @nleobj, else #FALSE.
+
+
+
+
+ the #GESClip controlling the track elements
+
+
+
+ the #GESTrackElement
+
+
+
+ the GNonLin object that needs to be filled.
+
+
+
+
+
+ Base class for timeline data serialization and deserialization.
+
+
+ Checks if there is a #GESFormatter available which can load a #GESTimeline
+from the given URI.
+
+ TRUE if there is a #GESFormatter that can support the given uri
+or FALSE if not.
+
+
+
+
+ a #gchar * pointing to the URI
+
+
+
+
+
+ Returns TRUE if there is a #GESFormatter available which can save a
+#GESTimeline to the given URI.
+
+ TRUE if the given @uri is supported, else FALSE.
+
+
+
+
+ a #gchar * pointing to a URI
+
+
+
+
+
+ Get the default #GESAsset to use as formatter. It will return
+the asset for the #GESFormatter that has the highest @rank
+
+ The #GESAsset for the formatter with highest @rank
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Load data from the given URI into timeline.
+
+ TRUE if the timeline data was successfully loaded from the URI,
+else FALSE.
+
+
+
+
+ a #GESFormatter
+
+
+
+ a #GESTimeline
+
+
+
+ a #gchar * pointing to a URI
+
+
+
+
+
+ Save data from timeline to the given URI.
+
+ TRUE if the timeline data was successfully saved to the URI
+else FALSE.
+
+
+
+
+ a #GESFormatter
+
+
+
+ a #GESTimeline
+
+
+
+ a #gchar * pointing to a URI
+
+
+
+ %TRUE to overwrite file if it exists
+
+
+
+
+
+ Load data from the given URI into timeline.
+
+ TRUE if the timeline data was successfully loaded from the URI,
+else FALSE.
+
+
+
+
+ a #GESFormatter
+
+
+
+ a #GESTimeline
+
+
+
+ a #gchar * pointing to a URI
+
+
+
+
+
+ Save data from timeline to the given URI.
+
+ TRUE if the timeline data was successfully saved to the URI
+else FALSE.
+
+
+
+
+ a #GESFormatter
+
+
+
+ a #GESTimeline
+
+
+
+ a #gchar * pointing to a URI
+
+
+
+ %TRUE to overwrite file if it exists
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ GES Formatter class. Override the vmethods to implement the formatter functionnality.
+
+ the parent class structure
+
+
+
+ Whether the URI can be loaded
+
+
+
+ class method to deserialize data from a URI
+
+
+
+ class method to serialize data to a URI
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Virtual method for loading a timeline from a given URI.
+
+Every #GESFormatter subclass needs to implement this method.
+
+ TRUE if the timeline data was successfully loaded from the URI,
+else FALSE.
+
+
+
+
+ a #GESFormatter
+
+
+
+ a #GESTimeline
+
+
+
+ a #gchar * pointing to a URI
+
+
+
+
+
+
+
+ Virtual method for saving a timeline to a uri.
+
+Every #GESFormatter subclass needs to implement this method.
+
+ TRUE if the timeline data was successfully saved to the URI
+else FALSE.
+
+
+
+
+ a #GESFormatter
+
+
+
+ a #GESTimeline
+
+
+
+ a #gchar * pointing to a URI
+
+
+
+ %TRUE to overwrite file if it exists
+
+
+
+
+
+ A #GESGroup is an object which controls one or more
+#GESClips in one or more #GESLayer(s).
+
+To instanciate a group, you should use the ges_container_group method,
+this will be responsible for deciding what subclass of #GESContainer
+should be instaciated to group the various #GESTimelineElement passed
+in parametter.
+
+
+
+ Created a new empty #GESGroup, if you want to group several container
+together, it is recommanded to use the #ges_container_group method so the
+proper subclass is selected.
+
+ The new empty group.
+
+
+
+
+ The duration (in nanoseconds) which will be used in the container
+
+
+
+ The in-point at which this #GESGroup will start outputting data
+from its contents (in nanoseconds).
+
+Ex : an in-point of 5 seconds means that the first outputted buffer will
+be the one located 5 seconds in the controlled resource.
+
+
+
+ The maximum duration (in nanoseconds) of the #GESGroup.
+
+
+
+
+
+
+ The position of the object in its container (in nanoseconds).
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Outputs the video stream from a given file as a still frame. The frame
+chosen will be determined by the in-point property on the track element. For
+image files, do not set the in-point property.
+
+
+
+ The location of the file/resource to use.
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Responsible for the ordering of the various contained Clip(s). A
+timeline layer has a "priority" property, which is used to manage the
+priorities of individual Clips. Two layers should not have the
+same priority within a given timeline.
+
+
+
+ Creates a new #GESLayer.
+
+ A new #GESLayer
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Creates Clip from asset, adds it to layer and
+returns a reference to it.
+
+ Created #GESClip
+
+
+
+
+ a #GESLayer
+
+
+
+ The asset to add to
+
+
+
+ The start value to set on the new #GESClip,
+if @start == GST_CLOCK_TIME_NONE, it will be set to
+the current duration of @layer
+
+
+
+ The inpoint value to set on the new #GESClip
+
+
+
+ The duration value to set on the new #GESClip
+
+
+
+ The #GESTrackType to set on the the new #GESClip
+
+
+
+
+
+ Adds the given clip to the layer. Sets the clip's parent, and thus
+takes ownership of the clip.
+
+An clip can only be added to one layer.
+
+Calling this method will construct and properly set all the media related
+elements on @clip. If you need to know when those objects (actually #GESTrackElement)
+are constructed, you should connect to the container::child-added signal which
+is emited right after those elements are ready to be used.
+
+ %TRUE if the clip was properly added to the layer, or %FALSE
+if the @layer refuses to add the clip.
+
+
+
+
+ a #GESLayer
+
+
+
+ the #GESClip to add.
+
+
+
+
+
+ Gets whether transitions are automatically added when objects
+overlap or not.
+
+ %TRUE if transitions are automatically added, else %FALSE.
+
+
+
+
+ a #GESLayer
+
+
+
+
+
+ Get the clips this layer contains.
+
+ a #GList of
+clips. The user is responsible for
+unreffing the contained objects and freeing the list.
+
+
+
+
+
+
+ a #GESLayer
+
+
+
+
+
+ Gets the clips which appear between @start and @end on @layer.
+
+ a #GList of clips intersecting [@start, @end) interval on @layer.
+
+
+
+
+
+
+ a #GESLayer
+
+
+
+ start of the interval
+
+
+
+ end of the interval
+
+
+
+
+
+ Lets you retrieve the duration of the layer, which means
+the end time of the last clip inside it
+
+ The duration of a layer
+
+
+
+
+ The #GESLayer to get the duration from
+
+
+
+
+
+ Get the priority of @layer within the timeline.
+
+ The priority of the @layer within the timeline.
+
+
+
+
+ a #GESLayer
+
+
+
+
+
+ Get the #GESTimeline in which #GESLayer currently is.
+
+ the #GESTimeline in which #GESLayer
+currently is or %NULL if not in any timeline yet.
+
+
+
+
+ The #GESLayer to get the parent #GESTimeline from
+
+
+
+
+
+ Convenience method to check if @layer is empty (doesn't contain any clip),
+or not.
+
+ %TRUE if @layer is empty, %FALSE if it already contains at least
+one #GESClip
+
+
+
+
+ The #GESLayer to check
+
+
+
+
+
+ Removes the given @clip from the @layer and unparents it.
+Unparenting it means the reference owned by @layer on the @clip will be
+removed. If you wish to use the @clip after this function, make sure you
+call gst_object_ref() before removing it from the @layer.
+
+ %TRUE if the clip could be removed, %FALSE if the layer does
+not want to remove the clip.
+
+
+
+
+ a #GESLayer
+
+
+
+ the #GESClip to remove
+
+
+
+
+
+ Sets the layer to the given @auto_transition. See the documentation of the
+property auto_transition for more information.
+
+
+
+
+
+ a #GESLayer
+
+
+
+ whether the auto_transition is active
+
+
+
+
+
+ Sets the layer to the given @priority. See the documentation of the
+priority property for more information.
+
+
+
+
+
+ a #GESLayer
+
+
+
+ the priority to set
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Sets whether transitions are added automagically when clips overlap.
+
+
+
+ The priority of the layer in the #GESTimeline. 0 is the highest
+priority. Conceptually, a #GESTimeline is a stack of GESLayers,
+and the priority of the layer represents its position in the stack. Two
+layers should not have the same priority within a given GESTimeline.
+
+Note that the timeline needs to be commited (with #ges_timeline_commit)
+for the change to be taken into account.
+
+
+
+
+
+
+ the #GESTimeline where this layer is being used.
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Will be emitted after the clip was added to the layer.
+
+
+
+
+
+ the #GESClip that was added.
+
+
+
+
+
+ Will be emitted after the clip was removed from the layer.
+
+
+
+
+
+ the #GESClip that was removed
+
+
+
+
+
+
+ Subclasses can override the @get_objects if they can provide a more
+efficient way of providing the list of contained #GESClip(s).
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ The description of an object, can be used in various context (string)
+
+The description
+
+
+
+ The extension of the files produced by a formatter (string)
+
+
+
+ Mimetype used for the file produced by a formatter (string)
+
+The mime type
+
+
+
+ Name of a formatter it is used as ID of Formater assets (string)
+
+The name of the formatter
+
+
+
+ The rank of a formatter (GstRank)
+
+The rank of a formatter
+
+
+
+ The version of a formatter (double)
+
+The formatter version
+
+
+
+ The version of the format in which a project is serialized
+
+
+
+ The volume, can be used for audio track or layers
+
+The volume for a track or a layer, it is register as a float
+
+
+
+ The default volume
+
+The default volume for a track or a layer as a float
+
+
+
+
+
+
+ Interface that allows reading and writing meta
+
+ Deserializes a meta container.
+
+ TRUE on success, FALSE if there was an error.
+
+
+
+
+ Target container
+
+
+
+ a string created with ges_meta_container_metas_to_string()
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Calls the given function for each metadata inside the meta container. Note
+that if there is no metadata, the function won't be called at all.
+
+
+
+
+
+ container to iterate over
+
+
+
+ function to be called for each metadata
+
+
+
+ user specified data
+
+
+
+
+
+
+
+
+
+
+ Target container
+
+
+
+ Name of the meta item to get
+
+
+
+ Destination to which value of meta item will be copied
+Gets the value of a given meta item, returns NULL if @meta_item
+can not be found.
+
+
+
+
+
+
+
+
+
+
+ Target container
+
+
+
+ Name of the meta item to get
+
+
+
+ Destination to which value of meta item will be copied
+Gets the value of a given meta item, returns NULL if @meta_item
+can not be found.
+
+
+
+
+
+
+
+
+
+
+ Target container
+
+
+
+ Name of the meta item to get
+
+
+
+ Destination to which value of meta item will be copied
+Gets the value of a given meta item, returns NULL if @meta_item
+can not be found.
+
+
+
+
+
+
+
+
+
+
+ Target container
+
+
+
+ Name of the meta item to get
+
+
+
+ Destination to which value of meta item will be copied
+Gets the value of a given meta item, returns NULL if @meta_item
+can not be found.
+
+
+
+
+
+
+
+
+
+
+ Target container
+
+
+
+ Name of the meta item to get
+
+
+
+ Destination to which value of meta item will be copied
+Gets the value of a given meta item, returns FALSE if @meta_item
+can not be found.
+
+
+
+
+
+
+
+
+
+
+ Target container
+
+
+
+ Name of the meta item to get
+
+
+
+ Destination to which value of meta item will be copied
+Gets the value of a given meta item, returns NULL if @meta_item
+can not be found.
+
+
+
+
+
+
+
+
+
+
+ Target container
+
+
+
+ Name of the meta item to get
+
+
+
+ Destination to which value of meta item will be copied
+Gets the value of a given meta item, returns %FALSE if @meta_item
+can not be found.
+
+
+
+
+
+ Gets the value of a given meta item, returns NULL if @key
+can not be found.
+
+ the #GValue corresponding to the meta with the given @key.
+
+
+
+
+ Target container
+
+
+
+ The key name of the meta to retrieve
+
+
+
+
+
+
+
+
+
+
+ Target container
+
+
+
+ Name of the meta item to get
+Gets the value of a given meta item, returns NULL if @meta_item
+can not be found.
+
+
+
+
+
+
+
+
+
+
+ Target container
+
+
+
+ Name of the meta item to get
+
+
+
+ Destination to which value of meta item will be copied
+Gets the value of a given meta item, returns NULL if @meta_item
+can not be found.
+
+
+
+
+
+
+
+
+
+
+ Target container
+
+
+
+ Name of the meta item to get
+
+
+
+ Destination to which value of meta item will be copied
+Gets the value of a given meta item, returns NULL if @meta_item
+can not be found.
+
+
+
+
+
+ Serializes a meta container to a string.
+
+ a newly-allocated string, or NULL in case of an error.
+The string must be freed with g_free() when no longer needed.
+
+
+
+
+ a #GESMetaContainer
+
+
+
+
+
+ Sets a static meta on @container. This method lets you define static
+metadatas, which means that the type of the registered will be the only
+type accepted for this meta on that particular @container.
+
+ %TRUE if the static meta could be added, %FALSE otherwize
+
+
+
+
+ Target container
+
+
+
+ The #GESMetaFlag to be used
+
+
+
+ Name of the meta item to set
+
+
+
+ Value to set
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Sets a static meta on @container. This method lets you define static
+metadatas, which means that the type of the registered will be the only
+type accepted for this meta on that particular @container.
+
+ %TRUE if the meta could be register, %FALSE otherwize
+
+
+
+
+ Target container
+
+
+
+ The #GESMetaFlag to be used
+
+
+
+ Name of the meta item to set
+
+
+
+ Value to set
+
+
+
+
+
+ Sets a static meta on @container. This method lets you define static
+metadatas, which means that the type of the registered will be the only
+type accepted for this meta on that particular @container.
+
+ %TRUE if the meta could be register, %FALSE otherwize
+
+
+
+
+ Target container
+
+
+
+ The #GESMetaFlag to be used
+
+
+
+ Name of the meta item to set
+
+
+
+ Value to set
+
+
+
+
+
+ Sets a static meta on @container. This method lets you define static
+metadatas, which means that the type of the registered will be the only
+type accepted for this meta on that particular @container.
+
+ %TRUE if the meta could be register, %FALSE otherwize
+
+
+
+
+ Target container
+
+
+
+ The #GESMetaFlag to be used
+
+
+
+ Name of the meta item to set
+
+
+
+ Value to set
+
+
+
+
+
+ Sets a static meta on @container. This method lets you define static
+metadatas, which means that the type of the registered will be the only
+type accepted for this meta on that particular @container.
+
+ %TRUE if the meta could be register, %FALSE otherwize
+
+
+
+
+ Target container
+
+
+
+ The #GESMetaFlag to be used
+
+
+
+ Name of the meta item to set
+
+
+
+ Value to set
+
+
+
+
+
+ Sets a static meta on @container. This method lets you define static
+metadatas, which means that the type of the registered will be the only
+type accepted for this meta on that particular @container.
+
+ %TRUE if the meta could be register, %FALSE otherwize
+
+
+
+
+ Target container
+
+
+
+ The #GESMetaFlag to be used
+
+
+
+ Name of the meta item to set
+
+
+
+ Value to set
+
+
+
+
+
+ Sets a static meta on @container. This method lets you define static
+metadatas, which means that the type of the registered will be the only
+type accepted for this meta on that particular @container.
+
+ %TRUE if the meta could be register, %FALSE otherwize
+
+
+
+
+ Target container
+
+
+
+ The #GESMetaFlag to be used
+
+
+
+ Name of the meta item to set
+
+
+
+ Value to set
+
+
+
+
+
+ Sets a static meta on @container. This method lets you define static
+metadatas, which means that the type of the registered will be the only
+type accepted for this meta on that particular @container.
+
+ %TRUE if the meta could be register, %FALSE otherwize
+
+
+
+
+ Target container
+
+
+
+ The #GESMetaFlag to be used
+
+
+
+ Name of the meta item to set
+
+
+
+ Value to set
+
+
+
+
+
+ Sets a static meta on @container. This method lets you define static
+metadatas, which means that the type of the registered will be the only
+type accepted for this meta on that particular @container.
+
+ %TRUE if the meta could be register, %FALSE otherwize
+
+
+
+
+ Target container
+
+
+
+ The #GESMetaFlag to be used
+
+
+
+ Name of the meta item to set
+
+
+
+ Value to set
+
+
+
+
+
+ Sets a static meta on @container. This method lets you define static
+metadatas, which means that the type of the registered will be the only
+type accepted for this meta on that particular @container.
+
+ %TRUE if the meta could be register, %FALSE otherwize
+
+
+
+
+ Target container
+
+
+
+ The #GESMetaFlag to be used
+
+
+
+ Name of the meta item to set
+
+
+
+ Value to set
+
+
+
+
+
+ Sets the value of a given meta item
+
+ %TRUE if the meta could be added, %FALSE otherwize
+
+
+
+
+ Target container
+
+
+
+ Name of the meta item to set
+
+
+
+ Value to set
+
+
+
+
+
+ Sets the value of a given meta item
+
+ %TRUE if the meta could be added, %FALSE otherwize
+
+
+
+
+ Target container
+
+
+
+ Name of the meta item to set
+
+
+
+ Value to set
+
+
+
+
+
+ Sets the value of a given meta item
+
+ %TRUE if the meta could be added, %FALSE otherwize
+
+
+
+
+ Target container
+
+
+
+ Name of the meta item to set
+
+
+
+ Value to set
+
+
+
+
+
+ Sets the value of a given meta item
+
+ %TRUE if the meta could be added, %FALSE otherwize
+
+
+
+
+ Target container
+
+
+
+ Name of the meta item to set
+
+
+
+ Value to set
+
+
+
+
+
+ Sets the value of a given meta item
+
+ %TRUE if the meta could be added, %FALSE otherwize
+
+
+
+
+ Target container
+
+
+
+ Name of the meta item to set
+
+
+
+ Value to set
+
+
+
+
+
+ Sets the value of a given meta item
+
+ %TRUE if the meta could be added, %FALSE otherwize
+
+
+
+
+ Target container
+
+
+
+ Name of the meta item to set
+
+
+
+ Value to set
+
+
+
+
+
+ Sets the value of a given meta item
+
+ %TRUE if the meta could be added, %FALSE otherwize
+
+
+
+
+ Target container
+
+
+
+ Name of the meta item to set
+
+
+
+ Value to set
+
+
+
+
+
+
+ %TRUE if the meta could be added, %FALSE otherwize
+
+
+
+
+ Target container
+
+
+
+ Name of the meta item to set
+
+
+
+ Value to set
+Sets the value of a given meta item
+
+
+
+
+
+ Sets the value of a given meta item
+
+ %TRUE if the meta could be added, %FALSE otherwize
+
+
+
+
+ Target container
+
+
+
+ Name of the meta item to set
+
+
+
+ Value to set
+
+
+
+
+
+ Sets the value of a given meta item
+
+ %TRUE if the meta could be added, %FALSE otherwize
+
+
+
+
+ Target container
+
+
+
+ Name of the meta item to set
+
+
+
+ Value to set
+
+
+
+
+
+ Sets the value of a given meta item
+
+ %TRUE if the meta could be added, %FALSE otherwize
+
+
+
+
+ Target container
+
+
+
+ Name of the meta item to set
+
+
+
+ Value to set
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ The metadata is readable
+
+
+ The metadata is writable
+
+
+ The metadata is readable and writable
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Outputs the video stream from a given image sequence. The start frame
+chosen will be determined by the in-point property on the track element.
+
+
+
+ Creates a new #GESMultiFileSource for the provided @uri.
+
+ A new #GESMultiFileSource.
+
+
+
+
+ the URI the source should control
+
+
+
+
+
+ The uri of the file/resource to use. You can set a start index,
+a stop index and a sequence pattern.
+The format is <multifile://start:stop\@location-pattern>.
+The pattern uses printf string formating.
+
+Example uris:
+
+multifile:///home/you/image\%03d.jpg
+
+multifile://20:50@/home/you/sequence/\%04d.png
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Base class for overlays, transitions, and effects
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Operations are any kind of object that both outputs AND consumes data.
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Overlays are objects which modify the underlying layer(s).
+
+Examples of overlays include text, image watermarks, or audio dubbing.
+
+Transitions, which change from one source to another over time, are
+not considered overlays.
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ parent class
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ #GESPipeline allows developers to view and render #GESTimeline
+in a simple fashion.
+Its usage is inspired by the 'playbin' element from gst-plugins-base.
+
+
+
+ Creates a new conveninence #GESPipeline.
+
+ the new #GESPipeline.
+
+
+
+
+
+ the #GESPipelineFlags currently in use.
+
+
+
+
+ a #GESPipeline
+
+
+
+
+
+ Returns a #GstSample with the currently playing image in the format specified by
+caps. The caller should free the sample with #gst_sample_unref when finished. If ANY
+caps are specified, the information will be returned in the whatever format
+is currently used by the sink. This information can be retrieve from caps
+associated with the buffer.
+
+ a #GstSample or %NULL
+
+
+
+
+ a #GESPipeline in %GST_STATE_PLAYING or %GST_STATE_PAUSED
+
+
+
+ caps specifying current format. Use %GST_CAPS_ANY
+for native size.
+
+
+
+
+
+ A convenience method for @ges_pipeline_get_thumbnail which
+returns a buffer in 24-bit RGB, optionally scaled to the specified width
+and height. If -1 is specified for either dimension, it will be left at
+native size. You can retreive this information from the caps associated
+with the buffer.
+
+The caller is responsible for unreffing the returned sample with
+#gst_sample_unref.
+
+ a #GstSample or %NULL
+
+
+
+
+ a #GESPipeline in %GST_STATE_PLAYING or %GST_STATE_PAUSED
+
+
+
+ the requested width or -1 for native size
+
+
+
+ the requested height or -1 for native size
+
+
+
+
+
+ Obtains a pointer to playsink's audio sink element that is used for
+displaying audio when the #GESPipeline is in %GES_PIPELINE_MODE_PREVIEW
+
+The caller is responsible for unreffing the returned element with
+#gst_object_unref.
+
+ a pointer to the playsink audio sink #GstElement
+
+
+
+
+ a #GESPipeline
+
+
+
+
+
+ Obtains a pointer to playsink's video sink element that is used for
+displaying video when the #GESPipeline is in %GES_PIPELINE_MODE_PREVIEW
+
+The caller is responsible for unreffing the returned element with
+#gst_object_unref.
+
+ a pointer to the playsink video sink #GstElement
+
+
+
+
+ a #GESPipeline
+
+
+
+
+
+ Sets playsink's audio sink element that is used for displaying audio when
+the #GESPipeline is in %GES_PIPELINE_MODE_PREVIEW
+
+
+
+
+
+ a #GESPipeline in %GST_STATE_NULL
+
+
+
+ a audio sink #GstElement
+
+
+
+
+
+ Sets playsink's video sink element that is used for displaying video when
+the #GESPipeline is in %GES_PIPELINE_MODE_PREVIEW
+
+
+
+
+
+ a #GESPipeline in %GST_STATE_NULL
+
+
+
+ a video sink #GstElement
+
+
+
+
+
+ Saves the current frame to the specified @location.
+
+ %TRUE if the thumbnail was properly save, else %FALSE.
+
+
+
+
+ a #GESPipeline in %GST_STATE_PLAYING or %GST_STATE_PAUSED
+
+
+
+ the requested width or -1 for native size
+
+
+
+ the requested height or -1 for native size
+
+
+
+ a string specifying the desired mime type (for example,
+image/jpeg)
+
+
+
+ the path to save the thumbnail
+
+
+
+
+
+ switches the @pipeline to the specified @mode. The default mode when
+creating a #GESPipeline is #GES_PIPELINE_MODE_PREVIEW.
+
+Note: The @pipeline will be set to #GST_STATE_NULL during this call due to
+the internal changes that happen. The caller will therefore have to
+set the @pipeline to the requested state after calling this method.
+
+ %TRUE if the mode was properly set, else %FALSE.
+
+
+
+
+ a #GESPipeline
+
+
+
+ the #GESPipelineFlags to use
+
+
+
+
+
+ Specify where the pipeline shall be rendered and with what settings.
+
+A copy of @profile and @output_uri will be done internally, the caller can
+safely free those values afterwards.
+
+This method must be called before setting the pipeline mode to
+#GES_PIPELINE_MODE_RENDER
+
+ %TRUE if the settings were aknowledged properly, else %FALSE
+
+
+
+
+ a #GESPipeline
+
+
+
+ the URI to which the timeline will be rendered
+
+
+
+ the #GstEncodingProfile to use to render the timeline.
+
+
+
+
+
+ Sets the timeline to use in this pipeline.
+
+The reference to the @timeline will be stolen by the @pipeline.
+
+ %TRUE if the @timeline could be successfully set on the @pipeline,
+else %FALSE.
+
+
+
+
+ a #GESPipeline
+
+
+
+ the #GESTimeline to set on the @pipeline.
+
+
+
+
+
+
+
+
+ Audio sink for the preview.
+
+
+
+ Pipeline mode. See ges_pipeline_set_mode() for more
+info.
+
+
+
+ Timeline to use in this pipeline. See also
+ges_pipeline_set_timeline() for more info.
+
+
+
+
+
+
+ Video sink for the preview.
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ parent class
+
+
+
+
+
+
+
+
+
+ The various modes the #GESPipeline can be configured to.
+
+ output audio to the soundcard
+
+
+ output video to the screen
+
+
+ output audio/video to soundcard/screen (default)
+
+
+ render timeline (forces decoding)
+
+
+ render timeline (tries to avoid decoding/reencoding)
+
+
+
+
+
+ This is a legacy format and you should avoid to use it. The formatter
+is really not in good shape and is deprecated.
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ The #GESProject is used to control a set of #GESAsset and is a
+#GESAsset with #GES_TYPE_TIMELINE as @extractable_type itself. That
+means that you can extract #GESTimeline from a project as followed:
+
+|[
+ GESProject *project;
+ GESTimeline *timeline;
+
+ project = ges_project_new ("file:///path/to/a/valid/project/uri");
+
+ // Here you can connect to the various signal to get more infos about
+ // what is happening and recover from errors if possible
+ ...
+
+ timeline = ges_asset_extract (GES_ASSET (project));
+]|
+
+The #GESProject class offers a higher level API to handle #GESAsset-s.
+It lets you request new asset, and it informs you about new assets through
+a set of signals. Also it handles problem such as missing files/missing
+#GstElement and lets you try to recover from those.
+
+
+
+
+ Creates a new #GESProject and sets its uri to @uri if provided. Note that
+if @uri is not valid or %NULL, the uri of the project will then be set
+the first time you save the project. If you then save the project to
+other locations, it will never be updated again and the first valid URI is
+the URI it will keep refering to.
+
+ A newly created #GESProject
+
+
+
+
+ The uri to be set after creating the project.
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Adds a #Asset to @project, the project will keep a reference on
+@asset.
+
+ %TRUE if the asset could be added %FALSE it was already
+in the project
+
+
+
+
+ A #GESProject
+
+
+
+ A #GESAsset to add to @project
+
+
+
+
+
+ Adds @profile to the project. It lets you save in what format
+the project has been renders and keep a reference to those formats.
+Also, those formats will be saves to the project file when possible.
+
+ %TRUE if @profile could be added, %FALSE otherwize
+
+
+
+
+ A #GESProject
+
+
+
+ A #GstEncodingProfile to add to the project. If a profile with
+the same name already exists, it will be replaced
+
+
+
+
+
+ Create and add a #GESAsset to @project. You should connect to the
+"asset-added" signal to get the asset when it finally gets added to
+@project
+
+ %TRUE if the asset started to be added %FALSE it was already
+in the project
+
+
+
+
+ A #GESProject
+
+
+
+ The id of the asset to create and add to @project
+
+
+
+ The #GType of the asset to create
+
+
+
+
+
+ Create and add a #GESAsset to @project. You should connect to the
+"asset-added" signal to get the asset when it finally gets added to
+@project
+
+ The newly created #GESAsset or %NULL.
+
+
+
+
+ A #GESProject
+
+
+
+ The id of the asset to create and add to @project
+
+
+
+ The #GType of the asset to create
+
+
+
+
+
+
+ The #GESAsset with
+@id or %NULL if no asset with @id as an ID
+
+
+
+
+ A #GESProject
+
+
+
+ The id of the asset to retrieve
+
+
+
+ The extractable_type of the asset
+to retrieve from @object
+
+
+
+
+
+ Get the assets that are being loaded
+
+ A set of loading asset
+that will be added to @project. Note that those Asset are *not* loaded yet,
+and thus can not be used
+
+
+
+
+
+
+ A #GESProject
+
+
+
+
+
+ Retrieve the uri that is currently set on @project
+
+ a newly allocated string representing uri.
+
+
+
+
+ A #GESProject
+
+
+
+
+
+ List all @asset contained in @project filtering per extractable_type
+as defined by @filter. It copies the asset and thus will not be updated
+in time.
+
+ The list of
+#GESAsset the object contains
+
+
+
+
+
+
+ A #GESProject
+
+
+
+ Type of assets to list, #GES_TYPE_EXTRACTABLE will list
+all assets
+
+
+
+
+
+ Lists the encoding profile that have been set to @project. The first one
+is the latest added.
+
+ The
+list of #GstEncodingProfile used in @project
+
+
+
+
+
+
+ A #GESProject
+
+
+
+
+
+ Loads @project into @timeline
+
+ %TRUE if the project could be loaded %FALSE otherwize.
+
+
+
+
+ A #GESProject that has an @uri set already
+
+
+
+ A blank timeline to load @project into
+
+
+
+
+
+ remove a @asset to from @project.
+
+ %TRUE if the asset could be removed %FALSE otherwise
+
+
+
+
+ A #GESProject
+
+
+
+ A #GESAsset to remove from @project
+
+
+
+
+
+ Save the timeline of @project to @uri. You should make sure that @timeline
+is one of the timelines that have been extracted from @project
+(using ges_asset_extract (@project);)
+
+ %TRUE if the project could be save, %FALSE otherwize
+
+
+
+
+ A #GESProject to save
+
+
+
+ The #GESTimeline to save, it must have been extracted from @project
+
+
+
+ The uri where to save @project and @timeline
+
+
+
+ The formatter asset to use or %NULL. If %NULL,
+will try to save in the same format as the one from which the timeline as been loaded
+or default to the formatter with highest rank
+
+
+
+ %TRUE to overwrite file if it exists
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ The #GESAsset that has been added to @project
+
+
+
+
+
+
+
+
+
+
+ The #GESAsset that started loading
+
+
+
+
+
+
+
+
+
+
+ The #GESAsset that has been removed from @project
+
+
+
+
+
+ Informs you that a #GESAsset could not be created. In case of
+missing GStreamer plugins, the error will be set to #GST_CORE_ERROR
+#GST_CORE_ERROR_MISSING_PLUGIN
+
+
+
+
+
+ The #GError defining the error that accured, might be %NULL
+
+
+
+ The @id of the asset that failed loading
+
+
+
+ The @extractable_type of the asset that
+failed loading
+
+
+
+
+
+
+
+
+
+
+ The #GESTimeline that complete loading
+
+
+
+
+
+ |[
+static gchar
+source_moved_cb (GESProject *project, GError *error, GESAsset *asset_with_error)
+{
+ return g_strdup ("file:///the/new/uri.ogg");
+}
+
+static int
+main (int argc, gchar ** argv)
+{
+ GESTimeline *timeline;
+ GESProject *project = ges_project_new ("file:///some/uri.xges");
+
+ g_signal_connect (project, "missing-uri", source_moved_cb, NULL);
+ timeline = ges_asset_extract (GES_ASSET (project));
+}
+]|
+
+ The new URI of @wrong_asset
+
+
+
+
+ The error that happened
+
+
+
+ The asset with the wrong ID, you should us it and its content
+only to find out what the new location is.
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Base class for single-media sources
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Base class for sources of a #GESLayer
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Useful for testing purposes.
+
+You can use the ges_asset_request_simple API to create an Asset
+capable of extracting GESTestClip-s
+
+
+
+ Creates a new #GESTestClip.
+
+ The newly created #GESTestClip,
+or %NULL if there was an error.
+
+
+
+
+ Creates a new #GESTestClip for the provided @nick.
+
+ The newly created #GESTestClip,
+or %NULL if there was an error.
+
+
+
+
+ the nickname for which to create the #GESTestClip
+
+
+
+
+
+ Get the frequency @self generates.
+
+ The frequency @self generates. See audiotestsrc element.
+
+
+
+
+ a #GESTestClip
+
+
+
+
+
+ Get the volume of the test audio signal applied on @self.
+
+ The volume of the test audio signal applied on @self.
+
+
+
+
+ a #GESTestClip
+
+
+
+
+
+ Get the #GESVideoTestPattern which is applied on @self.
+
+ The #GESVideoTestPattern which is applied on @self.
+
+
+
+
+ a #GESTestClip
+
+
+
+
+
+ Let you know if the audio track of @self is muted or not.
+
+ Whether the audio track of @self is muted or not.
+
+
+
+
+ a #GESTestClip
+
+
+
+
+
+ Sets the frequency to generate. See audiotestsrc element.
+
+
+
+
+
+ the #GESTestClip to set the frequency on
+
+
+
+ the frequency you want to use on @self
+
+
+
+
+
+ Sets whether the audio track of this clip is muted or not.
+
+
+
+
+
+ the #GESTestClip on which to mute or unmute the audio track
+
+
+
+ %TRUE to mute the audio track, %FALSE to unmute it
+
+
+
+
+
+ Sets the volume of the test audio signal.
+
+
+
+
+
+ the #GESTestClip to set the volume on
+
+
+
+ the volume of the audio signal you want to use on @self
+
+
+
+
+
+ Sets which video pattern to display on @self.
+
+
+
+
+
+ the #GESTestClip to set the pattern on
+
+
+
+ the #GESVideoTestPattern to use on @self
+
+
+
+
+
+ The frequency to generate for audio track elements.
+
+
+
+ Whether the sound will be played or not.
+
+
+
+ The volume for the audio track elements.
+
+
+
+ Video pattern to display in video track elements.
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Horizontal alignment of the text.
+
+ align text left
+
+
+ align text center
+
+
+ align text right
+
+
+ align text on xpos position
+
+
+
+
+
+
+
+
+ Creates a new #GESTextOverlay.
+
+ The newly created #GESTextOverlay or
+%NULL if something went wrong.
+
+
+
+
+ Get the color used by @source.
+
+ The color used by @source.
+
+
+
+
+ a GESTextOverlay
+
+
+
+
+
+ Get the pango font description currently set on @source.
+
+ The pango font description currently set on @source.
+
+
+
+
+ a GESTextOverlay
+
+
+
+
+
+ Get the horizontal aligment used by @source.
+
+ The horizontal aligment used by @source.
+
+
+
+
+ a GESTextOverlay
+
+
+
+
+
+ Get the text currently set on @source.
+
+ The text currently set on @source.
+
+
+
+
+ a GESTextOverlay
+
+
+
+
+
+ Get the vertical aligment used by @source.
+
+ The vertical aligment used by @source.
+
+
+
+
+ a GESTextOverlay
+
+
+
+
+
+ Get the horizontal position used by @source.
+
+ The horizontal position used by @source.
+
+
+
+
+ a GESTextOverlay
+
+
+
+
+
+ Get the vertical position used by @source.
+
+ The vertical position used by @source.
+
+
+
+
+ a GESTextOverlay
+
+
+
+
+
+ Sets the color of the text.
+
+
+
+
+
+ the #GESTextOverlay* to set
+
+
+
+ The color @self is being set to
+
+
+
+
+
+ Sets the pango font description of the text this track element
+will render.
+
+
+
+
+
+ the #GESTextOverlay
+
+
+
+ the pango font description
+
+
+
+
+
+ Sets the horizontal aligment of the text.
+
+
+
+
+
+ the #GESTextOverlay* to set text on
+
+
+
+ The #GESTextHAlign defining the horizontal alignment
+of the text render by @self.
+
+
+
+
+
+ Sets the text this track element will render.
+
+
+
+
+
+ the #GESTextOverlay* to set text on
+
+
+
+ the text to render. an internal copy of this text will be
+made.
+
+
+
+
+
+ Sets the vertical aligment of the text.
+
+
+
+
+
+ the #GESTextOverlay* to set text on
+
+
+
+ The #GESTextVAlign defining the vertical alignment
+of the text render by @self.
+
+
+
+
+
+ Sets the horizontal position of the text.
+
+
+
+
+
+ the #GESTextOverlay* to set
+
+
+
+ The horizontal position @self is being set to
+
+
+
+
+
+ Sets the vertical position of the text.
+
+
+
+
+
+ the #GESTextOverlay* to set
+
+
+
+ The vertical position @self is being set to
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Renders text onto the next lower priority stream using textrender.
+
+
+
+ Creates a new #GESTextOverlayClip
+
+ The newly created
+#GESTextOverlayClip, or %NULL if there was an error.
+
+
+
+
+ Get the color used by @source.
+
+ The color used by @source.
+
+
+
+
+ a #GESTextOverlayClip
+
+
+
+
+
+ Get the pango font description used by @self.
+
+ The pango font description used by @self.
+
+
+
+
+ a #GESTextOverlayClip
+
+
+
+
+
+ Get the horizontal aligment used by @self.
+
+ The horizontal aligment used by @self.
+
+
+
+
+ a #GESTextOverlayClip
+
+
+
+
+
+ Get the text currently set on @self.
+
+ The text currently set on @self.
+
+
+
+
+ a #GESTextOverlayClip
+
+
+
+
+
+ Get the vertical aligment used by @self.
+
+ The vertical aligment used by @self.
+
+
+
+
+ a #GESTextOverlayClip
+
+
+
+
+
+ Get the horizontal position used by @source.
+
+ The horizontal position used by @source.
+
+
+
+
+ a #GESTextOverlayClip
+
+
+
+
+
+ Get the vertical position used by @source.
+
+ The vertical position used by @source.
+
+
+
+
+ a #GESTextOverlayClip
+
+
+
+
+
+ Sets the color of the text.
+
+
+
+
+
+ the #GESTextOverlayClip* to set
+
+
+
+ The color @self is being set to
+
+
+
+
+
+ Sets the pango font description of the text
+
+
+
+
+
+ the #GESTextOverlayClip*
+
+
+
+ the pango font description
+
+
+
+
+
+ Sets the horizontal aligment of the text.
+
+
+
+
+
+ the #GESTextOverlayClip* to set horizontal alignement of text on
+
+
+
+ #GESTextHAlign
+
+
+
+
+
+ Sets the text this clip will render.
+
+
+
+
+
+ the #GESTextOverlayClip* to set text on
+
+
+
+ the text to render. an internal copy of this text will be
+made.
+
+
+
+
+
+ Sets the vertical aligment of the text.
+
+
+
+
+
+ the #GESTextOverlayClip* to set vertical alignement of text on
+
+
+
+ #GESTextVAlign
+
+
+
+
+
+ Sets the horizontal position of the text.
+
+
+
+
+
+ the #GESTextOverlayClip* to set
+
+
+
+ The horizontal position @self is being set to
+
+
+
+
+
+ Sets the vertical position of the text.
+
+
+
+
+
+ the #GESTextOverlayClip* to set
+
+
+
+ The vertical position @self is being set to
+
+
+
+
+
+ The color of the text
+
+
+
+ Pango font description string
+
+
+
+ Horizontal alignment of the text
+
+
+
+ The text to diplay
+
+
+
+ Vertical alignent of the text
+
+
+
+ The horizontal position of the text
+
+
+
+ The vertical position of the text
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Vertical alignment of the text.
+
+ draw text on the baseline
+
+
+ draw text on the bottom
+
+
+ draw text on top
+
+
+ draw text on ypos position
+
+
+ draw text on the center
+
+
+
+
+
+ #GESTimeline is the central object for any multimedia timeline.
+
+Contains a list of #GESLayer which users should use to arrange the
+various clips through time.
+
+The output type is determined by the #GESTrack that are set on
+the #GESTimeline.
+
+To save/load a timeline, you can use the ges_timeline_load_from_uri() and
+ges_timeline_save_to_uri() methods to use the default format. If you wish
+
+Note that any change you make in the timeline will not actually be taken
+into account until you call the #ges_timeline_commit method.
+
+
+
+
+ Creates a new empty #GESTimeline.
+
+ The new timeline.
+
+
+
+
+ Creates a new #GESTimeline containing a raw audio and a
+raw video track.
+
+ The newly created #GESTimeline.
+
+
+
+
+ Creates a timeline from the given URI.
+
+ A new timeline if the uri was loaded
+successfully, or %NULL if the uri could not be loaded.
+
+
+
+
+ the URI to load from
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Add the layer to the timeline. The reference to the @layer will be stolen
+by the @timeline.
+
+ %TRUE if the layer was properly added, else %FALSE.
+
+
+
+
+ a #GESTimeline
+
+
+
+ the #GESLayer to add
+
+
+
+
+
+ Add a track to the timeline. The reference to the track will be stolen by the
+pipeline.
+
+ %TRUE if the track was properly added, else %FALSE.
+
+
+
+
+ a #GESTimeline
+
+
+
+ the #GESTrack to add
+
+
+
+
+
+ Append a newly created #GESLayer to @timeline
+Note that you do not own any reference to the returned layer.
+
+ The newly created #GESLayer, or the last (empty)
+#GESLayer of @timeline.
+
+
+
+
+ a #GESTimeline
+
+
+
+
+
+ Commit all the pending changes of the clips contained in the
+@timeline.
+
+When changes happen in a timeline, they are not
+directly executed in the non-linear engine. Call this method once you are
+done with a set of changes and want it to be executed.
+
+The #GESTimeline::commited signal will be emitted when the (possibly updated)
+#GstPipeline is ready to output data again, except if the state of the
+timeline was #GST_STATE_READY or #GST_STATE_NULL.
+
+Note that all the pending changes will automatically be executed when the
+timeline goes from #GST_STATE_READY to #GST_STATE_PAUSED, which usually is
+triggered by corresponding state changes in a containing #GESPipeline.
+
+You should not try to change the state of the timeline, seek it or add
+tracks to it during a commit operation, that is between a call to this
+function and after receiving the #GESTimeline::commited signal.
+
+See #ges_timeline_commit_sync if you don't want to bother with waiting
+for the signal.
+
+ %TRUE if pending changes were commited or %FALSE if nothing needed
+to be commited
+
+
+
+
+ a #GESTimeline
+
+
+
+
+
+ Commit all the pending changes of the #GESClips contained in the
+@timeline.
+
+Will return once the update is complete, that is when the
+(possibly updated) #GstPipeline is ready to output data again, or if the
+state of the timeline was #GST_STATE_READY or #GST_STATE_NULL.
+
+This function will wait for any pending state change of the timeline by
+calling #gst_element_get_state with a #GST_CLOCK_TIME_NONE timeout, you
+should not try to change the state from another thread before this function
+has returned.
+
+See #ges_timeline_commit for more information.
+
+ %TRUE if pending changes were commited or %FALSE if nothing needed
+to be commited
+
+
+
+
+ a #GESTimeline
+
+
+
+
+
+ Gets whether transitions are automatically added when objects
+overlap or not.
+
+ %TRUE if transitions are automatically added, else %FALSE.
+
+
+
+
+ a #GESTimeline
+
+
+
+
+
+ Get the current duration of @timeline
+
+ The current duration of @timeline
+
+
+
+
+ a #GESTimeline
+
+
+
+
+
+ Gets a #GESTimelineElement contained in the timeline
+
+ The #GESTimelineElement or %NULL if
+not found.
+
+
+
+
+ a #GESTimeline
+
+
+
+
+
+
+
+
+ Get the list of #GESGroup present in the Timeline.
+
+ the list of
+#GESGroup that contain clips present in the timeline's layers.
+Must not be changed.
+
+
+
+
+
+
+ a #GESTimeline
+
+
+
+
+
+ Retrieve the layer with @priority as a priority
+
+ A #GESLayer or %NULL if no layer with
+@priority was found
+
+Since 1.6
+
+
+
+
+ The #GESTimeline to retrive a layer from
+
+
+
+ The priority of the layer to find
+
+
+
+
+
+ Get the list of #GESLayer present in the Timeline.
+
+ the list of
+#GESLayer present in the Timeline sorted by priority.
+The caller should unref each Layer once he is done with them.
+
+
+
+
+
+
+ a #GESTimeline
+
+
+
+
+
+ Search the #GstPad corresponding to the given @timeline's @track.
+
+ The corresponding #GstPad if it is
+found, or %NULL if there is an error.
+
+
+
+
+ The #GESTimeline
+
+
+
+ The #GESTrack
+
+
+
+
+
+ Gets the configured snapping distance of the timeline. See
+the documentation of the property snapping_distance for more
+information.
+
+ The @snapping_distance property of the timeline
+
+
+
+
+ a #GESTimeline
+
+
+
+
+
+ Search the #GESTrack corresponding to the given @timeline's @pad.
+
+ The corresponding #GESTrack if it is
+found, or %NULL if there is an error.
+
+
+
+
+ The #GESTimeline
+
+
+
+ The #GstPad
+
+
+
+
+
+ Returns the list of #GESTrack used by the Timeline.
+
+ A list of #GESTrack.
+The caller should unref each track once he is done with them.
+
+
+
+
+
+
+ a #GESTimeline
+
+
+
+
+
+ Check whether a #GESTimeline is empty or not
+
+ %TRUE if the timeline is empty %FALSE otherwize
+
+
+
+
+ a #GESTimeline
+
+
+
+
+
+ Loads the contents of URI into the given timeline.
+
+ %TRUE if the timeline was loaded successfully, or %FALSE if the uri
+could not be loaded.
+
+
+
+
+ an empty #GESTimeline into which to load the formatter
+
+
+
+ The URI to load from
+
+
+
+
+
+ Paste @element inside the timeline. @element must have been
+created using ges_timeline_element_copy with deep=TRUE set,
+i.e. it must be a deep copy, otherwise it will fail.
+
+ Shallow copy of the @element pasted
+
+
+
+
+ The #GESTimeline onto which the #GESTimelineElement should be pasted
+
+
+
+ The #GESTimelineElement to paste
+
+
+
+ The position in the timeline the element should
+be pasted to, meaning it will become the start of @element
+
+
+
+ The #GESLayer to which the element should be pasted to.
+-1 means paste to the same layer from which the @element has been copied from.
+
+
+
+
+
+ Removes the layer from the timeline. The reference that the @timeline holds on
+the layer will be dropped. If you wish to use the @layer after calling this
+method, you need to take a reference before calling.
+
+ %TRUE if the layer was properly removed, else %FALSE.
+
+
+
+
+ a #GESTimeline
+
+
+
+ the #GESLayer to remove
+
+
+
+
+
+ Remove the @track from the @timeline. The reference stolen when adding the
+@track will be removed. If you wish to use the @track after calling this
+function you must ensure that you have a reference to it.
+
+ %TRUE if the @track was properly removed, else %FALSE.
+
+
+
+
+ a #GESTimeline
+
+
+
+ the #GESTrack to remove
+
+
+
+
+
+ Saves the timeline to the given location
+
+ %TRUE if the timeline was successfully saved to the given location,
+else %FALSE.
+
+
+
+
+ a #GESTimeline
+
+
+
+ The location to save to
+
+
+
+ The formatter asset to use or %NULL. If %NULL,
+will try to save in the same format as the one from which the timeline as been loaded
+or default to the formatter with highest rank
+
+
+
+ %TRUE to overwrite file if it exists
+
+
+
+
+
+ Sets the layer to the given @auto_transition. See the documentation of the
+property auto_transition for more information.
+
+
+
+
+
+ a #GESLayer
+
+
+
+ whether the auto_transition is active
+
+
+
+
+
+ Sets the @snapping_distance of the timeline. See the documentation of the
+property snapping_distance for more information.
+
+
+
+
+
+ a #GESLayer
+
+
+
+ whether the snapping_distance is active
+
+
+
+
+
+ Sets whether transitions are added automagically when clips overlap.
+
+
+
+ Current duration (in nanoseconds) of the #GESTimeline
+
+
+
+ Distance (in nanoseconds) from which a moving object will snap
+with it neighboors. 0 means no snapping.
+
+
+
+
+
+
+ A list of #GESLayer sorted by priority NOTE: Do not modify.
+
+
+
+
+
+ A list of #GESTrack sorted by priority NOTE: Do not modify.
+
+
+
+
+
+
+
+
+
+
+
+
+
+ This signal will be emitted once the changes initiated by #ges_timeline_commit
+have been executed in the backend. Use #ges_timeline_commit_sync if you
+don't need to do anything in the meantime.
+
+
+
+
+
+ Will be emitted after a new group is added to to the timeline.
+
+
+
+
+
+ the #GESGroup
+
+
+
+
+
+ Will be emitted after a group has been removed from the timeline.
+
+
+
+
+
+ the #GESGroup
+
+
+
+ a list of #GESContainer
+
+
+
+
+
+
+
+ Will be emitted after a new layer is added to the timeline.
+
+
+
+
+
+ the #GESLayer that was added to the timeline
+
+
+
+
+
+ Will be emitted after the layer was removed from the timeline.
+
+
+
+
+
+ the #GESLayer that was removed from the timeline
+
+
+
+
+
+
+ a #GPtrArray of #GESTrack-s where that object should be added
+
+
+
+
+
+
+ The #GESClip on which @track_element will land
+
+
+
+ The #GESTrackElement for which to choose the tracks it should land into
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Will be emitted after the track was added to the timeline.
+
+
+
+
+
+ the #GESTrack that was added to the timeline
+
+
+
+
+
+ Will be emitted after the track was removed from the timeline.
+
+
+
+
+
+ the #GESTrack that was removed from the timeline
+
+
+
+
+
+
+
+ parent class
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ The GESTimelineElement base class implements the notion of timing as well
+as priority. A GESTimelineElement can have a parent object which will be
+responsible for controlling its timing properties.
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Gets all the TrackTypes @self will interact with
+
+
+
+
+
+ A #GESTimelineElement
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Looks up which @element and @pspec would be effected by the given @name. If various
+contained elements have this property name you will get the first one, unless you
+specify the class name in @name.
+
+ TRUE if @element and @pspec could be found. FALSE otherwise. In that
+case the values for @pspec and @element are not modified. Unref @element after
+usage.
+
+
+
+
+ object to lookup the property in
+
+
+
+ name of the property to look up. You can specify the name of the
+ class as such: "ClassName::property-name", to guarantee that you get the
+ proper GParamSpec in case various GstElement-s contain the same property
+ name. If you don't do so, you will get the first element found, having
+ this property and the and the corresponding GParamSpec.
+
+
+
+ pointer to a #GstElement that
+ takes the real object to set property on
+
+
+
+ pointer to take the #GParamSpec
+ describing the property
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Edits @self in ripple mode. It allows you to modify the
+start of @self and move the following neighbours accordingly.
+This will change the overall timeline duration.
+
+ %TRUE if the self as been rippled properly, %FALSE if an error
+occured
+
+
+
+
+ The #GESTimelineElement to ripple.
+
+
+
+ The new start of @self in ripple mode.
+
+
+
+
+
+ Edits @self in ripple mode. It allows you to modify the
+duration of a @self and move the following neighbours accordingly.
+This will change the overall timeline duration.
+
+ %TRUE if the self as been rippled properly, %FALSE if an error
+occured
+
+
+
+
+ The #GESTimelineElement to ripple.
+
+
+
+ The new end (start + duration) of @self in ripple mode. It will
+ basically only change the duration of @self.
+
+
+
+
+
+ Edits @self in roll mode. It allows you to modify the
+duration of a @self and trim (basicly change the start + inpoint
+in this case) the following neighbours accordingly.
+This will not change the overall timeline duration.
+
+ %TRUE if the self as been rolled properly, %FALSE if an error
+occured
+
+
+
+
+ The #GESTimelineElement to roll.
+
+
+
+ The new end (start + duration) of @self in roll mode
+
+
+
+
+
+ Edits @self in roll mode. It allows you to modify the
+start and inpoint of a @self and "resize" (basicly change the duration
+in this case) of the previous neighbours accordingly.
+This will not change the overall timeline duration.
+
+ %TRUE if the self as been roll properly, %FALSE if an error
+occured
+
+
+
+
+ The #GESTimelineElement to roll
+
+
+
+ The new start of @self in roll mode, it will also adapat
+the in-point of @self according
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Sets the parent of @self to @parent. The parents needs to already
+own a hard reference on @self.
+
+ %TRUE if @parent could be set or %FALSE when @self
+already had a parent or @self and @parent are the same.
+
+
+
+
+ a #GESTimelineElement
+
+
+
+ new parent of self
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Edits @self in trim mode. It allows you to modify the
+inpoint and start of @self.
+This will not change the overall timeline duration.
+
+Note that to trim the end of an self you can just set its duration. The same way
+as this method, it will take into account the snapping-distance property of the
+timeline in which @self is.
+
+ %TRUE if the self as been trimmed properly, %FALSE if an error
+occured
+
+
+
+
+ The #GESTimelineElement to trim.
+
+
+
+ The new start of @self in trim mode, will adapt the inpoint
+of @self accordingly
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Copies @self
+
+ The newly create #GESTimelineElement, copied from @self
+
+
+
+
+ The #GESTimelineElement to copy
+
+
+
+ whether we want to create the elements @self contains or not
+
+
+
+
+
+ Gets properties of a child of @self.
+
+
+
+
+
+ The origin #GESTimelineElement
+
+
+
+ The name of the first property to get
+
+
+
+ return location for the first property, followed optionally by more
+name/return location pairs, followed by NULL
+
+
+
+
+
+ In general, a copy is made of the property contents and
+the caller is responsible for freeing the memory by calling
+g_value_unset().
+
+Gets a property of a GstElement contained in @object.
+
+Note that #ges_timeline_element_get_child_property is really
+intended for language bindings, #ges_timeline_element_get_child_properties
+is much more convenient for C programming.
+
+ %TRUE if the property was found, %FALSE otherwize
+
+
+
+
+ The origin #GESTimelineElement
+
+
+
+ The name of the property
+
+
+
+ return location for the property value, it will
+be initialized if it is initialized with 0
+
+
+
+
+
+ Gets a property of a child of @self.
+
+
+
+
+
+ a #GESTrackElement
+
+
+
+ The #GParamSpec that specifies the property you want to get
+
+
+
+ return location for the value
+
+
+
+
+
+ Gets a property of a child of @self. If there are various child elements
+that have the same property name, you can distinguish them using the following
+syntax: 'ClasseName::property_name' as property name. If you don't, the
+corresponding property of the first element found will be set.
+
+
+
+
+
+ The #GESTimelineElement parent object
+
+
+
+ The name of the first property to get
+
+
+
+ value for the first property, followed optionally by more
+name/return location pairs, followed by NULL
+
+
+
+
+
+
+ The @duration of @self
+
+
+
+
+ a #GESTimelineElement
+
+
+
+
+
+
+ The @inpoint of @self
+
+
+
+
+ a #GESTimelineElement
+
+
+
+
+
+
+ The @maxduration of @self
+
+
+
+
+ a #GESTimelineElement
+
+
+
+
+
+ Returns a copy of the name of @self.
+Caller should g_free() the return value after usage.
+
+ The name of @self
+
+
+
+
+ a #GESTimelineElement
+
+
+
+
+
+ Returns the parent of @self. This function increases the refcount
+of the parent object so you should gst_object_unref() it after usage.
+
+ parent of @self, this can be %NULL if
+@self has no parent. unref after usage.
+
+
+
+
+ a #GESTimelineElement
+
+
+
+
+
+
+ The @priority of @self
+
+
+
+
+ a #GESTimelineElement
+
+
+
+
+
+
+ The @start of @self
+
+
+
+
+ a #GESTimelineElement
+
+
+
+
+
+ Returns the timeline of @self. This function increases the refcount
+of the timeline so you should gst_object_unref() it after usage.
+
+ timeline of @self, this can be %NULL if
+@self has no timeline. unref after usage.
+
+
+
+
+ a #GESTimelineElement
+
+
+
+
+
+ Gets the toplevel #GESTimelineElement controlling @self
+
+ The toplevel controlling parent of @self
+
+
+
+
+ The #GESTimelineElement to get the toplevel parent from
+
+
+
+
+
+ Gets all the TrackTypes @self will interact with
+
+
+
+
+
+ A #GESTimelineElement
+
+
+
+
+
+ Gets an array of #GParamSpec* for all configurable properties of the
+children of @self.
+
+ an array of #GParamSpec* which should be freed after use or
+%NULL if something went wrong
+
+
+
+
+
+
+ The #GESTimelineElement to get the list of children properties from
+
+
+
+ return location for the length of the returned array
+
+
+
+
+
+ Looks up which @element and @pspec would be effected by the given @name. If various
+contained elements have this property name you will get the first one, unless you
+specify the class name in @name.
+
+ TRUE if @element and @pspec could be found. FALSE otherwise. In that
+case the values for @pspec and @element are not modified. Unref @element after
+usage.
+
+
+
+
+ object to lookup the property in
+
+
+
+ name of the property to look up. You can specify the name of the
+ class as such: "ClassName::property-name", to guarantee that you get the
+ proper GParamSpec in case various GstElement-s contain the same property
+ name. If you don't do so, you will get the first element found, having
+ this property and the and the corresponding GParamSpec.
+
+
+
+ pointer to a #GstElement that
+ takes the real object to set property on
+
+
+
+ pointer to take the #GParamSpec
+ describing the property
+
+
+
+
+
+ Paste @self inside the timeline. @self must have been created
+using ges_timeline_element_copy with recurse=TRUE set,
+otherwise it will fail.
+
+ Paste @self copying the element
+
+
+
+
+ The #GESTimelineElement to paste
+
+
+
+ The position in the timeline the element should
+be copied to, meaning it will become the start of @self
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Edits @self in ripple mode. It allows you to modify the
+start of @self and move the following neighbours accordingly.
+This will change the overall timeline duration.
+
+ %TRUE if the self as been rippled properly, %FALSE if an error
+occured
+
+
+
+
+ The #GESTimelineElement to ripple.
+
+
+
+ The new start of @self in ripple mode.
+
+
+
+
+
+ Edits @self in ripple mode. It allows you to modify the
+duration of a @self and move the following neighbours accordingly.
+This will change the overall timeline duration.
+
+ %TRUE if the self as been rippled properly, %FALSE if an error
+occured
+
+
+
+
+ The #GESTimelineElement to ripple.
+
+
+
+ The new end (start + duration) of @self in ripple mode. It will
+ basically only change the duration of @self.
+
+
+
+
+
+ Edits @self in roll mode. It allows you to modify the
+duration of a @self and trim (basicly change the start + inpoint
+in this case) the following neighbours accordingly.
+This will not change the overall timeline duration.
+
+ %TRUE if the self as been rolled properly, %FALSE if an error
+occured
+
+
+
+
+ The #GESTimelineElement to roll.
+
+
+
+ The new end (start + duration) of @self in roll mode
+
+
+
+
+
+ Edits @self in roll mode. It allows you to modify the
+start and inpoint of a @self and "resize" (basicly change the duration
+in this case) of the previous neighbours accordingly.
+This will not change the overall timeline duration.
+
+ %TRUE if the self as been roll properly, %FALSE if an error
+occured
+
+
+
+
+ The #GESTimelineElement to roll
+
+
+
+ The new start of @self in roll mode, it will also adapat
+the in-point of @self according
+
+
+
+
+
+ Sets a property of a child of @self. If there are various child elements
+that have the same property name, you can distinguish them using the following
+syntax: 'ClasseName::property_name' as property name. If you don't, the
+corresponding property of the first element found will be set.
+
+
+
+
+
+ The #GESTimelineElement parent object
+
+
+
+ The name of the first property to set
+
+
+
+ value for the first property, followed optionally by more
+name/return location pairs, followed by NULL
+
+
+
+
+
+ Sets a property of a child of @self
+
+Note that #ges_timeline_element_set_child_property is really
+intended for language bindings, #ges_timeline_element_set_child_properties
+is much more convenient for C programming.
+
+ %TRUE if the property was set, %FALSE otherwize
+
+
+
+
+ The origin #GESTimelineElement
+
+
+
+ The name of the property
+
+
+
+ the value
+
+
+
+
+
+ Sets a property of a child of @self.
+
+
+
+
+
+ a #GESTimelineElement
+
+
+
+ The #GParamSpec that specifies the property you want to set
+
+
+
+ the value
+
+
+
+
+
+ Sets a property of a child of @self. If there are various child elements
+that have the same property name, you can distinguish them using the following
+syntax: 'ClasseName::property_name' as property name. If you don't, the
+corresponding property of the first element found will be set.
+
+
+
+
+
+ The #GESTimelineElement parent object
+
+
+
+ The name of the first property to set
+
+
+
+ value for the first property, followed optionally by more
+name/return location pairs, followed by NULL
+
+
+
+
+
+ Set the duration of the object
+
+Note that if the timeline snap-distance property of the timeline containing
+@self is set, @self will properly snap to its neighboors.
+
+
+
+
+
+ a #GESTimelineElement
+
+
+
+ the duration in #GstClockTime
+
+
+
+
+
+ Set the in-point, that is the moment at which the @self will start
+outputting data from its contents.
+
+
+
+
+
+ a #GESTimelineElement
+
+
+
+ the in-point in #GstClockTime
+
+
+
+
+
+ Set the maximun duration of the object
+
+
+
+
+
+ a #GESTimelineElement
+
+
+
+ the maximum duration in #GstClockTime
+
+
+
+
+
+ Sets the name of object, or gives @self a guaranteed unique name (if name is NULL).
+This function makes a copy of the provided name, so the caller retains ownership
+of the name it sent.
+
+
+
+
+
+ a #GESTimelineElement
+
+
+
+ The name @self should take (if avalaible<)
+
+
+
+
+
+ Sets the parent of @self to @parent. The parents needs to already
+own a hard reference on @self.
+
+ %TRUE if @parent could be set or %FALSE when @self
+already had a parent or @self and @parent are the same.
+
+
+
+
+ a #GESTimelineElement
+
+
+
+ new parent of self
+
+
+
+
+
+ Sets the priority of the object within the containing layer
+ All priority management is done by GES itself now.
+To set #GESEffect priorities #ges_clip_set_top_effect_index should
+be used.
+
+
+
+
+
+ a #GESTimelineElement
+
+
+
+ the priority
+
+
+
+
+
+ Set the position of the object in its containing layer.
+
+Note that if the snapping-distance property of the timeline containing
+@self is set, @self will properly snap to the edges around @start.
+
+
+
+
+
+ a #GESTimelineElement
+
+
+
+ the position in #GstClockTime
+
+
+
+
+
+ Sets the timeline of @self to @timeline.
+
+ %TRUE if @timeline could be set or %FALSE when @timeline
+already had a timeline.
+
+
+
+
+ a #GESTimelineElement
+
+
+
+ The #GESTimeline @self is in
+
+
+
+
+
+ Edits @self in trim mode. It allows you to modify the
+inpoint and start of @self.
+This will not change the overall timeline duration.
+
+Note that to trim the end of an self you can just set its duration. The same way
+as this method, it will take into account the snapping-distance property of the
+timeline in which @self is.
+
+ %TRUE if the self as been trimmed properly, %FALSE if an error
+occured
+
+
+
+
+ The #GESTimelineElement to trim.
+
+
+
+ The new start of @self in trim mode, will adapt the inpoint
+of @self accordingly
+
+
+
+
+
+ The duration (in nanoseconds) which will be used in the container
+
+
+
+ The in-point at which this #GESTimelineElement will start outputting data
+from its contents (in nanoseconds).
+
+Ex : an in-point of 5 seconds means that the first outputted buffer will
+be the one located 5 seconds in the controlled resource.
+
+
+
+ The maximum duration (in nanoseconds) of the #GESTimelineElement.
+
+
+
+ The name of the object
+
+
+
+ The parent container of the object
+
+
+
+ The priority of the object.
+
+Setting GESTimelineElement priorities is deprecated
+as all priority management is done by GES itself now.
+
+
+
+ Whether the element should be serialized.
+
+
+
+ The position of the object in its container (in nanoseconds).
+
+
+
+ The timeline in which @element is
+
+
+
+
+
+
+ The #GESTimelineElement that controls the object
+
+
+
+ The #GESAsset from which the object has been extracted
+
+
+
+ position (in time) of the object
+
+
+
+ Position in the media from which the object should be used
+
+
+
+ duration of the object to be used
+
+
+
+ The maximum duration the object can have
+
+
+
+ priority of the object in the layer (0:top priority)
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ The deep notify signal is used to be notified of property changes of all
+the childs of @timeline_element
+
+
+
+
+
+ the object that originated the signal
+
+
+
+ the property that changed
+
+
+
+
+
+
+ The GESTimelineElement base class. Subclasses should override at least
+@set_start @set_inpoint @set_duration @ripple @ripple_end @roll_start
+@roll_end and @trim.
+
+Vmethods in subclasses should apply all the operation they need to but
+the real method implementation is in charge of setting the proper field,
+and emit the notify signal.
+
+
+
+
+
+
+ %TRUE if @parent could be set or %FALSE when @self
+already had a parent or @self and @parent are the same.
+
+
+
+
+ a #GESTimelineElement
+
+
+
+ new parent of self
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ %TRUE if the self as been rippled properly, %FALSE if an error
+occured
+
+
+
+
+ The #GESTimelineElement to ripple.
+
+
+
+ The new start of @self in ripple mode.
+
+
+
+
+
+
+
+
+ %TRUE if the self as been rippled properly, %FALSE if an error
+occured
+
+
+
+
+ The #GESTimelineElement to ripple.
+
+
+
+ The new end (start + duration) of @self in ripple mode. It will
+ basically only change the duration of @self.
+
+
+
+
+
+
+
+
+ %TRUE if the self as been roll properly, %FALSE if an error
+occured
+
+
+
+
+ The #GESTimelineElement to roll
+
+
+
+ The new start of @self in roll mode, it will also adapat
+the in-point of @self according
+
+
+
+
+
+
+
+
+ %TRUE if the self as been rolled properly, %FALSE if an error
+occured
+
+
+
+
+ The #GESTimelineElement to roll.
+
+
+
+ The new end (start + duration) of @self in roll mode
+
+
+
+
+
+
+
+
+ %TRUE if the self as been trimmed properly, %FALSE if an error
+occured
+
+
+
+
+ The #GESTimelineElement to trim.
+
+
+
+ The new start of @self in trim mode, will adapt the inpoint
+of @self accordingly
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ TRUE if @element and @pspec could be found. FALSE otherwise. In that
+case the values for @pspec and @element are not modified. Unref @element after
+usage.
+
+
+
+
+ object to lookup the property in
+
+
+
+ name of the property to look up. You can specify the name of the
+ class as such: "ClassName::property-name", to guarantee that you get the
+ proper GParamSpec in case various GstElement-s contain the same property
+ name. If you don't do so, you will get the first element found, having
+ this property and the and the corresponding GParamSpec.
+
+
+
+ pointer to a #GstElement that
+ takes the real object to set property on
+
+
+
+ pointer to take the #GParamSpec
+ describing the property
+
+
+
+
+
+
+
+
+
+
+
+
+ A #GESTimelineElement
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Renders the given text in the specified font, at specified position, and
+with the specified background pattern.
+
+
+
+ Creates a new #GESTitleClip
+
+ The newly created #GESTitleClip,
+or %NULL if there was an error.
+
+
+
+
+ Get the background used by @self.
+ use ges_track_element_get/set_children_properties on the
+underlying GESTrackElement instead
+
+ The color used by @self.
+
+
+
+
+ a #GESTitleClip
+
+
+
+
+
+ Get the pango font description used by @self.
+ use ges_track_element_get/set_children_properties on the
+underlying GESTrackElement instead
+
+ The pango font description used by @self.
+
+
+
+
+ a #GESTitleClip
+
+
+
+
+
+ Get the horizontal aligment used by @self.
+ use ges_track_element_get/set_children_properties on the
+underlying GESTrackElement instead
+
+ The horizontal aligment used by @self.
+
+
+
+
+ a #GESTitleClip
+
+
+
+
+
+ Get the text currently set on @self.
+ use ges_track_element_get/set_children_properties on the
+underlying GESTrackElement instead
+
+ The text currently set on @self.
+
+
+
+
+ a #GESTitleClip
+
+
+
+
+
+ Get the color used by @self.
+ use ges_track_element_get/set_children_properties on the
+underlying GESTrackElement instead
+
+ The color used by @self.
+
+
+
+
+ a #GESTitleClip
+
+
+
+
+
+ Get the vertical aligment used by @self.
+ use ges_track_element_get/set_children_properties on the
+underlying GESTrackElement instead
+
+ The vertical aligment used by @self.
+
+
+
+
+ a #GESTitleClip
+
+
+
+
+
+ Get the horizontal position used by @self.
+ use ges_track_element_get/set_children_properties on the
+underlying GESTrackElement instead
+
+ The horizontal position used by @self.
+
+
+
+
+ a #GESTitleClip
+
+
+
+
+
+ Get the vertical position used by @self.
+ use ges_track_element_get/set_children_properties on the
+underlying GESTrackElement instead
+
+ The vertical position used by @self.
+
+
+
+
+ a #GESTitleClip
+
+
+
+
+
+ Sets the background of the text.
+ use ges_track_element_get/set_children_properties on the
+underlying GESTrackElement instead
+
+
+
+
+
+ the #GESTitleClip* to set
+
+
+
+ The color @self is being set to
+
+
+
+
+
+ Sets the color of the text.
+ use ges_track_element_get/set_children_properties on the
+underlying GESTrackElement instead
+
+
+
+
+
+ the #GESTitleClip* to set
+
+
+
+ The color @self is being set to
+
+
+
+
+
+ Sets the pango font description of the text.
+ use ges_track_element_get/set_children_properties on the
+underlying GESTrackElement instead
+
+
+
+
+
+ the #GESTitleClip*
+
+
+
+ the pango font description
+
+
+
+
+
+ Sets the horizontal aligment of the text.
+ use ges_track_element_get/set_children_properties on the
+underlying GESTrackElement instead
+
+
+
+
+
+ the #GESTitleClip* to set horizontal alignement of text on
+
+
+
+ #GESTextHAlign
+
+
+
+
+
+ Sets the text this clip will render.
+ use ges_track_element_get/set_children_properties on the
+underlying GESTrackElement instead
+
+
+
+
+
+ the #GESTitleClip* to set text on
+
+
+
+ the text to render. an internal copy of this text will be
+made.
+
+
+
+
+
+ Sets the vertical aligment of the text.
+ use ges_track_element_get/set_children_properties on the
+underlying GESTrackElement instead
+
+
+
+
+
+ the #GESTitleClip* to set vertical alignement of text on
+
+
+
+ #GESTextVAlign
+
+
+
+
+
+ Sets the horizontal position of the text.
+ use ges_track_element_get/set_children_properties on the
+underlying GESTrackElement instead
+
+
+
+
+
+ the #GESTitleClip* to set
+
+
+
+ The horizontal position @self is being set to
+
+
+
+
+
+ Sets the vertical position of the text.
+ use ges_track_element_get/set_children_properties on the
+underlying GESTrackElement instead
+
+
+
+
+
+ the #GESTitleClip* to set
+
+
+
+ The vertical position @self is being set to
+
+
+
+
+
+ The background of the text
+ use ges_track_element_get/set_children_properties on the
+underlying GESTrackElement instead
+
+
+
+ The color of the text
+ use ges_track_element_get/set_children_properties on the
+underlying GESTrackElement instead
+
+
+
+ Pango font description string
+ use ges_track_element_get/set_children_properties on the
+underlying GESTrackElement instead
+
+
+
+ Horizontal alignment of the text
+ use ges_track_element_get/set_children_properties on the
+underlying GESTrackElement instead
+
+
+
+ The text to diplay
+ use ges_track_element_get/set_children_properties on the
+underlying GESTrackElement instead
+
+
+
+ Vertical alignent of the text
+ use ges_track_element_get/set_children_properties on the
+underlying GESTrackElement instead
+
+
+
+ The horizontal position of the text
+ use ges_track_element_get/set_children_properties on the
+underlying GESTrackElement instead
+
+
+
+ The vertical position of the text
+ use ges_track_element_get/set_children_properties on the
+underlying GESTrackElement instead
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ #GESTitleSource is a GESTimelineElement that implements the notion
+of titles in GES.
+
+## Children Properties
+
+You can use the following children properties through the
+#ges_track_element_set_child_property and alike set of methods:
+<informaltable frame="none">
+<tgroup cols="3">
+<colspec colname="properties_type" colwidth="150px"/>
+<colspec colname="properties_name" colwidth="200px"/>
+<colspec colname="properties_flags" colwidth="400px"/>
+<tbody>
+<row>
+ <entry role="property_type"><link linkend="guint"><type>guint</type></link></entry>
+ <entry role="property_name"><link linkend="GESTileSource--background">background</link></entry>
+ <entry>The color of the background</entry>
+</row>
+<row>
+ <entry role="property_type"><link linkend="guint"><type>guint</type></link></entry>
+ <entry role="property_name"><link linkend="GESTileSource--color">color</link></entry>
+ <entry>The color of the text</entry>
+</row>
+<row>
+ <entry role="property_type"><link linkend="gchar"><type>gchar</type></link></entry>
+ <entry role="property_name"><link linkend="GESTileSource--font-desc">font-desc</link></entry>
+ <entry>Pango font description string</entry>
+</row>
+<row>
+ <entry role="property_type"><link linkend="GESTextHAlign"><type>GESTextHAlign</type></link></entry>
+ <entry role="property_name"><link linkend="GESTileSource--halignment">halignment</link></entry>
+ <entry>Horizontal alignment of the text</entry>
+</row>
+<row>
+ <entry role="property_type"><link linkend="gchar"><type>gchar</type></link></entry>
+ <entry role="property_name"><link linkend="GESTileSource--text">text</link></entry>
+ <entry>The text to be rendered</entry>
+</row>
+<row>
+ <entry role="property_type"><link linkend="GESTextVAlign"><type>GESTextVAlign</type></link>
+ </entry><entry role="property_name"><link linkend="GESTileSource--valignment">valignment</link>
+ </entry><entry>Vertical alignent of the text</entry>
+</row>
+<row>
+ <entry role="property_type"><link linkend="gdouble"><type>gdouble</type></link></entry>
+ <entry role="property_name"><link linkend="GESTileSource--xpos">xpos</link></entry>
+ <entry>The horizontal position of the text</entry>
+</row>
+<row><entry role="property_type"><link linkend="gdouble"><type>gdouble</type></link></entry>
+ <entry role="property_name"><link linkend="GESTileSource--ypos">ypos</link></entry>
+ <entry>The vertical position of the text</entry>
+</row>
+<row><entry role="property_type"><link linkend="gboolean"><type>gboolean</type></link></entry>
+ <entry role="property_name"><link linkend="GESTileSource--shaded-background">shaded-background</link></entry>
+ <entry>Whether to shade the background under the text area</entry>
+</row>
+<row><entry role="property_type"><link linkend="guint"><type>guint</type></link></entry>
+ <entry role="property_name"><link linkend="GESTileSource--outline-color">outline-color</link></entry>
+ <entry>Color to use for outline the text (big-endian ARGB).</entry>
+</row>
+</tbody>
+</tgroup>
+</informaltable>
+
+
+
+ Get the background used by @source.
+
+ The background used by @source.
+
+
+
+
+ a #GESTitleSource
+
+
+
+
+
+ Get the pango font description used by @source.
+
+ The pango font description used by this
+@source.
+
+
+
+
+ a #GESTitleSource
+
+
+
+
+
+ Get the horizontal aligment used by @source.
+
+ The horizontal aligment used by @source.
+
+
+
+
+ a #GESTitleSource
+
+
+
+
+
+ Get the text currently set on the @source.
+
+ The text currently set on the @source.
+
+
+
+
+ a #GESTitleSource
+
+
+
+
+
+ Get the color used by @source.
+
+ The color used by @source.
+
+
+
+
+ a #GESTitleSource
+
+
+
+
+
+ Get the vertical aligment used by @source.
+
+ The vertical aligment used by @source.
+
+
+
+
+ a #GESTitleSource
+
+
+
+
+
+ Get the horizontal position used by @source.
+
+ The horizontal position used by @source.
+
+
+
+
+ a #GESTitleSource
+
+
+
+
+
+ Get the vertical position used by @source.
+
+ The vertical position used by @source.
+
+
+
+
+ a #GESTitleSource
+
+
+
+
+
+ Sets the color of the background
+
+
+
+
+
+ the #GESTitleSource* to set
+
+
+
+ the color @self is being set to
+
+
+
+
+
+ Set the pango font description this source will use to render
+the text.
+
+
+
+
+
+ the #GESTitleSource
+
+
+
+ the pango font description
+
+
+
+
+
+ Sets the vertical aligment of the text.
+
+
+
+
+
+ the #GESTitleSource* to set text on
+
+
+
+ #GESTextHAlign
+
+
+
+
+
+ Sets the text this track element will render.
+ use ges_track_element_get/set_children_properties on the
+GESTrackElement instead
+
+
+
+
+
+ the #GESTitleSource* to set text on
+
+
+
+ the text to render. an internal copy of this text will be
+made.
+
+
+
+
+
+ Sets the color of the text.
+
+
+
+
+
+ the #GESTitleSource* to set
+
+
+
+ the color @self is being set to
+
+
+
+
+
+ Sets the vertical aligment of the text.
+
+
+
+
+
+ the #GESTitleSource* to set text on
+
+
+
+ #GESTextVAlign
+
+
+
+
+
+ Sets the horizontal position of the text.
+
+
+
+
+
+ the #GESTitleSource* to set
+
+
+
+ the horizontal position @self is being set to
+
+
+
+
+
+ Sets the vertical position of the text.
+
+
+
+
+
+ the #GESTitleSource* to set
+
+
+
+ the color @self is being set to
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ parent class
+
+
+
+
+
+
+
+
+
+
+
+ Corresponds to one output format (i.e. audio OR video).
+
+Contains the compatible TrackElement(s).
+
+
+
+ Creates a new #GESTrack with the given @type and @caps.
+
+The newly created track will steal a reference to the caps. If you wish to
+use those caps elsewhere, you will have to take an extra reference.
+
+ A new #GESTrack.
+
+
+
+
+ The type of track
+
+
+
+ The caps to restrict the output of the track to.
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Adds the given object to the track. Sets the object's controlling track,
+and thus takes ownership of the @object.
+
+An object can only be added to one track.
+
+ #TRUE if the object was properly added. #FALSE if the track does not
+want to accept the object.
+
+
+
+
+ a #GESTrack
+
+
+
+ the #GESTrackElement to add
+
+
+
+
+
+ Commits all the pending changes of the TrackElement contained in the
+track.
+
+When timing changes happen in a timeline, the changes are not
+directly done inside NLE. This method needs to be called so any changes
+on a clip contained in the timeline actually happen at the media
+processing level.
+
+ %TRUE if something as been commited %FALSE if nothing needed
+to be commited
+
+
+
+
+ a #GESTrack
+
+
+
+
+
+ Get the #GstCaps this track is configured to output.
+
+ The #GstCaps this track is configured to output.
+
+
+
+
+ a #GESTrack
+
+
+
+
+
+ Gets the #GESTrackElement contained in @track
+
+ the list of
+#GESTrackElement present in the Track sorted by priority and start.
+
+
+
+
+
+
+ a #GESTrack
+
+
+
+
+
+ Gets if the underlying #NleComposition contains an expandable mixer.
+
+ #True if there is a mixer, #False otherwise.
+
+
+
+
+ a #GESTrack
+
+
+
+
+
+ Get the #GESTimeline this track belongs to. Can be %NULL.
+
+ The #GESTimeline this track belongs to. Can be %NULL.
+
+
+
+
+ a #GESTrack
+
+
+
+
+
+ Removes the object from the track and unparents it.
+Unparenting it means the reference owned by @track on the @object will be
+removed. If you wish to use the @object after this function, make sure you
+call gst_object_ref() before removing it from the @track.
+
+ #TRUE if the object was removed, else #FALSE if the track
+could not remove the object (like if it didn't belong to the track).
+
+
+
+
+ a #GESTrack
+
+
+
+ the #GESTrackElement to remove
+
+
+
+
+
+ Sets the function that should be used to create the GstElement used to fill gaps.
+To avoid to provide such a function we advice you to use the
+#ges_audio_track_new and #ges_video_track_new constructor when possible.
+
+
+
+
+
+ a #GESTrack
+
+
+
+ The #GESCreateElementForGapFunc that will be used
+to create #GstElement to fill gaps
+
+
+
+
+
+ Sets if the #GESTrack should be mixing.
+
+
+
+
+
+ a #GESTrack
+
+
+
+ TRUE if the track should be mixing, FALSE otherwise.
+
+
+
+
+
+ Sets the given @caps as the caps the track has to output.
+
+
+
+
+
+ a #GESTrack
+
+
+
+ the #GstCaps to set
+
+
+
+
+
+ Sets @timeline as the timeline controlling @track.
+
+
+
+
+
+ a #GESTrack
+
+
+
+ a #GESTimeline
+
+
+
+
+
+ Updates the restriction caps by modifying all the fields present in @caps
+in the original restriction caps. If for example the current restriction caps
+are video/x-raw, format=I420, width=360 and @caps is video/x-raw, format=RGB,
+the restriction caps will be updated to video/x-raw, format=RGB, width=360.
+
+Modification happens for each structure in the new caps, and
+one can add new fields or structures through that function.
+
+
+
+
+
+ a #GESTrack
+
+
+
+ the #GstCaps to update with
+
+
+
+
+
+ Caps used to filter/choose the output stream. This is generally set to
+a generic set of caps like 'video/x-raw' for raw video.
+
+Default value: #GST_CAPS_ANY.
+
+
+
+ Current duration of the track
+
+Default value: O
+
+
+
+ Whether layer mixing is activated or not on the track.
+
+
+
+ Caps used to filter/choose the output stream.
+
+Default value: #GST_CAPS_ANY.
+
+
+
+ Type of stream the track outputs. This is used when creating the #GESTrack
+to specify in generic terms what type of content will be outputted.
+
+It also serves as a 'fast' way to check what type of data will be outputted
+from the #GESTrack without having to actually check the #GESTrack's caps
+property.
+
+
+
+
+
+
+ a #GESTrackType indicting the basic type of the track.
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Will be emitted after a track element was added to the track.
+
+
+
+
+
+ the #GESTrackElement that was added.
+
+
+
+
+
+ Will be emitted after a track element was removed from the track.
+
+
+
+
+
+ the #GESTrackElement that was removed.
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ #GESTrackElement is the Base Class for any object that can be contained in a
+#GESTrack.
+
+It contains the basic information as to the location of the object within
+its container, like the start position, the inpoint, the duration and the
+priority.
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Looks up which @element and @pspec would be effected by the given @name. If various
+contained elements have this property name you will get the first one, unless you
+specify the class name in @name.
+ Use #ges_timeline_element_lookup_child
+
+ TRUE if @element and @pspec could be found. FALSE otherwise. In that
+case the values for @pspec and @element are not modified. Unref @element after
+usage.
+
+
+
+
+ object to lookup the property in
+
+
+
+ name of the property to look up. You can specify the name of the
+ class as such: "ClassName::property-name", to guarantee that you get the
+ proper GParamSpec in case various GstElement-s contain the same property
+ name. If you don't do so, you will get the first element found, having
+ this property and the and the corresponding GParamSpec.
+
+
+
+ pointer to a #GstElement that
+ takes the real object to set property on
+
+
+
+ pointer to take the #GParamSpec
+ describing the property
+
+
+
+
+
+ Looks for the properties defines with the various parametters and add
+them to the hashtable of children properties.
+
+To be used by subclasses only
+
+
+
+
+
+ The #GESTrackElement to set chidlren props on
+
+
+
+ The GstElement to retrieve properties from
+
+
+
+
+An array of categories of GstElement to
+take into account (as defined in the factory meta "klass" field)
+
+
+
+
+
+ A
+blacklist of elements factory names to not take into account
+
+
+
+
+
+ A list
+of propery names to add as children properties
+
+
+
+
+
+
+
+ Edit @object in the different exisiting #GESEditMode modes. In the case of
+slide, and roll, you need to specify a #GESEdge
+
+ %TRUE if the object as been edited properly, %FALSE if an error
+occured
+
+
+
+
+ the #GESTrackElement to edit
+
+
+
+ The layers you want the edit to
+ happen in, %NULL means that the edition is done in all the
+ #GESLayers contained in the current timeline.
+ FIXME: This is not implemented yet.
+
+
+
+
+
+ The #GESEditMode in which the edition will happen.
+
+
+
+ The #GESEdge the edit should happen on.
+
+
+
+ The position at which to edit @object (in nanosecond)
+
+
+
+
+
+
+ A
+#GHashTable containing all property_name: GstControlBinding
+
+
+
+
+
+
+
+ The #TrackElement from which to get all set bindings
+
+
+
+
+
+ Gets properties of a child of @object.
+ Use #ges_timeline_element_get_child_properties
+
+
+
+
+
+ The origin #GESTrackElement
+
+
+
+ The name of the first property to get
+
+
+
+ return location for the first property, followed optionally by more
+name/return location pairs, followed by NULL
+
+
+
+
+
+ In general, a copy is made of the property contents and
+the caller is responsible for freeing the memory by calling
+g_value_unset().
+
+Gets a property of a GstElement contained in @object.
+
+Note that #ges_track_element_get_child_property is really
+intended for language bindings, #ges_track_element_get_child_properties
+is much more convenient for C programming.
+ Use #ges_timeline_element_get_child_property
+
+ %TRUE if the property was found, %FALSE otherwize
+
+
+
+
+ The origin #GESTrackElement
+
+
+
+ The name of the property
+
+
+
+ return location for the property value, it will
+be initialized if it is initialized with 0
+
+
+
+
+
+ Gets a property of a child of @object.
+ Use #ges_timeline_element_get_child_property_by_pspec
+
+
+
+
+
+ a #GESTrackElement
+
+
+
+ The #GParamSpec that specifies the property you want to get
+
+
+
+ return location for the value
+
+
+
+
+
+ Gets a property of a child of @object. If there are various child elements
+that have the same property name, you can distinguish them using the following
+syntax: 'ClasseName::property_name' as property name. If you don't, the
+corresponding property of the first element found will be set.
+ Use #ges_timeline_element_get_child_property_valist
+
+
+
+
+
+ The #GESTrackElement parent object
+
+
+
+ The name of the first property to get
+
+
+
+ value for the first property, followed optionally by more
+name/return location pairs, followed by NULL
+
+
+
+
+
+ Looks up the various controlled properties for that #GESTrackElement,
+and returns the #GstControlBinding which controls @property_name.
+
+ the #GstControlBinding associated with
+@property_name, or %NULL if that property is not controlled.
+
+
+
+
+ the #GESTrackElement in which to lookup the bindings.
+
+
+
+ The property_name to which the binding is associated.
+
+
+
+
+
+ Get the #GstElement this track element is controlling within GNonLin.
+
+ the #GstElement this track element is controlling
+within GNonLin.
+
+
+
+
+ a #GESTrackElement
+
+
+
+
+
+ Get the NleObject object this object is controlling.
+ use #ges_track_element_get_nleobject instead.
+
+ the NleObject object this object is controlling.
+
+
+
+
+ a #GESTrackElement
+
+
+
+
+
+ Get the GNonLin object this object is controlling.
+
+ the GNonLin object this object is controlling.
+
+
+
+
+ a #GESTrackElement
+
+
+
+
+
+ Get the #GESTrack to which this object belongs.
+
+ The #GESTrack to which this object
+belongs. Can be %NULL if it is not in any track
+
+
+
+
+ a #GESTrackElement
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Lets you know if @object will be used for playback and rendering,
+or not.
+
+ %TRUE if @object is active, %FALSE otherwize
+
+
+
+
+ a #GESTrackElement
+
+
+
+
+
+ Gets an array of #GParamSpec* for all configurable properties of the
+children of @object.
+ Use #ges_timeline_element_list_children_properties
+
+ an array of #GParamSpec* which should be freed after use or
+%NULL if something went wrong
+
+
+
+
+
+
+ The #GESTrackElement to get the list of children properties from
+
+
+
+ return location for the length of the returned array
+
+
+
+
+
+ Looks up which @element and @pspec would be effected by the given @name. If various
+contained elements have this property name you will get the first one, unless you
+specify the class name in @name.
+ Use #ges_timeline_element_lookup_child
+
+ TRUE if @element and @pspec could be found. FALSE otherwise. In that
+case the values for @pspec and @element are not modified. Unref @element after
+usage.
+
+
+
+
+ object to lookup the property in
+
+
+
+ name of the property to look up. You can specify the name of the
+ class as such: "ClassName::property-name", to guarantee that you get the
+ proper GParamSpec in case various GstElement-s contain the same property
+ name. If you don't do so, you will get the first element found, having
+ this property and the and the corresponding GParamSpec.
+
+
+
+ pointer to a #GstElement that
+ takes the real object to set property on
+
+
+
+ pointer to take the #GParamSpec
+ describing the property
+
+
+
+
+
+ Removes a #GstControlBinding from @object.
+
+ %TRUE if the binding could be removed, %FALSE if an error
+occured
+
+
+
+
+ the #GESTrackElement on which to set a control binding
+
+
+
+ The name of the property to control.
+
+
+
+
+
+ Sets the usage of the @object. If @active is %TRUE, the object will be used for
+playback and rendering, else it will be ignored.
+
+ %TRUE if the property was toggled, else %FALSE
+
+
+
+
+ a #GESTrackElement
+
+
+
+ visibility
+
+
+
+
+
+ Sets a property of a child of @object. If there are various child elements
+that have the same property name, you can distinguish them using the following
+syntax: 'ClasseName::property_name' as property name. If you don't, the
+corresponding property of the first element found will be set.
+ Use #ges_timeline_element_set_child_properties
+
+
+
+
+
+ The #GESTrackElement parent object
+
+
+
+ The name of the first property to set
+
+
+
+ value for the first property, followed optionally by more
+name/return location pairs, followed by NULL
+
+
+
+
+
+ Sets a property of a GstElement contained in @object.
+
+Note that #ges_track_element_set_child_property is really
+intended for language bindings, #ges_track_element_set_child_properties
+is much more convenient for C programming.
+ use #ges_timeline_element_set_child_property instead
+
+ %TRUE if the property was set, %FALSE otherwize
+
+
+
+
+ The origin #GESTrackElement
+
+
+
+ The name of the property
+
+
+
+ the value
+
+
+
+
+
+ Sets a property of a child of @object.
+ Use #ges_timeline_element_set_child_property_by_spec
+
+
+
+
+
+ a #GESTrackElement
+
+
+
+ The #GParamSpec that specifies the property you want to set
+
+
+
+ the value
+
+
+
+
+
+ Sets a property of a child of @object. If there are various child elements
+that have the same property name, you can distinguish them using the following
+syntax: 'ClasseName::property_name' as property name. If you don't, the
+corresponding property of the first element found will be set.
+ Use #ges_timeline_element_set_child_property_valist
+
+
+
+
+
+ The #GESTrackElement parent object
+
+
+
+ The name of the first property to set
+
+
+
+ value for the first property, followed optionally by more
+name/return location pairs, followed by NULL
+
+
+
+
+
+ Creates a #GstControlBinding and adds it to the #GstElement concerned by the
+property. Use the same syntax as #ges_track_element_lookup_child for
+the property name.
+
+ %TRUE if the binding could be created and added, %FALSE if an error
+occured
+
+
+
+
+ the #GESTrackElement on which to set a control binding
+
+
+
+ the #GstControlSource to set on the binding.
+
+
+
+ The name of the property to control.
+
+
+
+ The type of binding to create. Only "direct" is available for now.
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Whether the object should be taken into account in the #GESTrack output.
+If #FALSE, then its contents will not be used in the resulting track.
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ The control-binding-added signal is emitted each time a control binding
+is added for a child property of @track_element
+
+
+
+
+
+ the #GstControlBinding that has been added
+
+
+
+
+
+ The control-binding-removed signal is emitted each time a control binding
+is removed for a child property of @track_element
+
+
+
+
+
+ the #GstControlBinding that has been removed
+
+
+
+
+
+
+
+
+
+
+ Get the GESAssetTrackType the #GESTrackElement extracted from @self
+should get into
+
+ a #GESTrackType
+
+
+
+
+ A #GESAssetObject
+
+
+
+
+
+ Set the #GESAssetTrackType the #GESTrackElement extracted from @self
+should get into
+
+
+
+
+
+ A #GESAssetObject
+
+
+
+ A #GESTrackType
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Subclasses can override the @create_gnl_object method to override what type
+of GNonLin object will be created.
+
+
+
+
+ name of the GNonLin GStElementFactory type to use.
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ TRUE if @element and @pspec could be found. FALSE otherwise. In that
+case the values for @pspec and @element are not modified. Unref @element after
+usage.
+
+
+
+
+ object to lookup the property in
+
+
+
+ name of the property to look up. You can specify the name of the
+ class as such: "ClassName::property-name", to guarantee that you get the
+ proper GParamSpec in case various GstElement-s contain the same property
+ name. If you don't do so, you will get the first element found, having
+ this property and the and the corresponding GParamSpec.
+
+
+
+ pointer to a #GstElement that
+ takes the real object to set property on
+
+
+
+ pointer to take the #GParamSpec
+ describing the property
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Types of content handled by a track. If the content is not one of
+@GES_TRACK_TYPE_AUDIO, @GES_TRACK_TYPE_VIDEO or @GES_TRACK_TYPE_TEXT,
+the user of the #GESTrack must set the type to @GES_TRACK_TYPE_CUSTOM.
+
+@GES_TRACK_TYPE_UNKNOWN is for internal purposes and should not be used
+by users
+
+ A track of unknown type (i.e. invalid)
+
+
+ An audio track
+
+
+ A video track
+
+
+ A text (subtitle) track
+
+
+ A custom-content track
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Base class for media transitions.
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Creates an object that mixes together the two underlying objects, A and B.
+The A object is assumed to have a higher prioirity (lower number) than the
+B object. At the transition in point, only A will be visible, and by the
+end only B will be visible.
+
+The shape of the video transition depends on the value of the "vtype"
+property. The default value is "crossfade". For audio, only "crossfade" is
+supported.
+
+The ID of the ExtractableType is the nickname of the vtype property value. Note
+that this value can be changed after creation and the GESExtractable.asset value
+will be updated when needed.
+
+
+
+ Creates a new #GESTransitionClip.
+
+ a newly created #GESTransitionClip,
+or %NULL if something went wrong.
+
+
+
+
+ the type of transition to create
+
+
+
+
+
+ Creates a new #GESTransitionClip for the provided @nick.
+
+ The newly created #GESTransitionClip,
+or %NULL if something went wrong
+
+
+
+
+ a string representing the type of transition to create
+
+
+
+
+
+ a #GESVideoStandardTransitionType representing the wipe to use
+
+
+
+
+
+
+ a #GESVideoStandardTransitionType indicating the type of video transition
+to apply.
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Represents all the output streams from a particular uri. It is assumed that
+the URI points to a file of some type.
+
+
+
+ Creates a new #GESUriClip for the provided @uri.
+
+ The newly created #GESUriClip, or
+%NULL if there was an error.
+
+
+
+
+ the URI the source should control
+
+
+
+
+
+ Get the location of the resource.
+
+ The location of the resource.
+
+
+
+
+ the #GESUriClip
+
+
+
+
+
+ Lets you know if @self is an image or not.
+
+ %TRUE if @self is a still image %FALSE otherwise.
+
+
+
+
+ the #GESUriClip
+
+
+
+
+
+ Lets you know if the audio track of @self is muted or not.
+
+ %TRUE if the audio track of @self is muted, %FALSE otherwise.
+
+
+
+
+ the #GESUriClip
+
+
+
+
+
+ Sets whether the clip is a still image or not.
+
+
+
+
+
+ the #GESUriClip
+
+
+
+ %TRUE if @self is a still image, %FALSE otherwise
+
+
+
+
+
+ Sets whether the audio track of this clip is muted or not.
+
+
+
+
+
+ the #GESUriClip on which to mute or unmute the audio track
+
+
+
+ %TRUE to mute @self audio track, %FALSE to unmute it
+
+
+
+
+
+ Whether this uri clip represents a still image or not. This must be set
+before create_track_elements is called.
+
+
+
+ Whether the sound will be played or not.
+
+
+
+
+
+
+ The location of the file/resource to use.
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ The #GESUriClipAsset is a special #GESAsset that lets you handle
+the media file to use inside the GStreamer Editing Services. It has APIs that
+let you get information about the medias. Also, the tags found in the media file are
+set as Metadatas of the Asser.
+
+
+
+
+ Creates a #GESUriClipAsset for @uri
+
+Example of request of a GESUriClipAsset:
+|[
+// The request callback
+static void
+filesource_asset_loaded_cb (GESAsset * source, GAsyncResult * res, gpointer user_data)
+{
+ GError *error = NULL;
+ GESUriClipAsset *filesource_asset;
+
+ filesource_asset = GES_URI_CLIP_ASSET (ges_asset_request_finish (res, &error));
+ if (filesource_asset) {
+ g_print ("The file: %s is usable as a FileSource, it is%s an image and lasts %" GST_TIME_FORMAT,
+ ges_asset_get_id (GES_ASSET (filesource_asset))
+ ges_uri_clip_asset_is_image (filesource_asset) ? "" : " not",
+ GST_TIME_ARGS (ges_uri_clip_asset_get_duration (filesource_asset));
+ } else {
+ g_print ("The file: %s is *not* usable as a FileSource because: %s",
+ ges_asset_get_id (source), error->message);
+ }
+
+ gst_object_unref (mfs);
+}
+
+// The request:
+ges_uri_clip_asset_new (uri, (GAsyncReadyCallback) filesource_asset_loaded_cb, user_data);
+]|
+
+
+
+
+
+ The URI of the file for which to create a #GESUriClipAsset
+
+
+
+ optional %GCancellable object, %NULL to ignore.
+
+
+
+ a #GAsyncReadyCallback to call when the initialization is finished
+
+
+
+ The user data to pass when @callback is called
+
+
+
+
+
+ Creates a #GESUriClipAsset for @uri syncronously. You should avoid
+to use it in application, and rather create #GESUriClipAsset asynchronously
+
+ A reference to the requested asset or
+%NULL if an error happened
+
+
+
+
+ The URI of the file for which to create a #GESUriClipAsset.
+You can also use multi file uris for #GESMultiFileSource.
+
+
+
+
+
+ Gets duration of the file represented by @self
+
+ The duration of @self
+
+
+
+
+ a #GESUriClipAsset
+
+
+
+
+
+ Gets #GstDiscovererInfo about the file
+
+ #GstDiscovererInfo of specified asset
+
+
+
+
+ Target asset
+
+
+
+
+
+ Get the GESUriSourceAsset @self containes
+
+ a
+#GList of #GESUriSourceAsset
+
+
+
+
+
+
+ A #GESUriClipAsset
+
+
+
+
+
+ Gets Whether the file represented by @self is an image or not
+
+ Whether the file represented by @self is an image or not
+
+
+
+
+ a #indent: Standard input:311: Error:Unexpected end of file
+GESUriClipAsset
+
+
+
+
+
+ The duration (in nanoseconds) of the media file
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Sets the timeout of #GESUriClipAsset loading
+
+
+
+
+
+ The #GESUriClipAssetClass on which to set the discoverer timeout
+
+
+
+ The timeout to set
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ NOTE: You should never request such a #GESAsset as they will be created automatically
+by #GESUriClipAsset-s.
+
+
+
+
+ Get the #GESUriClipAsset @self is contained in
+
+ a #GESUriClipAsset
+
+
+
+
+ A #GESUriClipAsset
+
+
+
+
+
+ Get the #GstDiscovererStreamInfo user by @asset
+
+ a #GESUriClipAsset
+
+
+
+
+ A #GESUriClipAsset
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ # Children Properties:
+You can use the following children properties through the
+#ges_track_element_set_child_property and alike set of methods:
+
+<informaltable frame="none">
+<tgroup cols="3">
+<colspec colname="properties_type" colwidth="150px"/>
+<colspec colname="properties_name" colwidth="200px"/>
+<colspec colname="properties_flags" colwidth="400px"/>
+<tbody>
+<row>
+ <entry role="property_type"><link linkend="gdouble"><type>double</type></link></entry>
+ <entry role="property_name"><link linkend="GESVideoSource--alpha">alpha</link></entry>
+ <entry>The desired alpha for the stream.</entry>
+</row>
+<row>
+ <entry role="property_type"><link linkend="gint"><type>gint</type></link></entry>
+ <entry role="property_name"><link linkend="GESVideoSource--posx">posx</link></entry>
+ <entry>The desired x position for the stream.</entry>
+</row>
+<row>
+ <entry role="property_type"><link linkend="gint"><type>gint</type></link></entry>
+ <entry role="property_name"><link linkend="GESVideoSource--posy">posy</link></entry>
+ <entry>The desired y position for the stream</entry>
+</row>
+<row>
+ <entry role="property_type"><link linkend="gint"><type>gint</type></link></entry>
+ <entry role="property_name"><link linkend="GESVideoSource--width">width</link></entry>
+ <entry>The desired width for that source. Set to 0 if size is not mandatory, will be set to width of the current track.</entry>
+</row>
+<row>
+ <entry role="property_type"><link linkend="gint"><type>gint</type></link></entry>
+ <entry role="property_name"><link linkend="GESVideoSource--height">height</link></entry>
+ <entry>The desired height for that source. Set to 0 if size is not mandatory, will be set to height of the current track.</entry>
+</row>
+<row>
+ <entry role="property_type"><link linkend="GstDeinterlaceModes"><type>GstDeinterlaceModes</type></link></entry>
+ <entry role="property_name"><link linkend="GESVideoSource--deinterlace-mode">deinterlace-mode</link></entry>
+ <entry>Deinterlace Mode</entry>
+</row>
+<row>
+ <entry role="property_type"><link linkend="GstDeinterlaceFields"><type>GstDeinterlaceFields</type></link></entry>
+ <entry role="property_name"><link linkend="GESVideoSource--deinterlace-fields">deinterlace-fields</link></entry>
+ <entry>Fields to use for deinterlacing</entry>
+</row>
+<row>
+ <entry role="property_type"><link linkend="GstDeinterlaceFieldLayout"><type>GstDeinterlaceFieldLayout</type></link></entry>
+ <entry role="property_name"><link linkend="GESVideoSource--deinterlace-tff">deinterlace-tff</link></entry>
+ <entry>Deinterlace top field first</entry>
+</row>
+</tbody>
+</tgroup>
+</informaltable>
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Transition type has not been set,
+
+
+ A bar moves from left to right,
+
+
+ A bar moves from top to bottom,
+
+
+ A box expands from the upper-left corner to the lower-right corner,
+
+
+ A box expands from the upper-right corner to the lower-left corner,
+
+
+ A box expands from the lower-right corner to the upper-left corner,
+
+
+ A box expands from the lower-left corner to the upper-right corner,
+
+
+ A box shape expands from each of the four corners toward the center,
+
+
+ A box shape expands from the center of each quadrant toward the corners of each quadrant,
+
+
+ A central, vertical line splits and expands toward the left and right edges,
+
+
+ A central, horizontal line splits and expands toward the top and bottom edges,
+
+
+ A box expands from the top edge's midpoint to the bottom corners,
+
+
+ A box expands from the right edge's midpoint to the left corners,
+
+
+ A box expands from the bottom edge's midpoint to the top corners,
+
+
+ A box expands from the left edge's midpoint to the right corners,
+
+
+ A diagonal line moves from the upper-left corner to the lower-right corner,
+
+
+ A diagonal line moves from the upper right corner to the lower-left corner,
+
+
+ Two wedge shapes slide in from the top and bottom edges toward the center,
+
+
+ Two wedge shapes slide in from the left and right edges toward the center,
+
+
+ A diagonal line from the lower-left to upper-right corners splits and expands toward the opposite corners,
+
+
+ A diagonal line from upper-left to lower-right corners splits and expands toward the opposite corners,
+
+
+ Four wedge shapes split from the center and retract toward the four edges,
+
+
+ A diamond connecting the four edge midpoints simultaneously contracts toward the center and expands toward the edges,
+
+
+ A wedge shape moves from top to bottom,
+
+
+ A wedge shape moves from right to left,
+
+
+ A wedge shape moves from bottom to top,
+
+
+ A wedge shape moves from left to right,
+
+
+ A 'V' shape extending from the bottom edge's midpoint to the opposite corners contracts toward the center and expands toward the edges,
+
+
+ A 'V' shape extending from the left edge's midpoint to the opposite corners contracts toward the center and expands toward the edges,
+
+
+ A 'V' shape extending from the top edge's midpoint to the opposite corners contracts toward the center and expands toward the edges,
+
+
+ A 'V' shape extending from the right edge's midpoint to the opposite corners contracts toward the center and expands toward the edges,
+
+
+ A rectangle expands from the center.,
+
+
+ A radial hand sweeps clockwise from the twelve o'clock position,
+
+
+ A radial hand sweeps clockwise from the three o'clock position,
+
+
+ A radial hand sweeps clockwise from the six o'clock position,
+
+
+ A radial hand sweeps clockwise from the nine o'clock position,
+
+
+ Two radial hands sweep clockwise from the twelve and six o'clock positions,
+
+
+ Two radial hands sweep clockwise from the nine and three o'clock positions,
+
+
+ Four radial hands sweep clockwise,
+
+
+ A fan unfolds from the top edge, the fan axis at the center,
+
+
+ A fan unfolds from the right edge, the fan axis at the center,
+
+
+ Two fans, their axes at the center, unfold from the top and bottom,
+
+
+ Two fans, their axes at the center, unfold from the left and right,
+
+
+ A radial hand sweeps clockwise from the top edge's midpoint,
+
+
+ A radial hand sweeps clockwise from the right edge's midpoint,
+
+
+ A radial hand sweeps clockwise from the bottom edge's midpoint,
+
+
+ A radial hand sweeps clockwise from the left edge's midpoint,
+
+
+ Two radial hands sweep clockwise and counter-clockwise from the top and bottom edges' midpoints,
+
+
+ Two radial hands sweep clockwise and counter-clockwise from the left and right edges' midpoints,
+
+
+ Two radial hands attached at the top and bottom edges' midpoints sweep from right to left,
+
+
+ Two radial hands attached at the left and right edges' midpoints sweep from top to bottom,
+
+
+ A fan unfolds from the bottom, the fan axis at the top edge's midpoint,
+
+
+ A fan unfolds from the left, the fan axis at the right edge's midpoint,
+
+
+ A fan unfolds from the top, the fan axis at the bottom edge's midpoint,
+
+
+ A fan unfolds from the right, the fan axis at the left edge's midpoint,
+
+
+ Two fans, their axes at the top and bottom, unfold from the center,
+
+
+ Two fans, their axes at the left and right, unfold from the center,
+
+
+ A radial hand sweeps clockwise from the upper-left corner,
+
+
+ A radial hand sweeps counter-clockwise from the lower-left corner.,
+
+
+ A radial hand sweeps clockwise from the lower-right corner,
+
+
+ A radial hand sweeps counter-clockwise from the upper-right corner,
+
+
+ Two radial hands attached at the upper-left and lower-right corners sweep down and up,
+
+
+ Two radial hands attached at the lower-left and upper-right corners sweep down and up,
+
+
+ Two radial hands attached at the upper-left and upper-right corners sweep down,
+
+
+ Two radial hands attached at the upper-left and lower-left corners sweep to the right,
+
+
+ Two radial hands attached at the lower-left and lower-right corners sweep up,
+
+
+ Two radial hands attached at the upper-right and lower-right corners sweep to the left,
+
+
+ Two radial hands attached at the midpoints of the top and bottom halves sweep from right to left,
+
+
+ Two radial hands attached at the midpoints of the left and right halves sweep from top to bottom,
+
+
+ Two sets of radial hands attached at the midpoints of the top and bottom halves sweep from top to bottom and bottom to top,
+
+
+ Two sets of radial hands attached at the midpoints of the left and right halves sweep from left to right and right to left,
+
+
+ Crossfade
+
+
+
+ The test pattern to produce
+
+ A standard SMPTE test pattern
+
+
+ Random noise
+
+
+ A black image
+
+
+ A white image
+
+
+ A red image
+
+
+ A green image
+
+
+ A blue image
+
+
+ Checkers pattern (1px)
+
+
+ Checkers pattern (2px)
+
+
+ Checkers pattern (4px)
+
+
+ Checkers pattern (8px)
+
+
+ Circular pattern
+
+
+ Alternate between black and white
+
+
+ SMPTE test pattern (75% color bars)
+
+
+ Zone plate
+
+
+ Gamut checkers
+
+
+ Chroma zone plate
+
+
+ Solid color
+
+
+
+
+
+
+ Get the video pattern used by the @source.
+
+ The video pattern used by the @source.
+
+
+
+
+ a #GESVideoTestPattern
+
+
+
+
+
+ Sets the source to use the given @pattern.
+
+
+
+
+
+ a #GESVideoTestSource
+
+
+
+ a #GESVideoTestPattern
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Creates a new #GESVideoTrack of type #GES_TRACK_TYPE_VIDEO and with generic
+raw video caps ("video/x-raw");
+
+ A new #GESTrack.
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Creates a new #GESVideoTransition.
+
+ The newly created
+#GESVideoTransition, or %NULL if there was an error.
+
+
+
+
+ Get the border property of @self, this value represents
+the border width of the transition.
+
+ The border values of @self or -1 if not meaningful
+(this will happen when not using a smpte transition).
+
+
+
+
+ The #GESVideoTransition to get the border from
+
+
+
+
+
+ Get the transition type used by @trans.
+
+ The transition type used by @trans.
+
+
+
+
+ a #GESVideoTransition
+
+
+
+
+
+ Get the invert property of @self, this value represents
+the direction of the transition.
+
+ The invert value of @self
+
+
+
+
+ The #GESVideoTransition to get the inversion from
+
+
+
+
+
+ Set the border property of @self, this value represents
+the border width of the transition. In case this value does
+not make sense for the current transition type, it is cached
+for later use.
+
+
+
+
+
+ The #GESVideoTransition to set the border to
+
+
+
+ The value of the border to set on @object
+
+
+
+
+
+ Set the invert property of @self, this value represents
+the direction of the transition. In case this value does
+not make sense for the current transition type, it is cached
+for later use.
+
+
+
+
+
+ The #GESVideoTransition to set invert on
+
+
+
+ %TRUE if the transition should be inverted %FALSE otherwise
+
+
+
+
+
+ Sets the transition being used to @type.
+
+ %TRUE if the transition type was properly changed, else %FALSE.
+
+
+
+
+ a #GESVideoTransition
+
+
+
+ a #GESVideoStandardTransitionType
+
+
+
+
+
+ This value represents the border width of the transition.
+
+
+
+ This value represents the direction of the transition.
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ parent class
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ The location of the file/resource to use.
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Clean up any resources created by GES in ges_init().
+
+It is normally not needed to call this function in a normal application as the
+resources will automatically be freed when the program terminates.
+This function is therefore mostly used by testsuites and other memory profiling tools.
+
+After this call GES (including this method) should not be used anymore.
+
+
+
+
+
+ Initialize the GStreamer Editing Service. Call this before any usage of
+GES. You should take care of initilizing GStreamer before calling this
+function.
+
+
+
+
+
+ Initializes the GStreamer Editing Services library, setting up internal path lists,
+and loading evrything needed.
+
+This function will return %FALSE if GES could not be initialized
+for some reason.
+
+ %TRUE if GES could be initialized.
+
+
+
+
+ pointer to application's argc
+
+
+
+ pointer to application's argv
+
+
+
+
+
+
+
+ Returns a #GOptionGroup with GES's argument specifications. The
+group is set up to use standard GOption callbacks, so when using this
+group in combination with GOption parsing methods, all argument parsing
+and initialization is automated.
+
+This function is useful if you want to integrate GES with other
+libraries that use GOption (see g_option_context_add_group() ).
+
+If you use this function, you should make sure you initialise the GStreamer
+as one of the very first things in your program. That means you need to
+use gst_init_get_option_group() and add it to the option context before
+using the ges_init_get_option_group() result.
+
+ a pointer to GES's option group.
+
+
+
+
+ List all @asset filtering per filter as defined by @filter.
+It copies the asset and thus will not be updated in time.
+
+ The list of
+#GESAsset the object contains
+
+
+
+
+
+
+ Type of assets to list, #GES_TYPE_EXTRACTABLE will list
+all assets
+
+
+
+
+
+ Get the last buffer @playsink showed
+
+ A #GstSample containing the last frame from
+@playsink in the format defined by the @caps
+
+
+
+
+ The playsink to get last frame from
+
+
+
+ The caps defining the format the return value will have
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Gets the version number of the GStreamer Editing Services library.
+
+
+
+
+
+ pointer to a guint to store the major version number
+
+
+
+ pointer to a guint to store the minor version number
+
+
+
+ pointer to a guint to store the micro version number
+
+
+
+ pointer to a guint to store the nano version number
+
+
+
+
+
+
diff --git a/girs/Gst-1.0.gir b/girs/Gst-1.0.gir
new file mode 100644
index 0000000000..f46ee1bd0a
--- /dev/null
+++ b/girs/Gst-1.0.gir
@@ -0,0 +1,49338 @@
+
+
+
+
+
+
+
+
+
+
+ A datatype to hold the handle to an outstanding sync or async clock callback.
+
+
+
+ A datatype to hold a time, measured in nanoseconds.
+
+
+
+ A datatype to hold a time difference, measured in nanoseconds.
+
+
+
+
+
+
+ The allocator name for the default system memory allocator
+
+
+
+ Parameters to control the allocation of memory
+
+ flags to control allocation
+
+
+
+ the desired alignment of the memory
+
+
+
+ the desired prefix
+
+
+
+ the desired padding
+
+
+
+
+
+
+
+
+ Create a copy of @params.
+
+Free-function: gst_allocation_params_free
+
+ a new ##GstAllocationParams, free with
+gst_allocation_params_free().
+
+
+
+
+ a #GstAllocationParams
+
+
+
+
+
+ Free @params
+
+
+
+
+
+ a #GstAllocationParams
+
+
+
+
+
+ Initialize @params to its default values
+
+
+
+
+
+ a #GstAllocationParams
+
+
+
+
+
+
+ Memory is usually created by allocators with a gst_allocator_alloc()
+method call. When %NULL is used as the allocator, the default allocator will
+be used.
+
+New allocators can be registered with gst_allocator_register().
+Allocators are identified by name and can be retrieved with
+gst_allocator_find(). gst_allocator_set_default() can be used to change the
+default allocator.
+
+New memory can be created with gst_memory_new_wrapped() that wraps the memory
+allocated elsewhere.
+
+ Find a previously registered allocator with @name. When @name is %NULL, the
+default allocator will be returned.
+
+ a #GstAllocator or %NULL when
+the allocator with @name was not registered. Use gst_object_unref()
+to release the allocator after usage.
+
+
+
+
+ the name of the allocator
+
+
+
+
+
+ Registers the memory @allocator with @name. This function takes ownership of
+@allocator.
+
+
+
+
+
+ the name of the allocator
+
+
+
+ #GstAllocator
+
+
+
+
+
+ Use @allocator to allocate a new memory block with memory that is at least
+@size big.
+
+The optional @params can specify the prefix and padding for the memory. If
+%NULL is passed, no flags, no extra prefix/padding and a default alignment is
+used.
+
+The prefix/padding will be filled with 0 if flags contains
+#GST_MEMORY_FLAG_ZERO_PREFIXED and #GST_MEMORY_FLAG_ZERO_PADDED respectively.
+
+When @allocator is %NULL, the default allocator will be used.
+
+The alignment in @params is given as a bitmask so that @align + 1 equals
+the amount of bytes to align to. For example, to align to 8 bytes,
+use an alignment of 7.
+
+ a new #GstMemory.
+
+
+
+
+ a #GstAllocator to use
+
+
+
+ size of the visible memory area
+
+
+
+ optional parameters
+
+
+
+
+
+ Free @memory that was previously allocated with gst_allocator_alloc().
+
+
+
+
+
+ a #GstAllocator to use
+
+
+
+ the memory to free
+
+
+
+
+
+ Use @allocator to allocate a new memory block with memory that is at least
+@size big.
+
+The optional @params can specify the prefix and padding for the memory. If
+%NULL is passed, no flags, no extra prefix/padding and a default alignment is
+used.
+
+The prefix/padding will be filled with 0 if flags contains
+#GST_MEMORY_FLAG_ZERO_PREFIXED and #GST_MEMORY_FLAG_ZERO_PADDED respectively.
+
+When @allocator is %NULL, the default allocator will be used.
+
+The alignment in @params is given as a bitmask so that @align + 1 equals
+the amount of bytes to align to. For example, to align to 8 bytes,
+use an alignment of 7.
+
+ a new #GstMemory.
+
+
+
+
+ a #GstAllocator to use
+
+
+
+ size of the visible memory area
+
+
+
+ optional parameters
+
+
+
+
+
+ Free @memory that was previously allocated with gst_allocator_alloc().
+
+
+
+
+
+ a #GstAllocator to use
+
+
+
+ the memory to free
+
+
+
+
+
+ Set the default allocator. This function takes ownership of @allocator.
+
+
+
+
+
+ a #GstAllocator
+
+
+
+
+
+
+
+
+
+
+
+ the implementation of the GstMemoryMapFunction
+
+
+
+ the implementation of the GstMemoryUnmapFunction
+
+
+
+ the implementation of the GstMemoryCopyFunction
+
+
+
+ the implementation of the GstMemoryShareFunction
+
+
+
+ the implementation of the GstMemoryIsSpanFunction
+
+
+
+ the implementation of the GstMemoryMapFullFunction.
+ Will be used instead of @mem_map if present. (Since 1.6)
+
+
+
+ the implementation of the GstMemoryUnmapFullFunction.
+ Will be used instead of @mem_unmap if present. (Since 1.6)
+
+
+
+
+
+
+
+
+
+
+
+
+ The #GstAllocator is used to create new memory.
+
+ Object parent class
+
+
+
+
+
+ a new #GstMemory.
+
+
+
+
+ a #GstAllocator to use
+
+
+
+ size of the visible memory area
+
+
+
+ optional parameters
+
+
+
+
+
+
+
+
+
+
+
+
+ a #GstAllocator to use
+
+
+
+ the memory to free
+
+
+
+
+
+
+
+
+
+
+
+
+ Flags for allocators.
+
+ The allocator has a custom alloc function.
+
+
+ first flag that can be used for custom purposes
+
+
+
+
+
+ The #GstAtomicQueue object implements a queue that can be used from multiple
+threads without performing any blocking operations.
+
+ Create a new atomic queue instance. @initial_size will be rounded up to the
+nearest power of 2 and used as the initial size of the queue.
+
+ a new #GstAtomicQueue
+
+
+
+
+ initial queue size
+
+
+
+
+
+ Get the amount of items in the queue.
+
+ the number of elements in the queue.
+
+
+
+
+ a #GstAtomicQueue
+
+
+
+
+
+ Peek the head element of the queue without removing it from the queue.
+
+ the head element of @queue or
+%NULL when the queue is empty.
+
+
+
+
+ a #GstAtomicQueue
+
+
+
+
+
+ Get the head element of the queue.
+
+ the head element of @queue or %NULL when
+the queue is empty.
+
+
+
+
+ a #GstAtomicQueue
+
+
+
+
+
+ Append @data to the tail of the queue.
+
+
+
+
+
+ a #GstAtomicQueue
+
+
+
+ the data
+
+
+
+
+
+ Increase the refcount of @queue.
+
+
+
+
+
+ a #GstAtomicQueue
+
+
+
+
+
+ Unref @queue and free the memory when the refcount reaches 0.
+
+
+
+
+
+ a #GstAtomicQueue
+
+
+
+
+
+
+ Combination of all possible fields that can be copied with
+gst_buffer_copy_into().
+
+
+
+ Combination of all possible metadata fields that can be copied with
+gst_buffer_copy_into().
+
+
+
+ Constant for no-offset return results.
+
+
+
+ #GstBin is an element that can contain other #GstElement, allowing them to be
+managed as a group.
+Pads from the child elements can be ghosted to the bin, see #GstGhostPad.
+This makes the bin look like any other elements and enables creation of
+higher-level abstraction elements.
+
+A new #GstBin is created with gst_bin_new(). Use a #GstPipeline instead if you
+want to create a toplevel bin because a normal bin doesn't have a bus or
+handle clock distribution of its own.
+
+After the bin has been created you will typically add elements to it with
+gst_bin_add(). You can remove elements with gst_bin_remove().
+
+An element can be retrieved from a bin with gst_bin_get_by_name(), using the
+elements name. gst_bin_get_by_name_recurse_up() is mainly used for internal
+purposes and will query the parent bins when the element is not found in the
+current bin.
+
+An iterator of elements in a bin can be retrieved with
+gst_bin_iterate_elements(). Various other iterators exist to retrieve the
+elements in a bin.
+
+gst_object_unref() is used to drop your reference to the bin.
+
+The #GstBin::element-added signal is fired whenever a new element is added to
+the bin. Likewise the #GstBin::element-removed signal is fired whenever an
+element is removed from the bin.
+
+## Notes
+
+A #GstBin internally intercepts every #GstMessage posted by its children and
+implements the following default behaviour for each of them:
+
+* GST_MESSAGE_EOS: This message is only posted by sinks in the PLAYING
+state. If all sinks posted the EOS message, this bin will post and EOS
+message upwards.
+
+* GST_MESSAGE_SEGMENT_START: Just collected and never forwarded upwards.
+The messages are used to decide when all elements have completed playback
+of their segment.
+
+* GST_MESSAGE_SEGMENT_DONE: Is posted by #GstBin when all elements that posted
+a SEGMENT_START have posted a SEGMENT_DONE.
+
+* GST_MESSAGE_DURATION_CHANGED: Is posted by an element that detected a change
+in the stream duration. The default bin behaviour is to clear any
+cached duration values so that the next duration query will perform
+a full duration recalculation. The duration change is posted to the
+application so that it can refetch the new duration with a duration
+query. Note that these messages can be posted before the bin is
+prerolled, in which case the duration query might fail.
+
+* GST_MESSAGE_CLOCK_LOST: This message is posted by an element when it
+can no longer provide a clock. The default bin behaviour is to
+check if the lost clock was the one provided by the bin. If so and
+the bin is currently in the PLAYING state, the message is forwarded to
+the bin parent.
+This message is also generated when a clock provider is removed from
+the bin. If this message is received by the application, it should
+PAUSE the pipeline and set it back to PLAYING to force a new clock
+distribution.
+
+* GST_MESSAGE_CLOCK_PROVIDE: This message is generated when an element
+can provide a clock. This mostly happens when a new clock
+provider is added to the bin. The default behaviour of the bin is to
+mark the currently selected clock as dirty, which will perform a clock
+recalculation the next time the bin is asked to provide a clock.
+This message is never sent tot the application but is forwarded to
+the parent of the bin.
+
+* OTHERS: posted upwards.
+
+A #GstBin implements the following default behaviour for answering to a
+#GstQuery:
+
+* GST_QUERY_DURATION:If the query has been asked before with the same format
+and the bin is a toplevel bin (ie. has no parent),
+use the cached previous value. If no previous value was cached, the
+query is sent to all sink elements in the bin and the MAXIMUM of all
+values is returned. If the bin is a toplevel bin the value is cached.
+If no sinks are available in the bin, the query fails.
+
+* GST_QUERY_POSITION:The query is sent to all sink elements in the bin and the
+MAXIMUM of all values is returned. If no sinks are available in the bin,
+the query fails.
+
+* OTHERS:the query is forwarded to all sink elements, the result
+of the first sink that answers the query successfully is returned. If no
+sink is in the bin, the query fails.
+
+A #GstBin will by default forward any event sent to it to all sink
+(#GST_EVENT_TYPE_DOWNSTREAM) or source (#GST_EVENT_TYPE_UPSTREAM) elements
+depending on the event type.
+If all the elements return %TRUE, the bin will also return %TRUE, else %FALSE
+is returned. If no elements of the required type are in the bin, the event
+handler will return %TRUE.
+
+
+ Creates a new bin with the given name.
+
+ a new #GstBin
+
+
+
+
+ the name of the new bin
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Adds the given element to the bin. Sets the element's parent, and thus
+takes ownership of the element. An element can only be added to one bin.
+
+If the element's pads are linked to other pads, the pads will be unlinked
+before the element is added to the bin.
+
+> When you add an element to an already-running pipeline, you will have to
+> take care to set the state of the newly-added element to the desired
+> state (usually PLAYING or PAUSED, same you set the pipeline to originally)
+> with gst_element_set_state(), or use gst_element_sync_state_with_parent().
+> The bin or pipeline will not take care of this for you.
+
+MT safe.
+
+ %TRUE if the element could be added, %FALSE if
+the bin does not want to accept the element.
+
+
+
+
+ a #GstBin
+
+
+
+ the #GstElement to add
+
+
+
+
+
+ Adds a %NULL-terminated list of elements to a bin. This function is
+equivalent to calling gst_bin_add() for each member of the list. The return
+value of each gst_bin_add() is ignored.
+
+
+
+
+
+ a #GstBin
+
+
+
+ the #GstElement element to add to the bin
+
+
+
+ additional elements to add to the bin
+
+
+
+
+
+ Recursively looks for elements with an unlinked pad of the given
+direction within the specified bin and returns an unlinked pad
+if one is found, or %NULL otherwise. If a pad is found, the caller
+owns a reference to it and should use gst_object_unref() on the
+pad when it is not needed any longer.
+
+ unlinked pad of the given
+direction, %NULL.
+
+
+
+
+ bin in which to look for elements with unlinked pads
+
+
+
+ whether to look for an unlinked source or sink pad
+
+
+
+
+
+ Looks for an element inside the bin that implements the given
+interface. If such an element is found, it returns the element.
+You can cast this element to the given interface afterwards. If you want
+all elements that implement the interface, use
+gst_bin_iterate_all_by_interface(). This function recurses into child bins.
+
+MT safe. Caller owns returned reference.
+
+ A #GstElement inside the bin
+implementing the interface
+
+
+
+
+ a #GstBin
+
+
+
+ the #GType of an interface
+
+
+
+
+
+ Gets the element with the given name from a bin. This
+function recurses into child bins.
+
+Returns %NULL if no element with the given name is found in the bin.
+
+MT safe. Caller owns returned reference.
+
+ the #GstElement with the given
+name, or %NULL
+
+
+
+
+ a #GstBin
+
+
+
+ the element name to search for
+
+
+
+
+
+ Gets the element with the given name from this bin. If the
+element is not found, a recursion is performed on the parent bin.
+
+Returns %NULL if:
+- no element with the given name is found in the bin
+
+MT safe. Caller owns returned reference.
+
+ the #GstElement with the given
+name, or %NULL
+
+
+
+
+ a #GstBin
+
+
+
+ the element name to search for
+
+
+
+
+
+ Return the suppressed flags of the bin.
+
+MT safe.
+
+ the bin's suppressed #GstElementFlags.
+
+
+
+
+ a #GstBin
+
+
+
+
+
+ Looks for all elements inside the bin that implements the given
+interface. You can safely cast all returned elements to the given interface.
+The function recurses inside child bins. The iterator will yield a series
+of #GstElement that should be unreffed after use.
+
+MT safe. Caller owns returned value.
+
+ a #GstIterator of #GstElement
+ for all elements in the bin implementing the given interface,
+ or %NULL
+
+
+
+
+ a #GstBin
+
+
+
+ the #GType of an interface
+
+
+
+
+
+ Gets an iterator for the elements in this bin.
+
+MT safe. Caller owns returned value.
+
+ a #GstIterator of #GstElement,
+or %NULL
+
+
+
+
+ a #GstBin
+
+
+
+
+
+ Gets an iterator for the elements in this bin.
+This iterator recurses into GstBin children.
+
+MT safe. Caller owns returned value.
+
+ a #GstIterator of #GstElement,
+or %NULL
+
+
+
+
+ a #GstBin
+
+
+
+
+
+ Gets an iterator for all elements in the bin that have the
+#GST_ELEMENT_FLAG_SINK flag set.
+
+MT safe. Caller owns returned value.
+
+ a #GstIterator of #GstElement,
+or %NULL
+
+
+
+
+ a #GstBin
+
+
+
+
+
+ Gets an iterator for the elements in this bin in topologically
+sorted order. This means that the elements are returned from
+the most downstream elements (sinks) to the sources.
+
+This function is used internally to perform the state changes
+of the bin elements and for clock selection.
+
+MT safe. Caller owns returned value.
+
+ a #GstIterator of #GstElement,
+or %NULL
+
+
+
+
+ a #GstBin
+
+
+
+
+
+ Gets an iterator for all elements in the bin that have the
+#GST_ELEMENT_FLAG_SOURCE flag set.
+
+MT safe. Caller owns returned value.
+
+ a #GstIterator of #GstElement,
+or %NULL
+
+
+
+
+ a #GstBin
+
+
+
+
+
+ Query @bin for the current latency using and reconfigures this latency to all the
+elements with a LATENCY event.
+
+This method is typically called on the pipeline when a #GST_MESSAGE_LATENCY
+is posted on the bus.
+
+This function simply emits the 'do-latency' signal so any custom latency
+calculations will be performed.
+
+ %TRUE if the latency could be queried and reconfigured.
+
+
+
+
+ a #GstBin
+
+
+
+
+
+ Removes the element from the bin, unparenting it as well.
+Unparenting the element means that the element will be dereferenced,
+so if the bin holds the only reference to the element, the element
+will be freed in the process of removing it from the bin. If you
+want the element to still exist after removing, you need to call
+gst_object_ref() before removing it from the bin.
+
+If the element's pads are linked to other pads, the pads will be unlinked
+before the element is removed from the bin.
+
+MT safe.
+
+ %TRUE if the element could be removed, %FALSE if
+the bin does not want to remove the element.
+
+
+
+
+ a #GstBin
+
+
+
+ the #GstElement to remove
+
+
+
+
+
+ Remove a list of elements from a bin. This function is equivalent
+to calling gst_bin_remove() with each member of the list.
+
+
+
+
+
+ a #GstBin
+
+
+
+ the first #GstElement to remove from the bin
+
+
+
+ %NULL-terminated list of elements to remove from the bin
+
+
+
+
+
+ Suppress the given flags on the bin. #GstElementFlags of a
+child element are propagated when it is added to the bin.
+When suppressed flags are set, those specified flags will
+not be propagated to the bin.
+
+MT safe.
+
+
+
+
+
+ a #GstBin
+
+
+
+ the #GstElementFlags to suppress
+
+
+
+
+
+ Synchronizes the state of every child of @bin with the state
+of @bin. See also gst_element_sync_state_with_parent().
+
+ %TRUE if syncing the state was successful for all children,
+ otherwise %FALSE.
+
+
+
+
+ a #GstBin
+
+
+
+
+
+ If set to %TRUE, the bin will handle asynchronous state changes.
+This should be used only if the bin subclass is modifying the state
+of its children on its own.
+
+
+
+ Forward all children messages, even those that would normally be filtered by
+the bin. This can be interesting when one wants to be notified of the EOS
+state of individual elements, for example.
+
+The messages are converted to an ELEMENT message with the bin as the
+source. The structure of the message is named 'GstBinForwarded' and contains
+a field named 'message' of type GST_TYPE_MESSAGE that contains the original
+forwarded message.
+
+
+
+
+
+
+ the number of children in this bin
+
+
+
+ the list of children in this bin
+
+
+
+
+
+ updated whenever @children changes
+
+
+
+ internal bus for handling child messages
+
+
+
+ queued and cached messages
+
+
+
+
+
+ the bin is currently calculating its state
+
+
+
+ the bin needs to recalculate its state (deprecated)
+
+
+
+ the bin needs to select a new clock
+
+
+
+ the last clock selected
+
+
+
+ the element that provided @provided_clock
+
+
+
+
+
+
+
+
+
+
+
+ Will be emitted after the element was added to sub_bin.
+
+
+
+
+
+ the #GstBin the element was added to
+
+
+
+ the #GstElement that was added to @sub_bin
+
+
+
+
+
+ Will be emitted after the element was removed from sub_bin.
+
+
+
+
+
+ the #GstBin the element was removed from
+
+
+
+ the #GstElement that was removed from @sub_bin
+
+
+
+
+
+ Will be emitted when the bin needs to perform latency calculations. This
+signal is only emitted for toplevel bins or when async-handling is
+enabled.
+
+Only one signal handler is invoked. If no signals are connected, the
+default handler is invoked, which will query and distribute the lowest
+possible latency to all sinks.
+
+Connect to this signal if the default latency calculations are not
+sufficient, like when you need different latencies for different sinks in
+the same pipeline.
+
+
+
+
+
+ Will be emitted after the element was added to the bin.
+
+
+
+
+
+ the #GstElement that was added to the bin
+
+
+
+
+
+ Will be emitted after the element was removed from the bin.
+
+
+
+
+
+ the #GstElement that was removed from the bin
+
+
+
+
+
+
+ Subclasses can override the @add_element and @remove_element to
+update the list of children in the bin.
+
+The @handle_message method can be overridden to implement custom
+message handling. @handle_message takes ownership of the message, just like
+#gst_element_post_message.
+
+The @deep_element_added vfunc will be called when a new element has been
+added to any bin inside this bin, so it will also be called if a new child
+was added to a sub-bin of this bin. #GstBin implementations that override
+this message should chain up to the parent class implementation so the
+element-added-deep signal is emitted on all parents.
+
+ bin parent class
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ GstBinFlags are a set of flags specific to bins. Most are set/used
+internally. They can be checked using the GST_OBJECT_FLAG_IS_SET () macro,
+and (un)set using GST_OBJECT_FLAG_SET () and GST_OBJECT_FLAG_UNSET ().
+
+ don't resync a state change when elements are
+ added or linked in the bin (Since 1.0.5)
+
+
+ Indicates whether the bin can handle elements
+ that add/remove source pads at any point in time without
+ first posting a no-more-pads signal (Since 1.10)
+
+
+ the last enum in the series of flags for bins.
+Derived classes can use this as first value in a list of flags.
+
+
+
+
+
+
+
+ Buffers are the basic unit of data transfer in GStreamer. They contain the
+timing and offset along with other arbitrary metadata that is associated
+with the #GstMemory blocks that the buffer contains.
+
+Buffers are usually created with gst_buffer_new(). After a buffer has been
+created one will typically allocate memory for it and add it to the buffer.
+The following example creates a buffer that can hold a given video frame
+with a given width, height and bits per plane.
+|[<!-- language="C" -->
+ GstBuffer *buffer;
+ GstMemory *memory;
+ gint size, width, height, bpp;
+ ...
+ size = width * height * bpp;
+ buffer = gst_buffer_new ();
+ memory = gst_allocator_alloc (NULL, size, NULL);
+ gst_buffer_insert_memory (buffer, -1, memory);
+ ...
+]|
+
+Alternatively, use gst_buffer_new_allocate() to create a buffer with
+preallocated data of a given size.
+
+Buffers can contain a list of #GstMemory objects. You can retrieve how many
+memory objects with gst_buffer_n_memory() and you can get a pointer
+to memory with gst_buffer_peek_memory()
+
+A buffer will usually have timestamps, and a duration, but neither of these
+are guaranteed (they may be set to #GST_CLOCK_TIME_NONE). Whenever a
+meaningful value can be given for these, they should be set. The timestamps
+and duration are measured in nanoseconds (they are #GstClockTime values).
+
+The buffer DTS refers to the timestamp when the buffer should be decoded and
+is usually monotonically increasing. The buffer PTS refers to the timestamp when
+the buffer content should be presented to the user and is not always
+monotonically increasing.
+
+A buffer can also have one or both of a start and an end offset. These are
+media-type specific. For video buffers, the start offset will generally be
+the frame number. For audio buffers, it will be the number of samples
+produced so far. For compressed data, it could be the byte offset in a
+source or destination file. Likewise, the end offset will be the offset of
+the end of the buffer. These can only be meaningfully interpreted if you
+know the media type of the buffer (the preceding CAPS event). Either or both
+can be set to #GST_BUFFER_OFFSET_NONE.
+
+gst_buffer_ref() is used to increase the refcount of a buffer. This must be
+done when you want to keep a handle to the buffer after pushing it to the
+next element. The buffer refcount determines the writability of the buffer, a
+buffer is only writable when the refcount is exactly 1, i.e. when the caller
+has the only reference to the buffer.
+
+To efficiently create a smaller buffer out of an existing one, you can
+use gst_buffer_copy_region(). This method tries to share the memory objects
+between the two buffers.
+
+If a plug-in wants to modify the buffer data or metadata in-place, it should
+first obtain a buffer that is safe to modify by using
+gst_buffer_make_writable(). This function is optimized so that a copy will
+only be made when it is necessary.
+
+Several flags of the buffer can be set and unset with the
+GST_BUFFER_FLAG_SET() and GST_BUFFER_FLAG_UNSET() macros. Use
+GST_BUFFER_FLAG_IS_SET() to test if a certain #GstBufferFlags flag is set.
+
+Buffers can be efficiently merged into a larger buffer with
+gst_buffer_append(). Copying of memory will only be done when absolutely
+needed.
+
+Arbitrary extra metadata can be set on a buffer with gst_buffer_add_meta().
+Metadata can be retrieved with gst_buffer_get_meta(). See also #GstMeta
+
+An element should either unref the buffer or push it out on a src pad
+using gst_pad_push() (see #GstPad).
+
+Buffers are usually freed by unreffing them with gst_buffer_unref(). When
+the refcount drops to 0, any memory and metadata pointed to by the buffer is
+unreffed as well. Buffers allocated from a #GstBufferPool will be returned to
+the pool when the refcount drops to 0.
+
+The #GstParentBufferMeta is a meta which can be attached to a #GstBuffer
+to hold a reference to another buffer that is only released when the child
+#GstBuffer is released.
+
+Typically, #GstParentBufferMeta is used when the child buffer is directly
+using the #GstMemory of the parent buffer, and wants to prevent the parent
+buffer from being returned to a buffer pool until the #GstMemory is available
+for re-use. (Since 1.6)
+
+ the parent structure
+
+
+
+ pointer to the pool owner of the buffer
+
+
+
+ presentation timestamp of the buffer, can be #GST_CLOCK_TIME_NONE when the
+ pts is not known or relevant. The pts contains the timestamp when the
+ media should be presented to the user.
+
+
+
+ decoding timestamp of the buffer, can be #GST_CLOCK_TIME_NONE when the
+ dts is not known or relevant. The dts contains the timestamp when the
+ media should be processed.
+
+
+
+ duration in time of the buffer data, can be #GST_CLOCK_TIME_NONE
+ when the duration is not known or relevant.
+
+
+
+ a media specific offset for the buffer data.
+ For video frames, this is the frame number of this buffer.
+ For audio samples, this is the offset of the first sample in this buffer.
+ For file data or compressed data this is the byte offset of the first
+ byte in this buffer.
+
+
+
+ the last offset contained in this buffer. It has the same
+ format as @offset.
+
+
+
+ Creates a newly allocated buffer without any data.
+
+MT safe.
+
+ the new #GstBuffer.
+
+
+
+
+ Tries to create a newly allocated buffer with data of the given size and
+extra parameters from @allocator. If the requested amount of memory can't be
+allocated, %NULL will be returned. The allocated buffer memory is not cleared.
+
+When @allocator is %NULL, the default memory allocator will be used.
+
+Note that when @size == 0, the buffer will not have memory associated with it.
+
+MT safe.
+
+ a new #GstBuffer, or %NULL if
+ the memory couldn't be allocated.
+
+
+
+
+ the #GstAllocator to use, or %NULL to use the
+ default allocator
+
+
+
+ the size in bytes of the new buffer's data.
+
+
+
+ optional parameters
+
+
+
+
+
+ Creates a new buffer that wraps the given @data. The memory will be freed
+with g_free and will be marked writable.
+
+MT safe.
+
+ a new #GstBuffer
+
+
+
+
+ data to wrap
+
+
+
+
+
+ allocated size of @data
+
+
+
+
+
+ Allocate a new buffer that wraps the given memory. @data must point to
+@maxsize of memory, the wrapped buffer will have the region from @offset and
+@size visible.
+
+When the buffer is destroyed, @notify will be called with @user_data.
+
+The prefix/padding must be filled with 0 if @flags contains
+#GST_MEMORY_FLAG_ZERO_PREFIXED and #GST_MEMORY_FLAG_ZERO_PADDED respectively.
+
+ a new #GstBuffer
+
+
+
+
+ #GstMemoryFlags
+
+
+
+ data to wrap
+
+
+
+
+
+ allocated size of @data
+
+
+
+ offset in @data
+
+
+
+ size of valid data
+
+
+
+ user_data
+
+
+
+ called with @user_data when the memory is freed
+
+
+
+
+
+ Add metadata for @info to @buffer using the parameters in @params.
+
+ the metadata for the api in @info on @buffer.
+
+
+
+
+ a #GstBuffer
+
+
+
+ a #GstMetaInfo
+
+
+
+ params for @info
+
+
+
+
+
+ Add a #GstParentBufferMeta to @buffer that holds a reference on
+@ref until the buffer is freed.
+
+ The #GstParentBufferMeta that was added to the buffer
+
+
+
+
+ a #GstBuffer
+
+
+
+ a #GstBuffer to ref
+
+
+
+
+
+ Attaches protection metadata to a #GstBuffer.
+
+ a pointer to the added #GstProtectionMeta if successful; %NULL if
+unsuccessful.
+
+
+
+
+ #GstBuffer holding an encrypted sample, to which protection
+ metadata should be added.
+
+
+
+ a #GstStructure holding cryptographic
+ information relating to the sample contained in @buffer. This
+ function takes ownership of @info.
+
+
+
+
+
+ Add a #GstReferenceTimestampMeta to @buffer that holds a @timestamp and
+optionally @duration based on a specific timestamp @reference. See the
+documentation of #GstReferenceTimestampMeta for details.
+
+ The #GstReferenceTimestampMeta that was added to the buffer
+
+
+
+
+ a #GstBuffer
+
+
+
+ identifier for the timestamp reference.
+
+
+
+ timestamp
+
+
+
+ duration, or %GST_CLOCK_TIME_NONE
+
+
+
+
+
+ Append all the memory from @buf2 to @buf1. The result buffer will contain a
+concatenation of the memory of @buf1 and @buf2.
+
+ the new #GstBuffer that contains the memory
+ of the two source buffers.
+
+
+
+
+ the first source #GstBuffer to append.
+
+
+
+ the second source #GstBuffer to append.
+
+
+
+
+
+ Append the memory block @mem to @buffer. This function takes
+ownership of @mem and thus doesn't increase its refcount.
+
+This function is identical to gst_buffer_insert_memory() with an index of -1.
+See gst_buffer_insert_memory() for more details.
+
+
+
+
+
+ a #GstBuffer.
+
+
+
+ a #GstMemory.
+
+
+
+
+
+ Append @size bytes at @offset from @buf2 to @buf1. The result buffer will
+contain a concatenation of the memory of @buf1 and the requested region of
+@buf2.
+
+ the new #GstBuffer that contains the memory
+ of the two source buffers.
+
+
+
+
+ the first source #GstBuffer to append.
+
+
+
+ the second source #GstBuffer to append.
+
+
+
+ the offset in @buf2
+
+
+
+ the size or -1 of @buf2
+
+
+
+
+
+ Create a copy of the given buffer. This will make a newly allocated
+copy of the data the source buffer contains.
+
+ a new copy of @buf.
+
+
+
+
+ a #GstBuffer.
+
+
+
+
+
+ Copies the information from @src into @dest.
+
+If @dest already contains memory and @flags contains GST_BUFFER_COPY_MEMORY,
+the memory from @src will be appended to @dest.
+
+@flags indicate which fields will be copied.
+
+ %TRUE if the copying succeeded, %FALSE otherwise.
+
+
+
+
+ a destination #GstBuffer
+
+
+
+ a source #GstBuffer
+
+
+
+ flags indicating what metadata fields should be copied.
+
+
+
+ offset to copy from
+
+
+
+ total size to copy. If -1, all data is copied.
+
+
+
+
+
+ Creates a sub-buffer from @parent at @offset and @size.
+This sub-buffer uses the actual memory space of the parent buffer.
+This function will copy the offset and timestamp fields when the
+offset is 0. If not, they will be set to #GST_CLOCK_TIME_NONE and
+#GST_BUFFER_OFFSET_NONE.
+If @offset equals 0 and @size equals the total size of @buffer, the
+duration and offset end fields are also copied. If not they will be set
+to #GST_CLOCK_TIME_NONE and #GST_BUFFER_OFFSET_NONE.
+
+MT safe.
+
+ the new #GstBuffer or %NULL if the arguments were
+ invalid.
+
+
+
+
+ a #GstBuffer.
+
+
+
+ the #GstBufferCopyFlags
+
+
+
+ the offset into parent #GstBuffer at which the new sub-buffer
+ begins.
+
+
+
+ the size of the new #GstBuffer sub-buffer, in bytes. If -1, all
+ data is copied.
+
+
+
+
+
+ Copy @size bytes starting from @offset in @buffer to @dest.
+
+ The amount of bytes extracted. This value can be lower than @size
+ when @buffer did not contain enough data.
+
+
+
+
+ a #GstBuffer.
+
+
+
+ the offset to extract
+
+
+
+
+ the destination address
+
+
+
+
+
+ the size to extract
+
+
+
+
+
+ Extracts a copy of at most @size bytes the data at @offset into
+newly-allocated memory. @dest must be freed using g_free() when done.
+
+
+
+
+
+ a #GstBuffer
+
+
+
+ the offset to extract
+
+
+
+ the size to extract
+
+
+
+ A pointer where
+ the destination array will be written. Might be %NULL if the size is 0.
+
+
+
+
+
+ A location where the size of @dest can be written
+
+
+
+
+
+ Copy @size bytes from @src to @buffer at @offset.
+
+ The amount of bytes copied. This value can be lower than @size
+ when @buffer did not contain enough data.
+
+
+
+
+ a #GstBuffer.
+
+
+
+ the offset to fill
+
+
+
+ the source address
+
+
+
+
+
+ the size to fill
+
+
+
+
+
+ Find the memory blocks that span @size bytes starting from @offset
+in @buffer.
+
+When this function returns %TRUE, @idx will contain the index of the first
+memory block where the byte for @offset can be found and @length contains the
+number of memory blocks containing the @size remaining bytes. @skip contains
+the number of bytes to skip in the memory block at @idx to get to the byte
+for @offset.
+
+@size can be -1 to get all the memory blocks after @idx.
+
+ %TRUE when @size bytes starting from @offset could be found in
+@buffer and @idx, @length and @skip will be filled.
+
+
+
+
+ a #GstBuffer.
+
+
+
+ an offset
+
+
+
+ a size
+
+
+
+ pointer to index
+
+
+
+ pointer to length
+
+
+
+ pointer to skip
+
+
+
+
+
+ Call @func with @user_data for each meta in @buffer.
+
+@func can modify the passed meta pointer or its contents. The return value
+of @func define if this function returns or if the remaining metadata items
+in the buffer should be skipped.
+
+ %FALSE when @func returned %FALSE for one of the metadata.
+
+
+
+
+ a #GstBuffer
+
+
+
+ a #GstBufferForeachMetaFunc to call
+
+
+
+ user data passed to @func
+
+
+
+
+
+ Get all the memory block in @buffer. The memory blocks will be merged
+into one large #GstMemory.
+
+ a #GstMemory that contains the merged memory.
+Use gst_memory_unref () after usage.
+
+
+
+
+ a #GstBuffer.
+
+
+
+
+
+ Get the #GstBufferFlags flags set on this buffer.
+
+ the flags set on this buffer.
+
+
+
+
+ a #GstBuffer
+
+
+
+
+
+ Get the memory block at index @idx in @buffer.
+
+ a #GstMemory that contains the data of the
+memory block at @idx. Use gst_memory_unref () after usage.
+
+
+
+
+ a #GstBuffer.
+
+
+
+ an index
+
+
+
+
+
+ Get @length memory blocks in @buffer starting at @idx. The memory blocks will
+be merged into one large #GstMemory.
+
+If @length is -1, all memory starting from @idx is merged.
+
+ a #GstMemory that contains the merged data of @length
+ blocks starting at @idx. Use gst_memory_unref () after usage.
+
+
+
+
+ a #GstBuffer.
+
+
+
+ an index
+
+
+
+ a length
+
+
+
+
+
+ Get the metadata for @api on buffer. When there is no such metadata, %NULL is
+returned. If multiple metadata with the given @api are attached to this
+buffer only the first one is returned. To handle multiple metadata with a
+given API use gst_buffer_iterate_meta() or gst_buffer_foreach_meta() instead
+and check the meta->info.api member for the API type.
+
+ the metadata for @api on
+@buffer.
+
+
+
+
+ a #GstBuffer
+
+
+
+ the #GType of an API
+
+
+
+
+
+
+ number of metas of type @api_type on @buffer.
+
+
+
+
+ a #GstBuffer
+
+
+
+ the #GType of an API
+
+
+
+
+
+ Find the first #GstReferenceTimestampMeta on @buffer that conforms to
+@reference. Conformance is tested by checking if the meta's reference is a
+subset of @reference.
+
+Buffers can contain multiple #GstReferenceTimestampMeta metadata items.
+
+ the #GstReferenceTimestampMeta or %NULL when there
+is no such metadata on @buffer.
+
+
+
+
+ a #GstBuffer
+
+
+
+ a reference #GstCaps
+
+
+
+
+
+ Get the total size of the memory blocks in @buffer.
+
+ total size of the memory blocks in @buffer.
+
+
+
+
+ a #GstBuffer.
+
+
+
+
+
+ Get the total size of the memory blocks in @b.
+
+When not %NULL, @offset will contain the offset of the data in the
+first memory block in @buffer and @maxsize will contain the sum of
+the size and @offset and the amount of extra padding on the last
+memory block. @offset and @maxsize can be used to resize the
+buffer memory blocks with gst_buffer_resize().
+
+ total size of the memory blocks in @buffer.
+
+
+
+
+ a #GstBuffer.
+
+
+
+ a pointer to the offset
+
+
+
+ a pointer to the maxsize
+
+
+
+
+
+ Get the total size of @length memory blocks stating from @idx in @buffer.
+
+When not %NULL, @offset will contain the offset of the data in the
+memory block in @buffer at @idx and @maxsize will contain the sum of the size
+and @offset and the amount of extra padding on the memory block at @idx +
+@length -1.
+@offset and @maxsize can be used to resize the buffer memory blocks with
+gst_buffer_resize_range().
+
+ total size of @length memory blocks starting at @idx in @buffer.
+
+
+
+
+ a #GstBuffer.
+
+
+
+ an index
+
+
+
+ a length
+
+
+
+ a pointer to the offset
+
+
+
+ a pointer to the maxsize
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Insert the memory block @mem to @buffer at @idx. This function takes ownership
+of @mem and thus doesn't increase its refcount.
+
+Only gst_buffer_get_max_memory() can be added to a buffer. If more memory is
+added, existing memory blocks will automatically be merged to make room for
+the new memory.
+
+
+
+
+
+ a #GstBuffer.
+
+
+
+ the index to add the memory at, or -1 to append it to the end
+
+
+
+ a #GstMemory.
+
+
+
+
+
+ Check if all memory blocks in @buffer are writable.
+
+Note that this function does not check if @buffer is writable, use
+gst_buffer_is_writable() to check that if needed.
+
+ %TRUE if all memory blocks in @buffer are writable
+
+
+
+
+ a #GstBuffer.
+
+
+
+
+
+ Check if @length memory blocks in @buffer starting from @idx are writable.
+
+@length can be -1 to check all the memory blocks after @idx.
+
+Note that this function does not check if @buffer is writable, use
+gst_buffer_is_writable() to check that if needed.
+
+ %TRUE if the memory range is writable
+
+
+
+
+ a #GstBuffer.
+
+
+
+ an index
+
+
+
+ a length should not be 0
+
+
+
+
+
+ Retrieve the next #GstMeta after @current. If @state points
+to %NULL, the first metadata is returned.
+
+@state will be updated with an opaque state pointer
+
+ The next #GstMeta or %NULL
+when there are no more items.
+
+
+
+
+ a #GstBuffer
+
+
+
+ an opaque state pointer
+
+
+
+
+
+ Retrieve the next #GstMeta of type @meta_api_type after the current one
+according to @state. If @state points to %NULL, the first metadata of
+type @meta_api_type is returned.
+
+@state will be updated with an opaque state pointer
+
+ The next #GstMeta of type
+@meta_api_type or %NULL when there are no more items.
+
+
+
+
+ a #GstBuffer
+
+
+
+ an opaque state pointer
+
+
+
+ only return #GstMeta of this type
+
+
+
+
+
+ This function fills @info with the #GstMapInfo of all merged memory
+blocks in @buffer.
+
+@flags describe the desired access of the memory. When @flags is
+#GST_MAP_WRITE, @buffer should be writable (as returned from
+gst_buffer_is_writable()).
+
+When @buffer is writable but the memory isn't, a writable copy will
+automatically be created and returned. The readonly copy of the
+buffer memory will then also be replaced with this writable copy.
+
+The memory in @info should be unmapped with gst_buffer_unmap() after
+usage.
+
+ %TRUE if the map succeeded and @info contains valid data.
+
+
+
+
+ a #GstBuffer.
+
+
+
+ info about the mapping
+
+
+
+ flags for the mapping
+
+
+
+
+
+ This function fills @info with the #GstMapInfo of @length merged memory blocks
+starting at @idx in @buffer. When @length is -1, all memory blocks starting
+from @idx are merged and mapped.
+
+@flags describe the desired access of the memory. When @flags is
+#GST_MAP_WRITE, @buffer should be writable (as returned from
+gst_buffer_is_writable()).
+
+When @buffer is writable but the memory isn't, a writable copy will
+automatically be created and returned. The readonly copy of the buffer memory
+will then also be replaced with this writable copy.
+
+The memory in @info should be unmapped with gst_buffer_unmap() after usage.
+
+ %TRUE if the map succeeded and @info contains valid
+data.
+
+
+
+
+ a #GstBuffer.
+
+
+
+ an index
+
+
+
+ a length
+
+
+
+ info about the mapping
+
+
+
+ flags for the mapping
+
+
+
+
+
+ Compare @size bytes starting from @offset in @buffer with the memory in @mem.
+
+ 0 if the memory is equal.
+
+
+
+
+ a #GstBuffer.
+
+
+
+ the offset in @buffer
+
+
+
+ the memory to compare
+
+
+
+
+
+ the size to compare
+
+
+
+
+
+ Fill @buf with @size bytes with @val starting from @offset.
+
+ The amount of bytes filled. This value can be lower than @size
+ when @buffer did not contain enough data.
+
+
+
+
+ a #GstBuffer.
+
+
+
+ the offset in @buffer
+
+
+
+ the value to set
+
+
+
+ the size to set
+
+
+
+
+
+ Get the amount of memory blocks that this buffer has. This amount is never
+larger than what gst_buffer_get_max_memory() returns.
+
+ the number of memory blocks this buffer is made of.
+
+
+
+
+ a #GstBuffer.
+
+
+
+
+
+ Get the memory block at @idx in @buffer. The memory block stays valid until
+the memory block in @buffer is removed, replaced or merged, typically with
+any call that modifies the memory in @buffer.
+
+ the #GstMemory at @idx.
+
+
+
+
+ a #GstBuffer.
+
+
+
+ an index
+
+
+
+
+
+ Prepend the memory block @mem to @buffer. This function takes
+ownership of @mem and thus doesn't increase its refcount.
+
+This function is identical to gst_buffer_insert_memory() with an index of 0.
+See gst_buffer_insert_memory() for more details.
+
+
+
+
+
+ a #GstBuffer.
+
+
+
+ a #GstMemory.
+
+
+
+
+
+ Remove all the memory blocks in @buffer.
+
+
+
+
+
+ a #GstBuffer.
+
+
+
+
+
+ Remove the memory block in @b at index @i.
+
+
+
+
+
+ a #GstBuffer.
+
+
+
+ an index
+
+
+
+
+
+ Remove @length memory blocks in @buffer starting from @idx.
+
+@length can be -1, in which case all memory starting from @idx is removed.
+
+
+
+
+
+ a #GstBuffer.
+
+
+
+ an index
+
+
+
+ a length
+
+
+
+
+
+ Remove the metadata for @meta on @buffer.
+
+ %TRUE if the metadata existed and was removed, %FALSE if no such
+metadata was on @buffer.
+
+
+
+
+ a #GstBuffer
+
+
+
+ a #GstMeta
+
+
+
+
+
+ Replaces all memory in @buffer with @mem.
+
+
+
+
+
+ a #GstBuffer.
+
+
+
+ a #GstMemory
+
+
+
+
+
+ Replaces the memory block at index @idx in @buffer with @mem.
+
+
+
+
+
+ a #GstBuffer.
+
+
+
+ an index
+
+
+
+ a #GstMemory
+
+
+
+
+
+ Replaces @length memory blocks in @buffer starting at @idx with @mem.
+
+If @length is -1, all memory starting from @idx will be removed and
+replaced with @mem.
+
+@buffer should be writable.
+
+
+
+
+
+ a #GstBuffer.
+
+
+
+ an index
+
+
+
+ a length should not be 0
+
+
+
+ a #GstMemory
+
+
+
+
+
+ Set the offset and total size of the memory blocks in @buffer.
+
+
+
+
+
+ a #GstBuffer.
+
+
+
+ the offset adjustment
+
+
+
+ the new size or -1 to just adjust the offset
+
+
+
+
+
+ Set the total size of the @length memory blocks starting at @idx in
+@buffer
+
+ %TRUE if resizing succeeded, %FALSE otherwise.
+
+
+
+
+ a #GstBuffer.
+
+
+
+ an index
+
+
+
+ a length
+
+
+
+ the offset adjustment
+
+
+
+ the new size or -1 to just adjust the offset
+
+
+
+
+
+ Sets one or more buffer flags on a buffer.
+
+ %TRUE if @flags were successfully set on buffer.
+
+
+
+
+ a #GstBuffer
+
+
+
+ the #GstBufferFlags to set.
+
+
+
+
+
+ Set the total size of the memory blocks in @buffer.
+
+
+
+
+
+ a #GstBuffer.
+
+
+
+ the new size
+
+
+
+
+
+ Release the memory previously mapped with gst_buffer_map().
+
+
+
+
+
+ a #GstBuffer.
+
+
+
+ a #GstMapInfo
+
+
+
+
+
+ Clears one or more buffer flags.
+
+ true if @flags is successfully cleared from buffer.
+
+
+
+
+ a #GstBuffer
+
+
+
+ the #GstBufferFlags to clear
+
+
+
+
+
+ Get the maximum amount of memory blocks that a buffer can hold. This is a
+compile time constant that can be queried with the function.
+
+When more memory blocks are added, existing memory blocks will be merged
+together to make room for the new block.
+
+ the maximum amount of memory blocks that a buffer can hold.
+
+
+
+
+
+ A set of flags that can be provided to the gst_buffer_copy_into()
+function to specify which items should be copied.
+
+ copy nothing
+
+
+ flag indicating that buffer flags should be copied
+
+
+ flag indicating that buffer pts, dts,
+ duration, offset and offset_end should be copied
+
+
+ flag indicating that buffer meta should be
+ copied
+
+
+ flag indicating that buffer memory should be reffed
+ and appended to already existing memory. Unless the memory is marked as
+ NO_SHARE, no actual copy of the memory is made but it is simply reffed.
+ Add @GST_BUFFER_COPY_DEEP to force a real copy.
+
+
+ flag indicating that buffer memory should be
+ merged
+
+
+ flag indicating that memory should always be
+ copied instead of reffed (Since 1.2)
+
+
+
+ A set of buffer flags used to describe properties of a #GstBuffer.
+
+ the buffer is live data and should be discarded in
+ the PAUSED state.
+
+
+ the buffer contains data that should be dropped
+ because it will be clipped against the segment
+ boundaries or because it does not contain data
+ that should be shown to the user.
+
+
+ the buffer marks a data discontinuity in the stream.
+ This typically occurs after a seek or a dropped buffer
+ from a live or network source.
+
+
+ the buffer timestamps might have a discontinuity
+ and this buffer is a good point to resynchronize.
+
+
+ the buffer data is corrupted.
+
+
+ the buffer contains a media specific marker. for
+ video this is typically the end of a frame boundary, for audio
+ this is usually the start of a talkspurt.
+
+
+ the buffer contains header information that is
+ needed to decode the following data.
+
+
+ the buffer has been created to fill a gap in the
+ stream and contains media neutral data (elements can
+ switch to optimized code path that ignores the buffer
+ content).
+
+
+ the buffer can be dropped without breaking the
+ stream, for example to reduce bandwidth.
+
+
+ this unit cannot be decoded independently.
+
+
+ this flag is set when memory of the buffer
+ is added/removed
+
+
+ Elements which write to disk or permanent
+ storage should ensure the data is synced after
+ writing the contents of this buffer. (Since 1.6)
+
+
+ This buffer is important and should not be dropped.
+ This can be used to mark important buffers, e.g. to flag
+ RTP packets carrying keyframes or codec setup data for RTP
+ Forward Error Correction purposes, or to prevent still video
+ frames from being dropped by elements due to QoS. (Since 1.14)
+
+
+ additional media specific flags can be added starting from
+ this flag.
+
+
+
+ A function that will be called from gst_buffer_foreach_meta(). The @meta
+field will point to a the reference of the meta.
+
+@buffer should not be modified from this callback.
+
+When this function returns %TRUE, the next meta will be
+returned. When %FALSE is returned, gst_buffer_foreach_meta() will return.
+
+When @meta is set to %NULL, the item will be removed from the buffer.
+
+ %FALSE when gst_buffer_foreach_meta() should stop
+
+
+
+
+ a #GstBuffer
+
+
+
+ a pointer to a #GstMeta
+
+
+
+ user data passed to gst_buffer_foreach_meta()
+
+
+
+
+
+ Buffer lists are an object containing a list of buffers.
+
+Buffer lists are created with gst_buffer_list_new() and filled with data
+using a gst_buffer_list_insert().
+
+Buffer lists can be pushed on a srcpad with gst_pad_push_list(). This is
+interesting when multiple buffers need to be pushed in one go because it
+can reduce the amount of overhead for pushing each buffer individually.
+
+ Creates a new, empty #GstBufferList. The caller is responsible for unreffing
+the returned #GstBufferList.
+
+Free-function: gst_buffer_list_unref
+
+ the new #GstBufferList. gst_buffer_list_unref()
+ after usage.
+
+
+
+
+ Creates a new, empty #GstBufferList. The caller is responsible for unreffing
+the returned #GstBufferList. The list will have @size space preallocated so
+that memory reallocations can be avoided.
+
+Free-function: gst_buffer_list_unref
+
+ the new #GstBufferList. gst_buffer_list_unref()
+ after usage.
+
+
+
+
+ an initial reserved size
+
+
+
+
+
+ Calculates the size of the data contained in buffer list by adding the
+size of all buffers.
+
+ the size of the data contained in buffer list in bytes.
+
+
+
+
+ a #GstBufferList
+
+
+
+
+
+ Create a copy of the given buffer list. This will make a newly allocated
+copy of the buffer that the source buffer list contains.
+
+ a new copy of @list.
+
+
+
+
+ a #GstBufferList
+
+
+
+
+
+ Call @func with @data for each buffer in @list.
+
+@func can modify the passed buffer pointer or its contents. The return value
+of @func define if this function returns or if the remaining buffers in
+the list should be skipped.
+
+ %TRUE when @func returned %TRUE for each buffer in @list or when
+@list is empty.
+
+
+
+
+ a #GstBufferList
+
+
+
+ a #GstBufferListFunc to call
+
+
+
+ user data passed to @func
+
+
+
+
+
+ Get the buffer at @idx.
+
+You must make sure that @idx does not exceed the number of
+buffers available.
+
+ the buffer at @idx in @group
+ or %NULL when there is no buffer. The buffer remains valid as
+ long as @list is valid and buffer is not removed from the list.
+
+
+
+
+ a #GstBufferList
+
+
+
+ the index
+
+
+
+
+
+ Gets the buffer at @idx, ensuring it is a writable buffer.
+
+You must make sure that @idx does not exceed the number of
+buffers available.
+
+ the buffer at @idx in @group.
+ The returned buffer remains valid as long as @list is valid and
+ the buffer is not removed from the list.
+
+
+
+
+ a (writable) #GstBufferList
+
+
+
+ the index
+
+
+
+
+
+ Insert @buffer at @idx in @list. Other buffers are moved to make room for
+this new buffer.
+
+A -1 value for @idx will append the buffer at the end.
+
+
+
+
+
+ a #GstBufferList
+
+
+
+ the index
+
+
+
+ a #GstBuffer
+
+
+
+
+
+ Returns the number of buffers in @list.
+
+ the number of buffers in the buffer list
+
+
+
+
+ a #GstBufferList
+
+
+
+
+
+ Remove @length buffers starting from @idx in @list. The following buffers
+are moved to close the gap.
+
+
+
+
+
+ a #GstBufferList
+
+
+
+ the index
+
+
+
+ the amount to remove
+
+
+
+
+
+
+ A function that will be called from gst_buffer_list_foreach(). The @buffer
+field will point to a the reference of the buffer at @idx.
+
+When this function returns %TRUE, the next buffer will be
+returned. When %FALSE is returned, gst_buffer_list_foreach() will return.
+
+When @buffer is set to %NULL, the item will be removed from the bufferlist.
+When @buffer has been made writable, the new buffer reference can be assigned
+to @buffer. This function is responsible for unreffing the old buffer when
+removing or modifying.
+
+ %FALSE when gst_buffer_list_foreach() should stop
+
+
+
+
+ pointer the buffer
+
+
+
+ the index of @buffer
+
+
+
+ user data passed to gst_buffer_list_foreach()
+
+
+
+
+
+ A #GstBufferPool is an object that can be used to pre-allocate and recycle
+buffers of the same size and with the same properties.
+
+A #GstBufferPool is created with gst_buffer_pool_new().
+
+Once a pool is created, it needs to be configured. A call to
+gst_buffer_pool_get_config() returns the current configuration structure from
+the pool. With gst_buffer_pool_config_set_params() and
+gst_buffer_pool_config_set_allocator() the bufferpool parameters and
+allocator can be configured. Other properties can be configured in the pool
+depending on the pool implementation.
+
+A bufferpool can have extra options that can be enabled with
+gst_buffer_pool_config_add_option(). The available options can be retrieved
+with gst_buffer_pool_get_options(). Some options allow for additional
+configuration properties to be set.
+
+After the configuration structure has been configured,
+gst_buffer_pool_set_config() updates the configuration in the pool. This can
+fail when the configuration structure is not accepted.
+
+After the a pool has been configured, it can be activated with
+gst_buffer_pool_set_active(). This will preallocate the configured resources
+in the pool.
+
+When the pool is active, gst_buffer_pool_acquire_buffer() can be used to
+retrieve a buffer from the pool.
+
+Buffers allocated from a bufferpool will automatically be returned to the
+pool with gst_buffer_pool_release_buffer() when their refcount drops to 0.
+
+The bufferpool can be deactivated again with gst_buffer_pool_set_active().
+All further gst_buffer_pool_acquire_buffer() calls will return an error. When
+all buffers are returned to the pool they will be freed.
+
+Use gst_object_unref() to release the reference to a bufferpool. If the
+refcount of the pool reaches 0, the pool will be freed.
+
+ Creates a new #GstBufferPool instance.
+
+ a new #GstBufferPool instance
+
+
+
+
+ Enabled the option in @config. This will instruct the @bufferpool to enable
+the specified option on the buffers that it allocates.
+
+The supported options by @pool can be retrieved with gst_buffer_pool_get_options().
+
+
+
+
+
+ a #GstBufferPool configuration
+
+
+
+ an option to add
+
+
+
+
+
+ Get the @allocator and @params from @config.
+
+ %TRUE, if the values are set.
+
+
+
+
+ a #GstBufferPool configuration
+
+
+
+ a #GstAllocator, or %NULL
+
+
+
+ #GstAllocationParams, or %NULL
+
+
+
+
+
+ Parse an available @config and get the option at @index of the options API
+array.
+
+ a #gchar of the option at @index.
+
+
+
+
+ a #GstBufferPool configuration
+
+
+
+ position in the option array to read
+
+
+
+
+
+ Get the configuration values from @config.
+
+ %TRUE if all parameters could be fetched.
+
+
+
+
+ a #GstBufferPool configuration
+
+
+
+ the caps of buffers
+
+
+
+ the size of each buffer, not including prefix and padding
+
+
+
+ the minimum amount of buffers to allocate.
+
+
+
+ the maximum amount of buffers to allocate or 0 for unlimited.
+
+
+
+
+
+ Check if @config contains @option.
+
+ %TRUE if the options array contains @option.
+
+
+
+
+ a #GstBufferPool configuration
+
+
+
+ an option
+
+
+
+
+
+ Retrieve the number of values currently stored in the options array of the
+@config structure.
+
+ the options array size as a #guint.
+
+
+
+
+ a #GstBufferPool configuration
+
+
+
+
+
+ Set the @allocator and @params on @config.
+
+One of @allocator and @params can be %NULL, but not both. When @allocator
+is %NULL, the default allocator of the pool will use the values in @param
+to perform its allocation. When @param is %NULL, the pool will use the
+provided @allocator with its default #GstAllocationParams.
+
+A call to gst_buffer_pool_set_config() can update the allocator and params
+with the values that it is able to do. Some pools are, for example, not able
+to operate with different allocators or cannot allocate with the values
+specified in @params. Use gst_buffer_pool_get_config() to get the currently
+used values.
+
+
+
+
+
+ a #GstBufferPool configuration
+
+
+
+ a #GstAllocator
+
+
+
+ #GstAllocationParams
+
+
+
+
+
+ Configure @config with the given parameters.
+
+
+
+
+
+ a #GstBufferPool configuration
+
+
+
+ caps for the buffers
+
+
+
+ the size of each buffer, not including prefix and padding
+
+
+
+ the minimum amount of buffers to allocate.
+
+
+
+ the maximum amount of buffers to allocate or 0 for unlimited.
+
+
+
+
+
+ Validate that changes made to @config are still valid in the context of the
+expected parameters. This function is a helper that can be used to validate
+changes made by a pool to a config when gst_buffer_pool_set_config()
+returns %FALSE. This expects that @caps haven't changed and that
+@min_buffers aren't lower then what we initially expected.
+This does not check if options or allocator parameters are still valid,
+won't check if size have changed, since changing the size is valid to adapt
+padding.
+
+ %TRUE, if the parameters are valid in this context.
+
+
+
+
+ a #GstBufferPool configuration
+
+
+
+ the excepted caps of buffers
+
+
+
+ the expected size of each buffer, not including prefix and padding
+
+
+
+ the expected minimum amount of buffers to allocate.
+
+
+
+ the expect maximum amount of buffers to allocate or 0 for unlimited.
+
+
+
+
+
+ Acquire a buffer from @pool. @buffer should point to a memory location that
+can hold a pointer to the new buffer.
+
+@params can be %NULL or contain optional parameters to influence the
+allocation.
+
+ a #GstFlowReturn such as %GST_FLOW_FLUSHING when the pool is
+inactive.
+
+
+
+
+ a #GstBufferPool
+
+
+
+ a location for a #GstBuffer
+
+
+
+ parameters.
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Get a %NULL terminated array of string with supported bufferpool options for
+@pool. An option would typically be enabled with
+gst_buffer_pool_config_add_option().
+
+ a %NULL terminated array
+ of strings.
+
+
+
+
+
+
+ a #GstBufferPool
+
+
+
+
+
+ Release @buffer to @pool. @buffer should have previously been allocated from
+@pool with gst_buffer_pool_acquire_buffer().
+
+This function is usually called automatically when the last ref on @buffer
+disappears.
+
+
+
+
+
+ a #GstBufferPool
+
+
+
+ a #GstBuffer
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Set the configuration of the pool. If the pool is already configured, and
+the configuration haven't change, this function will return %TRUE. If the
+pool is active, this method will return %FALSE and active configuration
+will remain. Buffers allocated form this pool must be returned or else this
+function will do nothing and return %FALSE.
+
+@config is a #GstStructure that contains the configuration parameters for
+the pool. A default and mandatory set of parameters can be configured with
+gst_buffer_pool_config_set_params(), gst_buffer_pool_config_set_allocator()
+and gst_buffer_pool_config_add_option().
+
+If the parameters in @config can not be set exactly, this function returns
+%FALSE and will try to update as much state as possible. The new state can
+then be retrieved and refined with gst_buffer_pool_get_config().
+
+This function takes ownership of @config.
+
+ %TRUE when the configuration could be set.
+
+
+
+
+ a #GstBufferPool
+
+
+
+ a #GstStructure
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Acquire a buffer from @pool. @buffer should point to a memory location that
+can hold a pointer to the new buffer.
+
+@params can be %NULL or contain optional parameters to influence the
+allocation.
+
+ a #GstFlowReturn such as %GST_FLOW_FLUSHING when the pool is
+inactive.
+
+
+
+
+ a #GstBufferPool
+
+
+
+ a location for a #GstBuffer
+
+
+
+ parameters.
+
+
+
+
+
+ Get a copy of the current configuration of the pool. This configuration
+can either be modified and used for the gst_buffer_pool_set_config() call
+or it must be freed after usage.
+
+ a copy of the current configuration of @pool. use
+gst_structure_free() after usage or gst_buffer_pool_set_config().
+
+
+
+
+ a #GstBufferPool
+
+
+
+
+
+ Get a %NULL terminated array of string with supported bufferpool options for
+@pool. An option would typically be enabled with
+gst_buffer_pool_config_add_option().
+
+ a %NULL terminated array
+ of strings.
+
+
+
+
+
+
+ a #GstBufferPool
+
+
+
+
+
+ Check if the bufferpool supports @option.
+
+ %TRUE if the buffer pool contains @option.
+
+
+
+
+ a #GstBufferPool
+
+
+
+ an option
+
+
+
+
+
+ Check if @pool is active. A pool can be activated with the
+gst_buffer_pool_set_active() call.
+
+ %TRUE when the pool is active.
+
+
+
+
+ a #GstBufferPool
+
+
+
+
+
+ Release @buffer to @pool. @buffer should have previously been allocated from
+@pool with gst_buffer_pool_acquire_buffer().
+
+This function is usually called automatically when the last ref on @buffer
+disappears.
+
+
+
+
+
+ a #GstBufferPool
+
+
+
+ a #GstBuffer
+
+
+
+
+
+ Control the active state of @pool. When the pool is inactive, new calls to
+gst_buffer_pool_acquire_buffer() will return with %GST_FLOW_FLUSHING.
+
+Activating the bufferpool will preallocate all resources in the pool based on
+the configuration of the pool.
+
+Deactivating will free the resources again when there are no outstanding
+buffers. When there are outstanding buffers, they will be freed as soon as
+they are all returned to the pool.
+
+ %FALSE when the pool was not configured or when preallocation of the
+buffers failed.
+
+
+
+
+ a #GstBufferPool
+
+
+
+ the new active state
+
+
+
+
+
+ Set the configuration of the pool. If the pool is already configured, and
+the configuration haven't change, this function will return %TRUE. If the
+pool is active, this method will return %FALSE and active configuration
+will remain. Buffers allocated form this pool must be returned or else this
+function will do nothing and return %FALSE.
+
+@config is a #GstStructure that contains the configuration parameters for
+the pool. A default and mandatory set of parameters can be configured with
+gst_buffer_pool_config_set_params(), gst_buffer_pool_config_set_allocator()
+and gst_buffer_pool_config_add_option().
+
+If the parameters in @config can not be set exactly, this function returns
+%FALSE and will try to update as much state as possible. The new state can
+then be retrieved and refined with gst_buffer_pool_get_config().
+
+This function takes ownership of @config.
+
+ %TRUE when the configuration could be set.
+
+
+
+
+ a #GstBufferPool
+
+
+
+ a #GstStructure
+
+
+
+
+
+ Enable or disable the flushing state of a @pool without freeing or
+allocating buffers.
+
+
+
+
+
+ a #GstBufferPool
+
+
+
+ whether to start or stop flushing
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Additional flags to control the allocation of a buffer
+
+ no flags
+
+
+ buffer is keyframe
+
+
+ when the bufferpool is empty, acquire_buffer
+will by default block until a buffer is released into the pool again. Setting
+this flag makes acquire_buffer return #GST_FLOW_EOS instead of blocking.
+
+
+ buffer is discont
+
+
+ last flag, subclasses can use private flags
+ starting from this value.
+
+
+
+ Parameters passed to the gst_buffer_pool_acquire_buffer() function to control the
+allocation of the buffer.
+
+The default implementation ignores the @start and @stop members but other
+implementations can use this extra information to decide what buffer to
+return.
+
+ the format of @start and @stop
+
+
+
+ the start position
+
+
+
+ the stop position
+
+
+
+ additional flags
+
+
+
+
+
+
+
+
+
+ The GstBufferPool class.
+
+ Object parent class
+
+
+
+
+
+ a %NULL terminated array
+ of strings.
+
+
+
+
+
+
+ a #GstBufferPool
+
+
+
+
+
+
+
+
+ %TRUE when the configuration could be set.
+
+
+
+
+ a #GstBufferPool
+
+
+
+ a #GstStructure
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ a #GstFlowReturn such as %GST_FLOW_FLUSHING when the pool is
+inactive.
+
+
+
+
+ a #GstBufferPool
+
+
+
+ a location for a #GstBuffer
+
+
+
+ parameters.
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ a #GstBufferPool
+
+
+
+ a #GstBuffer
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ The different types of buffering methods.
+
+ a small amount of data is buffered
+
+
+ the stream is being downloaded
+
+
+ the stream is being downloaded in a ringbuffer
+
+
+ the stream is a live stream
+
+
+
+ The #GstBus is an object responsible for delivering #GstMessage packets in
+a first-in first-out way from the streaming threads (see #GstTask) to the
+application.
+
+Since the application typically only wants to deal with delivery of these
+messages from one thread, the GstBus will marshall the messages between
+different threads. This is important since the actual streaming of media
+is done in another thread than the application.
+
+The GstBus provides support for #GSource based notifications. This makes it
+possible to handle the delivery in the glib mainloop.
+
+The #GSource callback function gst_bus_async_signal_func() can be used to
+convert all bus messages into signal emissions.
+
+A message is posted on the bus with the gst_bus_post() method. With the
+gst_bus_peek() and gst_bus_pop() methods one can look at or retrieve a
+previously posted message.
+
+The bus can be polled with the gst_bus_poll() method. This methods blocks
+up to the specified timeout value until one of the specified messages types
+is posted on the bus. The application can then gst_bus_pop() the messages
+from the bus to handle them.
+Alternatively the application can register an asynchronous bus function
+using gst_bus_add_watch_full() or gst_bus_add_watch(). This function will
+install a #GSource in the default glib main loop and will deliver messages
+a short while after they have been posted. Note that the main loop should
+be running for the asynchronous callbacks.
+
+It is also possible to get messages from the bus without any thread
+marshalling with the gst_bus_set_sync_handler() method. This makes it
+possible to react to a message in the same thread that posted the
+message on the bus. This should only be used if the application is able
+to deal with messages from different threads.
+
+Every #GstPipeline has one bus.
+
+Note that a #GstPipeline will set its bus into flushing state when changing
+from READY to NULL state.
+
+ Creates a new #GstBus instance.
+
+ a new #GstBus instance
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Adds a bus signal watch to the default main context with the default priority
+(%G_PRIORITY_DEFAULT). It is also possible to use a non-default
+main context set up using g_main_context_push_thread_default() (before
+one had to create a bus watch source and attach it to the desired main
+context 'manually').
+
+After calling this statement, the bus will emit the "message" signal for each
+message posted on the bus.
+
+This function may be called multiple times. To clean up, the caller is
+responsible for calling gst_bus_remove_signal_watch() as many times as this
+function is called.
+
+MT safe.
+
+
+
+
+
+ a #GstBus on which you want to receive the "message" signal
+
+
+
+
+
+ Adds a bus signal watch to the default main context with the given @priority
+(e.g. %G_PRIORITY_DEFAULT). It is also possible to use a non-default main
+context set up using g_main_context_push_thread_default()
+(before one had to create a bus watch source and attach it to the desired
+main context 'manually').
+
+After calling this statement, the bus will emit the "message" signal for each
+message posted on the bus when the main loop is running.
+
+This function may be called multiple times. To clean up, the caller is
+responsible for calling gst_bus_remove_signal_watch() as many times as this
+function is called.
+
+There can only be a single bus watch per bus, you must remove any signal
+watch before you can set another type of watch.
+
+MT safe.
+
+
+
+
+
+ a #GstBus on which you want to receive the "message" signal
+
+
+
+ The priority of the watch.
+
+
+
+
+
+ Adds a bus watch to the default main context with the default priority
+(%G_PRIORITY_DEFAULT). It is also possible to use a non-default main
+context set up using g_main_context_push_thread_default() (before
+one had to create a bus watch source and attach it to the desired main
+context 'manually').
+
+This function is used to receive asynchronous messages in the main loop.
+There can only be a single bus watch per bus, you must remove it before you
+can set a new one.
+
+The bus watch will only work if a GLib main loop is being run.
+
+The watch can be removed using gst_bus_remove_watch() or by returning %FALSE
+from @func. If the watch was added to the default main context it is also
+possible to remove the watch using g_source_remove().
+
+The bus watch will take its own reference to the @bus, so it is safe to unref
+@bus using gst_object_unref() after setting the bus watch.
+
+MT safe.
+
+ The event source id or 0 if @bus already got an event source.
+
+
+
+
+ a #GstBus to create the watch for
+
+
+
+ A function to call when a message is received.
+
+
+
+ user data passed to @func.
+
+
+
+
+
+ Adds a bus watch to the default main context with the given @priority (e.g.
+%G_PRIORITY_DEFAULT). It is also possible to use a non-default main
+context set up using g_main_context_push_thread_default() (before
+one had to create a bus watch source and attach it to the desired main
+context 'manually').
+
+This function is used to receive asynchronous messages in the main loop.
+There can only be a single bus watch per bus, you must remove it before you
+can set a new one.
+
+The bus watch will only work if a GLib main loop is being run.
+
+When @func is called, the message belongs to the caller; if you want to
+keep a copy of it, call gst_message_ref() before leaving @func.
+
+The watch can be removed using gst_bus_remove_watch() or by returning %FALSE
+from @func. If the watch was added to the default main context it is also
+possible to remove the watch using g_source_remove().
+
+The bus watch will take its own reference to the @bus, so it is safe to unref
+@bus using gst_object_unref() after setting the bus watch.
+
+MT safe.
+
+ The event source id or 0 if @bus already got an event source.
+
+
+
+
+ a #GstBus to create the watch for.
+
+
+
+ The priority of the watch.
+
+
+
+ A function to call when a message is received.
+
+
+
+ user data passed to @func.
+
+
+
+ the function to call when the source is removed.
+
+
+
+
+
+ A helper #GstBusFunc that can be used to convert all asynchronous messages
+into signals.
+
+ %TRUE
+
+
+
+
+ a #GstBus
+
+
+
+ the #GstMessage received
+
+
+
+ user data
+
+
+
+
+
+ Create watch for this bus. The GSource will be dispatched whenever
+a message is on the bus. After the GSource is dispatched, the
+message is popped off the bus and unreffed.
+
+ a #GSource that can be added to a mainloop.
+
+
+
+
+ a #GstBus to create the watch for
+
+
+
+
+
+ Instructs GStreamer to stop emitting the "sync-message" signal for this bus.
+See gst_bus_enable_sync_message_emission() for more information.
+
+In the event that multiple pieces of code have called
+gst_bus_enable_sync_message_emission(), the sync-message emissions will only
+be stopped after all calls to gst_bus_enable_sync_message_emission() were
+"cancelled" by calling this function. In this way the semantics are exactly
+the same as gst_object_ref() that which calls enable should also call
+disable.
+
+MT safe.
+
+
+
+
+
+ a #GstBus on which you previously called
+gst_bus_enable_sync_message_emission()
+
+
+
+
+
+ Instructs GStreamer to emit the "sync-message" signal after running the bus's
+sync handler. This function is here so that code can ensure that they can
+synchronously receive messages without having to affect what the bin's sync
+handler is.
+
+This function may be called multiple times. To clean up, the caller is
+responsible for calling gst_bus_disable_sync_message_emission() as many times
+as this function is called.
+
+While this function looks similar to gst_bus_add_signal_watch(), it is not
+exactly the same -- this function enables <emphasis>synchronous</emphasis> emission of
+signals when messages arrive; gst_bus_add_signal_watch() adds an idle callback
+to pop messages off the bus <emphasis>asynchronously</emphasis>. The sync-message signal
+comes from the thread of whatever object posted the message; the "message"
+signal is marshalled to the main thread via the main loop.
+
+MT safe.
+
+
+
+
+
+ a #GstBus on which you want to receive the "sync-message" signal
+
+
+
+
+
+ Gets the file descriptor from the bus which can be used to get notified about
+messages being available with functions like g_poll(), and allows integration
+into other event loops based on file descriptors.
+Whenever a message is available, the POLLIN / %G_IO_IN event is set.
+
+Warning: NEVER read or write anything to the returned fd but only use it
+for getting notifications via g_poll() or similar and then use the normal
+GstBus API, e.g. gst_bus_pop().
+
+
+
+
+
+ A #GstBus
+
+
+
+ A GPollFD to fill
+
+
+
+
+
+ Check if there are pending messages on the bus that
+should be handled.
+
+ %TRUE if there are messages on the bus to be handled, %FALSE
+otherwise.
+
+MT safe.
+
+
+
+
+ a #GstBus to check
+
+
+
+
+
+ Peek the message on the top of the bus' queue. The message will remain
+on the bus' message queue. A reference is returned, and needs to be unreffed
+by the caller.
+
+ the #GstMessage that is on the
+ bus, or %NULL if the bus is empty.
+
+MT safe.
+
+
+
+
+ a #GstBus
+
+
+
+
+
+ Poll the bus for messages. Will block while waiting for messages to come.
+You can specify a maximum time to poll with the @timeout parameter. If
+@timeout is negative, this function will block indefinitely.
+
+All messages not in @events will be popped off the bus and will be ignored.
+It is not possible to use message enums beyond #GST_MESSAGE_EXTENDED in the
+@events mask
+
+Because poll is implemented using the "message" signal enabled by
+gst_bus_add_signal_watch(), calling gst_bus_poll() will cause the "message"
+signal to be emitted for every message that poll sees. Thus a "message"
+signal handler will see the same messages that this function sees -- neither
+will steal messages from the other.
+
+This function will run a main loop from the default main context when
+polling.
+
+You should never use this function, since it is pure evil. This is
+especially true for GUI applications based on Gtk+ or Qt, but also for any
+other non-trivial application that uses the GLib main loop. As this function
+runs a GLib main loop, any callback attached to the default GLib main
+context may be invoked. This could be timeouts, GUI events, I/O events etc.;
+even if gst_bus_poll() is called with a 0 timeout. Any of these callbacks
+may do things you do not expect, e.g. destroy the main application window or
+some other resource; change other application state; display a dialog and
+run another main loop until the user clicks it away. In short, using this
+function may add a lot of complexity to your code through unexpected
+re-entrancy and unexpected changes to your application's state.
+
+For 0 timeouts use gst_bus_pop_filtered() instead of this function; for
+other short timeouts use gst_bus_timed_pop_filtered(); everything else is
+better handled by setting up an asynchronous bus watch and doing things
+from there.
+
+ the message that was received,
+ or %NULL if the poll timed out. The message is taken from the
+ bus and needs to be unreffed with gst_message_unref() after
+ usage.
+
+
+
+
+ a #GstBus
+
+
+
+ a mask of #GstMessageType, representing the set of message types to
+poll for (note special handling of extended message types below)
+
+
+
+ the poll timeout, as a #GstClockTime, or #GST_CLOCK_TIME_NONE to poll
+indefinitely.
+
+
+
+
+
+ Get a message from the bus.
+
+ the #GstMessage that is on the
+ bus, or %NULL if the bus is empty. The message is taken from
+ the bus and needs to be unreffed with gst_message_unref() after
+ usage.
+
+MT safe.
+
+
+
+
+ a #GstBus to pop
+
+
+
+
+
+ Get a message matching @type from the bus. Will discard all messages on
+the bus that do not match @type and that have been posted before the first
+message that does match @type. If there is no message matching @type on
+the bus, all messages will be discarded. It is not possible to use message
+enums beyond #GST_MESSAGE_EXTENDED in the @events mask.
+
+ the next #GstMessage matching
+ @type that is on the bus, or %NULL if the bus is empty or there
+ is no message matching @type. The message is taken from the bus
+ and needs to be unreffed with gst_message_unref() after usage.
+
+MT safe.
+
+
+
+
+ a #GstBus to pop
+
+
+
+ message types to take into account
+
+
+
+
+
+ Post a message on the given bus. Ownership of the message
+is taken by the bus.
+
+ %TRUE if the message could be posted, %FALSE if the bus is flushing.
+
+MT safe.
+
+
+
+
+ a #GstBus to post on
+
+
+
+ the #GstMessage to post
+
+
+
+
+
+ Removes a signal watch previously added with gst_bus_add_signal_watch().
+
+MT safe.
+
+
+
+
+
+ a #GstBus you previously added a signal watch to
+
+
+
+
+
+ Removes an installed bus watch from @bus.
+
+ %TRUE on success or %FALSE if @bus has no event source.
+
+
+
+
+ a #GstBus to remove the watch from.
+
+
+
+
+
+ If @flushing, flush out and unref any messages queued in the bus. Releases
+references to the message origin objects. Will flush future messages until
+gst_bus_set_flushing() sets @flushing to %FALSE.
+
+MT safe.
+
+
+
+
+
+ a #GstBus
+
+
+
+ whether or not to flush the bus
+
+
+
+
+
+ Sets the synchronous handler on the bus. The function will be called
+every time a new message is posted on the bus. Note that the function
+will be called in the same thread context as the posting object. This
+function is usually only called by the creator of the bus. Applications
+should handle messages asynchronously using the gst_bus watch and poll
+functions.
+
+You cannot replace an existing sync_handler. You can pass %NULL to this
+function, which will clear the existing handler.
+
+
+
+
+
+ a #GstBus to install the handler on
+
+
+
+ The handler function to install
+
+
+
+ User data that will be sent to the handler function.
+
+
+
+ called when @user_data becomes unused
+
+
+
+
+
+ A helper GstBusSyncHandler that can be used to convert all synchronous
+messages into signals.
+
+ GST_BUS_PASS
+
+
+
+
+ a #GstBus
+
+
+
+ the #GstMessage received
+
+
+
+ user data
+
+
+
+
+
+ Get a message from the bus, waiting up to the specified timeout.
+
+If @timeout is 0, this function behaves like gst_bus_pop(). If @timeout is
+#GST_CLOCK_TIME_NONE, this function will block forever until a message was
+posted on the bus.
+
+ the #GstMessage that is on the
+ bus after the specified timeout or %NULL if the bus is empty
+ after the timeout expired. The message is taken from the bus
+ and needs to be unreffed with gst_message_unref() after usage.
+
+MT safe.
+
+
+
+
+ a #GstBus to pop
+
+
+
+ a timeout
+
+
+
+
+
+ Get a message from the bus whose type matches the message type mask @types,
+waiting up to the specified timeout (and discarding any messages that do not
+match the mask provided).
+
+If @timeout is 0, this function behaves like gst_bus_pop_filtered(). If
+@timeout is #GST_CLOCK_TIME_NONE, this function will block forever until a
+matching message was posted on the bus.
+
+ a #GstMessage matching the
+ filter in @types, or %NULL if no matching message was found on
+ the bus until the timeout expired. The message is taken from
+ the bus and needs to be unreffed with gst_message_unref() after
+ usage.
+
+MT safe.
+
+
+
+
+ a #GstBus to pop from
+
+
+
+ a timeout in nanoseconds, or GST_CLOCK_TIME_NONE to wait forever
+
+
+
+ message types to take into account, GST_MESSAGE_ANY for any type
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ A message has been posted on the bus. This signal is emitted from a
+GSource added to the mainloop. this signal will only be emitted when
+there is a mainloop running.
+
+
+
+
+
+ the message that has been posted asynchronously
+
+
+
+
+
+ A message has been posted on the bus. This signal is emitted from the
+thread that posted the message so one has to be careful with locking.
+
+This signal will not be emitted by default, you have to call
+gst_bus_enable_sync_message_emission() before.
+
+
+
+
+
+ the message that has been posted synchronously
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ The standard flags that a bus may have.
+
+ The bus is currently dropping all messages
+
+
+ offset to define more flags
+
+
+
+ Specifies the type of function passed to gst_bus_add_watch() or
+gst_bus_add_watch_full(), which is called from the mainloop when a message
+is available on the bus.
+
+The message passed to the function will be unreffed after execution of this
+function so it should not be freed in the function.
+
+Note that this function is used as a GSourceFunc which means that returning
+%FALSE will remove the GSource from the mainloop.
+
+ %FALSE if the event source should be removed.
+
+
+
+
+ the #GstBus that sent the message
+
+
+
+ the #GstMessage
+
+
+
+ user data that has been given, when registering the handler
+
+
+
+
+
+
+
+ Handler will be invoked synchronously, when a new message has been injected
+into the bus. This function is mostly used internally. Only one sync handler
+can be attached to a given bus.
+
+If the handler returns GST_BUS_DROP, it should unref the message, else the
+message should not be unreffed by the sync handler.
+
+ #GstBusSyncReply stating what to do with the message
+
+
+
+
+ the #GstBus that sent the message
+
+
+
+ the #GstMessage
+
+
+
+ user data that has been given, when registering the handler
+
+
+
+
+
+ The result values for a GstBusSyncHandler.
+
+ drop the message
+
+
+ pass the message to the async queue
+
+
+ pass message to async queue, continue if message is handled
+
+
+
+
+
+
+
+
+
+ Constant to define an undefined clock time.
+
+
+
+ Caps (capabilities) are lightweight refcounted objects describing media types.
+They are composed of an array of #GstStructure.
+
+Caps are exposed on #GstPadTemplate to describe all possible types a
+given pad can handle. They are also stored in the #GstRegistry along with
+a description of the #GstElement.
+
+Caps are exposed on the element pads using the gst_pad_query_caps() pad
+function. This function describes the possible types that the pad can
+handle or produce at runtime.
+
+A #GstCaps can be constructed with the following code fragment:
+|[<!-- language="C" -->
+ GstCaps *caps = gst_caps_new_simple ("video/x-raw",
+ "format", G_TYPE_STRING, "I420",
+ "framerate", GST_TYPE_FRACTION, 25, 1,
+ "pixel-aspect-ratio", GST_TYPE_FRACTION, 1, 1,
+ "width", G_TYPE_INT, 320,
+ "height", G_TYPE_INT, 240,
+ NULL);
+]|
+
+A #GstCaps is fixed when it has no properties with ranges or lists. Use
+gst_caps_is_fixed() to test for fixed caps. Fixed caps can be used in a
+caps event to notify downstream elements of the current media type.
+
+Various methods exist to work with the media types such as subtracting
+or intersecting.
+
+Be aware that the current #GstCaps / #GstStructure serialization into string
+has limited support for nested #GstCaps / #GstStructure fields. It can only
+support one level of nesting. Using more levels will lead to unexpected
+behavior when using serialization features, such as gst_caps_to_string() or
+gst_value_serialize() and their counterparts.
+
+ the parent type
+
+
+
+ Creates a new #GstCaps that indicates that it is compatible with
+any media format.
+
+ the new #GstCaps
+
+
+
+
+ Creates a new #GstCaps that is empty. That is, the returned
+#GstCaps contains no media formats.
+The #GstCaps is guaranteed to be writable.
+Caller is responsible for unreffing the returned caps.
+
+ the new #GstCaps
+
+
+
+
+ Creates a new #GstCaps that contains one #GstStructure with name
+@media_type.
+Caller is responsible for unreffing the returned caps.
+
+ the new #GstCaps
+
+
+
+
+ the media type of the structure
+
+
+
+
+
+ Creates a new #GstCaps and adds all the structures listed as
+arguments. The list must be %NULL-terminated. The structures
+are not copied; the returned #GstCaps owns the structures.
+
+ the new #GstCaps
+
+
+
+
+ the first structure to add
+
+
+
+ additional structures to add
+
+
+
+
+
+ Creates a new #GstCaps and adds all the structures listed as
+arguments. The list must be %NULL-terminated. The structures
+are not copied; the returned #GstCaps owns the structures.
+
+ the new #GstCaps
+
+
+
+
+ the first structure to add
+
+
+
+ additional structures to add
+
+
+
+
+
+ Creates a new #GstCaps that contains one #GstStructure. The
+structure is defined by the arguments, which have the same format
+as gst_structure_new().
+Caller is responsible for unreffing the returned caps.
+
+ the new #GstCaps
+
+
+
+
+ the media type of the structure
+
+
+
+ first field to set
+
+
+
+ additional arguments
+
+
+
+
+
+ Appends the structures contained in @caps2 to @caps1. The structures in
+@caps2 are not copied -- they are transferred to @caps1, and then @caps2 is
+freed. If either caps is ANY, the resulting caps will be ANY.
+
+
+
+
+
+ the #GstCaps that will be appended to
+
+
+
+ the #GstCaps to append
+
+
+
+
+
+ Appends @structure to @caps. The structure is not copied; @caps
+becomes the owner of @structure.
+
+
+
+
+
+ the #GstCaps that will be appended to
+
+
+
+ the #GstStructure to append
+
+
+
+
+
+ Appends @structure with @features to @caps. The structure is not copied; @caps
+becomes the owner of @structure.
+
+
+
+
+
+ the #GstCaps that will be appended to
+
+
+
+ the #GstStructure to append
+
+
+
+ the #GstCapsFeatures to append
+
+
+
+
+
+ Tries intersecting @caps1 and @caps2 and reports whether the result would not
+be empty
+
+ %TRUE if intersection would be not empty
+
+
+
+
+ a #GstCaps to intersect
+
+
+
+ a #GstCaps to intersect
+
+
+
+
+
+ Creates a new #GstCaps and appends a copy of the nth structure
+contained in @caps.
+
+ the new #GstCaps
+
+
+
+
+ the #GstCaps to copy
+
+
+
+ the nth structure to copy
+
+
+
+
+
+ Calls the provided function once for each structure and caps feature in the
+#GstCaps. In contrast to gst_caps_foreach(), the function may modify the
+structure and features. In contrast to gst_caps_filter_and_map_in_place(),
+the structure and features are removed from the caps if %FALSE is returned
+from the function.
+The caps must be mutable.
+
+
+
+
+
+ a #GstCaps
+
+
+
+ a function to call for each field
+
+
+
+ private data
+
+
+
+
+
+ Modifies the given @caps into a representation with only fixed
+values. First the caps will be truncated and then the first structure will be
+fixated with gst_structure_fixate().
+
+This function takes ownership of @caps and will call gst_caps_make_writable()
+on it so you must not use @caps afterwards unless you keep an additional
+reference to it with gst_caps_ref().
+
+ the fixated caps
+
+
+
+
+ a #GstCaps to fixate
+
+
+
+
+
+ Calls the provided function once for each structure and caps feature in the
+#GstCaps. The function must not modify the fields.
+Also see gst_caps_map_in_place() and gst_caps_filter_and_map_in_place().
+
+ %TRUE if the supplied function returns %TRUE for each call,
+%FALSE otherwise.
+
+
+
+
+ a #GstCaps
+
+
+
+ a function to call for each field
+
+
+
+ private data
+
+
+
+
+
+ Finds the features in @caps that has the index @index, and
+returns it.
+
+WARNING: This function takes a const GstCaps *, but returns a
+non-const GstCapsFeatures *. This is for programming convenience --
+the caller should be aware that structures inside a constant
+#GstCaps should not be modified. However, if you know the caps
+are writable, either because you have just copied them or made
+them writable with gst_caps_make_writable(), you may modify the
+features returned in the usual way, e.g. with functions like
+gst_caps_features_add().
+
+You do not need to free or unref the structure returned, it
+belongs to the #GstCaps.
+
+ a pointer to the #GstCapsFeatures
+ corresponding to @index
+
+
+
+
+ a #GstCaps
+
+
+
+ the index of the structure
+
+
+
+
+
+ Gets the number of structures contained in @caps.
+
+ the number of structures that @caps contains
+
+
+
+
+ a #GstCaps
+
+
+
+
+
+ Finds the structure in @caps that has the index @index, and
+returns it.
+
+WARNING: This function takes a const GstCaps *, but returns a
+non-const GstStructure *. This is for programming convenience --
+the caller should be aware that structures inside a constant
+#GstCaps should not be modified. However, if you know the caps
+are writable, either because you have just copied them or made
+them writable with gst_caps_make_writable(), you may modify the
+structure returned in the usual way, e.g. with functions like
+gst_structure_set().
+
+You do not need to free or unref the structure returned, it
+belongs to the #GstCaps.
+
+ a pointer to the #GstStructure corresponding
+ to @index
+
+
+
+
+ a #GstCaps
+
+
+
+ the index of the structure
+
+
+
+
+
+ Creates a new #GstCaps that contains all the formats that are common
+to both @caps1 and @caps2. Defaults to %GST_CAPS_INTERSECT_ZIG_ZAG mode.
+
+ the new #GstCaps
+
+
+
+
+ a #GstCaps to intersect
+
+
+
+ a #GstCaps to intersect
+
+
+
+
+
+ Creates a new #GstCaps that contains all the formats that are common
+to both @caps1 and @caps2, the order is defined by the #GstCapsIntersectMode
+used.
+
+ the new #GstCaps
+
+
+
+
+ a #GstCaps to intersect
+
+
+
+ a #GstCaps to intersect
+
+
+
+ The intersection algorithm/mode to use
+
+
+
+
+
+ A given #GstCaps structure is always compatible with another if
+every media format that is in the first is also contained in the
+second. That is, @caps1 is a subset of @caps2.
+
+ %TRUE if @caps1 is a subset of @caps2.
+
+
+
+
+ the #GstCaps to test
+
+
+
+ the #GstCaps to test
+
+
+
+
+
+ Determines if @caps represents any media format.
+
+ %TRUE if @caps represents any format.
+
+
+
+
+ the #GstCaps to test
+
+
+
+
+
+ Determines if @caps represents no media formats.
+
+ %TRUE if @caps represents no formats.
+
+
+
+
+ the #GstCaps to test
+
+
+
+
+
+ Checks if the given caps represent the same set of caps.
+
+ %TRUE if both caps are equal.
+
+
+
+
+ a #GstCaps
+
+
+
+ another #GstCaps
+
+
+
+
+
+ Tests if two #GstCaps are equal. This function only works on fixed
+#GstCaps.
+
+ %TRUE if the arguments represent the same format
+
+
+
+
+ the #GstCaps to test
+
+
+
+ the #GstCaps to test
+
+
+
+
+
+ Fixed #GstCaps describe exactly one format, that is, they have exactly
+one structure, and each field in the structure describes a fixed type.
+Examples of non-fixed types are GST_TYPE_INT_RANGE and GST_TYPE_LIST.
+
+ %TRUE if @caps is fixed
+
+
+
+
+ the #GstCaps to test
+
+
+
+
+
+ Checks if the given caps are exactly the same set of caps.
+
+ %TRUE if both caps are strictly equal.
+
+
+
+
+ a #GstCaps
+
+
+
+ another #GstCaps
+
+
+
+
+
+ Checks if all caps represented by @subset are also represented by @superset.
+
+ %TRUE if @subset is a subset of @superset
+
+
+
+
+ a #GstCaps
+
+
+
+ a potentially greater #GstCaps
+
+
+
+
+
+ Checks if @structure is a subset of @caps. See gst_caps_is_subset()
+for more information.
+
+ %TRUE if @structure is a subset of @caps
+
+
+
+
+ a #GstCaps
+
+
+
+ a potential #GstStructure subset of @caps
+
+
+
+
+
+ Checks if @structure is a subset of @caps. See gst_caps_is_subset()
+for more information.
+
+ %TRUE if @structure is a subset of @caps
+
+
+
+
+ a #GstCaps
+
+
+
+ a potential #GstStructure subset of @caps
+
+
+
+ a #GstCapsFeatures for @structure
+
+
+
+
+
+ Calls the provided function once for each structure and caps feature in the
+#GstCaps. In contrast to gst_caps_foreach(), the function may modify but not
+delete the structures and features. The caps must be mutable.
+
+ %TRUE if the supplied function returns %TRUE for each call,
+%FALSE otherwise.
+
+
+
+
+ a #GstCaps
+
+
+
+ a function to call for each field
+
+
+
+ private data
+
+
+
+
+
+ Appends the structures contained in @caps2 to @caps1 if they are not yet
+expressed by @caps1. The structures in @caps2 are not copied -- they are
+transferred to a writable copy of @caps1, and then @caps2 is freed.
+If either caps is ANY, the resulting caps will be ANY.
+
+ the merged caps.
+
+
+
+
+ the #GstCaps that will take the new entries
+
+
+
+ the #GstCaps to merge in
+
+
+
+
+
+ Appends @structure to @caps if its not already expressed by @caps.
+
+ the merged caps.
+
+
+
+
+ the #GstCaps to merge into
+
+
+
+ the #GstStructure to merge
+
+
+
+
+
+ Appends @structure with @features to @caps if its not already expressed by @caps.
+
+ the merged caps.
+
+
+
+
+ the #GstCaps to merge into
+
+
+
+ the #GstStructure to merge
+
+
+
+ the #GstCapsFeatures to merge
+
+
+
+
+
+ Returns a #GstCaps that represents the same set of formats as
+@caps, but contains no lists. Each list is expanded into separate
+@GstStructures.
+
+This function takes ownership of @caps and will call gst_caps_make_writable()
+on it so you must not use @caps afterwards unless you keep an additional
+reference to it with gst_caps_ref().
+
+ the normalized #GstCaps
+
+
+
+
+ a #GstCaps to normalize
+
+
+
+
+
+ removes the structure with the given index from the list of structures
+contained in @caps.
+
+
+
+
+
+ the #GstCaps to remove from
+
+
+
+ Index of the structure to remove
+
+
+
+
+
+ Sets the #GstCapsFeatures @features for the structure at @index.
+
+
+
+
+
+ a #GstCaps
+
+
+
+ the index of the structure
+
+
+
+ the #GstCapsFeatures to set
+
+
+
+
+
+ Sets fields in a #GstCaps. The arguments must be passed in the same
+manner as gst_structure_set(), and be %NULL-terminated.
+
+
+
+
+
+ the #GstCaps to set
+
+
+
+ first field to set
+
+
+
+ additional parameters
+
+
+
+
+
+ Sets fields in a #GstCaps. The arguments must be passed in the same
+manner as gst_structure_set(), and be %NULL-terminated.
+
+
+
+
+
+ the #GstCaps to set
+
+
+
+ first field to set
+
+
+
+ additional parameters
+
+
+
+
+
+ Sets the given @field on all structures of @caps to the given @value.
+This is a convenience function for calling gst_structure_set_value() on
+all structures of @caps.
+
+
+
+
+
+ a writable caps
+
+
+
+ name of the field to set
+
+
+
+ value to set the field to
+
+
+
+
+
+ Converts the given @caps into a representation that represents the
+same set of formats, but in a simpler form. Component structures that are
+identical are merged. Component structures that have values that can be
+merged are also merged.
+
+This function takes ownership of @caps and will call gst_caps_make_writable()
+on it if necessary, so you must not use @caps afterwards unless you keep an
+additional reference to it with gst_caps_ref().
+
+This method does not preserve the original order of @caps.
+
+ The simplified caps.
+
+
+
+
+ a #GstCaps to simplify
+
+
+
+
+
+ Retrieves the structure with the given index from the list of structures
+contained in @caps. The caller becomes the owner of the returned structure.
+
+ a pointer to the #GstStructure
+ corresponding to @index.
+
+
+
+
+ the #GstCaps to retrieve from
+
+
+
+ Index of the structure to retrieve
+
+
+
+
+
+ Subtracts the @subtrahend from the @minuend.
+> This function does not work reliably if optional properties for caps
+> are included on one caps and omitted on the other.
+
+ the resulting caps
+
+
+
+
+ #GstCaps to subtract from
+
+
+
+ #GstCaps to subtract
+
+
+
+
+
+ Converts @caps to a string representation. This string representation
+can be converted back to a #GstCaps by gst_caps_from_string().
+
+For debugging purposes its easier to do something like this:
+|[<!-- language="C" -->
+GST_LOG ("caps are %" GST_PTR_FORMAT, caps);
+]|
+This prints the caps in human readable form.
+
+The current implementation of serialization will lead to unexpected results
+when there are nested #GstCaps / #GstStructure deeper than one level.
+
+ a newly allocated string representing @caps.
+
+
+
+
+ a #GstCaps
+
+
+
+
+
+ Discard all but the first structure from @caps. Useful when
+fixating.
+
+This function takes ownership of @caps and will call gst_caps_make_writable()
+on it if necessary, so you must not use @caps afterwards unless you keep an
+additional reference to it with gst_caps_ref().
+
+ truncated caps
+
+
+
+
+ the #GstCaps to truncate
+
+
+
+
+
+ Converts @caps from a string representation.
+
+The current implementation of serialization will lead to unexpected results
+when there are nested #GstCaps / #GstStructure deeper than one level.
+
+ a newly allocated #GstCaps
+
+
+
+
+ a string to convert to #GstCaps
+
+
+
+
+
+
+ #GstCapsFeatures can optionally be set on a #GstCaps to add requirements
+for additional features for a specific #GstStructure. Caps structures with
+the same name but with a non-equal set of caps features are not compatible.
+If a pad supports multiple sets of features it has to add multiple equal
+structures with different feature sets to the caps.
+
+Empty #GstCapsFeatures are equivalent with the #GstCapsFeatures that only
+contain #GST_CAPS_FEATURE_MEMORY_SYSTEM_MEMORY. ANY #GstCapsFeatures as
+created by gst_caps_features_new_any() are equal to any other #GstCapsFeatures
+and can be used to specify that any #GstCapsFeatures would be supported, e.g.
+for elements that don't touch buffer memory. #GstCaps with ANY #GstCapsFeatures
+are considered non-fixed and during negotiation some #GstCapsFeatures have
+to be selected.
+
+Examples for caps features would be the requirement of a specific #GstMemory
+types or the requirement of having a specific #GstMeta on the buffer. Features
+are given as a string of the format "memory:GstMemoryTypeName" or
+"meta:GstMetaAPIName".
+
+ Creates a new #GstCapsFeatures with the given features.
+The last argument must be %NULL.
+
+Free-function: gst_caps_features_free
+
+ a new, empty #GstCapsFeatures
+
+
+
+
+ name of first feature to set
+
+
+
+ additional features
+
+
+
+
+
+ Creates a new, ANY #GstCapsFeatures. This will be equal
+to any other #GstCapsFeatures but caps with these are
+unfixed.
+
+Free-function: gst_caps_features_free
+
+ a new, ANY #GstCapsFeatures
+
+
+
+
+ Creates a new, empty #GstCapsFeatures.
+
+Free-function: gst_caps_features_free
+
+ a new, empty #GstCapsFeatures
+
+
+
+
+ Creates a new #GstCapsFeatures with the given features.
+The last argument must be 0.
+
+Free-function: gst_caps_features_free
+
+ a new, empty #GstCapsFeatures
+
+
+
+
+ name of first feature to set
+
+
+
+ additional features
+
+
+
+
+
+ Creates a new #GstCapsFeatures with the given features.
+
+Free-function: gst_caps_features_free
+
+ a new, empty #GstCapsFeatures
+
+
+
+
+ name of first feature to set
+
+
+
+ variable argument list
+
+
+
+
+
+ Creates a new #GstCapsFeatures with the given features.
+
+Free-function: gst_caps_features_free
+
+ a new, empty #GstCapsFeatures
+
+
+
+
+ name of first feature to set
+
+
+
+ variable argument list
+
+
+
+
+
+ Adds @feature to @features.
+
+
+
+
+
+ a #GstCapsFeatures.
+
+
+
+ a feature.
+
+
+
+
+
+ Adds @feature to @features.
+
+
+
+
+
+ a #GstCapsFeatures.
+
+
+
+ a feature.
+
+
+
+
+
+ Check if @features contains @feature.
+
+ %TRUE if @features contains @feature.
+
+
+
+
+ a #GstCapsFeatures.
+
+
+
+ a feature
+
+
+
+
+
+ Check if @features contains @feature.
+
+ %TRUE if @features contains @feature.
+
+
+
+
+ a #GstCapsFeatures.
+
+
+
+ a feature
+
+
+
+
+
+ Duplicates a #GstCapsFeatures and all its values.
+
+Free-function: gst_caps_features_free
+
+ a new #GstCapsFeatures.
+
+
+
+
+ a #GstCapsFeatures to duplicate
+
+
+
+
+
+ Frees a #GstCapsFeatures and all its values. The caps features must not
+have a parent when this function is called.
+
+
+
+
+
+ the #GstCapsFeatures to free
+
+
+
+
+
+ Returns the @i-th feature of @features.
+
+ The @i-th feature of @features.
+
+
+
+
+ a #GstCapsFeatures.
+
+
+
+ index of the feature
+
+
+
+
+
+ Returns the @i-th feature of @features.
+
+ The @i-th feature of @features.
+
+
+
+
+ a #GstCapsFeatures.
+
+
+
+ index of the feature
+
+
+
+
+
+ Returns the number of features in @features.
+
+ The number of features in @features.
+
+
+
+
+ a #GstCapsFeatures.
+
+
+
+
+
+ Check if @features is %GST_CAPS_FEATURES_ANY.
+
+ %TRUE if @features is %GST_CAPS_FEATURES_ANY.
+
+
+
+
+ a #GstCapsFeatures.
+
+
+
+
+
+ Check if @features1 and @features2 are equal.
+
+ %TRUE if @features1 and @features2 are equal.
+
+
+
+
+ a #GstCapsFeatures.
+
+
+
+ a #GstCapsFeatures.
+
+
+
+
+
+ Removes @feature from @features.
+
+
+
+
+
+ a #GstCapsFeatures.
+
+
+
+ a feature.
+
+
+
+
+
+ Removes @feature from @features.
+
+
+
+
+
+ a #GstCapsFeatures.
+
+
+
+ a feature.
+
+
+
+
+
+ Sets the parent_refcount field of #GstCapsFeatures. This field is used to
+determine whether a caps features is mutable or not. This function should only be
+called by code implementing parent objects of #GstCapsFeatures, as described in
+the MT Refcounting section of the design documents.
+
+ %TRUE if the parent refcount could be set.
+
+
+
+
+ a #GstCapsFeatures
+
+
+
+ a pointer to the parent's refcount
+
+
+
+
+
+ Converts @features to a human-readable string representation.
+
+For debugging purposes its easier to do something like this:
+|[<!-- language="C" -->
+GST_LOG ("features is %" GST_PTR_FORMAT, features);
+]|
+This prints the features in human readable form.
+
+Free-function: g_free
+
+ a pointer to string allocated by g_malloc().
+ g_free() after usage.
+
+
+
+
+ a #GstCapsFeatures
+
+
+
+
+
+ Creates a #GstCapsFeatures from a string representation.
+
+Free-function: gst_caps_features_free
+
+ a new #GstCapsFeatures or
+ %NULL when the string could not be parsed. Free with
+ gst_caps_features_free() after use.
+
+
+
+
+ a string representation of a #GstCapsFeatures.
+
+
+
+
+
+
+ A function that will be called in gst_caps_filter_and_map_in_place().
+The function may modify @features and @structure, and both will be
+removed from the caps if %FALSE is returned.
+
+ %TRUE if the features and structure should be preserved,
+%FALSE if it should be removed.
+
+
+
+
+ the #GstCapsFeatures
+
+
+
+ the #GstStructure
+
+
+
+ user data
+
+
+
+
+
+ Extra flags for a caps.
+
+ Caps has no specific content, but can contain
+ anything.
+
+
+
+ A function that will be called in gst_caps_foreach(). The function may
+not modify @features or @structure.
+
+ %TRUE if the foreach operation should continue, %FALSE if
+the foreach operation should stop with %FALSE.
+
+
+
+
+ the #GstCapsFeatures
+
+
+
+ the #GstStructure
+
+
+
+ user data
+
+
+
+
+
+ Modes of caps intersection
+
+@GST_CAPS_INTERSECT_ZIG_ZAG tries to preserve overall order of both caps
+by iterating on the caps' structures as the following matrix shows:
+|[
+ caps1
+ +-------------
+ | 1 2 4 7
+caps2 | 3 5 8 10
+ | 6 9 11 12
+]|
+Used when there is no explicit precedence of one caps over the other. e.g.
+tee's sink pad getcaps function, it will probe its src pad peers' for their
+caps and intersect them with this mode.
+
+@GST_CAPS_INTERSECT_FIRST is useful when an element wants to preserve
+another element's caps priority order when intersecting with its own caps.
+Example: If caps1 is [A, B, C] and caps2 is [E, B, D, A], the result
+would be [A, B], maintaining the first caps priority on the intersection.
+
+ Zig-zags over both caps.
+
+
+ Keeps the first caps order.
+
+
+
+ A function that will be called in gst_caps_map_in_place(). The function
+may modify @features and @structure.
+
+ %TRUE if the map operation should continue, %FALSE if
+the map operation should stop with %FALSE.
+
+
+
+
+ the #GstCapsFeatures
+
+
+
+ the #GstStructure
+
+
+
+ user data
+
+
+
+
+
+ This interface abstracts handling of property sets for elements with
+children. Imagine elements such as mixers or polyphonic generators. They all
+have multiple #GstPad or some kind of voice objects. Another use case are
+container elements like #GstBin.
+The element implementing the interface acts as a parent for those child
+objects.
+
+By implementing this interface the child properties can be accessed from the
+parent element by using gst_child_proxy_get() and gst_child_proxy_set().
+
+Property names are written as "child-name::property-name". The whole naming
+scheme is recursive. Thus "child1::child2::property" is valid too, if
+"child1" and "child2" implement the #GstChildProxy interface.
+
+ Emits the "child-added" signal.
+
+
+
+
+
+ the parent object
+
+
+
+ the newly added child
+
+
+
+ the name of the new child
+
+
+
+
+
+ Emits the "child-removed" signal.
+
+
+
+
+
+ the parent object
+
+
+
+ the removed child
+
+
+
+ the name of the old child
+
+
+
+
+
+ Fetches a child by its number.
+
+ the child object or %NULL if
+ not found (index too high). Unref after usage.
+
+MT safe.
+
+
+
+
+ the parent object to get the child from
+
+
+
+ the child's position in the child list
+
+
+
+
+
+ Looks up a child element by the given name.
+
+This virtual method has a default implementation that uses #GstObject
+together with gst_object_get_name(). If the interface is to be used with
+#GObjects, this methods needs to be overridden.
+
+ the child object or %NULL if
+ not found. Unref after usage.
+
+MT safe.
+
+
+
+
+ the parent object to get the child from
+
+
+
+ the child's name
+
+
+
+
+
+ Gets the number of child objects this parent contains.
+
+ the number of child objects
+
+MT safe.
+
+
+
+
+ the parent object
+
+
+
+
+
+ Emits the "child-added" signal.
+
+
+
+
+
+ the parent object
+
+
+
+ the newly added child
+
+
+
+ the name of the new child
+
+
+
+
+
+ Emits the "child-removed" signal.
+
+
+
+
+
+ the parent object
+
+
+
+ the removed child
+
+
+
+ the name of the old child
+
+
+
+
+
+ Gets properties of the parent object and its children.
+
+
+
+
+
+ the parent object
+
+
+
+ name of the first property to get
+
+
+
+ return location for the first property, followed optionally by more name/return location pairs, followed by %NULL
+
+
+
+
+
+ Fetches a child by its number.
+
+ the child object or %NULL if
+ not found (index too high). Unref after usage.
+
+MT safe.
+
+
+
+
+ the parent object to get the child from
+
+
+
+ the child's position in the child list
+
+
+
+
+
+ Looks up a child element by the given name.
+
+This virtual method has a default implementation that uses #GstObject
+together with gst_object_get_name(). If the interface is to be used with
+#GObjects, this methods needs to be overridden.
+
+ the child object or %NULL if
+ not found. Unref after usage.
+
+MT safe.
+
+
+
+
+ the parent object to get the child from
+
+
+
+ the child's name
+
+
+
+
+
+ Gets the number of child objects this parent contains.
+
+ the number of child objects
+
+MT safe.
+
+
+
+
+ the parent object
+
+
+
+
+
+ Gets a single property using the GstChildProxy mechanism.
+You are responsible for freeing it by calling g_value_unset()
+
+
+
+
+
+ object to query
+
+
+
+ name of the property
+
+
+
+ a #GValue that should take the result.
+
+
+
+
+
+ Gets properties of the parent object and its children.
+
+
+
+
+
+ the object to query
+
+
+
+ name of the first property to get
+
+
+
+ return location for the first property, followed optionally by more name/return location pairs, followed by %NULL
+
+
+
+
+
+ Looks up which object and #GParamSpec would be effected by the given @name.
+
+MT safe.
+
+ %TRUE if @target and @pspec could be found. %FALSE otherwise. In that
+case the values for @pspec and @target are not modified. Unref @target after
+usage. For plain GObjects @target is the same as @object.
+
+
+
+
+ child proxy object to lookup the property in
+
+
+
+ name of the property to look up
+
+
+
+ pointer to a #GObject that
+ takes the real object to set property on
+
+
+
+ pointer to take the #GParamSpec
+ describing the property
+
+
+
+
+
+ Sets properties of the parent object and its children.
+
+
+
+
+
+ the parent object
+
+
+
+ name of the first property to set
+
+
+
+ value for the first property, followed optionally by more name/value pairs, followed by %NULL
+
+
+
+
+
+ Sets a single property using the GstChildProxy mechanism.
+
+
+
+
+
+ the parent object
+
+
+
+ name of the property to set
+
+
+
+ new #GValue for the property
+
+
+
+
+
+ Sets properties of the parent object and its children.
+
+
+
+
+
+ the parent object
+
+
+
+ name of the first property to set
+
+
+
+ value for the first property, followed optionally by more name/value pairs, followed by %NULL
+
+
+
+
+
+ Will be emitted after the @object was added to the @child_proxy.
+
+
+
+
+
+ the #GObject that was added
+
+
+
+ the name of the new child
+
+
+
+
+
+ Will be emitted after the @object was removed from the @child_proxy.
+
+
+
+
+
+ the #GObject that was removed
+
+
+
+ the name of the old child
+
+
+
+
+
+
+ #GstChildProxy interface.
+
+ parent interface type.
+
+
+
+
+
+ the child object or %NULL if
+ not found. Unref after usage.
+
+MT safe.
+
+
+
+
+ the parent object to get the child from
+
+
+
+ the child's name
+
+
+
+
+
+
+
+
+ the child object or %NULL if
+ not found (index too high). Unref after usage.
+
+MT safe.
+
+
+
+
+ the parent object to get the child from
+
+
+
+ the child's position in the child list
+
+
+
+
+
+
+
+
+ the number of child objects
+
+MT safe.
+
+
+
+
+ the parent object
+
+
+
+
+
+
+
+
+
+
+
+
+ the parent object
+
+
+
+ the newly added child
+
+
+
+ the name of the new child
+
+
+
+
+
+
+
+
+
+
+
+
+ the parent object
+
+
+
+ the removed child
+
+
+
+ the name of the old child
+
+
+
+
+
+
+
+
+
+
+
+
+ GStreamer uses a global clock to synchronize the plugins in a pipeline.
+Different clock implementations are possible by implementing this abstract
+base class or, more conveniently, by subclassing #GstSystemClock.
+
+The #GstClock returns a monotonically increasing time with the method
+gst_clock_get_time(). Its accuracy and base time depend on the specific
+clock implementation but time is always expressed in nanoseconds. Since the
+baseline of the clock is undefined, the clock time returned is not
+meaningful in itself, what matters are the deltas between two clock times.
+The time returned by a clock is called the absolute time.
+
+The pipeline uses the clock to calculate the running time. Usually all
+renderers synchronize to the global clock using the buffer timestamps, the
+newsegment events and the element's base time, see #GstPipeline.
+
+A clock implementation can support periodic and single shot clock
+notifications both synchronous and asynchronous.
+
+One first needs to create a #GstClockID for the periodic or single shot
+notification using gst_clock_new_single_shot_id() or
+gst_clock_new_periodic_id().
+
+To perform a blocking wait for the specific time of the #GstClockID use the
+gst_clock_id_wait(). To receive a callback when the specific time is reached
+in the clock use gst_clock_id_wait_async(). Both these calls can be
+interrupted with the gst_clock_id_unschedule() call. If the blocking wait is
+unscheduled a return value of #GST_CLOCK_UNSCHEDULED is returned.
+
+Periodic callbacks scheduled async will be repeatedly called automatically
+until it is unscheduled. To schedule a sync periodic callback,
+gst_clock_id_wait() should be called repeatedly.
+
+The async callbacks can happen from any thread, either provided by the core
+or from a streaming thread. The application should be prepared for this.
+
+A #GstClockID that has been unscheduled cannot be used again for any wait
+operation, a new #GstClockID should be created and the old unscheduled one
+should be destroyed with gst_clock_id_unref().
+
+It is possible to perform a blocking wait on the same #GstClockID from
+multiple threads. However, registering the same #GstClockID for multiple
+async notifications is not possible, the callback will only be called for
+the thread registering the entry last.
+
+None of the wait operations unref the #GstClockID, the owner is responsible
+for unreffing the ids itself. This holds for both periodic and single shot
+notifications. The reason being that the owner of the #GstClockID has to
+keep a handle to the #GstClockID to unblock the wait on FLUSHING events or
+state changes and if the entry would be unreffed automatically, the handle
+might become invalid without any notification.
+
+These clock operations do not operate on the running time, so the callbacks
+will also occur when not in PLAYING state as if the clock just keeps on
+running. Some clocks however do not progress when the element that provided
+the clock is not PLAYING.
+
+When a clock has the #GST_CLOCK_FLAG_CAN_SET_MASTER flag set, it can be
+slaved to another #GstClock with the gst_clock_set_master(). The clock will
+then automatically be synchronized to this master clock by repeatedly
+sampling the master clock and the slave clock and recalibrating the slave
+clock with gst_clock_set_calibration(). This feature is mostly useful for
+plugins that have an internal clock but must operate with another clock
+selected by the #GstPipeline. They can track the offset and rate difference
+of their internal clock relative to the master clock by using the
+gst_clock_get_calibration() function.
+
+The master/slave synchronisation can be tuned with the #GstClock:timeout,
+#GstClock:window-size and #GstClock:window-threshold properties.
+The #GstClock:timeout property defines the interval to sample the master
+clock and run the calibration functions. #GstClock:window-size defines the
+number of samples to use when calibrating and #GstClock:window-threshold
+defines the minimum number of samples before the calibration is performed.
+
+ Compares the two #GstClockID instances. This function can be used
+as a GCompareFunc when sorting ids.
+
+ negative value if a < b; zero if a = b; positive value if a > b
+
+MT safe.
+
+
+
+
+ A #GstClockID
+
+
+
+ A #GstClockID to compare with
+
+
+
+
+
+ Get the time of the clock ID
+
+ the time of the given clock id.
+
+MT safe.
+
+
+
+
+ The #GstClockID to query
+
+
+
+
+
+ Increase the refcount of given @id.
+
+ The same #GstClockID with increased refcount.
+
+MT safe.
+
+
+
+
+ The #GstClockID to ref
+
+
+
+
+
+ Unref given @id. When the refcount reaches 0 the
+#GstClockID will be freed.
+
+MT safe.
+
+
+
+
+
+ The #GstClockID to unref
+
+
+
+
+
+ Cancel an outstanding request with @id. This can either
+be an outstanding async notification or a pending sync notification.
+After this call, @id cannot be used anymore to receive sync or
+async notifications, you need to create a new #GstClockID.
+
+MT safe.
+
+
+
+
+
+ The id to unschedule
+
+
+
+
+
+ Perform a blocking wait on @id.
+@id should have been created with gst_clock_new_single_shot_id()
+or gst_clock_new_periodic_id() and should not have been unscheduled
+with a call to gst_clock_id_unschedule().
+
+If the @jitter argument is not %NULL and this function returns #GST_CLOCK_OK
+or #GST_CLOCK_EARLY, it will contain the difference
+against the clock and the time of @id when this method was
+called.
+Positive values indicate how late @id was relative to the clock
+(in which case this function will return #GST_CLOCK_EARLY).
+Negative values indicate how much time was spent waiting on the clock
+before this function returned.
+
+ the result of the blocking wait. #GST_CLOCK_EARLY will be returned
+if the current clock time is past the time of @id, #GST_CLOCK_OK if
+@id was scheduled in time. #GST_CLOCK_UNSCHEDULED if @id was
+unscheduled with gst_clock_id_unschedule().
+
+MT safe.
+
+
+
+
+ The #GstClockID to wait on
+
+
+
+ a pointer that will contain the jitter,
+ can be %NULL.
+
+
+
+
+
+ Register a callback on the given #GstClockID @id with the given
+function and user_data. When passing a #GstClockID with an invalid
+time to this function, the callback will be called immediately
+with a time set to GST_CLOCK_TIME_NONE. The callback will
+be called when the time of @id has been reached.
+
+The callback @func can be invoked from any thread, either provided by the
+core or from a streaming thread. The application should be prepared for this.
+
+ the result of the non blocking wait.
+
+MT safe.
+
+
+
+
+ a #GstClockID to wait on
+
+
+
+ The callback function
+
+
+
+ User data passed in the callback
+
+
+
+ #GDestroyNotify for user_data
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Gets the current internal time of the given clock. The time is returned
+unadjusted for the offset and the rate.
+
+ the internal time of the clock. Or GST_CLOCK_TIME_NONE when
+given invalid input.
+
+MT safe.
+
+
+
+
+ a #GstClock to query
+
+
+
+
+
+ Get the accuracy of the clock. The accuracy of the clock is the granularity
+of the values returned by gst_clock_get_time().
+
+ the resolution of the clock in units of #GstClockTime.
+
+MT safe.
+
+
+
+
+ a #GstClock
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ The time @master of the master clock and the time @slave of the slave
+clock are added to the list of observations. If enough observations
+are available, a linear regression algorithm is run on the
+observations and @clock is recalibrated.
+
+If this functions returns %TRUE, @r_squared will contain the
+correlation coefficient of the interpolation. A value of 1.0
+means a perfect regression was performed. This value can
+be used to control the sampling frequency of the master and slave
+clocks.
+
+ %TRUE if enough observations were added to run the
+regression algorithm.
+
+MT safe.
+
+
+
+
+ a #GstClock
+
+
+
+ a time on the slave
+
+
+
+ a time on the master
+
+
+
+ a pointer to hold the result
+
+
+
+
+
+ Add a clock observation to the internal slaving algorithm the same as
+gst_clock_add_observation(), and return the result of the master clock
+estimation, without updating the internal calibration.
+
+The caller can then take the results and call gst_clock_set_calibration()
+with the values, or some modified version of them.
+
+
+
+
+
+ a #GstClock
+
+
+
+ a time on the slave
+
+
+
+ a time on the master
+
+
+
+ a pointer to hold the result
+
+
+
+ a location to store the internal time
+
+
+
+ a location to store the external time
+
+
+
+ a location to store the rate numerator
+
+
+
+ a location to store the rate denominator
+
+
+
+
+
+ Converts the given @internal clock time to the external time, adjusting for the
+rate and reference time set with gst_clock_set_calibration() and making sure
+that the returned time is increasing. This function should be called with the
+clock's OBJECT_LOCK held and is mainly used by clock subclasses.
+
+This function is the reverse of gst_clock_unadjust_unlocked().
+
+ the converted time of the clock.
+
+
+
+
+ a #GstClock to use
+
+
+
+ a clock time
+
+
+
+
+
+ Converts the given @internal_target clock time to the external time,
+using the passed calibration parameters. This function performs the
+same calculation as gst_clock_adjust_unlocked() when called using the
+current calibration parameters, but doesn't ensure a monotonically
+increasing result as gst_clock_adjust_unlocked() does.
+
+Note: The @clock parameter is unused and can be NULL
+
+ the converted time of the clock.
+
+
+
+
+ a #GstClock to use
+
+
+
+ a clock time
+
+
+
+ a reference internal time
+
+
+
+ a reference external time
+
+
+
+ the numerator of the rate of the clock relative to its
+ internal time
+
+
+
+ the denominator of the rate of the clock
+
+
+
+
+
+ Gets the internal rate and reference time of @clock. See
+gst_clock_set_calibration() for more information.
+
+@internal, @external, @rate_num, and @rate_denom can be left %NULL if the
+caller is not interested in the values.
+
+MT safe.
+
+
+
+
+
+ a #GstClock
+
+
+
+ a location to store the internal time
+
+
+
+ a location to store the external time
+
+
+
+ a location to store the rate numerator
+
+
+
+ a location to store the rate denominator
+
+
+
+
+
+ Gets the current internal time of the given clock. The time is returned
+unadjusted for the offset and the rate.
+
+ the internal time of the clock. Or GST_CLOCK_TIME_NONE when
+given invalid input.
+
+MT safe.
+
+
+
+
+ a #GstClock to query
+
+
+
+
+
+ Get the master clock that @clock is slaved to or %NULL when the clock is
+not slaved to any master clock.
+
+ a master #GstClock or %NULL
+ when this clock is not slaved to a master clock. Unref after
+ usage.
+
+MT safe.
+
+
+
+
+ a #GstClock
+
+
+
+
+
+ Get the accuracy of the clock. The accuracy of the clock is the granularity
+of the values returned by gst_clock_get_time().
+
+ the resolution of the clock in units of #GstClockTime.
+
+MT safe.
+
+
+
+
+ a #GstClock
+
+
+
+
+
+ Gets the current time of the given clock. The time is always
+monotonically increasing and adjusted according to the current
+offset and rate.
+
+ the time of the clock. Or GST_CLOCK_TIME_NONE when
+given invalid input.
+
+MT safe.
+
+
+
+
+ a #GstClock to query
+
+
+
+
+
+ Get the amount of time that master and slave clocks are sampled.
+
+ the interval between samples.
+
+
+
+
+ a #GstClock
+
+
+
+
+
+ Checks if the clock is currently synced.
+
+This returns if GST_CLOCK_FLAG_NEEDS_STARTUP_SYNC is not set on the clock.
+
+ %TRUE if the clock is currently synced
+
+
+
+
+ a GstClock
+
+
+
+
+
+ Get an ID from @clock to trigger a periodic notification.
+The periodic notifications will start at time @start_time and
+will then be fired with the given @interval. @id should be unreffed
+after usage.
+
+Free-function: gst_clock_id_unref
+
+ a #GstClockID that can be used to request the
+ time notification.
+
+MT safe.
+
+
+
+
+ The #GstClockID to get a periodic notification id from
+
+
+
+ the requested start time
+
+
+
+ the requested interval
+
+
+
+
+
+ Get a #GstClockID from @clock to trigger a single shot
+notification at the requested time. The single shot id should be
+unreffed after usage.
+
+Free-function: gst_clock_id_unref
+
+ a #GstClockID that can be used to request the
+ time notification.
+
+MT safe.
+
+
+
+
+ The #GstClockID to get a single shot notification from
+
+
+
+ the requested time
+
+
+
+
+
+ Reinitializes the provided periodic @id to the provided start time and
+interval. Does not modify the reference count.
+
+ %TRUE if the GstClockID could be reinitialized to the provided
+@time, else %FALSE.
+
+
+
+
+ a #GstClock
+
+
+
+ a #GstClockID
+
+
+
+ the requested start time
+
+
+
+ the requested interval
+
+
+
+
+
+ Adjusts the rate and time of @clock. A rate of 1/1 is the normal speed of
+the clock. Values bigger than 1/1 make the clock go faster.
+
+@internal and @external are calibration parameters that arrange that
+gst_clock_get_time() should have been @external at internal time @internal.
+This internal time should not be in the future; that is, it should be less
+than the value of gst_clock_get_internal_time() when this function is called.
+
+Subsequent calls to gst_clock_get_time() will return clock times computed as
+follows:
+
+|[
+ time = (internal_time - internal) * rate_num / rate_denom + external
+]|
+
+This formula is implemented in gst_clock_adjust_unlocked(). Of course, it
+tries to do the integer arithmetic as precisely as possible.
+
+Note that gst_clock_get_time() always returns increasing values so when you
+move the clock backwards, gst_clock_get_time() will report the previous value
+until the clock catches up.
+
+MT safe.
+
+
+
+
+
+ a #GstClock to calibrate
+
+
+
+ a reference internal time
+
+
+
+ a reference external time
+
+
+
+ the numerator of the rate of the clock relative to its
+ internal time
+
+
+
+ the denominator of the rate of the clock
+
+
+
+
+
+ Set @master as the master clock for @clock. @clock will be automatically
+calibrated so that gst_clock_get_time() reports the same time as the
+master clock.
+
+A clock provider that slaves its clock to a master can get the current
+calibration values with gst_clock_get_calibration().
+
+@master can be %NULL in which case @clock will not be slaved anymore. It will
+however keep reporting its time adjusted with the last configured rate
+and time offsets.
+
+ %TRUE if the clock is capable of being slaved to a master clock.
+Trying to set a master on a clock without the
+#GST_CLOCK_FLAG_CAN_SET_MASTER flag will make this function return %FALSE.
+
+MT safe.
+
+
+
+
+ a #GstClock
+
+
+
+ a master #GstClock
+
+
+
+
+
+ Set the accuracy of the clock. Some clocks have the possibility to operate
+with different accuracy at the expense of more resource usage. There is
+normally no need to change the default resolution of a clock. The resolution
+of a clock can only be changed if the clock has the
+#GST_CLOCK_FLAG_CAN_SET_RESOLUTION flag set.
+
+ the new resolution of the clock.
+
+
+
+
+ a #GstClock
+
+
+
+ The resolution to set
+
+
+
+
+
+ Sets @clock to synced and emits the GstClock::synced signal, and wakes up any
+thread waiting in gst_clock_wait_for_sync().
+
+This function must only be called if GST_CLOCK_FLAG_NEEDS_STARTUP_SYNC
+is set on the clock, and is intended to be called by subclasses only.
+
+
+
+
+
+ a GstClock
+
+
+
+ if the clock is synced
+
+
+
+
+
+ Set the amount of time, in nanoseconds, to sample master and slave
+clocks
+
+
+
+
+
+ a #GstClock
+
+
+
+ a timeout
+
+
+
+
+
+ Reinitializes the provided single shot @id to the provided time. Does not
+modify the reference count.
+
+ %TRUE if the GstClockID could be reinitialized to the provided
+@time, else %FALSE.
+
+
+
+
+ a #GstClock
+
+
+
+ a #GstClockID
+
+
+
+ The requested time.
+
+
+
+
+
+ Converts the given @external clock time to the internal time of @clock,
+using the rate and reference time set with gst_clock_set_calibration().
+This function should be called with the clock's OBJECT_LOCK held and
+is mainly used by clock subclasses.
+
+This function is the reverse of gst_clock_adjust_unlocked().
+
+ the internal time of the clock corresponding to @external.
+
+
+
+
+ a #GstClock to use
+
+
+
+ an external clock time
+
+
+
+
+
+ Converts the given @external_target clock time to the internal time,
+using the passed calibration parameters. This function performs the
+same calculation as gst_clock_unadjust_unlocked() when called using the
+current calibration parameters.
+
+Note: The @clock parameter is unused and can be NULL
+
+ the converted time of the clock.
+
+
+
+
+ a #GstClock to use
+
+
+
+ a clock time
+
+
+
+ a reference internal time
+
+
+
+ a reference external time
+
+
+
+ the numerator of the rate of the clock relative to its
+ internal time
+
+
+
+ the denominator of the rate of the clock
+
+
+
+
+
+ Waits until @clock is synced for reporting the current time. If @timeout
+is %GST_CLOCK_TIME_NONE it will wait forever, otherwise it will time out
+after @timeout nanoseconds.
+
+For asynchronous waiting, the GstClock::synced signal can be used.
+
+This returns immediately with TRUE if GST_CLOCK_FLAG_NEEDS_STARTUP_SYNC
+is not set on the clock, or if the clock is already synced.
+
+ %TRUE if waiting was successful, or %FALSE on timeout
+
+
+
+
+ a GstClock
+
+
+
+ timeout for waiting or %GST_CLOCK_TIME_NONE
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Signaled on clocks with GST_CLOCK_FLAG_NEEDS_STARTUP_SYNC set once
+the clock is synchronized, or when it completely lost synchronization.
+This signal will not be emitted on clocks without the flag.
+
+This signal will be emitted from an arbitrary thread, most likely not
+the application's main thread.
+
+
+
+
+
+ if the clock is synced now
+
+
+
+
+
+
+ The function prototype of the callback.
+
+ %TRUE or %FALSE (currently unused)
+
+
+
+
+ The clock that triggered the callback
+
+
+
+ The time it was triggered
+
+
+
+ The #GstClockID that expired
+
+
+
+ user data passed in the gst_clock_id_wait_async() function
+
+
+
+
+
+ GStreamer clock class. Override the vmethods to implement the clock
+functionality.
+
+ the parent class structure
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ the resolution of the clock in units of #GstClockTime.
+
+MT safe.
+
+
+
+
+ a #GstClock
+
+
+
+
+
+
+
+
+ the internal time of the clock. Or GST_CLOCK_TIME_NONE when
+given invalid input.
+
+MT safe.
+
+
+
+
+ a #GstClock to query
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ All pending timeouts or periodic notifies are converted into
+an entry.
+Note that GstClockEntry should be treated as an opaque structure. It must
+not be extended or allocated using a custom allocator.
+
+ reference counter (read-only)
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ The type of the clock entry
+
+ a single shot timeout
+
+
+ a periodic timeout request
+
+
+
+ The capabilities of this clock
+
+ clock can do a single sync timeout request
+
+
+ clock can do a single async timeout request
+
+
+ clock can do sync periodic timeout requests
+
+
+ clock can do async periodic timeout callbacks
+
+
+ clock's resolution can be changed
+
+
+ clock can be slaved to a master clock
+
+
+ clock needs to be synced before it can be used
+ (Since 1.6)
+
+
+ subclasses can add additional flags starting from this flag
+
+
+
+
+
+ The return value of a clock operation.
+
+ The operation succeeded.
+
+
+ The operation was scheduled too late.
+
+
+ The clockID was unscheduled
+
+
+ The ClockID is busy
+
+
+ A bad time was provided to a function.
+
+
+ An error occurred
+
+
+ Operation is not supported
+
+
+ The ClockID is done waiting
+
+
+
+ The different kind of clocks.
+
+ time since Epoch
+
+
+ monotonic time since some unspecified starting
+ point
+
+
+ some other time source is used (Since 1.0.5)
+
+
+
+ #GstContext is a container object used to store contexts like a device
+context, a display server connection and similar concepts that should
+be shared between multiple elements.
+
+Applications can set a context on a complete pipeline by using
+gst_element_set_context(), which will then be propagated to all
+child elements. Elements can handle these in #GstElementClass.set_context()
+and merge them with the context information they already have.
+
+When an element needs a context it will do the following actions in this
+order until one step succeeds:
+1. Check if the element already has a context
+2. Query downstream with GST_QUERY_CONTEXT for the context
+3. Query upstream with GST_QUERY_CONTEXT for the context
+4. Post a GST_MESSAGE_NEED_CONTEXT message on the bus with the required
+ context types and afterwards check if a usable context was set now
+5. Create a context by itself and post a GST_MESSAGE_HAVE_CONTEXT message
+ on the bus.
+
+Bins will catch GST_MESSAGE_NEED_CONTEXT messages and will set any previously
+known context on the element that asks for it if possible. Otherwise the
+application should provide one if it can.
+
+#GstContext<!-- -->s can be persistent.
+A persistent #GstContext is kept in elements when they reach
+%GST_STATE_NULL, non-persistent ones will be removed.
+Also, a non-persistent context won't override a previous persistent
+context set to an element.
+
+ Create a new context.
+
+ The new context.
+
+
+
+
+ Context type
+
+
+
+ Persistent context
+
+
+
+
+
+ Get the type of @context.
+
+ The type of the context.
+
+
+
+
+ The #GstContext.
+
+
+
+
+
+ Access the structure of the context.
+
+ The structure of the context. The structure is
+still owned by the context, which means that you should not modify it,
+free it and that the pointer becomes invalid when you free the context.
+
+
+
+
+ The #GstContext.
+
+
+
+
+
+ Checks if @context has @context_type.
+
+ %TRUE if @context has @context_type.
+
+
+
+
+ The #GstContext.
+
+
+
+ Context type to check.
+
+
+
+
+
+ Check if @context is persistent.
+
+ %TRUE if the context is persistent.
+
+
+
+
+ The #GstContext.
+
+
+
+
+
+ Get a writable version of the structure.
+
+ The structure of the context. The structure is still
+owned by the context, which means that you should not free it and
+that the pointer becomes invalid when you free the context.
+This function checks if @context is writable.
+
+
+
+
+ The #GstContext.
+
+
+
+
+
+
+ A base class for value mapping objects that attaches control sources to gobject
+properties. Such an object is taking one or more #GstControlSource instances,
+combines them and maps the resulting value to the type and value range of the
+bound property.
+
+ Gets a number of #GValues for the given controlled property starting at the
+requested time. The array @values need to hold enough space for @n_values of
+#GValue.
+
+This function is useful if one wants to e.g. draw a graph of the control
+curve or apply a control curve sample by sample.
+
+ %TRUE if the given array could be filled, %FALSE otherwise
+
+
+
+
+ the control binding
+
+
+
+ the time that should be processed
+
+
+
+ the time spacing between subsequent values
+
+
+
+ the number of values
+
+
+
+ array to put control-values in
+
+
+
+
+
+
+
+ Gets the value for the given controlled property at the requested time.
+
+ the GValue of the property at the given time,
+or %NULL if the property isn't controlled.
+
+
+
+
+ the control binding
+
+
+
+ the time the control-change should be read from
+
+
+
+
+
+ Gets a number of values for the given controlled property starting at the
+requested time. The array @values need to hold enough space for @n_values of
+the same type as the objects property's type.
+
+This function is useful if one wants to e.g. draw a graph of the control
+curve or apply a control curve sample by sample.
+
+The values are unboxed and ready to be used. The similar function
+gst_control_binding_get_g_value_array() returns the array as #GValues and is
+more suitable for bindings.
+
+ %TRUE if the given array could be filled, %FALSE otherwise
+
+
+
+
+ the control binding
+
+
+
+ the time that should be processed
+
+
+
+ the time spacing between subsequent values
+
+
+
+ the number of values
+
+
+
+ array to put control-values in
+
+
+
+
+
+
+
+ Sets the property of the @object, according to the #GstControlSources that
+handle them and for the given timestamp.
+
+If this function fails, it is most likely the application developers fault.
+Most probably the control sources are not setup correctly.
+
+ %TRUE if the controller value could be applied to the object
+property, %FALSE otherwise
+
+
+
+
+ the control binding
+
+
+
+ the object that has controlled properties
+
+
+
+ the time that should be processed
+
+
+
+ the last time this was called
+
+
+
+
+
+ Gets a number of #GValues for the given controlled property starting at the
+requested time. The array @values need to hold enough space for @n_values of
+#GValue.
+
+This function is useful if one wants to e.g. draw a graph of the control
+curve or apply a control curve sample by sample.
+
+ %TRUE if the given array could be filled, %FALSE otherwise
+
+
+
+
+ the control binding
+
+
+
+ the time that should be processed
+
+
+
+ the time spacing between subsequent values
+
+
+
+ the number of values
+
+
+
+ array to put control-values in
+
+
+
+
+
+
+
+ Gets the value for the given controlled property at the requested time.
+
+ the GValue of the property at the given time,
+or %NULL if the property isn't controlled.
+
+
+
+
+ the control binding
+
+
+
+ the time the control-change should be read from
+
+
+
+
+
+ Gets a number of values for the given controlled property starting at the
+requested time. The array @values need to hold enough space for @n_values of
+the same type as the objects property's type.
+
+This function is useful if one wants to e.g. draw a graph of the control
+curve or apply a control curve sample by sample.
+
+The values are unboxed and ready to be used. The similar function
+gst_control_binding_get_g_value_array() returns the array as #GValues and is
+more suitable for bindings.
+
+ %TRUE if the given array could be filled, %FALSE otherwise
+
+
+
+
+ the control binding
+
+
+
+ the time that should be processed
+
+
+
+ the time spacing between subsequent values
+
+
+
+ the number of values
+
+
+
+ array to put control-values in
+
+
+
+
+
+
+
+ Check if the control binding is disabled.
+
+ %TRUE if the binding is inactive
+
+
+
+
+ the control binding
+
+
+
+
+
+ This function is used to disable a control binding for some time, i.e.
+gst_object_sync_values() will do nothing.
+
+
+
+
+
+ the control binding
+
+
+
+ boolean that specifies whether to disable the controller
+or not.
+
+
+
+
+
+ Sets the property of the @object, according to the #GstControlSources that
+handle them and for the given timestamp.
+
+If this function fails, it is most likely the application developers fault.
+Most probably the control sources are not setup correctly.
+
+ %TRUE if the controller value could be applied to the object
+property, %FALSE otherwise
+
+
+
+
+ the control binding
+
+
+
+ the object that has controlled properties
+
+
+
+ the time that should be processed
+
+
+
+ the last time this was called
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ name of the property of this binding
+
+
+
+ #GParamSpec for this property
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ The class structure of #GstControlBinding.
+
+ Parent class
+
+
+
+
+
+ %TRUE if the controller value could be applied to the object
+property, %FALSE otherwise
+
+
+
+
+ the control binding
+
+
+
+ the object that has controlled properties
+
+
+
+ the time that should be processed
+
+
+
+ the last time this was called
+
+
+
+
+
+
+
+
+ the GValue of the property at the given time,
+or %NULL if the property isn't controlled.
+
+
+
+
+ the control binding
+
+
+
+ the time the control-change should be read from
+
+
+
+
+
+
+
+
+ %TRUE if the given array could be filled, %FALSE otherwise
+
+
+
+
+ the control binding
+
+
+
+ the time that should be processed
+
+
+
+ the time spacing between subsequent values
+
+
+
+ the number of values
+
+
+
+ array to put control-values in
+
+
+
+
+
+
+
+
+
+
+ %TRUE if the given array could be filled, %FALSE otherwise
+
+
+
+
+ the control binding
+
+
+
+ the time that should be processed
+
+
+
+ the time spacing between subsequent values
+
+
+
+ the number of values
+
+
+
+ array to put control-values in
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ The #GstControlSource is a base class for control value sources that could
+be used to get timestamp-value pairs. A control source essentially is a
+function over time.
+
+A #GstControlSource is used by first getting an instance of a specific
+control-source, creating a binding for the control-source to the target property
+of the element and then adding the binding to the element. The binding will
+convert the data types and value range to fit to the bound property.
+
+For implementing a new #GstControlSource one has to implement
+#GstControlSourceGetValue and #GstControlSourceGetValueArray functions.
+These are then used by gst_control_source_get_value() and
+gst_control_source_get_value_array() to get values for specific timestamps.
+
+ Gets the value for this #GstControlSource at a given timestamp.
+
+ %FALSE if the value couldn't be returned, %TRUE otherwise.
+
+
+
+
+ the #GstControlSource object
+
+
+
+ the time for which the value should be returned
+
+
+
+ the value
+
+
+
+
+
+ Gets an array of values for for this #GstControlSource. Values that are
+undefined contain NANs.
+
+ %TRUE if the given array could be filled, %FALSE otherwise
+
+
+
+
+ the #GstControlSource object
+
+
+
+ the first timestamp
+
+
+
+ the time steps
+
+
+
+ the number of values to fetch
+
+
+
+ array to put control-values in
+
+
+
+
+
+
+
+
+
+
+ Function for returning a value for a given timestamp
+
+
+
+ Function for returning a values array for a given timestamp
+
+
+
+
+
+
+
+
+
+ The class structure of #GstControlSource.
+
+ Parent class
+
+
+
+
+
+
+
+
+
+ Function for returning a value for a given timestamp.
+
+ %TRUE if the value was successfully calculated.
+
+
+
+
+ the #GstControlSource instance
+
+
+
+ timestamp for which a value should be calculated
+
+
+
+ a value which will be set to the result.
+
+
+
+
+
+ Function for returning an array of values for starting at a given timestamp.
+
+ %TRUE if the values were successfully calculated.
+
+
+
+
+ the #GstControlSource instance
+
+
+
+ timestamp for which a value should be calculated
+
+
+
+ the time spacing between subsequent values
+
+
+
+ the number of values
+
+
+
+ array to put control-values in
+
+
+
+
+
+ Core errors are errors inside the core GStreamer library.
+
+ a general error which doesn't fit in any other
+category. Make sure you add a custom message to the error call.
+
+
+ do not use this except as a placeholder for
+deciding where to go while developing code.
+
+
+ use this when you do not want to implement
+this functionality yet.
+
+
+ used for state change errors.
+
+
+ used for pad-related errors.
+
+
+ used for thread-related errors.
+
+
+ used for negotiation-related errors.
+
+
+ used for event-related errors.
+
+
+ used for seek-related errors.
+
+
+ used for caps-related errors.
+
+
+ used for negotiation-related errors.
+
+
+ used if a plugin is missing.
+
+
+ used for clock related errors.
+
+
+ used if functionality has been disabled at
+ compile time.
+
+
+ the number of core error types.
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Struct to store date, time and timezone information altogether.
+#GstDateTime is refcounted and immutable.
+
+Date information is handled using the proleptic Gregorian calendar.
+
+Provides basic creation functions and accessor functions to its fields.
+
+ Creates a new #GstDateTime using the date and times in the gregorian calendar
+in the supplied timezone.
+
+@year should be from 1 to 9999, @month should be from 1 to 12, @day from
+1 to 31, @hour from 0 to 23, @minutes and @seconds from 0 to 59.
+
+Note that @tzoffset is a float and was chosen so for being able to handle
+some fractional timezones, while it still keeps the readability of
+representing it in hours for most timezones.
+
+If value is -1 then all over value will be ignored. For example
+if @month == -1, then #GstDateTime will created only for @year. If
+@day == -1, then #GstDateTime will created for @year and @month and
+so on.
+
+Free-function: gst_date_time_unref
+
+ the newly created #GstDateTime
+
+
+
+
+ Offset from UTC in hours.
+
+
+
+ the gregorian year
+
+
+
+ the gregorian month
+
+
+
+ the day of the gregorian month
+
+
+
+ the hour of the day
+
+
+
+ the minute of the hour
+
+
+
+ the second of the minute
+
+
+
+
+
+ Creates a new #GstDateTime from a #GDateTime object.
+
+Free-function: gst_date_time_unref
+
+ a newly created #GstDateTime,
+or %NULL on error
+
+
+
+
+ the #GDateTime. The new #GstDateTime takes ownership.
+
+
+
+
+
+ Tries to parse common variants of ISO-8601 datetime strings into a
+#GstDateTime. Possible input formats are (for example):
+2012-06-30T22:46:43Z, 2012, 2012-06, 2012-06-30, 2012-06-30T22:46:43-0430,
+2012-06-30T22:46Z, 2012-06-30T22:46-0430, 2012-06-30 22:46,
+2012-06-30 22:46:43, 2012-06-00, 2012-00-00, 2012-00-30, 22:46:43Z, 22:46Z,
+22:46:43-0430, 22:46-0430, 22:46:30, 22:46
+If no date is provided, it is assumed to be "today" in the timezone
+provided (if any), otherwise UTC.
+
+Free-function: gst_date_time_unref
+
+ a newly created #GstDateTime,
+or %NULL on error
+
+
+
+
+ ISO 8601-formatted datetime string.
+
+
+
+
+
+ Creates a new #GstDateTime using the time since Jan 1, 1970 specified by
+@secs. The #GstDateTime is in the local timezone.
+
+Free-function: gst_date_time_unref
+
+ the newly created #GstDateTime
+
+
+
+
+ seconds from the Unix epoch
+
+
+
+
+
+ Creates a new #GstDateTime using the time since Jan 1, 1970 specified by
+@secs. The #GstDateTime is in the UTC timezone.
+
+Free-function: gst_date_time_unref
+
+ the newly created #GstDateTime
+
+
+
+
+ seconds from the Unix epoch
+
+
+
+
+
+ Creates a new #GstDateTime using the date and times in the gregorian calendar
+in the local timezone.
+
+@year should be from 1 to 9999, @month should be from 1 to 12, @day from
+1 to 31, @hour from 0 to 23, @minutes and @seconds from 0 to 59.
+
+If @month is -1, then the #GstDateTime created will only contain @year,
+and all other fields will be considered not set.
+
+If @day is -1, then the #GstDateTime created will only contain @year and
+@month and all other fields will be considered not set.
+
+If @hour is -1, then the #GstDateTime created will only contain @year and
+@month and @day, and the time fields will be considered not set. In this
+case @minute and @seconds should also be -1.
+
+Free-function: gst_date_time_unref
+
+ the newly created #GstDateTime
+
+
+
+
+ the gregorian year
+
+
+
+ the gregorian month, or -1
+
+
+
+ the day of the gregorian month, or -1
+
+
+
+ the hour of the day, or -1
+
+
+
+ the minute of the hour, or -1
+
+
+
+ the second of the minute, or -1
+
+
+
+
+
+ Creates a new #GstDateTime representing the current date and time.
+
+Free-function: gst_date_time_unref
+
+ the newly created #GstDateTime which should
+ be freed with gst_date_time_unref().
+
+
+
+
+ Creates a new #GstDateTime that represents the current instant at Universal
+coordinated time.
+
+Free-function: gst_date_time_unref
+
+ the newly created #GstDateTime which should
+ be freed with gst_date_time_unref().
+
+
+
+
+ Creates a new #GstDateTime using the date and times in the gregorian calendar
+in the local timezone.
+
+@year should be from 1 to 9999.
+
+Free-function: gst_date_time_unref
+
+ the newly created #GstDateTime
+
+
+
+
+ the gregorian year
+
+
+
+
+
+ Creates a new #GstDateTime using the date and times in the gregorian calendar
+in the local timezone.
+
+@year should be from 1 to 9999, @month should be from 1 to 12.
+
+If value is -1 then all over value will be ignored. For example
+if @month == -1, then #GstDateTime will created only for @year.
+
+Free-function: gst_date_time_unref
+
+ the newly created #GstDateTime
+
+
+
+
+ the gregorian year
+
+
+
+ the gregorian month
+
+
+
+
+
+ Creates a new #GstDateTime using the date and times in the gregorian calendar
+in the local timezone.
+
+@year should be from 1 to 9999, @month should be from 1 to 12, @day from
+1 to 31.
+
+If value is -1 then all over value will be ignored. For example
+if @month == -1, then #GstDateTime will created only for @year. If
+@day == -1, then #GstDateTime will created for @year and @month and
+so on.
+
+Free-function: gst_date_time_unref
+
+ the newly created #GstDateTime
+
+
+
+
+ the gregorian year
+
+
+
+ the gregorian month
+
+
+
+ the day of the gregorian month
+
+
+
+
+
+ Returns the day of the month of this #GstDateTime.
+Call gst_date_time_has_day before, to avoid warnings.
+
+ The day of this #GstDateTime
+
+
+
+
+ a #GstDateTime
+
+
+
+
+
+ Retrieves the hour of the day represented by @datetime in the gregorian
+calendar. The return is in the range of 0 to 23.
+Call gst_date_time_has_haur before, to avoid warnings.
+
+ the hour of the day
+
+
+
+
+ a #GstDateTime
+
+
+
+
+
+ Retrieves the fractional part of the seconds in microseconds represented by
+@datetime in the gregorian calendar.
+
+ the microsecond of the second
+
+
+
+
+ a #GstDateTime
+
+
+
+
+
+ Retrieves the minute of the hour represented by @datetime in the gregorian
+calendar.
+Call gst_date_time_has_minute before, to avoid warnings.
+
+ the minute of the hour
+
+
+
+
+ a #GstDateTime
+
+
+
+
+
+ Returns the month of this #GstDateTime. January is 1, February is 2, etc..
+Call gst_date_time_has_month before, to avoid warnings.
+
+ The month of this #GstDateTime
+
+
+
+
+ a #GstDateTime
+
+
+
+
+
+ Retrieves the second of the minute represented by @datetime in the gregorian
+calendar.
+Call gst_date_time_has_second before, to avoid warnings.
+
+ the second represented by @datetime
+
+
+
+
+ a #GstDateTime
+
+
+
+
+
+ Retrieves the offset from UTC in hours that the timezone specified
+by @datetime represents. Timezones ahead (to the east) of UTC have positive
+values, timezones before (to the west) of UTC have negative values.
+If @datetime represents UTC time, then the offset is zero.
+
+ the offset from UTC in hours
+
+
+
+
+ a #GstDateTime
+
+
+
+
+
+ Returns the year of this #GstDateTime
+Call gst_date_time_has_year before, to avoid warnings.
+
+ The year of this #GstDateTime
+
+
+
+
+ a #GstDateTime
+
+
+
+
+
+
+ %TRUE if @datetime<!-- -->'s day field is set, otherwise %FALSE
+
+
+
+
+ a #GstDateTime
+
+
+
+
+
+
+ %TRUE if @datetime<!-- -->'s month field is set, otherwise %FALSE
+
+
+
+
+ a #GstDateTime
+
+
+
+
+
+
+ %TRUE if @datetime<!-- -->'s second field is set, otherwise %FALSE
+
+
+
+
+ a #GstDateTime
+
+
+
+
+
+
+ %TRUE if @datetime<!-- -->'s hour and minute fields are set,
+ otherwise %FALSE
+
+
+
+
+ a #GstDateTime
+
+
+
+
+
+
+ %TRUE if @datetime<!-- -->'s year field is set (which should always
+ be the case), otherwise %FALSE
+
+
+
+
+ a #GstDateTime
+
+
+
+
+
+ Atomically increments the reference count of @datetime by one.
+
+ the reference @datetime
+
+
+
+
+ a #GstDateTime
+
+
+
+
+
+ Creates a new #GDateTime from a fully defined #GstDateTime object.
+
+Free-function: g_date_time_unref
+
+ a newly created #GDateTime, or
+%NULL on error
+
+
+
+
+ GstDateTime.
+
+
+
+
+
+ Create a minimal string compatible with ISO-8601. Possible output formats
+are (for example): 2012, 2012-06, 2012-06-23, 2012-06-23T23:30Z,
+2012-06-23T23:30+0100, 2012-06-23T23:30:59Z, 2012-06-23T23:30:59+0100
+
+ a newly allocated string formatted according
+ to ISO 8601 and only including the datetime fields that are
+ valid, or %NULL in case there was an error. The string should
+ be freed with g_free().
+
+
+
+
+ GstDateTime.
+
+
+
+
+
+ Atomically decrements the reference count of @datetime by one. When the
+reference count reaches zero, the structure is freed.
+
+
+
+
+
+ a #GstDateTime
+
+
+
+
+
+
+ This is the struct that describes the categories. Once initialized with
+#GST_DEBUG_CATEGORY_INIT, its values can't be changed anymore.
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Removes and frees the category and all associated resources.
+
+
+
+
+
+ #GstDebugCategory to free.
+
+
+
+
+
+ Returns the color of a debug category used when printing output in this
+category.
+
+ the color of the category.
+
+
+
+
+ a #GstDebugCategory to get the color of.
+
+
+
+
+
+ Returns the description of a debug category.
+
+ the description of the category.
+
+
+
+
+ a #GstDebugCategory to get the description of.
+
+
+
+
+
+ Returns the name of a debug category.
+
+ the name of the category.
+
+
+
+
+ a #GstDebugCategory to get name of.
+
+
+
+
+
+ Returns the threshold of a #GstDebugCategory.
+
+ the #GstDebugLevel that is used as threshold.
+
+
+
+
+ a #GstDebugCategory to get threshold of.
+
+
+
+
+
+ Resets the threshold of the category to the default level. Debug information
+will only be output if the threshold is lower or equal to the level of the
+debugging message.
+Use this function to set the threshold back to where it was after using
+gst_debug_category_set_threshold().
+
+
+
+
+
+ a #GstDebugCategory to reset threshold of.
+
+
+
+
+
+ Sets the threshold of the category to the given level. Debug information will
+only be output if the threshold is lower or equal to the level of the
+debugging message.
+> Do not use this function in production code, because other functions may
+> change the threshold of categories as side effect. It is however a nice
+> function to use when debugging (even from gdb).
+
+
+
+
+
+ a #GstDebugCategory to set threshold of.
+
+
+
+ the #GstDebugLevel threshold to set.
+
+
+
+
+
+
+ These are some terminal style flags you can use when creating your
+debugging categories to make them stand out in debugging output.
+
+ Use black as foreground color.
+
+
+ Use red as foreground color.
+
+
+ Use green as foreground color.
+
+
+ Use yellow as foreground color.
+
+
+ Use blue as foreground color.
+
+
+ Use magenta as foreground color.
+
+
+ Use cyan as foreground color.
+
+
+ Use white as foreground color.
+
+
+ Use black as background color.
+
+
+ Use red as background color.
+
+
+ Use green as background color.
+
+
+ Use yellow as background color.
+
+
+ Use blue as background color.
+
+
+ Use magenta as background color.
+
+
+ Use cyan as background color.
+
+
+ Use white as background color.
+
+
+ Make the output bold.
+
+
+ Underline the output.
+
+
+
+
+ Do not use colors in logs.
+
+
+ Paint logs in a platform-specific way.
+
+
+ Paint logs with UNIX terminal color codes
+ no matter what platform GStreamer is running on.
+
+
+
+
+
+
+
+
+ Available details for pipeline graphs produced by GST_DEBUG_BIN_TO_DOT_FILE()
+and GST_DEBUG_BIN_TO_DOT_FILE_WITH_TS().
+
+ show caps-name on edges
+
+
+ show caps-details on edges
+
+
+ show modified parameters on
+ elements
+
+
+ show element states
+
+
+ show full element parameter values even
+ if they are very long
+
+
+ show all the typical details that one might want
+
+
+ show all details regardless of how large or
+ verbose they make the resulting output
+
+
+
+ The level defines the importance of a debugging message. The more important a
+message is, the greater the probability that the debugging system outputs it.
+
+ No debugging level specified or desired. Used to deactivate
+ debugging output.
+
+
+ Error messages are to be used only when an error occurred
+ that stops the application from keeping working correctly.
+ An examples is gst_element_error, which outputs a message with this priority.
+ It does not mean that the application is terminating as with g_error.
+
+
+ Warning messages are to inform about abnormal behaviour
+ that could lead to problems or weird behaviour later on. An example of this
+ would be clocking issues ("your computer is pretty slow") or broken input
+ data ("Can't synchronize to stream.")
+
+
+ Fixme messages are messages that indicate that something
+ in the executed code path is not fully implemented or handled yet. Note
+ that this does not replace proper error handling in any way, the purpose
+ of this message is to make it easier to spot incomplete/unfinished pieces
+ of code when reading the debug log.
+
+
+ Informational messages should be used to keep the developer
+ updated about what is happening.
+ Examples where this should be used are when a typefind function has
+ successfully determined the type of the stream or when an mp3 plugin detects
+ the format to be used. ("This file has mono sound.")
+
+
+ Debugging messages should be used when something common
+ happens that is not the expected default behavior, or something that's
+ useful to know but doesn't happen all the time (ie. per loop iteration or
+ buffer processed or event handled).
+ An example would be notifications about state changes or receiving/sending
+ of events.
+
+
+ Log messages are messages that are very common but might be
+ useful to know. As a rule of thumb a pipeline that is running as expected
+ should never output anything else but LOG messages whilst processing data.
+ Use this log level to log recurring information in chain functions and
+ loop functions, for example.
+
+
+ Tracing-related messages.
+ Examples for this are referencing/dereferencing of objects.
+
+
+ memory dump messages are used to log (small) chunks of
+ data as memory dumps in the log. They will be displayed as hexdump with
+ ASCII characters.
+
+
+ The number of defined debugging levels.
+
+
+ Get the string representation of a debugging level
+
+ the name
+
+
+
+
+ the level to get the name for
+
+
+
+
+
+
+
+ Gets the string representation of a #GstDebugMessage. This function is used
+in debug handlers to extract the message.
+
+ the string representation of a #GstDebugMessage.
+
+
+
+
+ a debug message
+
+
+
+
+
+
+ #GstDevice are objects representing a device, they contain
+relevant metadata about the device, such as its class and the #GstCaps
+representing the media types it can produce or handle.
+
+#GstDevice are created by #GstDeviceProvider objects which can be
+aggregated by #GstDeviceMonitor objects.
+
+ Creates the element with all of the required parameters set to use
+this device.
+
+ a new #GstElement configured to use
+this device
+
+
+
+
+ a #GstDevice
+
+
+
+ name of new element, or %NULL to automatically
+create a unique name.
+
+
+
+
+
+ Tries to reconfigure an existing element to use the device. If this
+function fails, then one must destroy the element and create a new one
+using gst_device_create_element().
+
+Note: This should only be implemented for elements can change their
+device in the PLAYING state.
+
+ %TRUE if the element could be reconfigured to use this device,
+%FALSE otherwise.
+
+
+
+
+ a #GstDevice
+
+
+
+ a #GstElement
+
+
+
+
+
+ Creates the element with all of the required parameters set to use
+this device.
+
+ a new #GstElement configured to use
+this device
+
+
+
+
+ a #GstDevice
+
+
+
+ name of new element, or %NULL to automatically
+create a unique name.
+
+
+
+
+
+ Getter for the #GstCaps that this device supports.
+
+ The #GstCaps supported by this device. Unref with
+gst_caps_unref() when done.
+
+
+
+
+ a #GstDevice
+
+
+
+
+
+ Gets the "class" of a device. This is a "/" separated list of
+classes that represent this device. They are a subset of the
+classes of the #GstDeviceProvider that produced this device.
+
+ The device class. Free with g_free() after use.
+
+
+
+
+ a #GstDevice
+
+
+
+
+
+ Gets the user-friendly name of the device.
+
+ The device name. Free with g_free() after use.
+
+
+
+
+ a #GstDevice
+
+
+
+
+
+ Gets the extra properties of a device.
+
+ The extra properties or %NULL when there are none.
+ Free with gst_structure_free() after use.
+
+
+
+
+ a #GstDevice
+
+
+
+
+
+ Check if @device matches all of the given classes
+
+ %TRUE if @device matches.
+
+
+
+
+ a #GstDevice
+
+
+
+ a "/"-separated list of device classes to match, only match if
+ all classes are matched
+
+
+
+
+
+ Check if @factory matches all of the given classes
+
+ %TRUE if @device matches.
+
+
+
+
+ a #GstDevice
+
+
+
+ a %NULL terminated array of classes
+ to match, only match if all classes are matched
+
+
+
+
+
+
+
+ Tries to reconfigure an existing element to use the device. If this
+function fails, then one must destroy the element and create a new one
+using gst_device_create_element().
+
+Note: This should only be implemented for elements can change their
+device in the PLAYING state.
+
+ %TRUE if the element could be reconfigured to use this device,
+%FALSE otherwise.
+
+
+
+
+ a #GstDevice
+
+
+
+ a #GstElement
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ The parent #GstObject strucuture.
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ The class structure for a #GstDevice object.
+
+ The parent #GstObjectClass strucuture.
+
+
+
+
+
+ a new #GstElement configured to use
+this device
+
+
+
+
+ a #GstDevice
+
+
+
+ name of new element, or %NULL to automatically
+create a unique name.
+
+
+
+
+
+
+
+
+ %TRUE if the element could be reconfigured to use this device,
+%FALSE otherwise.
+
+
+
+
+ a #GstDevice
+
+
+
+ a #GstElement
+
+
+
+
+
+
+
+
+
+
+
+
+ Applications should create a #GstDeviceMonitor when they want
+to probe, list and monitor devices of a specific type. The
+#GstDeviceMonitor will create the appropriate
+#GstDeviceProvider objects and manage them. It will then post
+messages on its #GstBus for devices that have been added and
+removed.
+
+The device monitor will monitor all devices matching the filters that
+the application has set.
+
+The basic use pattern of a device monitor is as follows:
+|[
+ static gboolean
+ my_bus_func (GstBus * bus, GstMessage * message, gpointer user_data)
+ {
+ GstDevice *device;
+ gchar *name;
+
+ switch (GST_MESSAGE_TYPE (message)) {
+ case GST_MESSAGE_DEVICE_ADDED:
+ gst_message_parse_device_added (message, &device);
+ name = gst_device_get_display_name (device);
+ g_print("Device added: %s\n", name);
+ g_free (name);
+ gst_object_unref (device);
+ break;
+ case GST_MESSAGE_DEVICE_REMOVED:
+ gst_message_parse_device_removed (message, &device);
+ name = gst_device_get_display_name (device);
+ g_print("Device removed: %s\n", name);
+ g_free (name);
+ gst_object_unref (device);
+ break;
+ default:
+ break;
+ }
+
+ return G_SOURCE_CONTINUE;
+ }
+
+ GstDeviceMonitor *
+ setup_raw_video_source_device_monitor (void) {
+ GstDeviceMonitor *monitor;
+ GstBus *bus;
+ GstCaps *caps;
+
+ monitor = gst_device_monitor_new ();
+
+ bus = gst_device_monitor_get_bus (monitor);
+ gst_bus_add_watch (bus, my_bus_func, NULL);
+ gst_object_unref (bus);
+
+ caps = gst_caps_new_empty_simple ("video/x-raw");
+ gst_device_monitor_add_filter (monitor, "Video/Source", caps);
+ gst_caps_unref (caps);
+
+ gst_device_monitor_start (monitor);
+
+ return monitor;
+ }
+]|
+
+ Create a new #GstDeviceMonitor
+
+ a new device monitor.
+
+
+
+
+ Adds a filter for which #GstDevice will be monitored, any device that matches
+all these classes and the #GstCaps will be returned.
+
+If this function is called multiple times to add more filters, each will be
+matched independently. That is, adding more filters will not further restrict
+what devices are matched.
+
+The #GstCaps supported by the device as returned by gst_device_get_caps() are
+not intersected with caps filters added using this function.
+
+Filters must be added before the #GstDeviceMonitor is started.
+
+ The id of the new filter or 0 if no provider matched the filter's
+ classes.
+
+
+
+
+ a device monitor
+
+
+
+ device classes to use as filter or %NULL for any class
+
+
+
+ the #GstCaps to filter or %NULL for ANY
+
+
+
+
+
+ Gets the #GstBus of this #GstDeviceMonitor
+
+ a #GstBus
+
+
+
+
+ a #GstDeviceProvider
+
+
+
+
+
+ Gets a list of devices from all of the relevant monitors. This may actually
+probe the hardware if the monitor is not currently started.
+
+ a #GList of
+ #GstDevice
+
+
+
+
+
+
+ A #GstDeviceProvider
+
+
+
+
+
+ Get a list of the currently selected device provider factories.
+
+This
+
+
+ A list of device provider factory names that are currently being
+ monitored by @monitor or %NULL when nothing is being monitored.
+
+
+
+
+
+
+ a #GstDeviceMonitor
+
+
+
+
+
+ Get if @monitor is curretly showing all devices, even those from hidden
+providers.
+
+ %TRUE when all devices will be shown.
+
+
+
+
+ a #GstDeviceMonitor
+
+
+
+
+
+ Removes a filter from the #GstDeviceMonitor using the id that was returned
+by gst_device_monitor_add_filter().
+
+ %TRUE of the filter id was valid, %FALSE otherwise
+
+
+
+
+ a device monitor
+
+
+
+ the id of the filter
+
+
+
+
+
+ Set if all devices should be visible, even those devices from hidden
+providers. Setting @show_all to true might show some devices multiple times.
+
+
+
+
+
+ a #GstDeviceMonitor
+
+
+
+ show all devices
+
+
+
+
+
+ Starts monitoring the devices, one this has succeeded, the
+%GST_MESSAGE_DEVICE_ADDED and %GST_MESSAGE_DEVICE_REMOVED messages
+will be emitted on the bus when the list of devices changes.
+
+ %TRUE if the device monitoring could be started
+
+
+
+
+ A #GstDeviceMonitor
+
+
+
+
+
+ Stops monitoring the devices.
+
+
+
+
+
+ A #GstDeviceProvider
+
+
+
+
+
+
+
+
+ the parent #GstObject structure
+
+
+
+
+
+
+
+
+
+
+
+
+ Opaque device monitor class structure.
+
+ the parent #GstObjectClass structure
+
+
+
+
+
+
+
+
+
+
+
+
+
+ A #GstDeviceProvider subclass is provided by a plugin that handles devices
+if there is a way to programatically list connected devices. It can also
+optionally provide updates to the list of connected devices.
+
+Each #GstDeviceProvider subclass is a singleton, a plugin should
+normally provide a single subclass for all devices.
+
+Applications would normally use a #GstDeviceMonitor to monitor devices
+from all relevant providers.
+
+ Create a new device providerfactory capable of instantiating objects of the
+@type and add the factory to @plugin.
+
+ %TRUE, if the registering succeeded, %FALSE on error
+
+
+
+
+ #GstPlugin to register the device provider with, or %NULL for
+ a static device provider.
+
+
+
+ name of device providers of this type
+
+
+
+ rank of device provider (higher rank means more importance when autoplugging)
+
+
+
+ GType of device provider to register
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Starts providering the devices. This will cause #GST_MESSAGE_DEVICE_ADDED
+and #GST_MESSAGE_DEVICE_REMOVED messages to be posted on the provider's bus
+when devices are added or removed from the system.
+
+Since the #GstDeviceProvider is a singleton,
+gst_device_provider_start() may already have been called by another
+user of the object, gst_device_provider_stop() needs to be called the same
+number of times.
+
+ %TRUE if the device providering could be started
+
+
+
+
+ A #GstDeviceProvider
+
+
+
+
+
+ Decreases the use-count by one. If the use count reaches zero, this
+#GstDeviceProvider will stop providering the devices. This needs to be
+called the same number of times that gst_device_provider_start() was called.
+
+
+
+
+
+ A #GstDeviceProvider
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Posts a message on the provider's #GstBus to inform applications that
+a new device has been added.
+
+This is for use by subclasses.
+
+@device's reference count will be incremented, and any floating reference
+will be removed (see gst_object_ref_sink()).
+
+
+
+
+
+ a #GstDeviceProvider
+
+
+
+ a #GstDevice that has been added
+
+
+
+
+
+ Posts a message on the provider's #GstBus to inform applications that
+a device has been removed.
+
+This is for use by subclasses.
+
+
+
+
+
+ a #GstDeviceProvider
+
+
+
+ a #GstDevice that has been removed
+
+
+
+
+
+ Gets the #GstBus of this #GstDeviceProvider
+
+ a #GstBus
+
+
+
+
+ a #GstDeviceProvider
+
+
+
+
+
+ Gets a list of devices that this provider understands. This may actually
+probe the hardware if the provider is not currently started.
+
+ a #GList of
+ #GstDevice
+
+
+
+
+
+
+ A #GstDeviceProvider
+
+
+
+
+
+ Retrieves the factory that was used to create this device provider.
+
+ the #GstDeviceProviderFactory used for
+ creating this device provider. no refcounting is needed.
+
+
+
+
+ a #GstDeviceProvider to request the device provider factory of.
+
+
+
+
+
+ Get the provider factory names of the #GstDeviceProvider instances that
+are hidden by @provider.
+
+
+ a list of hidden providers factory names or %NULL when
+ nothing is hidden by @provider. Free with g_strfreev.
+
+
+
+
+
+
+ a #GstDeviceProvider
+
+
+
+
+
+ Get metadata with @key in @provider.
+
+ the metadata for @key.
+
+
+
+
+ provider to get metadata for
+
+
+
+ the key to get
+
+
+
+
+
+ Make @provider hide the devices from the factory with @name.
+
+This function is used when @provider will also provide the devices reported
+by provider factory @name. A monitor should stop monitoring the
+device provider with @name to avoid duplicate devices.
+
+
+
+
+
+ a #GstDeviceProvider
+
+
+
+ a provider factory name
+
+
+
+
+
+ Starts providering the devices. This will cause #GST_MESSAGE_DEVICE_ADDED
+and #GST_MESSAGE_DEVICE_REMOVED messages to be posted on the provider's bus
+when devices are added or removed from the system.
+
+Since the #GstDeviceProvider is a singleton,
+gst_device_provider_start() may already have been called by another
+user of the object, gst_device_provider_stop() needs to be called the same
+number of times.
+
+ %TRUE if the device providering could be started
+
+
+
+
+ A #GstDeviceProvider
+
+
+
+
+
+ Decreases the use-count by one. If the use count reaches zero, this
+#GstDeviceProvider will stop providering the devices. This needs to be
+called the same number of times that gst_device_provider_start() was called.
+
+
+
+
+
+ A #GstDeviceProvider
+
+
+
+
+
+ Make @provider unhide the devices from factory @name.
+
+This function is used when @provider will no longer provide the devices
+reported by provider factory @name. A monitor should start
+monitoring the devices from provider factory @name in order to see
+all devices again.
+
+
+
+
+
+ a #GstDeviceProvider
+
+
+
+ a provider factory name
+
+
+
+
+
+ The parent #GstObject
+
+
+
+ a #GList of the #GstDevice objects
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ The structure of the base #GstDeviceProviderClass
+
+ the parent #GstObjectClass structure
+
+
+
+ a pointer to the #GstDeviceProviderFactory that creates this
+ provider
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ %TRUE if the device providering could be started
+
+
+
+
+ A #GstDeviceProvider
+
+
+
+
+
+
+
+
+
+
+
+
+ A #GstDeviceProvider
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Set @key with @value as metadata in @klass.
+
+
+
+
+
+ class to set metadata for
+
+
+
+ the key to set
+
+
+
+ the value to set
+
+
+
+
+
+ Set @key with @value as metadata in @klass.
+
+Same as gst_device_provider_class_add_metadata(), but @value must be a static string
+or an inlined string, as it will not be copied. (GStreamer plugins will
+be made resident once loaded, so this function can be used even from
+dynamically loaded plugins.)
+
+
+
+
+
+ class to set metadata for
+
+
+
+ the key to set
+
+
+
+ the value to set
+
+
+
+
+
+ Get metadata with @key in @klass.
+
+ the metadata for @key.
+
+
+
+
+ class to get metadata for
+
+
+
+ the key to get
+
+
+
+
+
+ Sets the detailed information for a #GstDeviceProviderClass.
+
+> This function is for use in _class_init functions only.
+
+
+
+
+
+ class to set metadata for
+
+
+
+ The long English name of the device provider. E.g. "File Sink"
+
+
+
+ String describing the type of device provider, as an
+ unordered list separated with slashes ('/'). See draft-klass.txt of the
+ design docs
+for more details and common types. E.g: "Sink/File"
+
+
+
+ Sentence describing the purpose of the device provider.
+E.g: "Write stream to a file"
+
+
+
+ Name and contact details of the author(s). Use \n to separate
+multiple author metadata. E.g: "Joe Bloggs <joe.blogs at foo.com>"
+
+
+
+
+
+ Sets the detailed information for a #GstDeviceProviderClass.
+
+> This function is for use in _class_init functions only.
+
+Same as gst_device_provider_class_set_metadata(), but @longname, @classification,
+@description, and @author must be static strings or inlined strings, as
+they will not be copied. (GStreamer plugins will be made resident once
+loaded, so this function can be used even from dynamically loaded plugins.)
+
+
+
+
+
+ class to set metadata for
+
+
+
+ The long English name of the element. E.g. "File Sink"
+
+
+
+ String describing the type of element, as
+an unordered list separated with slashes ('/'). See draft-klass.txt of the
+design docs for more details and common types. E.g: "Sink/File"
+
+
+
+ Sentence describing the purpose of the
+element. E.g: "Write stream to a file"
+
+
+
+ Name and contact details of the author(s). Use \n
+to separate multiple author metadata. E.g: "Joe Bloggs <joe.blogs at
+foo.com>"
+
+
+
+
+
+
+ #GstDeviceProviderFactory is used to create instances of device providers. A
+GstDeviceProviderfactory can be added to a #GstPlugin as it is also a
+#GstPluginFeature.
+
+Use the gst_device_provider_factory_find() and
+gst_device_provider_factory_get() functions to create device
+provider instances or use gst_device_provider_factory_get_by_name() as a
+convenient shortcut.
+
+ Search for an device provider factory of the given name. Refs the returned
+device provider factory; caller is responsible for unreffing.
+
+ #GstDeviceProviderFactory if
+found, %NULL otherwise
+
+
+
+
+ name of factory to find
+
+
+
+
+
+ Returns the device provider of the type defined by the given device
+provider factory.
+
+ a #GstDeviceProvider or %NULL
+if unable to create device provider
+
+
+
+
+ a named factory to instantiate
+
+
+
+
+
+ Get a list of factories with a rank greater or equal to @minrank.
+The list of factories is returned by decreasing rank.
+
+
+a #GList of #GstDeviceProviderFactory device providers. Use
+gst_plugin_feature_list_free() after usage.
+
+
+
+
+
+
+ Minimum rank
+
+
+
+
+
+ Returns the device provider of the type defined by the given device
+providerfactory.
+
+ the #GstDeviceProvider or %NULL
+if the device provider couldn't be created
+
+
+
+
+ factory to instantiate
+
+
+
+
+
+ Get the #GType for device providers managed by this factory. The type can
+only be retrieved if the device provider factory is loaded, which can be
+assured with gst_plugin_feature_load().
+
+ the #GType for device providers managed by this factory.
+
+
+
+
+ factory to get managed #GType from
+
+
+
+
+
+ Get the metadata on @factory with @key.
+
+ the metadata with @key on @factory or %NULL
+when there was no metadata with the given @key.
+
+
+
+
+ a #GstDeviceProviderFactory
+
+
+
+ a key
+
+
+
+
+
+ Get the available keys for the metadata on @factory.
+
+
+a %NULL-terminated array of key strings, or %NULL when there is no
+metadata. Free with g_strfreev() when no longer needed.
+
+
+
+
+
+
+ a #GstDeviceProviderFactory
+
+
+
+
+
+ Check if @factory matches all of the given @classes
+
+ %TRUE if @factory matches or if @classes is %NULL.
+
+
+
+
+ a #GstDeviceProviderFactory
+
+
+
+ a "/" separate list of classes to match, only match
+ if all classes are matched
+
+
+
+
+
+ Check if @factory matches all of the given classes
+
+ %TRUE if @factory matches.
+
+
+
+
+ a #GstDeviceProviderFactory
+
+
+
+ a %NULL terminated array
+ of classes to match, only match if all classes are matched
+
+
+
+
+
+
+
+
+ The opaque #GstDeviceProviderFactoryClass data structure.
+
+
+
+
+
+
+ #GstDynamicTypeFactory is used to represent a type that can be
+automatically loaded the first time it is used. For example,
+a non-standard type for use in caps fields.
+
+In general, applications and plugins don't need to use the factory
+beyond registering the type in a plugin init function. Once that is
+done, the type is stored in the registry, and ready as soon as the
+registry is loaded.
+
+## Registering a type for dynamic loading
+
+|[<!-- language="C" -->
+
+static gboolean
+plugin_init (GstPlugin * plugin)
+{
+ return gst_dynamic_type_register (plugin, GST_TYPE_CUSTOM_CAPS_FIELD);
+}
+]|
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Elements of any of the defined GST_ELEMENT_FACTORY_LIST types
+
+
+
+ All sinks handling audio, video or image media types
+
+
+
+ All encoders handling audio media types
+
+
+
+ All elements used to 'decode' streams (decoders, demuxers, parsers, depayloaders)
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Elements matching any of the defined GST_ELEMENT_FACTORY_TYPE_MEDIA types
+
+Note: Do not use this if you wish to not filter against any of the defined
+media types. If you wish to do this, simply don't specify any
+GST_ELEMENT_FACTORY_TYPE_MEDIA flag.
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ All encoders handling video or image media types
+
+
+
+ Name and contact details of the author(s). Use \n to separate
+multiple author details.
+E.g: "Joe Bloggs <joe.blogs at foo.com>"
+
+
+
+ Sentence describing the purpose of the element.
+E.g: "Write stream to a file"
+
+
+
+ Set uri pointing to user documentation. Applications can use this to show
+help for e.g. effects to users.
+
+
+
+ Elements that bridge to certain other products can include an icon of that
+used product. Application can show the icon in menus/selectors to help
+identifying specific elements.
+
+
+
+ String describing the type of element, as an unordered list
+separated with slashes ('/'). See draft-klass.txt of the design docs
+for more details and common types. E.g: "Sink/File"
+
+
+
+ The long English name of the element. E.g. "File Sink"
+
+
+
+ Builds a string using errno describing the previously failed system
+call. To be used as the debug argument in #GST_ELEMENT_ERROR.
+
+
+
+
+
+
+ The same thing as #GST_EVENT_TYPE_UPSTREAM | #GST_EVENT_TYPE_DOWNSTREAM.
+
+
+
+ GstElement is the abstract base class needed to construct an element that
+can be used in a GStreamer pipeline. Please refer to the plugin writers
+guide for more information on creating #GstElement subclasses.
+
+The name of a #GstElement can be get with gst_element_get_name() and set with
+gst_element_set_name(). For speed, GST_ELEMENT_NAME() can be used in the
+core when using the appropriate locking. Do not use this in plug-ins or
+applications in order to retain ABI compatibility.
+
+Elements can have pads (of the type #GstPad). These pads link to pads on
+other elements. #GstBuffer flow between these linked pads.
+A #GstElement has a #GList of #GstPad structures for all their input (or sink)
+and output (or source) pads.
+Core and plug-in writers can add and remove pads with gst_element_add_pad()
+and gst_element_remove_pad().
+
+An existing pad of an element can be retrieved by name with
+gst_element_get_static_pad(). A new dynamic pad can be created using
+gst_element_request_pad() with a #GstPadTemplate.
+An iterator of all pads can be retrieved with gst_element_iterate_pads().
+
+Elements can be linked through their pads.
+If the link is straightforward, use the gst_element_link()
+convenience function to link two elements, or gst_element_link_many()
+for more elements in a row.
+Use gst_element_link_filtered() to link two elements constrained by
+a specified set of #GstCaps.
+For finer control, use gst_element_link_pads() and
+gst_element_link_pads_filtered() to specify the pads to link on
+each element by name.
+
+Each element has a state (see #GstState). You can get and set the state
+of an element with gst_element_get_state() and gst_element_set_state().
+Setting a state triggers a #GstStateChange. To get a string representation
+of a #GstState, use gst_element_state_get_name().
+
+You can get and set a #GstClock on an element using gst_element_get_clock()
+and gst_element_set_clock().
+Some elements can provide a clock for the pipeline if
+the #GST_ELEMENT_FLAG_PROVIDE_CLOCK flag is set. With the
+gst_element_provide_clock() method one can retrieve the clock provided by
+such an element.
+Not all elements require a clock to operate correctly. If the
+#GST_ELEMENT_FLAG_REQUIRE_CLOCK() flag is set, a clock should be set on the
+element with gst_element_set_clock().
+
+Note that clock selection and distribution is normally handled by the
+toplevel #GstPipeline so the clock functions are only to be used in very
+specific situations.
+
+ Creates an element for handling the given URI.
+
+ a new element or %NULL if none
+could be created
+
+
+
+
+ Whether to create a source or a sink
+
+
+
+ URI to create an element for
+
+
+
+ Name of created element, can be %NULL.
+
+
+
+
+
+ Create a new elementfactory capable of instantiating objects of the
+@type and add the factory to @plugin.
+
+ %TRUE, if the registering succeeded, %FALSE on error
+
+
+
+
+ #GstPlugin to register the element with, or %NULL for
+ a static element.
+
+
+
+ name of elements of this type
+
+
+
+ rank of element (higher rank means more importance when autoplugging)
+
+
+
+ GType of element to register
+
+
+
+
+
+ Gets a string representing the given state change result.
+
+ a string with the name of the state
+ result.
+
+
+
+
+ a #GstStateChangeReturn to get the name of.
+
+
+
+
+
+ Gets a string representing the given state.
+
+ a string with the name of the state.
+
+
+
+
+ a #GstState to get the name of.
+
+
+
+
+
+ Perform @transition on @element.
+
+This function must be called with STATE_LOCK held and is mainly used
+internally.
+
+ the #GstStateChangeReturn of the state transition.
+
+
+
+
+ a #GstElement
+
+
+
+ the requested transition
+
+
+
+
+
+ Gets the state of the element.
+
+For elements that performed an ASYNC state change, as reported by
+gst_element_set_state(), this function will block up to the
+specified timeout value for the state change to complete.
+If the element completes the state change or goes into
+an error, this function returns immediately with a return value of
+%GST_STATE_CHANGE_SUCCESS or %GST_STATE_CHANGE_FAILURE respectively.
+
+For elements that did not return %GST_STATE_CHANGE_ASYNC, this function
+returns the current and pending state immediately.
+
+This function returns %GST_STATE_CHANGE_NO_PREROLL if the element
+successfully changed its state but is not able to provide data yet.
+This mostly happens for live sources that only produce data in
+%GST_STATE_PLAYING. While the state change return is equivalent to
+%GST_STATE_CHANGE_SUCCESS, it is returned to the application to signal that
+some sink elements might not be able to complete their state change because
+an element is not producing data to complete the preroll. When setting the
+element to playing, the preroll will complete and playback will start.
+
+ %GST_STATE_CHANGE_SUCCESS if the element has no more pending state
+ and the last state change succeeded, %GST_STATE_CHANGE_ASYNC if the
+ element is still performing a state change or
+ %GST_STATE_CHANGE_FAILURE if the last state change failed.
+
+MT safe.
+
+
+
+
+ a #GstElement to get the state of.
+
+
+
+ a pointer to #GstState to hold the state.
+ Can be %NULL.
+
+
+
+ a pointer to #GstState to hold the pending
+ state. Can be %NULL.
+
+
+
+ a #GstClockTime to specify the timeout for an async
+ state change or %GST_CLOCK_TIME_NONE for infinite timeout.
+
+
+
+
+
+ Use this function to signal that the element does not expect any more pads
+to show up in the current pipeline. This function should be called whenever
+pads have been added by the element itself. Elements with #GST_PAD_SOMETIMES
+pad templates use this in combination with autopluggers to figure out that
+the element is done initializing its pads.
+
+This function emits the #GstElement::no-more-pads signal.
+
+MT safe.
+
+
+
+
+
+ a #GstElement
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Post a message on the element's #GstBus. This function takes ownership of the
+message; if you want to access the message after this call, you should add an
+additional reference before calling.
+
+ %TRUE if the message was successfully posted. The function returns
+%FALSE if the element did not have a bus.
+
+MT safe.
+
+
+
+
+ a #GstElement posting the message
+
+
+
+ a #GstMessage to post
+
+
+
+
+
+ Get the clock provided by the given element.
+> An element is only required to provide a clock in the PAUSED
+> state. Some elements can provide a clock in other states.
+
+ the GstClock provided by the
+element or %NULL if no clock could be provided. Unref after usage.
+
+MT safe.
+
+
+
+
+ a #GstElement to query
+
+
+
+
+
+ Performs a query on the given element.
+
+For elements that don't implement a query handler, this function
+forwards the query to a random srcpad or to the peer of a
+random linked sinkpad of this element.
+
+Please note that some queries might need a running pipeline to work.
+
+ %TRUE if the query could be performed.
+
+MT safe.
+
+
+
+
+ a #GstElement to perform the query on.
+
+
+
+ the #GstQuery.
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Retrieves a request pad from the element according to the provided template.
+Pad templates can be looked up using
+gst_element_factory_get_static_pad_templates().
+
+The pad should be released with gst_element_release_request_pad().
+
+ requested #GstPad if found,
+ otherwise %NULL. Release after usage.
+
+
+
+
+ a #GstElement to find a request pad of.
+
+
+
+ a #GstPadTemplate of which we want a pad of.
+
+
+
+ the name of the request #GstPad
+to retrieve. Can be %NULL.
+
+
+
+ the caps of the pad we want to
+request. Can be %NULL.
+
+
+
+
+
+ Sends an event to an element. If the element doesn't implement an
+event handler, the event will be pushed on a random linked sink pad for
+downstream events or a random linked source pad for upstream events.
+
+This function takes ownership of the provided event so you should
+gst_event_ref() it if you want to reuse the event after this call.
+
+MT safe.
+
+ %TRUE if the event was handled. Events that trigger a preroll (such
+as flushing seeks and steps) will emit %GST_MESSAGE_ASYNC_DONE.
+
+
+
+
+ a #GstElement to send the event to.
+
+
+
+ the #GstEvent to send to the element.
+
+
+
+
+
+ Sets the bus of the element. Increases the refcount on the bus.
+For internal use only, unless you're testing elements.
+
+MT safe.
+
+
+
+
+
+ a #GstElement to set the bus of.
+
+
+
+ the #GstBus to set.
+
+
+
+
+
+ Sets the clock for the element. This function increases the
+refcount on the clock. Any previously set clock on the object
+is unreffed.
+
+ %TRUE if the element accepted the clock. An element can refuse a
+clock when it, for example, is not able to slave its internal clock to the
+@clock or when it requires a specific clock to operate.
+
+MT safe.
+
+
+
+
+ a #GstElement to set the clock for.
+
+
+
+ the #GstClock to set for the element.
+
+
+
+
+
+ Sets the context of the element. Increases the refcount of the context.
+
+MT safe.
+
+
+
+
+
+ a #GstElement to set the context of.
+
+
+
+ the #GstContext to set.
+
+
+
+
+
+ Sets the state of the element. This function will try to set the
+requested state by going through all the intermediary states and calling
+the class's state change function for each.
+
+This function can return #GST_STATE_CHANGE_ASYNC, in which case the
+element will perform the remainder of the state change asynchronously in
+another thread.
+An application can use gst_element_get_state() to wait for the completion
+of the state change or it can wait for a %GST_MESSAGE_ASYNC_DONE or
+%GST_MESSAGE_STATE_CHANGED on the bus.
+
+State changes to %GST_STATE_READY or %GST_STATE_NULL never return
+#GST_STATE_CHANGE_ASYNC.
+
+ Result of the state change using #GstStateChangeReturn.
+
+MT safe.
+
+
+
+
+ a #GstElement to change state of.
+
+
+
+ the element's new #GstState.
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Abort the state change of the element. This function is used
+by elements that do asynchronous state changes and find out
+something is wrong.
+
+This function should be called with the STATE_LOCK held.
+
+MT safe.
+
+
+
+
+
+ a #GstElement to abort the state of.
+
+
+
+
+
+ Adds a pad (link point) to @element. @pad's parent will be set to @element;
+see gst_object_set_parent() for refcounting information.
+
+Pads are not automatically activated so elements should perform the needed
+steps to activate the pad in case this pad is added in the PAUSED or PLAYING
+state. See gst_pad_set_active() for more information about activating pads.
+
+The pad and the element should be unlocked when calling this function.
+
+This function will emit the #GstElement::pad-added signal on the element.
+
+ %TRUE if the pad could be added. This function can fail when
+a pad with the same name already existed or the pad already had another
+parent.
+
+MT safe.
+
+
+
+
+ a #GstElement to add the pad to.
+
+
+
+ the #GstPad to add to the element.
+
+
+
+
+
+
+ a watch id, which can be used in connection with
+ gst_element_remove_property_notify_watch() to remove the watch again.
+
+
+
+
+ a #GstElement to watch (recursively) for property changes
+
+
+
+ name of property to watch for changes, or
+ NULL to watch all properties
+
+
+
+ whether to include the new property value in the message
+
+
+
+
+
+
+ a watch id, which can be used in connection with
+ gst_element_remove_property_notify_watch() to remove the watch again.
+
+
+
+
+ a #GstElement to watch for property changes
+
+
+
+ name of property to watch for changes, or
+ NULL to watch all properties
+
+
+
+ whether to include the new property value in the message
+
+
+
+
+
+ Calls @func from another thread and passes @user_data to it. This is to be
+used for cases when a state change has to be performed from a streaming
+thread, directly via gst_element_set_state() or indirectly e.g. via SEEK
+events.
+
+Calling those functions directly from the streaming thread will cause
+deadlocks in many situations, as they might involve waiting for the
+streaming thread to shut down from this very streaming thread.
+
+MT safe.
+
+
+
+
+
+ a #GstElement
+
+
+
+ Function to call asynchronously from another thread
+
+
+
+ Data to pass to @func
+
+
+
+ GDestroyNotify for @user_data
+
+
+
+
+
+ Perform @transition on @element.
+
+This function must be called with STATE_LOCK held and is mainly used
+internally.
+
+ the #GstStateChangeReturn of the state transition.
+
+
+
+
+ a #GstElement
+
+
+
+ the requested transition
+
+
+
+
+
+ Commit the state change of the element and proceed to the next
+pending state if any. This function is used
+by elements that do asynchronous state changes.
+The core will normally call this method automatically when an
+element returned %GST_STATE_CHANGE_SUCCESS from the state change function.
+
+If after calling this method the element still has not reached
+the pending state, the next state change is performed.
+
+This method is used internally and should normally not be called by plugins
+or applications.
+
+This function must be called with STATE_LOCK held.
+
+ The result of the commit state change.
+
+MT safe.
+
+
+
+
+ a #GstElement to continue the state change of.
+
+
+
+ The previous state return value
+
+
+
+
+
+ Creates a pad for each pad template that is always available.
+This function is only useful during object initialization of
+subclasses of #GstElement.
+
+
+
+
+
+ a #GstElement to create pads for
+
+
+
+
+
+ Call @func with @user_data for each of @element's pads. @func will be called
+exactly once for each pad that exists at the time of this call, unless
+one of the calls to @func returns %FALSE in which case we will stop
+iterating pads and return early. If new pads are added or pads are removed
+while pads are being iterated, this will not be taken into account until
+next time this function is used.
+
+ %FALSE if @element had no pads or if one of the calls to @func
+ returned %FALSE.
+
+
+
+
+ a #GstElement to iterate pads of
+
+
+
+ function to call for each pad
+
+
+
+ user data passed to @func
+
+
+
+
+
+ Call @func with @user_data for each of @element's sink pads. @func will be
+called exactly once for each sink pad that exists at the time of this call,
+unless one of the calls to @func returns %FALSE in which case we will stop
+iterating pads and return early. If new sink pads are added or sink pads
+are removed while the sink pads are being iterated, this will not be taken
+into account until next time this function is used.
+
+ %FALSE if @element had no sink pads or if one of the calls to @func
+ returned %FALSE.
+
+
+
+
+ a #GstElement to iterate sink pads of
+
+
+
+ function to call for each sink pad
+
+
+
+ user data passed to @func
+
+
+
+
+
+ Call @func with @user_data for each of @element's source pads. @func will be
+called exactly once for each source pad that exists at the time of this call,
+unless one of the calls to @func returns %FALSE in which case we will stop
+iterating pads and return early. If new source pads are added or source pads
+are removed while the source pads are being iterated, this will not be taken
+into account until next time this function is used.
+
+ %FALSE if @element had no source pads or if one of the calls
+ to @func returned %FALSE.
+
+
+
+
+ a #GstElement to iterate source pads of
+
+
+
+ function to call for each source pad
+
+
+
+ user data passed to @func
+
+
+
+
+
+ Returns the base time of the element. The base time is the
+absolute time of the clock when this element was last put to
+PLAYING. Subtracting the base time from the clock time gives
+the running time of the element.
+
+ the base time of the element.
+
+MT safe.
+
+
+
+
+ a #GstElement.
+
+
+
+
+
+ Returns the bus of the element. Note that only a #GstPipeline will provide a
+bus for the application.
+
+ the element's #GstBus. unref after
+usage.
+
+MT safe.
+
+
+
+
+ a #GstElement to get the bus of.
+
+
+
+
+
+ Gets the currently configured clock of the element. This is the clock as was
+last set with gst_element_set_clock().
+
+Elements in a pipeline will only have their clock set when the
+pipeline is in the PLAYING state.
+
+ the #GstClock of the element. unref after usage.
+
+MT safe.
+
+
+
+
+ a #GstElement to get the clock of.
+
+
+
+
+
+ Looks for an unlinked pad to which the given pad can link. It is not
+guaranteed that linking the pads will work, though it should work in most
+cases.
+
+This function will first attempt to find a compatible unlinked ALWAYS pad,
+and if none can be found, it will request a compatible REQUEST pad by looking
+at the templates of @element.
+
+ the #GstPad to which a link
+ can be made, or %NULL if one cannot be found. gst_object_unref()
+ after usage.
+
+
+
+
+ a #GstElement in which the pad should be found.
+
+
+
+ the #GstPad to find a compatible one for.
+
+
+
+ the #GstCaps to use as a filter.
+
+
+
+
+
+ Retrieves a pad template from @element that is compatible with @compattempl.
+Pads from compatible templates can be linked together.
+
+ a compatible #GstPadTemplate,
+ or %NULL if none was found. No unreferencing is necessary.
+
+
+
+
+ a #GstElement to get a compatible pad template for
+
+
+
+ the #GstPadTemplate to find a compatible
+ template for
+
+
+
+
+
+ Gets the context with @context_type set on the element or NULL.
+
+MT safe.
+
+ A #GstContext or NULL
+
+
+
+
+ a #GstElement to get the context of.
+
+
+
+ a name of a context to retrieve
+
+
+
+
+
+ Gets the context with @context_type set on the element or NULL.
+
+ A #GstContext or NULL
+
+
+
+
+ a #GstElement to get the context of.
+
+
+
+ a name of a context to retrieve
+
+
+
+
+
+ Gets the contexts set on the element.
+
+MT safe.
+
+ List of #GstContext
+
+
+
+
+
+
+ a #GstElement to set the context of.
+
+
+
+
+
+ Retrieves the factory that was used to create this element.
+
+ the #GstElementFactory used for creating this
+ element. no refcounting is needed.
+
+
+
+
+ a #GstElement to request the element factory of.
+
+
+
+
+
+ Get metadata with @key in @klass.
+
+ the metadata for @key.
+
+
+
+
+ class to get metadata for
+
+
+
+ the key to get
+
+
+
+
+
+ Retrieves a padtemplate from @element with the given name.
+
+ the #GstPadTemplate with the
+ given name, or %NULL if none was found. No unreferencing is
+ necessary.
+
+
+
+
+ a #GstElement to get the pad template of.
+
+
+
+ the name of the #GstPadTemplate to get.
+
+
+
+
+
+ Retrieves a list of the pad templates associated with @element. The
+list must not be modified by the calling code.
+
+ the #GList of
+ pad templates.
+
+
+
+
+
+
+ a #GstElement to get pad templates of.
+
+
+
+
+
+ Retrieves a pad from the element by name (e.g. "src_\%d"). This version only
+retrieves request pads. The pad should be released with
+gst_element_release_request_pad().
+
+This method is slower than manually getting the pad template and calling
+gst_element_request_pad() if the pads should have a specific name (e.g.
+@name is "src_1" instead of "src_\%u").
+
+ requested #GstPad if found,
+ otherwise %NULL. Release after usage.
+
+
+
+
+ a #GstElement to find a request pad of.
+
+
+
+ the name of the request #GstPad to retrieve.
+
+
+
+
+
+ Returns the start time of the element. The start time is the
+running time of the clock when this element was last put to PAUSED.
+
+Usually the start_time is managed by a toplevel element such as
+#GstPipeline.
+
+MT safe.
+
+ the start time of the element.
+
+
+
+
+ a #GstElement.
+
+
+
+
+
+ Gets the state of the element.
+
+For elements that performed an ASYNC state change, as reported by
+gst_element_set_state(), this function will block up to the
+specified timeout value for the state change to complete.
+If the element completes the state change or goes into
+an error, this function returns immediately with a return value of
+%GST_STATE_CHANGE_SUCCESS or %GST_STATE_CHANGE_FAILURE respectively.
+
+For elements that did not return %GST_STATE_CHANGE_ASYNC, this function
+returns the current and pending state immediately.
+
+This function returns %GST_STATE_CHANGE_NO_PREROLL if the element
+successfully changed its state but is not able to provide data yet.
+This mostly happens for live sources that only produce data in
+%GST_STATE_PLAYING. While the state change return is equivalent to
+%GST_STATE_CHANGE_SUCCESS, it is returned to the application to signal that
+some sink elements might not be able to complete their state change because
+an element is not producing data to complete the preroll. When setting the
+element to playing, the preroll will complete and playback will start.
+
+ %GST_STATE_CHANGE_SUCCESS if the element has no more pending state
+ and the last state change succeeded, %GST_STATE_CHANGE_ASYNC if the
+ element is still performing a state change or
+ %GST_STATE_CHANGE_FAILURE if the last state change failed.
+
+MT safe.
+
+
+
+
+ a #GstElement to get the state of.
+
+
+
+ a pointer to #GstState to hold the state.
+ Can be %NULL.
+
+
+
+ a pointer to #GstState to hold the pending
+ state. Can be %NULL.
+
+
+
+ a #GstClockTime to specify the timeout for an async
+ state change or %GST_CLOCK_TIME_NONE for infinite timeout.
+
+
+
+
+
+ Retrieves a pad from @element by name. This version only retrieves
+already-existing (i.e. 'static') pads.
+
+ the requested #GstPad if
+ found, otherwise %NULL. unref after usage.
+
+MT safe.
+
+
+
+
+ a #GstElement to find a static pad of.
+
+
+
+ the name of the static #GstPad to retrieve.
+
+
+
+
+
+ Checks if the state of an element is locked.
+If the state of an element is locked, state changes of the parent don't
+affect the element.
+This way you can leave currently unused elements inside bins. Just lock their
+state before changing the state from #GST_STATE_NULL.
+
+MT safe.
+
+ %TRUE, if the element's state is locked.
+
+
+
+
+ a #GstElement.
+
+
+
+
+
+ Retrieves an iterator of @element's pads. The iterator should
+be freed after usage. Also more specialized iterators exists such as
+gst_element_iterate_src_pads() or gst_element_iterate_sink_pads().
+
+The order of pads returned by the iterator will be the order in which
+the pads were added to the element.
+
+ the #GstIterator of #GstPad.
+
+MT safe.
+
+
+
+
+ a #GstElement to iterate pads of.
+
+
+
+
+
+ Retrieves an iterator of @element's sink pads.
+
+The order of pads returned by the iterator will be the order in which
+the pads were added to the element.
+
+ the #GstIterator of #GstPad.
+
+MT safe.
+
+
+
+
+ a #GstElement.
+
+
+
+
+
+ Retrieves an iterator of @element's source pads.
+
+The order of pads returned by the iterator will be the order in which
+the pads were added to the element.
+
+ the #GstIterator of #GstPad.
+
+MT safe.
+
+
+
+
+ a #GstElement.
+
+
+
+
+
+ Links @src to @dest. The link must be from source to
+destination; the other direction will not be tried. The function looks for
+existing pads that aren't linked yet. It will request new pads if necessary.
+Such pads need to be released manually when unlinking.
+If multiple links are possible, only one is established.
+
+Make sure you have added your elements to a bin or pipeline with
+gst_bin_add() before trying to link them.
+
+ %TRUE if the elements could be linked, %FALSE otherwise.
+
+
+
+
+ a #GstElement containing the source pad.
+
+
+
+ the #GstElement containing the destination pad.
+
+
+
+
+
+ Links @src to @dest using the given caps as filtercaps.
+The link must be from source to
+destination; the other direction will not be tried. The function looks for
+existing pads that aren't linked yet. It will request new pads if necessary.
+If multiple links are possible, only one is established.
+
+Make sure you have added your elements to a bin or pipeline with
+gst_bin_add() before trying to link them.
+
+ %TRUE if the pads could be linked, %FALSE otherwise.
+
+
+
+
+ a #GstElement containing the source pad.
+
+
+
+ the #GstElement containing the destination pad.
+
+
+
+ the #GstCaps to filter the link,
+ or %NULL for no filter.
+
+
+
+
+
+ Chain together a series of elements. Uses gst_element_link().
+Make sure you have added your elements to a bin or pipeline with
+gst_bin_add() before trying to link them.
+
+ %TRUE on success, %FALSE otherwise.
+
+
+
+
+ the first #GstElement in the link chain.
+
+
+
+ the second #GstElement in the link chain.
+
+
+
+ the %NULL-terminated list of elements to link in order.
+
+
+
+
+
+ Links the two named pads of the source and destination elements.
+Side effect is that if one of the pads has no parent, it becomes a
+child of the parent of the other element. If they have different
+parents, the link fails.
+
+ %TRUE if the pads could be linked, %FALSE otherwise.
+
+
+
+
+ a #GstElement containing the source pad.
+
+
+
+ the name of the #GstPad in source element
+ or %NULL for any pad.
+
+
+
+ the #GstElement containing the destination pad.
+
+
+
+ the name of the #GstPad in destination element,
+or %NULL for any pad.
+
+
+
+
+
+ Links the two named pads of the source and destination elements. Side effect
+is that if one of the pads has no parent, it becomes a child of the parent of
+the other element. If they have different parents, the link fails. If @caps
+is not %NULL, makes sure that the caps of the link is a subset of @caps.
+
+ %TRUE if the pads could be linked, %FALSE otherwise.
+
+
+
+
+ a #GstElement containing the source pad.
+
+
+
+ the name of the #GstPad in source element
+ or %NULL for any pad.
+
+
+
+ the #GstElement containing the destination pad.
+
+
+
+ the name of the #GstPad in destination element
+ or %NULL for any pad.
+
+
+
+ the #GstCaps to filter the link,
+ or %NULL for no filter.
+
+
+
+
+
+ Links the two named pads of the source and destination elements.
+Side effect is that if one of the pads has no parent, it becomes a
+child of the parent of the other element. If they have different
+parents, the link fails.
+
+Calling gst_element_link_pads_full() with @flags == %GST_PAD_LINK_CHECK_DEFAULT
+is the same as calling gst_element_link_pads() and the recommended way of
+linking pads with safety checks applied.
+
+This is a convenience function for gst_pad_link_full().
+
+ %TRUE if the pads could be linked, %FALSE otherwise.
+
+
+
+
+ a #GstElement containing the source pad.
+
+
+
+ the name of the #GstPad in source element
+ or %NULL for any pad.
+
+
+
+ the #GstElement containing the destination pad.
+
+
+
+ the name of the #GstPad in destination element,
+or %NULL for any pad.
+
+
+
+ the #GstPadLinkCheck to be performed when linking pads.
+
+
+
+
+
+ Brings the element to the lost state. The current state of the
+element is copied to the pending state so that any call to
+gst_element_get_state() will return %GST_STATE_CHANGE_ASYNC.
+
+An ASYNC_START message is posted. If the element was PLAYING, it will
+go to PAUSED. The element will be restored to its PLAYING state by
+the parent pipeline when it prerolls again.
+
+This is mostly used for elements that lost their preroll buffer
+in the %GST_STATE_PAUSED or %GST_STATE_PLAYING state after a flush,
+they will go to their pending state again when a new preroll buffer is
+queued. This function can only be called when the element is currently
+not in error or an async state change.
+
+This function is used internally and should normally not be called from
+plugins or applications.
+
+
+
+
+
+ a #GstElement the state is lost of
+
+
+
+
+
+ Post an error, warning or info message on the bus from inside an element.
+
+@type must be of #GST_MESSAGE_ERROR, #GST_MESSAGE_WARNING or
+#GST_MESSAGE_INFO.
+
+MT safe.
+
+
+
+
+
+ a #GstElement to send message from
+
+
+
+ the #GstMessageType
+
+
+
+ the GStreamer GError domain this message belongs to
+
+
+
+ the GError code belonging to the domain
+
+
+
+ an allocated text string to be used
+ as a replacement for the default message connected to code,
+ or %NULL
+
+
+
+ an allocated debug message to be
+ used as a replacement for the default debugging information,
+ or %NULL
+
+
+
+ the source code file where the error was generated
+
+
+
+ the source code function where the error was generated
+
+
+
+ the source code line where the error was generated
+
+
+
+
+
+ Post an error, warning or info message on the bus from inside an element.
+
+@type must be of #GST_MESSAGE_ERROR, #GST_MESSAGE_WARNING or
+#GST_MESSAGE_INFO.
+
+
+
+
+
+ a #GstElement to send message from
+
+
+
+ the #GstMessageType
+
+
+
+ the GStreamer GError domain this message belongs to
+
+
+
+ the GError code belonging to the domain
+
+
+
+ an allocated text string to be used
+ as a replacement for the default message connected to code,
+ or %NULL
+
+
+
+ an allocated debug message to be
+ used as a replacement for the default debugging information,
+ or %NULL
+
+
+
+ the source code file where the error was generated
+
+
+
+ the source code function where the error was generated
+
+
+
+ the source code line where the error was generated
+
+
+
+ optional details structure
+
+
+
+
+
+ Use this function to signal that the element does not expect any more pads
+to show up in the current pipeline. This function should be called whenever
+pads have been added by the element itself. Elements with #GST_PAD_SOMETIMES
+pad templates use this in combination with autopluggers to figure out that
+the element is done initializing its pads.
+
+This function emits the #GstElement::no-more-pads signal.
+
+MT safe.
+
+
+
+
+
+ a #GstElement
+
+
+
+
+
+ Post a message on the element's #GstBus. This function takes ownership of the
+message; if you want to access the message after this call, you should add an
+additional reference before calling.
+
+ %TRUE if the message was successfully posted. The function returns
+%FALSE if the element did not have a bus.
+
+MT safe.
+
+
+
+
+ a #GstElement posting the message
+
+
+
+ a #GstMessage to post
+
+
+
+
+
+ Get the clock provided by the given element.
+> An element is only required to provide a clock in the PAUSED
+> state. Some elements can provide a clock in other states.
+
+ the GstClock provided by the
+element or %NULL if no clock could be provided. Unref after usage.
+
+MT safe.
+
+
+
+
+ a #GstElement to query
+
+
+
+
+
+ Performs a query on the given element.
+
+For elements that don't implement a query handler, this function
+forwards the query to a random srcpad or to the peer of a
+random linked sinkpad of this element.
+
+Please note that some queries might need a running pipeline to work.
+
+ %TRUE if the query could be performed.
+
+MT safe.
+
+
+
+
+ a #GstElement to perform the query on.
+
+
+
+ the #GstQuery.
+
+
+
+
+
+ Queries an element to convert @src_val in @src_format to @dest_format.
+
+ %TRUE if the query could be performed.
+
+
+
+
+ a #GstElement to invoke the convert query on.
+
+
+
+ a #GstFormat to convert from.
+
+
+
+ a value to convert.
+
+
+
+ the #GstFormat to convert to.
+
+
+
+ a pointer to the result.
+
+
+
+
+
+ Queries an element (usually top-level pipeline or playbin element) for the
+total stream duration in nanoseconds. This query will only work once the
+pipeline is prerolled (i.e. reached PAUSED or PLAYING state). The application
+will receive an ASYNC_DONE message on the pipeline bus when that is the case.
+
+If the duration changes for some reason, you will get a DURATION_CHANGED
+message on the pipeline bus, in which case you should re-query the duration
+using this function.
+
+ %TRUE if the query could be performed.
+
+
+
+
+ a #GstElement to invoke the duration query on.
+
+
+
+ the #GstFormat requested
+
+
+
+ A location in which to store the total duration, or %NULL.
+
+
+
+
+
+ Queries an element (usually top-level pipeline or playbin element) for the
+stream position in nanoseconds. This will be a value between 0 and the
+stream duration (if the stream duration is known). This query will usually
+only work once the pipeline is prerolled (i.e. reached PAUSED or PLAYING
+state). The application will receive an ASYNC_DONE message on the pipeline
+bus when that is the case.
+
+If one repeatedly calls this function one can also create a query and reuse
+it in gst_element_query().
+
+ %TRUE if the query could be performed.
+
+
+
+
+ a #GstElement to invoke the position query on.
+
+
+
+ the #GstFormat requested
+
+
+
+ a location in which to store the current
+ position, or %NULL.
+
+
+
+
+
+ Makes the element free the previously requested pad as obtained
+with gst_element_request_pad().
+
+This does not unref the pad. If the pad was created by using
+gst_element_request_pad(), gst_element_release_request_pad() needs to be
+followed by gst_object_unref() to free the @pad.
+
+MT safe.
+
+
+
+
+
+ a #GstElement to release the request pad of.
+
+
+
+ the #GstPad to release.
+
+
+
+
+
+ Removes @pad from @element. @pad will be destroyed if it has not been
+referenced elsewhere using gst_object_unparent().
+
+This function is used by plugin developers and should not be used
+by applications. Pads that were dynamically requested from elements
+with gst_element_request_pad() should be released with the
+gst_element_release_request_pad() function instead.
+
+Pads are not automatically deactivated so elements should perform the needed
+steps to deactivate the pad in case this pad is removed in the PAUSED or
+PLAYING state. See gst_pad_set_active() for more information about
+deactivating pads.
+
+The pad and the element should be unlocked when calling this function.
+
+This function will emit the #GstElement::pad-removed signal on the element.
+
+ %TRUE if the pad could be removed. Can return %FALSE if the
+pad does not belong to the provided element.
+
+MT safe.
+
+
+
+
+ a #GstElement to remove pad from.
+
+
+
+ the #GstPad to remove from the element.
+
+
+
+
+
+
+
+
+
+
+ a #GstElement being watched for property changes
+
+
+
+ watch id to remove
+
+
+
+
+
+ Retrieves a request pad from the element according to the provided template.
+Pad templates can be looked up using
+gst_element_factory_get_static_pad_templates().
+
+The pad should be released with gst_element_release_request_pad().
+
+ requested #GstPad if found,
+ otherwise %NULL. Release after usage.
+
+
+
+
+ a #GstElement to find a request pad of.
+
+
+
+ a #GstPadTemplate of which we want a pad of.
+
+
+
+ the name of the request #GstPad
+to retrieve. Can be %NULL.
+
+
+
+ the caps of the pad we want to
+request. Can be %NULL.
+
+
+
+
+
+ Sends a seek event to an element. See gst_event_new_seek() for the details of
+the parameters. The seek event is sent to the element using
+gst_element_send_event().
+
+MT safe.
+
+ %TRUE if the event was handled. Flushing seeks will trigger a
+preroll, which will emit %GST_MESSAGE_ASYNC_DONE.
+
+
+
+
+ a #GstElement to send the event to.
+
+
+
+ The new playback rate
+
+
+
+ The format of the seek values
+
+
+
+ The optional seek flags.
+
+
+
+ The type and flags for the new start position
+
+
+
+ The value of the new start position
+
+
+
+ The type and flags for the new stop position
+
+
+
+ The value of the new stop position
+
+
+
+
+
+ Simple API to perform a seek on the given element, meaning it just seeks
+to the given position relative to the start of the stream. For more complex
+operations like segment seeks (e.g. for looping) or changing the playback
+rate or seeking relative to the last configured playback segment you should
+use gst_element_seek().
+
+In a completely prerolled PAUSED or PLAYING pipeline, seeking is always
+guaranteed to return %TRUE on a seekable media type or %FALSE when the media
+type is certainly not seekable (such as a live stream).
+
+Some elements allow for seeking in the READY state, in this
+case they will store the seek event and execute it when they are put to
+PAUSED. If the element supports seek in READY, it will always return %TRUE when
+it receives the event in the READY state.
+
+ %TRUE if the seek operation succeeded. Flushing seeks will trigger a
+preroll, which will emit %GST_MESSAGE_ASYNC_DONE.
+
+
+
+
+ a #GstElement to seek on
+
+
+
+ a #GstFormat to execute the seek in, such as #GST_FORMAT_TIME
+
+
+
+ seek options; playback applications will usually want to use
+ GST_SEEK_FLAG_FLUSH | GST_SEEK_FLAG_KEY_UNIT here
+
+
+
+ position to seek to (relative to the start); if you are doing
+ a seek in #GST_FORMAT_TIME this value is in nanoseconds -
+ multiply with #GST_SECOND to convert seconds to nanoseconds or
+ with #GST_MSECOND to convert milliseconds to nanoseconds.
+
+
+
+
+
+ Sends an event to an element. If the element doesn't implement an
+event handler, the event will be pushed on a random linked sink pad for
+downstream events or a random linked source pad for upstream events.
+
+This function takes ownership of the provided event so you should
+gst_event_ref() it if you want to reuse the event after this call.
+
+MT safe.
+
+ %TRUE if the event was handled. Events that trigger a preroll (such
+as flushing seeks and steps) will emit %GST_MESSAGE_ASYNC_DONE.
+
+
+
+
+ a #GstElement to send the event to.
+
+
+
+ the #GstEvent to send to the element.
+
+
+
+
+
+ Set the base time of an element. See gst_element_get_base_time().
+
+MT safe.
+
+
+
+
+
+ a #GstElement.
+
+
+
+ the base time to set.
+
+
+
+
+
+ Sets the bus of the element. Increases the refcount on the bus.
+For internal use only, unless you're testing elements.
+
+MT safe.
+
+
+
+
+
+ a #GstElement to set the bus of.
+
+
+
+ the #GstBus to set.
+
+
+
+
+
+ Sets the clock for the element. This function increases the
+refcount on the clock. Any previously set clock on the object
+is unreffed.
+
+ %TRUE if the element accepted the clock. An element can refuse a
+clock when it, for example, is not able to slave its internal clock to the
+@clock or when it requires a specific clock to operate.
+
+MT safe.
+
+
+
+
+ a #GstElement to set the clock for.
+
+
+
+ the #GstClock to set for the element.
+
+
+
+
+
+ Sets the context of the element. Increases the refcount of the context.
+
+MT safe.
+
+
+
+
+
+ a #GstElement to set the context of.
+
+
+
+ the #GstContext to set.
+
+
+
+
+
+ Locks the state of an element, so state changes of the parent don't affect
+this element anymore.
+
+MT safe.
+
+ %TRUE if the state was changed, %FALSE if bad parameters were given
+or the elements state-locking needed no change.
+
+
+
+
+ a #GstElement
+
+
+
+ %TRUE to lock the element's state
+
+
+
+
+
+ Set the start time of an element. The start time of the element is the
+running time of the element when it last went to the PAUSED state. In READY
+or after a flushing seek, it is set to 0.
+
+Toplevel elements like #GstPipeline will manage the start_time and
+base_time on its children. Setting the start_time to #GST_CLOCK_TIME_NONE
+on such a toplevel element will disable the distribution of the base_time to
+the children and can be useful if the application manages the base_time
+itself, for example if you want to synchronize capture from multiple
+pipelines, and you can also ensure that the pipelines have the same clock.
+
+MT safe.
+
+
+
+
+
+ a #GstElement.
+
+
+
+ the base time to set.
+
+
+
+
+
+ Sets the state of the element. This function will try to set the
+requested state by going through all the intermediary states and calling
+the class's state change function for each.
+
+This function can return #GST_STATE_CHANGE_ASYNC, in which case the
+element will perform the remainder of the state change asynchronously in
+another thread.
+An application can use gst_element_get_state() to wait for the completion
+of the state change or it can wait for a %GST_MESSAGE_ASYNC_DONE or
+%GST_MESSAGE_STATE_CHANGED on the bus.
+
+State changes to %GST_STATE_READY or %GST_STATE_NULL never return
+#GST_STATE_CHANGE_ASYNC.
+
+ Result of the state change using #GstStateChangeReturn.
+
+MT safe.
+
+
+
+
+ a #GstElement to change state of.
+
+
+
+ the element's new #GstState.
+
+
+
+
+
+ Tries to change the state of the element to the same as its parent.
+If this function returns %FALSE, the state of element is undefined.
+
+ %TRUE, if the element's state could be synced to the parent's state.
+
+MT safe.
+
+
+
+
+ a #GstElement.
+
+
+
+
+
+ Unlinks all source pads of the source element with all sink pads
+of the sink element to which they are linked.
+
+If the link has been made using gst_element_link(), it could have created an
+requestpad, which has to be released using gst_element_release_request_pad().
+
+
+
+
+
+ the source #GstElement to unlink.
+
+
+
+ the sink #GstElement to unlink.
+
+
+
+
+
+ Unlinks a series of elements. Uses gst_element_unlink().
+
+
+
+
+
+ the first #GstElement in the link chain.
+
+
+
+ the second #GstElement in the link chain.
+
+
+
+ the %NULL-terminated list of elements to unlink in order.
+
+
+
+
+
+ Unlinks the two named pads of the source and destination elements.
+
+This is a convenience function for gst_pad_unlink().
+
+
+
+
+
+ a (transfer none): #GstElement containing the source pad.
+
+
+
+ the name of the #GstPad in source element.
+
+
+
+ a #GstElement containing the destination pad.
+
+
+
+ the name of the #GstPad in destination element.
+
+
+
+
+
+
+
+
+ Used to serialize execution of gst_element_set_state()
+
+
+
+ Used to signal completion of a state change
+
+
+
+ Used to detect concurrent execution of
+gst_element_set_state() and gst_element_get_state()
+
+
+
+ the target state of an element as set by the application
+
+
+
+ the current state of an element
+
+
+
+ the next state of an element, can be #GST_STATE_VOID_PENDING if
+the element is in the correct state.
+
+
+
+ the final state the element should go to, can be
+#GST_STATE_VOID_PENDING if the element is in the correct state
+
+
+
+ the last return value of an element state change
+
+
+
+ the bus of the element. This bus is provided to the element by the
+parent element or the application. A #GstPipeline has a bus of its own.
+
+
+
+ the clock of the element. This clock is usually provided to the
+element by the toplevel #GstPipeline.
+
+
+
+ the time of the clock right before the element is set to
+PLAYING. Subtracting @base_time from the current clock time in the PLAYING
+state will yield the running_time against the clock.
+
+
+
+ the running_time of the last PAUSED state
+
+
+
+ number of pads of the element, includes both source and sink pads.
+
+
+
+ list of pads
+
+
+
+
+
+ number of source pads of the element.
+
+
+
+ list of source pads
+
+
+
+
+
+ number of sink pads of the element.
+
+
+
+ list of sink pads
+
+
+
+
+
+ updated whenever the a pad is added or removed
+
+
+
+ list of contexts
+
+
+
+
+
+
+
+
+
+
+ This signals that the element will not generate more dynamic pads.
+Note that this signal will usually be emitted from the context of
+the streaming thread.
+
+
+
+
+
+ a new #GstPad has been added to the element. Note that this signal will
+usually be emitted from the context of the streaming thread. Also keep in
+mind that if you add new elements to the pipeline in the signal handler
+you will need to set them to the desired target state with
+gst_element_set_state() or gst_element_sync_state_with_parent().
+
+
+
+
+
+ the pad that has been added
+
+
+
+
+
+ a #GstPad has been removed from the element
+
+
+
+
+
+ the pad that has been removed
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ GStreamer element class. Override the vmethods to implement the element
+functionality.
+
+ the parent class structure
+
+
+
+ metadata for elements of this class
+
+
+
+ the #GstElementFactory that creates these elements
+
+
+
+ a #GList of #GstPadTemplate
+
+
+
+
+
+ the number of padtemplates
+
+
+
+ changed whenever the padtemplates change
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ a #GstElement
+
+
+
+
+
+
+
+
+ requested #GstPad if found,
+ otherwise %NULL. Release after usage.
+
+
+
+
+ a #GstElement to find a request pad of.
+
+
+
+ a #GstPadTemplate of which we want a pad of.
+
+
+
+ the name of the request #GstPad
+to retrieve. Can be %NULL.
+
+
+
+ the caps of the pad we want to
+request. Can be %NULL.
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ %GST_STATE_CHANGE_SUCCESS if the element has no more pending state
+ and the last state change succeeded, %GST_STATE_CHANGE_ASYNC if the
+ element is still performing a state change or
+ %GST_STATE_CHANGE_FAILURE if the last state change failed.
+
+MT safe.
+
+
+
+
+ a #GstElement to get the state of.
+
+
+
+ a pointer to #GstState to hold the state.
+ Can be %NULL.
+
+
+
+ a pointer to #GstState to hold the pending
+ state. Can be %NULL.
+
+
+
+ a #GstClockTime to specify the timeout for an async
+ state change or %GST_CLOCK_TIME_NONE for infinite timeout.
+
+
+
+
+
+
+
+
+ Result of the state change using #GstStateChangeReturn.
+
+MT safe.
+
+
+
+
+ a #GstElement to change state of.
+
+
+
+ the element's new #GstState.
+
+
+
+
+
+
+
+
+ the #GstStateChangeReturn of the state transition.
+
+
+
+
+ a #GstElement
+
+
+
+ the requested transition
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ a #GstElement to set the bus of.
+
+
+
+ the #GstBus to set.
+
+
+
+
+
+
+
+
+ the GstClock provided by the
+element or %NULL if no clock could be provided. Unref after usage.
+
+MT safe.
+
+
+
+
+ a #GstElement to query
+
+
+
+
+
+
+
+
+ %TRUE if the element accepted the clock. An element can refuse a
+clock when it, for example, is not able to slave its internal clock to the
+@clock or when it requires a specific clock to operate.
+
+MT safe.
+
+
+
+
+ a #GstElement to set the clock for.
+
+
+
+ the #GstClock to set for the element.
+
+
+
+
+
+
+
+
+ %TRUE if the event was handled. Events that trigger a preroll (such
+as flushing seeks and steps) will emit %GST_MESSAGE_ASYNC_DONE.
+
+
+
+
+ a #GstElement to send the event to.
+
+
+
+ the #GstEvent to send to the element.
+
+
+
+
+
+
+
+
+ %TRUE if the query could be performed.
+
+MT safe.
+
+
+
+
+ a #GstElement to perform the query on.
+
+
+
+ the #GstQuery.
+
+
+
+
+
+
+
+
+ %TRUE if the message was successfully posted. The function returns
+%FALSE if the element did not have a bus.
+
+MT safe.
+
+
+
+
+ a #GstElement posting the message
+
+
+
+ a #GstMessage to post
+
+
+
+
+
+
+
+
+
+
+
+
+ a #GstElement to set the context of.
+
+
+
+ the #GstContext to set.
+
+
+
+
+
+
+
+
+
+
+
+ Set @key with @value as metadata in @klass.
+
+
+
+
+
+ class to set metadata for
+
+
+
+ the key to set
+
+
+
+ the value to set
+
+
+
+
+
+ Adds a padtemplate to an element class. This is mainly used in the _class_init
+functions of classes. If a pad template with the same name as an already
+existing one is added the old one is replaced by the new one.
+
+@templ's reference count will be incremented, and any floating
+reference will be removed (see gst_object_ref_sink())
+
+
+
+
+
+ the #GstElementClass to add the pad template to.
+
+
+
+ a #GstPadTemplate to add to the element class.
+
+
+
+
+
+ Set @key with @value as metadata in @klass.
+
+Same as gst_element_class_add_metadata(), but @value must be a static string
+or an inlined string, as it will not be copied. (GStreamer plugins will
+be made resident once loaded, so this function can be used even from
+dynamically loaded plugins.)
+
+
+
+
+
+ class to set metadata for
+
+
+
+ the key to set
+
+
+
+ the value to set
+
+
+
+
+
+ Adds a pad template to an element class based on the static pad template
+@templ. This is mainly used in the _class_init functions of element
+implementations. If a pad template with the same name already exists,
+the old one is replaced by the new one.
+
+
+
+
+
+ the #GstElementClass to add the pad template to.
+
+
+
+ #GstStaticPadTemplate to add as pad template to the element class.
+
+
+
+
+
+ Adds a pad template to an element class based on the static pad template
+@templ. This is mainly used in the _class_init functions of element
+implementations. If a pad template with the same name already exists,
+the old one is replaced by the new one.
+
+
+
+
+
+ the #GstElementClass to add the pad template to.
+
+
+
+ #GstStaticPadTemplate to add as pad template to the element class.
+
+
+
+ The #GType of the pad to create
+
+
+
+
+
+ Get metadata with @key in @klass.
+
+ the metadata for @key.
+
+
+
+
+ class to get metadata for
+
+
+
+ the key to get
+
+
+
+
+
+ Retrieves a padtemplate from @element_class with the given name.
+> If you use this function in the #GInstanceInitFunc of an object class
+> that has subclasses, make sure to pass the g_class parameter of the
+> #GInstanceInitFunc here.
+
+ the #GstPadTemplate with the
+ given name, or %NULL if none was found. No unreferencing is
+ necessary.
+
+
+
+
+ a #GstElementClass to get the pad template of.
+
+
+
+ the name of the #GstPadTemplate to get.
+
+
+
+
+
+ Retrieves a list of the pad templates associated with @element_class. The
+list must not be modified by the calling code.
+> If you use this function in the #GInstanceInitFunc of an object class
+> that has subclasses, make sure to pass the g_class parameter of the
+> #GInstanceInitFunc here.
+
+ the #GList of
+ pad templates.
+
+
+
+
+
+
+ a #GstElementClass to get pad templates of.
+
+
+
+
+
+ Sets the detailed information for a #GstElementClass.
+> This function is for use in _class_init functions only.
+
+
+
+
+
+ class to set metadata for
+
+
+
+ The long English name of the element. E.g. "File Sink"
+
+
+
+ String describing the type of element, as an unordered list
+separated with slashes ('/'). See draft-klass.txt of the design docs
+for more details and common types. E.g: "Sink/File"
+
+
+
+ Sentence describing the purpose of the element.
+E.g: "Write stream to a file"
+
+
+
+ Name and contact details of the author(s). Use \n to separate
+multiple author metadata. E.g: "Joe Bloggs <joe.blogs at foo.com>"
+
+
+
+
+
+ Sets the detailed information for a #GstElementClass.
+
+> This function is for use in _class_init functions only.
+
+Same as gst_element_class_set_metadata(), but @longname, @classification,
+@description, and @author must be static strings or inlined strings, as
+they will not be copied. (GStreamer plugins will be made resident once
+loaded, so this function can be used even from dynamically loaded plugins.)
+
+
+
+
+
+ class to set metadata for
+
+
+
+ The long English name of the element. E.g. "File Sink"
+
+
+
+ String describing the type of element, as an unordered list
+separated with slashes ('/'). See draft-klass.txt of the design docs
+for more details and common types. E.g: "Sink/File"
+
+
+
+ Sentence describing the purpose of the element.
+E.g: "Write stream to a file"
+
+
+
+ Name and contact details of the author(s). Use \n to separate
+multiple author metadata. E.g: "Joe Bloggs <joe.blogs at foo.com>"
+
+
+
+
+
+
+ #GstElementFactory is used to create instances of elements. A
+GstElementFactory can be added to a #GstPlugin as it is also a
+#GstPluginFeature.
+
+Use the gst_element_factory_find() and gst_element_factory_create()
+functions to create element instances or use gst_element_factory_make() as a
+convenient shortcut.
+
+The following code example shows you how to create a GstFileSrc element.
+
+## Using an element factory
+|[<!-- language="C" -->
+ #include <gst/gst.h>
+
+ GstElement *src;
+ GstElementFactory *srcfactory;
+
+ gst_init (&argc, &argv);
+
+ srcfactory = gst_element_factory_find ("filesrc");
+ g_return_if_fail (srcfactory != NULL);
+ src = gst_element_factory_create (srcfactory, "src");
+ g_return_if_fail (src != NULL);
+ ...
+]|
+
+ Search for an element factory of the given name. Refs the returned
+element factory; caller is responsible for unreffing.
+
+ #GstElementFactory if found,
+%NULL otherwise
+
+
+
+
+ name of factory to find
+
+
+
+
+
+ Filter out all the elementfactories in @list that can handle @caps in
+the given direction.
+
+If @subsetonly is %TRUE, then only the elements whose pads templates
+are a complete superset of @caps will be returned. Else any element
+whose pad templates caps can intersect with @caps will be returned.
+
+ a #GList of
+ #GstElementFactory elements that match the given requisites.
+ Use #gst_plugin_feature_list_free after usage.
+
+
+
+
+
+
+ a #GList of
+ #GstElementFactory to filter
+
+
+
+
+
+ a #GstCaps
+
+
+
+ a #GstPadDirection to filter on
+
+
+
+ whether to filter on caps subsets or not.
+
+
+
+
+
+ Get a list of factories that match the given @type. Only elements
+with a rank greater or equal to @minrank will be returned.
+The list of factories is returned by decreasing rank.
+
+ a #GList of
+ #GstElementFactory elements. Use gst_plugin_feature_list_free() after
+ usage.
+
+
+
+
+
+
+ a #GstElementFactoryListType
+
+
+
+ Minimum rank
+
+
+
+
+
+ Create a new element of the type defined by the given element factory.
+If name is %NULL, then the element will receive a guaranteed unique name,
+consisting of the element factory name and a number.
+If name is given, it will be given the name supplied.
+
+ new #GstElement or %NULL
+if unable to create element
+
+
+
+
+ a named factory to instantiate
+
+
+
+ name of new element, or %NULL to automatically create
+ a unique name
+
+
+
+
+
+ Checks if the factory can sink all possible capabilities.
+
+ %TRUE if the caps are fully compatible.
+
+
+
+
+ factory to query
+
+
+
+ the caps to check
+
+
+
+
+
+ Checks if the factory can sink any possible capability.
+
+ %TRUE if the caps have a common subset.
+
+
+
+
+ factory to query
+
+
+
+ the caps to check
+
+
+
+
+
+ Checks if the factory can src all possible capabilities.
+
+ %TRUE if the caps are fully compatible.
+
+
+
+
+ factory to query
+
+
+
+ the caps to check
+
+
+
+
+
+ Checks if the factory can src any possible capability.
+
+ %TRUE if the caps have a common subset.
+
+
+
+
+ factory to query
+
+
+
+ the caps to check
+
+
+
+
+
+ Create a new element of the type defined by the given elementfactory.
+It will be given the name supplied, since all elements require a name as
+their first argument.
+
+ new #GstElement or %NULL
+ if the element couldn't be created
+
+
+
+
+ factory to instantiate
+
+
+
+ name of new element, or %NULL to automatically create
+ a unique name
+
+
+
+
+
+ Get the #GType for elements managed by this factory. The type can
+only be retrieved if the element factory is loaded, which can be
+assured with gst_plugin_feature_load().
+
+ the #GType for elements managed by this factory or 0 if
+the factory is not loaded.
+
+
+
+
+ factory to get managed #GType from
+
+
+
+
+
+ Get the metadata on @factory with @key.
+
+ the metadata with @key on @factory or %NULL
+when there was no metadata with the given @key.
+
+
+
+
+ a #GstElementFactory
+
+
+
+ a key
+
+
+
+
+
+ Get the available keys for the metadata on @factory.
+
+
+a %NULL-terminated array of key strings, or %NULL when there is no
+metadata. Free with g_strfreev() when no longer needed.
+
+
+
+
+
+
+ a #GstElementFactory
+
+
+
+
+
+ Gets the number of pad_templates in this factory.
+
+ the number of pad_templates
+
+
+
+
+ a #GstElementFactory
+
+
+
+
+
+ Gets the #GList of #GstStaticPadTemplate for this factory.
+
+ the
+ static pad templates
+
+
+
+
+
+
+ a #GstElementFactory
+
+
+
+
+
+ Gets a %NULL-terminated array of protocols this element supports or %NULL if
+no protocols are supported. You may not change the contents of the returned
+array, as it is still owned by the element factory. Use g_strdupv() to
+make a copy of the protocol string array if you need to.
+
+ the supported protocols
+ or %NULL
+
+
+
+
+
+
+ a #GstElementFactory
+
+
+
+
+
+ Gets the type of URIs the element supports or #GST_URI_UNKNOWN if none.
+
+ type of URIs this element supports
+
+
+
+
+ a #GstElementFactory
+
+
+
+
+
+ Check if @factory implements the interface with name @interfacename.
+
+ %TRUE when @factory implement the interface.
+
+
+
+
+ a #GstElementFactory
+
+
+
+ an interface name
+
+
+
+
+
+ Check if @factory is of the given types.
+
+ %TRUE if @factory is of @type.
+
+
+
+
+ a #GstElementFactory
+
+
+
+ a #GstElementFactoryListType
+
+
+
+
+
+
+
+
+ The standard flags that an element may have.
+
+ ignore state changes from parent
+
+
+ the element is a sink
+
+
+ the element is a source.
+
+
+ the element can provide a clock
+
+
+ the element requires a clock
+
+
+ the element can use an index
+
+
+ offset to define more flags
+
+
+
+ Function called for each pad when using gst_element_foreach_sink_pad(),
+gst_element_foreach_src_pad(), or gst_element_foreach_pad().
+
+ %FALSE to stop iterating pads, %TRUE to continue
+
+
+
+
+ the #GstElement
+
+
+
+ a #GstPad
+
+
+
+ user data passed to the foreach function
+
+
+
+
+
+ The event class provides factory methods to construct events for sending
+and functions to query (parse) received events.
+
+Events are usually created with gst_event_new_*() which takes event-type
+specific parameters as arguments.
+To send an event application will usually use gst_element_send_event() and
+elements will use gst_pad_send_event() or gst_pad_push_event().
+The event should be unreffed with gst_event_unref() if it has not been sent.
+
+Events that have been received can be parsed with their respective
+gst_event_parse_*() functions. It is valid to pass %NULL for unwanted details.
+
+Events are passed between elements in parallel to the data stream. Some events
+are serialized with buffers, others are not. Some events only travel downstream,
+others only upstream. Some events can travel both upstream and downstream.
+
+The events are used to signal special conditions in the datastream such as
+EOS (end of stream) or the start of a new stream-segment.
+Events are also used to flush the pipeline of any pending data.
+
+Most of the event API is used inside plugins. Applications usually only
+construct and use seek events.
+To do that gst_event_new_seek() is used to create a seek event. It takes
+the needed parameters to specify seeking time and mode.
+|[<!-- language="C" -->
+ GstEvent *event;
+ gboolean result;
+ ...
+ // construct a seek event to play the media from second 2 to 5, flush
+ // the pipeline to decrease latency.
+ event = gst_event_new_seek (1.0,
+ GST_FORMAT_TIME,
+ GST_SEEK_FLAG_FLUSH,
+ GST_SEEK_TYPE_SET, 2 * GST_SECOND,
+ GST_SEEK_TYPE_SET, 5 * GST_SECOND);
+ ...
+ result = gst_element_send_event (pipeline, event);
+ if (!result)
+ g_warning ("seek failed");
+ ...
+]|
+
+ the parent structure
+
+
+
+ the #GstEventType of the event
+
+
+
+ the timestamp of the event
+
+
+
+ the sequence number of the event
+
+
+
+ Create a new buffersize event. The event is sent downstream and notifies
+elements that they should provide a buffer of the specified dimensions.
+
+When the @async flag is set, a thread boundary is preferred.
+
+ a new #GstEvent
+
+
+
+
+ buffer format
+
+
+
+ minimum buffer size
+
+
+
+ maximum buffer size
+
+
+
+ thread behavior
+
+
+
+
+
+ Create a new CAPS event for @caps. The caps event can only travel downstream
+synchronized with the buffer flow and contains the format of the buffers
+that will follow after the event.
+
+ the new CAPS event.
+
+
+
+
+ a #GstCaps
+
+
+
+
+
+ Create a new custom-typed event. This can be used for anything not
+handled by other event-specific functions to pass an event to another
+element.
+
+Make sure to allocate an event type with the #GST_EVENT_MAKE_TYPE macro,
+assigning a free number and filling in the correct direction and
+serialization flags.
+
+New custom events can also be created by subclassing the event type if
+needed.
+
+ the new custom event.
+
+
+
+
+ The type of the new event
+
+
+
+ the structure for the event. The event will
+ take ownership of the structure.
+
+
+
+
+
+ Create a new EOS event. The eos event can only travel downstream
+synchronized with the buffer flow. Elements that receive the EOS
+event on a pad can return #GST_FLOW_EOS as a #GstFlowReturn
+when data after the EOS event arrives.
+
+The EOS event will travel down to the sink elements in the pipeline
+which will then post the #GST_MESSAGE_EOS on the bus after they have
+finished playing any buffered data.
+
+When all sinks have posted an EOS message, an EOS message is
+forwarded to the application.
+
+The EOS event itself will not cause any state transitions of the pipeline.
+
+ the new EOS event.
+
+
+
+
+ Allocate a new flush start event. The flush start event can be sent
+upstream and downstream and travels out-of-bounds with the dataflow.
+
+It marks pads as being flushing and will make them return
+#GST_FLOW_FLUSHING when used for data flow with gst_pad_push(),
+gst_pad_chain(), gst_pad_get_range() and gst_pad_pull_range().
+Any event (except a #GST_EVENT_FLUSH_STOP) received
+on a flushing pad will return %FALSE immediately.
+
+Elements should unlock any blocking functions and exit their streaming
+functions as fast as possible when this event is received.
+
+This event is typically generated after a seek to flush out all queued data
+in the pipeline so that the new media is played as soon as possible.
+
+ a new flush start event.
+
+
+
+
+ Allocate a new flush stop event. The flush stop event can be sent
+upstream and downstream and travels serialized with the dataflow.
+It is typically sent after sending a FLUSH_START event to make the
+pads accept data again.
+
+Elements can process this event synchronized with the dataflow since
+the preceding FLUSH_START event stopped the dataflow.
+
+This event is typically generated to complete a seek and to resume
+dataflow.
+
+ a new flush stop event.
+
+
+
+
+ if time should be reset
+
+
+
+
+
+ Create a new GAP event. A gap event can be thought of as conceptually
+equivalent to a buffer to signal that there is no data for a certain
+amount of time. This is useful to signal a gap to downstream elements
+which may wait for data, such as muxers or mixers or overlays, especially
+for sparse streams such as subtitle streams.
+
+ the new GAP event.
+
+
+
+
+ the start time (pts) of the gap
+
+
+
+ the duration of the gap
+
+
+
+
+
+ Create a new latency event. The event is sent upstream from the sinks and
+notifies elements that they should add an additional @latency to the
+running time before synchronising against the clock.
+
+The latency is mostly used in live sinks and is always expressed in
+the time format.
+
+ a new #GstEvent
+
+
+
+
+ the new latency value
+
+
+
+
+
+ Create a new navigation event from the given description.
+
+ a new #GstEvent
+
+
+
+
+ description of the event. The event will take
+ ownership of the structure.
+
+
+
+
+
+ Creates a new event containing information specific to a particular
+protection system (uniquely identified by @system_id), by which that
+protection system can acquire key(s) to decrypt a protected stream.
+
+In order for a decryption element to decrypt media
+protected using a specific system, it first needs all the
+protection system specific information necessary to acquire the decryption
+key(s) for that stream. The functions defined here enable this information
+to be passed in events from elements that extract it
+(e.g., ISOBMFF demuxers, MPEG DASH demuxers) to protection decrypter
+elements that use it.
+
+Events containing protection system specific information are created using
+#gst_event_new_protection, and they can be parsed by downstream elements
+using #gst_event_parse_protection.
+
+In Common Encryption, protection system specific information may be located
+within ISOBMFF files, both in movie (moov) boxes and movie fragment (moof)
+boxes; it may also be contained in ContentProtection elements within MPEG
+DASH MPDs. The events created by #gst_event_new_protection contain data
+identifying from which of these locations the encapsulated protection system
+specific information originated. This origin information is required as
+some protection systems use different encodings depending upon where the
+information originates.
+
+The events returned by gst_event_new_protection() are implemented
+in such a way as to ensure that the most recently-pushed protection info
+event of a particular @origin and @system_id will
+be stuck to the output pad of the sending element.
+
+ a #GST_EVENT_PROTECTION event, if successful; %NULL
+if unsuccessful.
+
+
+
+
+ a string holding a UUID that uniquely
+identifies a protection system.
+
+
+
+ a #GstBuffer holding protection system specific
+information. The reference count of the buffer will be incremented by one.
+
+
+
+ a string indicating where the protection
+information carried in the event was extracted from. The allowed values
+of this string will depend upon the protection scheme.
+
+
+
+
+
+ Allocate a new qos event with the given values.
+The QOS event is generated in an element that wants an upstream
+element to either reduce or increase its rate because of
+high/low CPU load or other resource usage such as network performance or
+throttling. Typically sinks generate these events for each buffer
+they receive.
+
+@type indicates the reason for the QoS event. #GST_QOS_TYPE_OVERFLOW is
+used when a buffer arrived in time or when the sink cannot keep up with
+the upstream datarate. #GST_QOS_TYPE_UNDERFLOW is when the sink is not
+receiving buffers fast enough and thus has to drop late buffers.
+#GST_QOS_TYPE_THROTTLE is used when the datarate is artificially limited
+by the application, for example to reduce power consumption.
+
+@proportion indicates the real-time performance of the streaming in the
+element that generated the QoS event (usually the sink). The value is
+generally computed based on more long term statistics about the streams
+timestamps compared to the clock.
+A value < 1.0 indicates that the upstream element is producing data faster
+than real-time. A value > 1.0 indicates that the upstream element is not
+producing data fast enough. 1.0 is the ideal @proportion value. The
+proportion value can safely be used to lower or increase the quality of
+the element.
+
+@diff is the difference against the clock in running time of the last
+buffer that caused the element to generate the QOS event. A negative value
+means that the buffer with @timestamp arrived in time. A positive value
+indicates how late the buffer with @timestamp was. When throttling is
+enabled, @diff will be set to the requested throttling interval.
+
+@timestamp is the timestamp of the last buffer that cause the element
+to generate the QOS event. It is expressed in running time and thus an ever
+increasing value.
+
+The upstream element can use the @diff and @timestamp values to decide
+whether to process more buffers. For positive @diff, all buffers with
+timestamp <= @timestamp + @diff will certainly arrive late in the sink
+as well. A (negative) @diff value so that @timestamp + @diff would yield a
+result smaller than 0 is not allowed.
+
+The application can use general event probes to intercept the QoS
+event and implement custom application specific QoS handling.
+
+ a new QOS event.
+
+
+
+
+ the QoS type
+
+
+
+ the proportion of the qos message
+
+
+
+ The time difference of the last Clock sync
+
+
+
+ The timestamp of the buffer
+
+
+
+
+
+ Create a new reconfigure event. The purpose of the reconfigure event is
+to travel upstream and make elements renegotiate their caps or reconfigure
+their buffer pools. This is useful when changing properties on elements
+or changing the topology of the pipeline.
+
+ a new #GstEvent
+
+
+
+
+ Allocate a new seek event with the given parameters.
+
+The seek event configures playback of the pipeline between @start to @stop
+at the speed given in @rate, also called a playback segment.
+The @start and @stop values are expressed in @format.
+
+A @rate of 1.0 means normal playback rate, 2.0 means double speed.
+Negatives values means backwards playback. A value of 0.0 for the
+rate is not allowed and should be accomplished instead by PAUSING the
+pipeline.
+
+A pipeline has a default playback segment configured with a start
+position of 0, a stop position of -1 and a rate of 1.0. The currently
+configured playback segment can be queried with #GST_QUERY_SEGMENT.
+
+@start_type and @stop_type specify how to adjust the currently configured
+start and stop fields in playback segment. Adjustments can be made relative
+or absolute to the last configured values. A type of #GST_SEEK_TYPE_NONE
+means that the position should not be updated.
+
+When the rate is positive and @start has been updated, playback will start
+from the newly configured start position.
+
+For negative rates, playback will start from the newly configured stop
+position (if any). If the stop position is updated, it must be different from
+-1 (#GST_CLOCK_TIME_NONE) for negative rates.
+
+It is not possible to seek relative to the current playback position, to do
+this, PAUSE the pipeline, query the current playback position with
+#GST_QUERY_POSITION and update the playback segment current position with a
+#GST_SEEK_TYPE_SET to the desired position.
+
+ a new seek event.
+
+
+
+
+ The new playback rate
+
+
+
+ The format of the seek values
+
+
+
+ The optional seek flags
+
+
+
+ The type and flags for the new start position
+
+
+
+ The value of the new start position
+
+
+
+ The type and flags for the new stop position
+
+
+
+ The value of the new stop position
+
+
+
+
+
+ Create a new SEGMENT event for @segment. The segment event can only travel
+downstream synchronized with the buffer flow and contains timing information
+and playback properties for the buffers that will follow.
+
+The segment event marks the range of buffers to be processed. All
+data not within the segment range is not to be processed. This can be
+used intelligently by plugins to apply more efficient methods of skipping
+unneeded data. The valid range is expressed with the @start and @stop
+values.
+
+The time value of the segment is used in conjunction with the start
+value to convert the buffer timestamps into the stream time. This is
+usually done in sinks to report the current stream_time.
+@time represents the stream_time of a buffer carrying a timestamp of
+@start. @time cannot be -1.
+
+@start cannot be -1, @stop can be -1. If there
+is a valid @stop given, it must be greater or equal the @start, including
+when the indicated playback @rate is < 0.
+
+The @applied_rate value provides information about any rate adjustment that
+has already been made to the timestamps and content on the buffers of the
+stream. (@rate * @applied_rate) should always equal the rate that has been
+requested for playback. For example, if an element has an input segment
+with intended playback @rate of 2.0 and applied_rate of 1.0, it can adjust
+incoming timestamps and buffer content by half and output a segment event
+with @rate of 1.0 and @applied_rate of 2.0
+
+After a segment event, the buffer stream time is calculated with:
+
+ time + (TIMESTAMP(buf) - start) * ABS (rate * applied_rate)
+
+ the new SEGMENT event.
+
+
+
+
+ a #GstSegment
+
+
+
+
+
+ Create a new segment-done event. This event is sent by elements that
+finish playback of a segment as a result of a segment seek.
+
+ a new #GstEvent
+
+
+
+
+ The format of the position being done
+
+
+
+ The position of the segment being done
+
+
+
+
+
+ Allocate a new select-streams event.
+
+The select-streams event requests the specified @streams to be activated.
+
+The list of @streams corresponds to the "Stream ID" of each stream to be
+activated. Those ID can be obtained via the #GstStream objects present
+in #GST_EVENT_STREAM_START, #GST_EVENT_STREAM_COLLECTION or
+#GST_MESSAGE_STREAM_COLLECTION.
+
+Note: The list of @streams can not be empty.
+
+ a new select-streams event or %NULL in case of
+an error (like an empty streams list).
+
+
+
+
+ the list of streams to
+activate
+
+
+
+
+
+
+
+ Create a new sink-message event. The purpose of the sink-message event is
+to instruct a sink to post the message contained in the event synchronized
+with the stream.
+
+@name is used to store multiple sticky events on one pad.
+
+ a new #GstEvent
+
+
+
+
+ a name for the event
+
+
+
+ the #GstMessage to be posted
+
+
+
+
+
+ Create a new step event. The purpose of the step event is to instruct a sink
+to skip @amount (expressed in @format) of media. It can be used to implement
+stepping through the video frame by frame or for doing fast trick modes.
+
+A rate of <= 0.0 is not allowed. Pause the pipeline, for the effect of rate
+= 0.0 or first reverse the direction of playback using a seek event to get
+the same effect as rate < 0.0.
+
+The @flush flag will clear any pending data in the pipeline before starting
+the step operation.
+
+The @intermediate flag instructs the pipeline that this step operation is
+part of a larger step operation.
+
+ a new #GstEvent
+
+
+
+
+ the format of @amount
+
+
+
+ the amount of data to step
+
+
+
+ the step rate
+
+
+
+ flushing steps
+
+
+
+ intermediate steps
+
+
+
+
+
+ Create a new STREAM_COLLECTION event. The stream collection event can only
+travel downstream synchronized with the buffer flow.
+
+Source elements, demuxers and other elements that manage collections
+of streams and post #GstStreamCollection messages on the bus also send
+this event downstream on each pad involved in the collection, so that
+activation of a new collection can be tracked through the downstream
+data flow.
+
+ the new STREAM_COLLECTION event.
+
+
+
+
+ Active collection for this data flow
+
+
+
+
+
+ Create a new Stream Group Done event. The stream-group-done event can
+only travel downstream synchronized with the buffer flow. Elements
+that receive the event on a pad should handle it mostly like EOS,
+and emit any data or pending buffers that would depend on more data
+arriving and unblock, since there won't be any more data.
+
+This event is followed by EOS at some point in the future, and is
+generally used when switching pads - to unblock downstream so that
+new pads can be exposed before sending EOS on the existing pads.
+
+ the new stream-group-done event.
+
+
+
+
+ the group id of the stream group which is ending
+
+
+
+
+
+ Create a new STREAM_START event. The stream start event can only
+travel downstream synchronized with the buffer flow. It is expected
+to be the first event that is sent for a new stream.
+
+Source elements, demuxers and other elements that create new streams
+are supposed to send this event as the first event of a new stream. It
+should not be sent after a flushing seek or in similar situations
+and is used to mark the beginning of a new logical stream. Elements
+combining multiple streams must ensure that this event is only forwarded
+downstream once and not for every single input stream.
+
+The @stream_id should be a unique string that consists of the upstream
+stream-id, / as separator and a unique stream-id for this specific
+stream. A new stream-id should only be created for a stream if the upstream
+stream is split into (potentially) multiple new streams, e.g. in a demuxer,
+but not for every single element in the pipeline.
+gst_pad_create_stream_id() or gst_pad_create_stream_id_printf() can be
+used to create a stream-id. There are no particular semantics for the
+stream-id, though it should be deterministic (to support stream matching)
+and it might be used to order streams (besides any information conveyed by
+stream flags).
+
+ the new STREAM_START event.
+
+
+
+
+ Identifier for this stream
+
+
+
+
+
+ Generates a metadata tag event from the given @taglist.
+
+The scope of the taglist specifies if the taglist applies to the
+complete medium or only to this specific stream. As the tag event
+is a sticky event, elements should merge tags received from
+upstream with a given scope with their own tags with the same
+scope and create a new tag event from it.
+
+ a new #GstEvent
+
+
+
+
+ metadata list. The event will take ownership
+ of the taglist.
+
+
+
+
+
+ Generate a TOC event from the given @toc. The purpose of the TOC event is to
+inform elements that some kind of the TOC was found.
+
+ a new #GstEvent.
+
+
+
+
+ #GstToc structure.
+
+
+
+ whether @toc was updated or not.
+
+
+
+
+
+ Generate a TOC select event with the given @uid. The purpose of the
+TOC select event is to start playback based on the TOC's entry with the
+given @uid.
+
+ a new #GstEvent.
+
+
+
+
+ UID in the TOC to start playback from.
+
+
+
+
+
+ Parses a segment @event and copies the #GstSegment into the location
+given by @segment.
+
+
+
+
+
+ The event to parse
+
+
+
+ a pointer to a #GstSegment
+
+
+
+
+
+ Retrieve the accumulated running time offset of the event.
+
+Events passing through #GstPads that have a running time
+offset set via gst_pad_set_offset() will get their offset
+adjusted according to the pad's offset.
+
+If the event contains any information that related to the
+running time, this information will need to be updated
+before usage with this offset.
+
+ The event's running time offset
+
+MT safe.
+
+
+
+
+ A #GstEvent.
+
+
+
+
+
+ Retrieve the sequence number of a event.
+
+Events have ever-incrementing sequence numbers, which may also be set
+explicitly via gst_event_set_seqnum(). Sequence numbers are typically used to
+indicate that a event corresponds to some other set of events or messages,
+for example an EOS event corresponding to a SEEK event. It is considered good
+practice to make this correspondence when possible, though it is not
+required.
+
+Note that events and messages share the same sequence number incrementor;
+two events or messages will never have the same sequence number unless
+that correspondence was made explicitly.
+
+ The event's sequence number.
+
+MT safe.
+
+
+
+
+ A #GstEvent.
+
+
+
+
+
+ Access the structure of the event.
+
+ The structure of the event. The
+structure is still owned by the event, which means that you should not free
+it and that the pointer becomes invalid when you free the event.
+
+MT safe.
+
+
+
+
+ The #GstEvent.
+
+
+
+
+
+ Checks if @event has the given @name. This function is usually used to
+check the name of a custom event.
+
+ %TRUE if @name matches the name of the event structure.
+
+
+
+
+ The #GstEvent.
+
+
+
+ name to check
+
+
+
+
+
+ Get the format, minsize, maxsize and async-flag in the buffersize event.
+
+
+
+
+
+ The event to query
+
+
+
+ A pointer to store the format in
+
+
+
+ A pointer to store the minsize in
+
+
+
+ A pointer to store the maxsize in
+
+
+
+ A pointer to store the async-flag in
+
+
+
+
+
+ Get the caps from @event. The caps remains valid as long as @event remains
+valid.
+
+
+
+
+
+ The event to parse
+
+
+
+ A pointer to the caps
+
+
+
+
+
+ Parse the FLUSH_STOP event and retrieve the @reset_time member.
+
+
+
+
+
+ The event to parse
+
+
+
+ if time should be reset
+
+
+
+
+
+ Extract timestamp and duration from a new GAP event.
+
+
+
+
+
+ a #GstEvent of type #GST_EVENT_GAP
+
+
+
+ location where to store the
+ start time (pts) of the gap, or %NULL
+
+
+
+ location where to store the duration of
+ the gap, or %NULL
+
+
+
+
+
+
+ %TRUE if a group id was set on the event and could be parsed,
+ %FALSE otherwise.
+
+
+
+
+ a stream-start event
+
+
+
+ address of variable where to store the group id
+
+
+
+
+
+ Get the latency in the latency event.
+
+
+
+
+
+ The event to query
+
+
+
+ A pointer to store the latency in.
+
+
+
+
+
+ Parses an event containing protection system specific information and stores
+the results in @system_id, @data and @origin. The data stored in @system_id,
+@origin and @data are valid until @event is released.
+
+
+
+
+
+ a #GST_EVENT_PROTECTION event.
+
+
+
+ pointer to store the UUID
+string uniquely identifying a content protection system.
+
+
+
+ pointer to store a #GstBuffer
+holding protection system specific information.
+
+
+
+ pointer to store a value that
+indicates where the protection information carried by @event was extracted
+from.
+
+
+
+
+
+ Get the type, proportion, diff and timestamp in the qos event. See
+gst_event_new_qos() for more information about the different QoS values.
+
+@timestamp will be adjusted for any pad offsets of pads it was passing through.
+
+
+
+
+
+ The event to query
+
+
+
+ A pointer to store the QoS type in
+
+
+
+ A pointer to store the proportion in
+
+
+
+ A pointer to store the diff in
+
+
+
+ A pointer to store the timestamp in
+
+
+
+
+
+ Parses a seek @event and stores the results in the given result locations.
+
+
+
+
+
+ a seek event
+
+
+
+ result location for the rate
+
+
+
+ result location for the stream format
+
+
+
+ result location for the #GstSeekFlags
+
+
+
+ result location for the #GstSeekType of the start position
+
+
+
+ result location for the start position expressed in @format
+
+
+
+ result location for the #GstSeekType of the stop position
+
+
+
+ result location for the stop position expressed in @format
+
+
+
+
+
+ Parses a segment @event and stores the result in the given @segment location.
+@segment remains valid only until the @event is freed. Don't modify the segment
+and make a copy if you want to modify it or store it for later use.
+
+
+
+
+
+ The event to parse
+
+
+
+ a pointer to a #GstSegment
+
+
+
+
+
+ Extracts the position and format from the segment done message.
+
+
+
+
+
+ A valid #GstEvent of type GST_EVENT_SEGMENT_DONE.
+
+
+
+ Result location for the format, or %NULL
+
+
+
+ Result location for the position, or %NULL
+
+
+
+
+
+ Parse the SELECT_STREAMS event and retrieve the contained streams.
+
+
+
+
+
+ The event to parse
+
+
+
+ the streams
+
+
+
+
+
+
+
+ Parse the sink-message event. Unref @msg after usage.
+
+
+
+
+
+ The event to query
+
+
+
+ a pointer to store the #GstMessage in.
+
+
+
+
+
+ Parse the step event.
+
+
+
+
+
+ The event to query
+
+
+
+ a pointer to store the format in
+
+
+
+ a pointer to store the amount in
+
+
+
+ a pointer to store the rate in
+
+
+
+ a pointer to store the flush boolean in
+
+
+
+ a pointer to store the intermediate
+ boolean in
+
+
+
+
+
+ Parse a stream-start @event and extract the #GstStream from it.
+
+
+
+
+
+ a stream-start event
+
+
+
+ adress of variable to store the stream
+
+
+
+
+
+ Retrieve new #GstStreamCollection from STREAM_COLLECTION event @event.
+
+
+
+
+
+ a stream-collection event
+
+
+
+ pointer to store the collection
+
+
+
+
+
+
+
+
+
+
+ a stream-start event
+
+
+
+ address of variable where to store the stream flags
+
+
+
+
+
+ Parse a stream-group-done @event and store the result in the given
+@group_id location.
+
+
+
+
+
+ a stream-group-done event.
+
+
+
+ address of variable to store the group id into
+
+
+
+
+
+ Parse a stream-id @event and store the result in the given @stream_id
+location. The string stored in @stream_id must not be modified and will
+remain valid only until @event gets freed. Make a copy if you want to
+modify it or store it for later use.
+
+
+
+
+
+ a stream-start event.
+
+
+
+ pointer to store the stream-id
+
+
+
+
+
+ Parses a tag @event and stores the results in the given @taglist location.
+No reference to the taglist will be returned, it remains valid only until
+the @event is freed. Don't modify or free the taglist, make a copy if you
+want to modify it or store it for later use.
+
+
+
+
+
+ a tag event
+
+
+
+ pointer to metadata list
+
+
+
+
+
+ Parse a TOC @event and store the results in the given @toc and @updated locations.
+
+
+
+
+
+ a TOC event.
+
+
+
+ pointer to #GstToc structure.
+
+
+
+ pointer to store TOC updated flag.
+
+
+
+
+
+ Parse a TOC select @event and store the results in the given @uid location.
+
+
+
+
+
+ a TOC select event.
+
+
+
+ storage for the selection UID.
+
+
+
+
+
+ All streams that have the same group id are supposed to be played
+together, i.e. all streams inside a container file should have the
+same group id but different stream ids. The group id should change
+each time the stream is started, resulting in different group ids
+each time a file is played for example.
+
+Use gst_util_group_id_next() to get a new group id.
+
+
+
+
+
+ a stream-start event
+
+
+
+ the group id to set
+
+
+
+
+
+ Set the running time offset of a event. See
+gst_event_get_running_time_offset() for more information.
+
+MT safe.
+
+
+
+
+
+ A #GstEvent.
+
+
+
+ A the new running time offset
+
+
+
+
+
+ Set the sequence number of a event.
+
+This function might be called by the creator of a event to indicate that the
+event relates to other events or messages. See gst_event_get_seqnum() for
+more information.
+
+MT safe.
+
+
+
+
+
+ A #GstEvent.
+
+
+
+ A sequence number.
+
+
+
+
+
+ Set the @stream on the stream-start @event
+
+
+
+
+
+ a stream-start event
+
+
+
+ the stream object to set
+
+
+
+
+
+
+
+
+
+
+ a stream-start event
+
+
+
+ the stream flags to set
+
+
+
+
+
+ Get a writable version of the structure.
+
+ The structure of the event. The structure
+is still owned by the event, which means that you should not free
+it and that the pointer becomes invalid when you free the event.
+This function checks if @event is writable and will never return
+%NULL.
+
+MT safe.
+
+
+
+
+ The #GstEvent.
+
+
+
+
+
+
+ #GstEventType lists the standard event types that can be sent in a pipeline.
+
+The custom event types can be used for private messages between elements
+that can't be expressed using normal
+GStreamer buffer passing semantics. Custom events carry an arbitrary
+#GstStructure.
+Specific custom events are distinguished by the name of the structure.
+
+ unknown event.
+
+
+ Start a flush operation. This event clears all data
+ from the pipeline and unblock all streaming threads.
+
+
+ Stop a flush operation. This event resets the
+ running-time of the pipeline.
+
+
+ Event to mark the start of a new stream. Sent before any
+ other serialized event and only sent at the start of a new stream,
+ not after flushing seeks.
+
+
+ #GstCaps event. Notify the pad of a new media type.
+
+
+ A new media segment follows in the dataflow. The
+ segment events contains information for clipping buffers and
+ converting buffer timestamps to running-time and
+ stream-time.
+
+
+ A new #GstStreamCollection is available (Since 1.10)
+
+
+ A new set of metadata tags has been found in the stream.
+
+
+ Notification of buffering requirements. Currently not
+ used yet.
+
+
+ An event that sinks turn into a message. Used to
+ send messages that should be emitted in sync with
+ rendering.
+
+
+ Indicates that there is no more data for
+ the stream group ID in the message. Sent before EOS
+ in some instances and should be handled mostly the same. (Since 1.10)
+
+
+ End-Of-Stream. No more data is to be expected to follow
+ without either a STREAM_START event, or a FLUSH_STOP and a SEGMENT
+ event.
+
+
+ An event which indicates that a new table of contents (TOC)
+ was found or updated.
+
+
+ An event which indicates that new or updated
+ encryption information has been found in the stream.
+
+
+ Marks the end of a segment playback.
+
+
+ Marks a gap in the datastream.
+
+
+ A quality message. Used to indicate to upstream elements
+ that the downstream elements should adjust their processing
+ rate.
+
+
+ A request for a new playback position and rate.
+
+
+ Navigation events are usually used for communicating
+ user requests, such as mouse or keyboard movements,
+ to upstream elements.
+
+
+ Notification of new latency adjustment. Sinks will use
+ the latency information to adjust their synchronisation.
+
+
+ A request for stepping through the media. Sinks will usually
+ execute the step operation.
+
+
+ A request for upstream renegotiating caps and reconfiguring.
+
+
+ A request for a new playback position based on TOC
+ entry's UID.
+
+
+ A request to select one or more streams (Since 1.10)
+
+
+ Upstream custom event
+
+
+ Downstream custom event that travels in the
+ data flow.
+
+
+ Custom out-of-band downstream event.
+
+
+ Custom sticky downstream event.
+
+
+ Custom upstream or downstream event.
+ In-band when travelling downstream.
+
+
+ Custom upstream or downstream out-of-band event.
+
+
+ Gets the #GstEventTypeFlags associated with @type.
+
+ a #GstEventTypeFlags.
+
+
+
+
+ a #GstEventType
+
+
+
+
+
+ Get a printable name for the given event type. Do not modify or free.
+
+ a reference to the static name of the event.
+
+
+
+
+ the event type
+
+
+
+
+
+ Get the unique quark for the given event type.
+
+ the quark associated with the event type
+
+
+
+
+ the event type
+
+
+
+
+
+
+ #GstEventTypeFlags indicate the aspects of the different #GstEventType
+values. You can get the type flags of a #GstEventType with the
+gst_event_type_get_flags() function.
+
+ Set if the event can travel upstream.
+
+
+ Set if the event can travel downstream.
+
+
+ Set if the event should be serialized with data
+ flow.
+
+
+ Set if the event is sticky on the pads.
+
+
+ Multiple sticky events can be on a pad, each
+ identified by the event name.
+
+
+
+ A mask value with all bits set, for use as a
+GstFlagSet mask where all flag bits must match
+exactly
+
+
+
+ The PERCENT format is between 0 and this value
+
+
+
+ The value used to scale down the reported PERCENT format value to
+its real value.
+
+
+
+ Can be used together with #GST_FOURCC_ARGS to properly output a
+#guint32 fourcc value in a printf()-style text message.
+
+|[
+printf ("fourcc: %" GST_FOURCC_FORMAT "\n", GST_FOURCC_ARGS (fcc));
+]|
+
+
+
+
+ Create a new sub-class of #GST_TYPE_FLAG_SET
+which will pretty-print the human-readable flags
+when serializing, for easier debugging.
+
+
+
+
+
+ a #GType of a #G_TYPE_FLAGS type.
+
+
+
+
+
+
+ The result of passing data to a pad.
+
+Note that the custom return values should not be exposed outside of the
+element scope.
+
+ Pre-defined custom success code.
+
+
+ Pre-defined custom success code (define your
+ custom success code to this to avoid compiler
+ warnings).
+
+
+ Elements can use values starting from
+ this (and higher) to define custom success
+ codes.
+
+
+ Data passing was ok.
+
+
+ Pad is not linked.
+
+
+ Pad is flushing.
+
+
+ Pad is EOS.
+
+
+ Pad is not negotiated.
+
+
+ Some (fatal) error occurred. Element generating
+ this error should post an error message with more
+ details.
+
+
+ This operation is not supported.
+
+
+ Elements can use values starting from
+ this (and lower) to define custom error codes.
+
+
+ Pre-defined custom error code (define your
+ custom error code to this to avoid compiler
+ warnings).
+
+
+ Pre-defined custom error code.
+
+
+
+ Standard predefined formats
+
+ undefined format
+
+
+ the default format of the pad/element. This can be
+ samples for raw audio, frames/fields for raw video (some, but not all,
+ elements support this; use @GST_FORMAT_TIME if you don't have a good
+ reason to query for samples/frames)
+
+
+ bytes
+
+
+ time in nanoseconds
+
+
+ buffers (few, if any, elements implement this as of
+ May 2009)
+
+
+ percentage of stream (few, if any, elements implement
+ this as of May 2009)
+
+
+ Return the format registered with the given nick.
+
+ The format with @nick or GST_FORMAT_UNDEFINED
+if the format was not registered.
+
+
+
+
+ The nick of the format
+
+
+
+
+
+ Get details about the given format.
+
+ The #GstFormatDefinition for @format or %NULL
+on failure.
+
+MT safe.
+
+
+
+
+ The format to get details of
+
+
+
+
+
+ Get a printable name for the given format. Do not modify or free.
+
+ a reference to the static name of the format
+or %NULL if the format is unknown.
+
+
+
+
+ a #GstFormat
+
+
+
+
+
+ Iterate all the registered formats. The format definition is read
+only.
+
+ a GstIterator of #GstFormatDefinition.
+
+
+
+
+ Create a new GstFormat based on the nick or return an
+already registered format with that nick.
+
+ A new GstFormat or an already registered format
+with the same nick.
+
+MT safe.
+
+
+
+
+ The nick of the new format
+
+
+
+ The description of the new format
+
+
+
+
+
+ Get the unique quark for the given format.
+
+ the quark associated with the format or 0 if the format
+is unknown.
+
+
+
+
+ a #GstFormat
+
+
+
+
+
+
+ A format definition
+
+ The unique id of this format
+
+
+
+ A short nick of the format
+
+
+
+ A longer description of the format
+
+
+
+ A quark for the nick
+
+
+
+
+
+
+
+
+ A value which is guaranteed to never be returned by
+gst_util_group_id_next().
+
+Can be used as a default value in variables used to store group_id.
+
+
+
+ GhostPads are useful when organizing pipelines with #GstBin like elements.
+The idea here is to create hierarchical element graphs. The bin element
+contains a sub-graph. Now one would like to treat the bin-element like any
+other #GstElement. This is where GhostPads come into play. A GhostPad acts as
+a proxy for another pad. Thus the bin can have sink and source ghost-pads
+that are associated with sink and source pads of the child elements.
+
+If the target pad is known at creation time, gst_ghost_pad_new() is the
+function to use to get a ghost-pad. Otherwise one can use gst_ghost_pad_new_no_target()
+to create the ghost-pad and use gst_ghost_pad_set_target() to establish the
+association later on.
+
+Note that GhostPads add overhead to the data processing of a pipeline.
+
+ Create a new ghostpad with @target as the target. The direction will be taken
+from the target pad. @target must be unlinked.
+
+Will ref the target.
+
+ a new #GstPad, or %NULL in
+case of an error.
+
+
+
+
+ the name of the new pad, or %NULL to assign a default name
+
+
+
+ the pad to ghost.
+
+
+
+
+
+ Create a new ghostpad with @target as the target. The direction will be taken
+from the target pad. The template used on the ghostpad will be @template.
+
+Will ref the target.
+
+ a new #GstPad, or %NULL in
+case of an error.
+
+
+
+
+ the name of the new pad, or %NULL to assign a default name.
+
+
+
+ the pad to ghost.
+
+
+
+ the #GstPadTemplate to use on the ghostpad.
+
+
+
+
+
+ Create a new ghostpad without a target with the given direction.
+A target can be set on the ghostpad later with the
+gst_ghost_pad_set_target() function.
+
+The created ghostpad will not have a padtemplate.
+
+ a new #GstPad, or %NULL in
+case of an error.
+
+
+
+
+ the name of the new pad, or %NULL to assign a default name.
+
+
+
+ the direction of the ghostpad
+
+
+
+
+
+ Create a new ghostpad based on @templ, without setting a target. The
+direction will be taken from the @templ.
+
+ a new #GstPad, or %NULL in
+case of an error.
+
+
+
+
+ the name of the new pad, or %NULL to assign a default name
+
+
+
+ the #GstPadTemplate to create the ghostpad from.
+
+
+
+
+
+ Invoke the default activate mode function of a ghost pad.
+
+ %TRUE if the operation was successful.
+
+
+
+
+ the #GstPad to activate or deactivate.
+
+
+
+ the parent of @pad or %NULL
+
+
+
+ the requested activation mode
+
+
+
+ whether the pad should be active or not.
+
+
+
+
+
+ Invoke the default activate mode function of a proxy pad that is
+owned by a ghost pad.
+
+ %TRUE if the operation was successful.
+
+
+
+
+ the #GstPad to activate or deactivate.
+
+
+
+ the parent of @pad or %NULL
+
+
+
+ the requested activation mode
+
+
+
+ whether the pad should be active or not.
+
+
+
+
+
+ Finish initialization of a newly allocated ghost pad.
+
+This function is most useful in language bindings and when subclassing
+#GstGhostPad; plugin and application developers normally will not call this
+function. Call this function directly after a call to g_object_new
+(GST_TYPE_GHOST_PAD, "direction", @dir, ..., NULL).
+
+ %TRUE if the construction succeeds, %FALSE otherwise.
+
+
+
+
+ the newly allocated ghost pad
+
+
+
+
+
+ Get the target pad of @gpad. Unref target pad after usage.
+
+ the target #GstPad, can be
+%NULL if the ghostpad has no target set. Unref target pad after
+usage.
+
+
+
+
+ the #GstGhostPad
+
+
+
+
+
+ Set the new target of the ghostpad @gpad. Any existing target
+is unlinked and links to the new target are established. if @newtarget is
+%NULL the target will be cleared.
+
+ %TRUE if the new target could be set. This function
+ can return %FALSE when the internal pads could not be linked.
+
+
+
+
+ the #GstGhostPad
+
+
+
+ the new pad target
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ A GstIterator is used to retrieve multiple objects from another object in
+a threadsafe way.
+
+Various GStreamer objects provide access to their internal structures using
+an iterator.
+
+Note that if calling a GstIterator function results in your code receiving
+a refcounted object (with, say, g_value_get_object()), the refcount for that
+object will not be increased. Your code is responsible for taking a reference
+if it wants to continue using it later.
+
+The basic use pattern of an iterator is as follows:
+|[<!-- language="C" -->
+ GstIterator *it = _get_iterator(object);
+ GValue item = G_VALUE_INIT;
+ done = FALSE;
+ while (!done) {
+ switch (gst_iterator_next (it, &item)) {
+ case GST_ITERATOR_OK:
+ ...get/use/change item here...
+ g_value_reset (&item);
+ break;
+ case GST_ITERATOR_RESYNC:
+ ...rollback changes to items...
+ gst_iterator_resync (it);
+ break;
+ case GST_ITERATOR_ERROR:
+ ...wrong parameters were given...
+ done = TRUE;
+ break;
+ case GST_ITERATOR_DONE:
+ done = TRUE;
+ break;
+ }
+ }
+ g_value_unset (&item);
+ gst_iterator_free (it);
+]|
+
+ The function to copy the iterator
+
+
+
+ The function to get the next item in the iterator
+
+
+
+ The function to be called for each item retrieved
+
+
+
+ The function to call when a resync is needed.
+
+
+
+ The function to call when the iterator is freed
+
+
+
+ The iterator that is currently pushed with gst_iterator_push()
+
+
+
+ The type of the object that this iterator will return
+
+
+
+ The lock protecting the data structure and the cookie.
+
+
+
+ The cookie; the value of the master_cookie when this iterator was
+ created.
+
+
+
+ A pointer to the master cookie.
+
+
+
+ the size of the iterator
+
+
+
+
+
+
+
+
+ Create a new iterator. This function is mainly used for objects
+implementing the next/resync/free function to iterate a data structure.
+
+For each item retrieved, the @item function is called with the lock
+held. The @free function is called when the iterator is freed.
+
+ the new #GstIterator.
+
+MT safe.
+
+
+
+
+ the size of the iterator structure
+
+
+
+ #GType of children
+
+
+
+ pointer to a #GMutex.
+
+
+
+ pointer to a guint32 that is changed when the items in the
+ iterator changed.
+
+
+
+ copy function
+
+
+
+ function to get next item
+
+
+
+ function to call on each item retrieved
+
+
+
+ function to resync the iterator
+
+
+
+ function to free the iterator
+
+
+
+
+
+ Create a new iterator designed for iterating @list.
+
+The list you iterate is usually part of a data structure @owner and is
+protected with @lock.
+
+The iterator will use @lock to retrieve the next item of the list and it
+will then call the @item function before releasing @lock again.
+
+When a concurrent update to the list is performed, usually by @owner while
+holding @lock, @master_cookie will be updated. The iterator implementation
+will notice the update of the cookie and will return %GST_ITERATOR_RESYNC to
+the user of the iterator in the next call to gst_iterator_next().
+
+ the new #GstIterator for @list.
+
+MT safe.
+
+
+
+
+ #GType of elements
+
+
+
+ pointer to a #GMutex protecting the list.
+
+
+
+ pointer to a guint32 that is incremented when the list
+ is changed.
+
+
+
+ pointer to the list
+
+
+
+
+
+ object owning the list
+
+
+
+ function to call on each item retrieved
+
+
+
+
+
+ This #GstIterator is a convenient iterator for the common
+case where a #GstIterator needs to be returned but only
+a single object has to be considered. This happens often
+for the #GstPadIterIntLinkFunction.
+
+ the new #GstIterator for @object.
+
+
+
+
+ #GType of the passed object
+
+
+
+ object that this iterator should return
+
+
+
+
+
+ Copy the iterator and its state.
+
+ a new copy of @it.
+
+
+
+
+ a #GstIterator
+
+
+
+
+
+ Create a new iterator from an existing iterator. The new iterator
+will only return those elements that match the given compare function @func.
+The first parameter that is passed to @func is the #GValue of the current
+iterator element and the second parameter is @user_data. @func should
+return 0 for elements that should be included in the filtered iterator.
+
+When this iterator is freed, @it will also be freed.
+
+ a new #GstIterator.
+
+MT safe.
+
+
+
+
+ The #GstIterator to filter
+
+
+
+ the compare function to select elements
+
+
+
+ user data passed to the compare function
+
+
+
+
+
+ Find the first element in @it that matches the compare function @func.
+@func should return 0 when the element is found. The first parameter
+to @func will be the current element of the iterator and the
+second parameter will be @user_data.
+The result will be stored in @elem if a result is found.
+
+The iterator will not be freed.
+
+This function will return %FALSE if an error happened to the iterator
+or if the element wasn't found.
+
+ Returns %TRUE if the element was found, else %FALSE.
+
+MT safe.
+
+
+
+
+ The #GstIterator to iterate
+
+
+
+ the compare function to use
+
+
+
+ pointer to a #GValue where to store the result
+
+
+
+ user data passed to the compare function
+
+
+
+
+
+ Folds @func over the elements of @iter. That is to say, @func will be called
+as @func (object, @ret, @user_data) for each object in @it. The normal use
+of this procedure is to accumulate the results of operating on the objects in
+@ret.
+
+This procedure can be used (and is used internally) to implement the
+gst_iterator_foreach() and gst_iterator_find_custom() operations.
+
+The fold will proceed as long as @func returns %TRUE. When the iterator has no
+more arguments, %GST_ITERATOR_DONE will be returned. If @func returns %FALSE,
+the fold will stop, and %GST_ITERATOR_OK will be returned. Errors or resyncs
+will cause fold to return %GST_ITERATOR_ERROR or %GST_ITERATOR_RESYNC as
+appropriate.
+
+The iterator will not be freed.
+
+ A #GstIteratorResult, as described above.
+
+MT safe.
+
+
+
+
+ The #GstIterator to fold over
+
+
+
+ the fold function
+
+
+
+ the seed value passed to the fold function
+
+
+
+ user data passed to the fold function
+
+
+
+
+
+ Iterate over all element of @it and call the given function @func for
+each element.
+
+ the result call to gst_iterator_fold(). The iterator will not be
+freed.
+
+MT safe.
+
+
+
+
+ The #GstIterator to iterate
+
+
+
+ the function to call for each element.
+
+
+
+ user data passed to the function
+
+
+
+
+
+ Free the iterator.
+
+MT safe.
+
+
+
+
+
+ The #GstIterator to free
+
+
+
+
+
+ Get the next item from the iterator in @elem.
+
+Only when this function returns %GST_ITERATOR_OK, @elem will contain a valid
+value. @elem must have been initialized to the type of the iterator or
+initialized to zeroes with g_value_unset(). The caller is responsible for
+unsetting or resetting @elem with g_value_unset() or g_value_reset()
+after usage.
+
+When this function returns %GST_ITERATOR_DONE, no more elements can be
+retrieved from @it.
+
+A return value of %GST_ITERATOR_RESYNC indicates that the element list was
+concurrently updated. The user of @it should call gst_iterator_resync() to
+get the newly updated list.
+
+A return value of %GST_ITERATOR_ERROR indicates an unrecoverable fatal error.
+
+ The result of the iteration. Unset @elem after usage.
+
+MT safe.
+
+
+
+
+ The #GstIterator to iterate
+
+
+
+ pointer to hold next element
+
+
+
+
+
+ Pushes @other iterator onto @it. All calls performed on @it are
+forwarded to @other. If @other returns %GST_ITERATOR_DONE, it is
+popped again and calls are handled by @it again.
+
+This function is mainly used by objects implementing the iterator
+next function to recurse into substructures.
+
+When gst_iterator_resync() is called on @it, @other will automatically be
+popped.
+
+MT safe.
+
+
+
+
+
+ The #GstIterator to use
+
+
+
+ The #GstIterator to push
+
+
+
+
+
+ Resync the iterator. this function is mostly called
+after gst_iterator_next() returned %GST_ITERATOR_RESYNC.
+
+When an iterator was pushed on @it, it will automatically be popped again
+with this function.
+
+MT safe.
+
+
+
+
+
+ The #GstIterator to resync
+
+
+
+
+
+
+ This function will be called when creating a copy of @it and should
+create a copy of all custom iterator fields or increase their
+reference counts.
+
+
+
+
+
+ The original iterator
+
+
+
+ The copied iterator
+
+
+
+
+
+ A function to be passed to gst_iterator_fold().
+
+ %TRUE if the fold should continue, %FALSE if it should stop.
+
+
+
+
+ the item to fold
+
+
+
+ a #GValue collecting the result
+
+
+
+ data passed to gst_iterator_fold()
+
+
+
+
+
+ A function that is called by gst_iterator_foreach() for every element.
+
+
+
+
+
+ The item
+
+
+
+ User data
+
+
+
+
+
+ This function will be called when the iterator is freed.
+
+Implementors of a #GstIterator should implement this
+function and pass it to the constructor of the custom iterator.
+The function will be called with the iterator lock held.
+
+
+
+
+
+ the iterator
+
+
+
+
+
+ The result of a #GstIteratorItemFunction.
+
+ Skip this item
+
+
+ Return item
+
+
+ Stop after this item.
+
+
+
+ The function that will be called after the next item of the iterator
+has been retrieved. This function can be used to skip items or stop
+the iterator.
+
+The function will be called with the iterator lock held.
+
+ the result of the operation.
+
+
+
+
+ the iterator
+
+
+
+ the item being retrieved.
+
+
+
+
+
+ The function that will be called when the next element of the iterator
+should be retrieved.
+
+Implementors of a #GstIterator should implement this
+function and pass it to the constructor of the custom iterator.
+The function will be called with the iterator lock held.
+
+ the result of the operation.
+
+
+
+
+ the iterator
+
+
+
+ a pointer to hold the next item
+
+
+
+
+
+ The result of gst_iterator_next().
+
+ No more items in the iterator
+
+
+ An item was retrieved
+
+
+ Datastructure changed while iterating
+
+
+ An error happened
+
+
+
+ This function will be called whenever a concurrent update happened
+to the iterated datastructure. The implementor of the iterator should
+restart the iterator from the beginning and clean up any state it might
+have.
+
+Implementors of a #GstIterator should implement this
+function and pass it to the constructor of the custom iterator.
+The function will be called with the iterator lock held.
+
+
+
+
+
+ the iterator
+
+
+
+
+
+ To be used in GST_PLUGIN_DEFINE if unsure about the licence.
+
+
+
+ GstLockFlags value alias for GST_LOCK_FLAG_READ | GST_LOCK_FLAG_WRITE
+
+
+
+ Library errors are for errors from the library being used by elements
+(initializing, finalizing, settings, ...)
+
+ a general error which doesn't fit in any other
+category. Make sure you add a custom message to the error call.
+
+
+ do not use this except as a placeholder for
+deciding where to go while developing code.
+
+
+ used when the library could not be opened.
+
+
+ used when the library could not be closed.
+
+
+ used when the library doesn't accept settings.
+
+
+ used when the library generated an encoding error.
+
+
+ the number of library error types.
+
+
+
+
+
+
+
+
+ Flags used when locking miniobjects
+
+ lock for read access
+
+
+ lock for write access
+
+
+ lock for exclusive access
+
+
+ first flag that can be used for custom purposes
+
+
+
+ Function prototype for a logging function that can be registered with
+gst_debug_add_log_function().
+Use G_GNUC_NO_INSTRUMENT on that function.
+
+
+
+
+
+ a #GstDebugCategory
+
+
+
+ a #GstDebugLevel
+
+
+
+ file name
+
+
+
+ function name
+
+
+
+ line number
+
+
+
+ a #GObject
+
+
+
+ the message
+
+
+
+ user data for the log function
+
+
+
+
+
+ GstMapFlags value alias for GST_MAP_READ | GST_MAP_WRITE
+
+
+
+ This metadata stays relevant as long as memory layout is unchanged.
+
+
+
+ Constant that defines one GStreamer millisecond.
+
+
+
+ Flags used when mapping memory
+
+ map for read access
+
+
+ map for write access
+
+
+ first flag that can be used for custom purposes
+
+
+
+ A structure containing the result of a map operation such as
+gst_memory_map(). It contains the data and size.
+
+ a pointer to the mapped memory
+
+
+
+ flags used when mapping the memory
+
+
+
+ a pointer to the mapped data
+
+
+
+
+
+ the valid size in @data
+
+
+
+ the maximum bytes in @data
+
+
+
+ extra private user_data that the implementation of the memory
+ can use to store extra info.
+
+
+
+
+
+
+
+
+
+
+
+ GstMemory is a lightweight refcounted object that wraps a region of memory.
+They are typically used to manage the data of a #GstBuffer.
+
+A GstMemory object has an allocated region of memory of maxsize. The maximum
+size does not change during the lifetime of the memory object. The memory
+also has an offset and size property that specifies the valid range of memory
+in the allocated region.
+
+Memory is usually created by allocators with a gst_allocator_alloc()
+method call. When %NULL is used as the allocator, the default allocator will
+be used.
+
+New allocators can be registered with gst_allocator_register().
+Allocators are identified by name and can be retrieved with
+gst_allocator_find(). gst_allocator_set_default() can be used to change the
+default allocator.
+
+New memory can be created with gst_memory_new_wrapped() that wraps the memory
+allocated elsewhere.
+
+Refcounting of the memory block is performed with gst_memory_ref() and
+gst_memory_unref().
+
+The size of the memory can be retrieved and changed with
+gst_memory_get_sizes() and gst_memory_resize() respectively.
+
+Getting access to the data of the memory is performed with gst_memory_map().
+The call will return a pointer to offset bytes into the region of memory.
+After the memory access is completed, gst_memory_unmap() should be called.
+
+Memory can be copied with gst_memory_copy(), which will return a writable
+copy. gst_memory_share() will create a new memory block that shares the
+memory with an existing memory block at a custom offset and with a custom
+size.
+
+Memory can be efficiently merged when gst_memory_is_span() returns %TRUE.
+
+ parent structure
+
+
+
+ pointer to the #GstAllocator
+
+
+
+ parent memory block
+
+
+
+ the maximum size allocated
+
+
+
+ the alignment of the memory
+
+
+
+ the offset where valid data starts
+
+
+
+ the size of valid data
+
+
+
+ Allocate a new memory block that wraps the given @data.
+
+The prefix/padding must be filled with 0 if @flags contains
+#GST_MEMORY_FLAG_ZERO_PREFIXED and #GST_MEMORY_FLAG_ZERO_PADDED respectively.
+
+ a new #GstMemory.
+
+
+
+
+ #GstMemoryFlags
+
+
+
+ data to
+ wrap
+
+
+
+
+
+ allocated size of @data
+
+
+
+ offset in @data
+
+
+
+ size of valid data
+
+
+
+ user_data
+
+
+
+ called with @user_data when the memory is freed
+
+
+
+
+
+ Return a copy of @size bytes from @mem starting from @offset. This copy is
+guaranteed to be writable. @size can be set to -1 to return a copy
+from @offset to the end of the memory region.
+
+ a new #GstMemory.
+
+
+
+
+ a #GstMemory
+
+
+
+ offset to copy from
+
+
+
+ size to copy, or -1 to copy to the end of the memory region
+
+
+
+
+
+ Get the current @size, @offset and @maxsize of @mem.
+
+ the current sizes of @mem
+
+
+
+
+ a #GstMemory
+
+
+
+ pointer to offset
+
+
+
+ pointer to maxsize
+
+
+
+
+
+ Initializes a newly allocated @mem with the given parameters. This function
+will call gst_mini_object_init() with the default memory parameters.
+
+
+
+
+
+ a #GstMemory
+
+
+
+ #GstMemoryFlags
+
+
+
+ the #GstAllocator
+
+
+
+ the parent of @mem
+
+
+
+ the total size of the memory
+
+
+
+ the alignment of the memory
+
+
+
+ The offset in the memory
+
+
+
+ the size of valid data in the memory
+
+
+
+
+
+ Check if @mem1 and mem2 share the memory with a common parent memory object
+and that the memory is contiguous.
+
+If this is the case, the memory of @mem1 and @mem2 can be merged
+efficiently by performing gst_memory_share() on the parent object from
+the returned @offset.
+
+ %TRUE if the memory is contiguous and of a common parent.
+
+
+
+
+ a #GstMemory
+
+
+
+ a #GstMemory
+
+
+
+ a pointer to a result offset
+
+
+
+
+
+ Check if @mem if allocated with an allocator for @mem_type.
+
+ %TRUE if @mem was allocated from an allocator for @mem_type.
+
+
+
+
+ a #GstMemory
+
+
+
+ a memory type
+
+
+
+
+
+ Create a #GstMemory object that is mapped with @flags. If @mem is mappable
+with @flags, this function returns the mapped @mem directly. Otherwise a
+mapped copy of @mem is returned.
+
+This function takes ownership of old @mem and returns a reference to a new
+#GstMemory.
+
+ a #GstMemory object mapped
+with @flags or %NULL when a mapping is not possible.
+
+
+
+
+ a #GstMemory
+
+
+
+ pointer for info
+
+
+
+ mapping flags
+
+
+
+
+
+ Fill @info with the pointer and sizes of the memory in @mem that can be
+accessed according to @flags.
+
+This function can return %FALSE for various reasons:
+- the memory backed by @mem is not accessible with the given @flags.
+- the memory was already mapped with a different mapping.
+
+@info and its contents remain valid for as long as @mem is valid and
+until gst_memory_unmap() is called.
+
+For each gst_memory_map() call, a corresponding gst_memory_unmap() call
+should be done.
+
+ %TRUE if the map operation was successful.
+
+
+
+
+ a #GstMemory
+
+
+
+ pointer for info
+
+
+
+ mapping flags
+
+
+
+
+
+ Resize the memory region. @mem should be writable and offset + size should be
+less than the maxsize of @mem.
+
+#GST_MEMORY_FLAG_ZERO_PREFIXED and #GST_MEMORY_FLAG_ZERO_PADDED will be
+cleared when offset or padding is increased respectively.
+
+
+
+
+
+ a #GstMemory
+
+
+
+ a new offset
+
+
+
+ a new size
+
+
+
+
+
+ Return a shared copy of @size bytes from @mem starting from @offset. No
+memory copy is performed and the memory region is simply shared. The result
+is guaranteed to be non-writable. @size can be set to -1 to return a shared
+copy from @offset to the end of the memory region.
+
+ a new #GstMemory.
+
+
+
+
+ a #GstMemory
+
+
+
+ offset to share from
+
+
+
+ size to share, or -1 to share to the end of the memory region
+
+
+
+
+
+ Release the memory obtained with gst_memory_map()
+
+
+
+
+
+ a #GstMemory
+
+
+
+ a #GstMapInfo
+
+
+
+
+
+
+ Copy @size bytes from @mem starting at @offset and return them wrapped in a
+new GstMemory object.
+If @size is set to -1, all bytes starting at @offset are copied.
+
+ a new #GstMemory object wrapping a copy of the requested region in
+@mem.
+
+
+
+
+ a #GstMemory
+
+
+
+ an offset
+
+
+
+ a size or -1
+
+
+
+
+
+ Flags for wrapped memory.
+
+ memory is readonly. It is not allowed to map the
+memory with #GST_MAP_WRITE.
+
+
+ memory must not be shared. Copies will have to be
+made when this memory needs to be shared between buffers.
+
+
+ the memory prefix is filled with 0 bytes
+
+
+ the memory padding is filled with 0 bytes
+
+
+ the memory is physically contiguous. (Since 1.2)
+
+
+ the memory can't be mapped via gst_memory_map() without any preconditions. (Since 1.2)
+
+
+ first flag that can be used for custom purposes
+
+
+
+ Check if @mem1 and @mem2 occupy contiguous memory and return the offset of
+@mem1 in the parent buffer in @offset.
+
+ %TRUE if @mem1 and @mem2 are in contiguous memory.
+
+
+
+
+ a #GstMemory
+
+
+
+ a #GstMemory
+
+
+
+ a result offset
+
+
+
+
+
+ Get the memory of @mem that can be accessed according to the mode specified
+in @info's flags. The function should return a pointer that contains at least
+@maxsize bytes.
+
+ a pointer to memory of which at least @maxsize bytes can be
+accessed according to the access pattern in @info's flags.
+
+
+
+
+ a #GstMemory
+
+
+
+ the #GstMapInfo to map with
+
+
+
+ size to map
+
+
+
+
+
+ Get the memory of @mem that can be accessed according to the mode specified
+in @flags. The function should return a pointer that contains at least
+@maxsize bytes.
+
+ a pointer to memory of which at least @maxsize bytes can be
+accessed according to the access pattern in @flags.
+
+
+
+
+ a #GstMemory
+
+
+
+ size to map
+
+
+
+ access mode for the memory
+
+
+
+
+
+ Share @size bytes from @mem starting at @offset and return them wrapped in a
+new GstMemory object. If @size is set to -1, all bytes starting at @offset are
+shared. This function does not make a copy of the bytes in @mem.
+
+ a new #GstMemory object sharing the requested region in @mem.
+
+
+
+
+ a #GstMemory
+
+
+
+ an offset
+
+
+
+ a size or -1
+
+
+
+
+
+ Return the pointer previously retrieved with gst_memory_map() with @info.
+
+
+
+
+
+ a #GstMemory
+
+
+
+ a #GstMapInfo
+
+
+
+
+
+ Return the pointer previously retrieved with gst_memory_map().
+
+
+
+
+
+ a #GstMemory
+
+
+
+
+
+ Messages are implemented as a subclass of #GstMiniObject with a generic
+#GstStructure as the content. This allows for writing custom messages without
+requiring an API change while allowing a wide range of different types
+of messages.
+
+Messages are posted by objects in the pipeline and are passed to the
+application using the #GstBus.
+
+The basic use pattern of posting a message on a #GstBus is as follows:
+|[<!-- language="C" -->
+ gst_bus_post (bus, gst_message_new_eos());
+]|
+
+A #GstElement usually posts messages on the bus provided by the parent
+container using gst_element_post_message().
+
+ the parent structure
+
+
+
+ the #GstMessageType of the message
+
+
+
+ the timestamp of the message
+
+
+
+ the src of the message
+
+
+
+ the sequence number of the message
+
+
+
+
+
+
+
+
+
+ Create a new application-typed message. GStreamer will never create these
+messages; they are a gift from us to you. Enjoy.
+
+ The new application message.
+
+MT safe.
+
+
+
+
+ The object originating the message.
+
+
+
+ the structure for the message. The message
+ will take ownership of the structure.
+
+
+
+
+
+ The message is posted when elements completed an ASYNC state change.
+@running_time contains the time of the desired running_time when this
+elements goes to PLAYING. A value of #GST_CLOCK_TIME_NONE for @running_time
+means that the element has no clock interaction and thus doesn't care about
+the running_time of the pipeline.
+
+ The new async_done message.
+
+MT safe.
+
+
+
+
+ The object originating the message.
+
+
+
+ the desired running_time
+
+
+
+
+
+ This message is posted by elements when they start an ASYNC state change.
+
+ The new async_start message.
+
+MT safe.
+
+
+
+
+ The object originating the message.
+
+
+
+
+
+ Create a new buffering message. This message can be posted by an element that
+needs to buffer data before it can continue processing. @percent should be a
+value between 0 and 100. A value of 100 means that the buffering completed.
+
+When @percent is < 100 the application should PAUSE a PLAYING pipeline. When
+@percent is 100, the application can set the pipeline (back) to PLAYING.
+The application must be prepared to receive BUFFERING messages in the
+PREROLLING state and may only set the pipeline to PLAYING after receiving a
+message with @percent set to 100, which can happen after the pipeline
+completed prerolling.
+
+MT safe.
+
+ The new buffering message.
+
+
+
+
+ The object originating the message.
+
+
+
+ The buffering percent
+
+
+
+
+
+ Create a clock lost message. This message is posted whenever the
+clock is not valid anymore.
+
+If this message is posted by the pipeline, the pipeline will
+select a new clock again when it goes to PLAYING. It might therefore
+be needed to set the pipeline to PAUSED and PLAYING again.
+
+ The new clock lost message.
+
+MT safe.
+
+
+
+
+ The object originating the message.
+
+
+
+ the clock that was lost
+
+
+
+
+
+ Create a clock provide message. This message is posted whenever an
+element is ready to provide a clock or lost its ability to provide
+a clock (maybe because it paused or became EOS).
+
+This message is mainly used internally to manage the clock
+selection.
+
+ the new provide clock message.
+
+MT safe.
+
+
+
+
+ The object originating the message.
+
+
+
+ the clock it provides
+
+
+
+ %TRUE if the sender can provide a clock
+
+
+
+
+
+ Create a new custom-typed message. This can be used for anything not
+handled by other message-specific functions to pass a message to the
+app. The structure field can be %NULL.
+
+ The new message.
+
+MT safe.
+
+
+
+
+ The #GstMessageType to distinguish messages
+
+
+
+ The object originating the message.
+
+
+
+ the structure for the
+ message. The message will take ownership of the structure.
+
+
+
+
+
+ Creates a new device-added message. The device-added message is produced by
+#GstDeviceProvider or a #GstDeviceMonitor. They announce the appearance
+of monitored devices.
+
+ a newly allocated #GstMessage
+
+
+
+
+ The #GstObject that created the message
+
+
+
+ The new #GstDevice
+
+
+
+
+
+ Creates a new device-removed message. The device-removed message is produced
+by #GstDeviceProvider or a #GstDeviceMonitor. They announce the
+disappearance of monitored devices.
+
+ a newly allocated #GstMessage
+
+
+
+
+ The #GstObject that created the message
+
+
+
+ The removed #GstDevice
+
+
+
+
+
+ Create a new duration changed message. This message is posted by elements
+that know the duration of a stream when the duration changes. This message
+is received by bins and is used to calculate the total duration of a
+pipeline.
+
+ The new duration-changed message.
+
+MT safe.
+
+
+
+
+ The object originating the message.
+
+
+
+
+
+ Create a new element-specific message. This is meant as a generic way of
+allowing one-way communication from an element to an application, for example
+"the firewire cable was unplugged". The format of the message should be
+documented in the element's documentation. The structure field can be %NULL.
+
+ The new element message.
+
+MT safe.
+
+
+
+
+ The object originating the message.
+
+
+
+ The structure for the
+ message. The message will take ownership of the structure.
+
+
+
+
+
+ Create a new eos message. This message is generated and posted in
+the sink elements of a GstBin. The bin will only forward the EOS
+message to the application if all sinks have posted an EOS message.
+
+ The new eos message.
+
+MT safe.
+
+
+
+
+ The object originating the message.
+
+
+
+
+
+ Create a new error message. The message will copy @error and
+@debug. This message is posted by element when a fatal event
+occurred. The pipeline will probably (partially) stop. The application
+receiving this message should stop the pipeline.
+
+ the new error message.
+
+MT safe.
+
+
+
+
+ The object originating the message.
+
+
+
+ The GError for this message.
+
+
+
+ A debugging string.
+
+
+
+
+
+ Create a new error message. The message will copy @error and
+@debug. This message is posted by element when a fatal event
+occurred. The pipeline will probably (partially) stop. The application
+receiving this message should stop the pipeline.
+
+ the new error message.
+
+
+
+
+ The object originating the message.
+
+
+
+ The GError for this message.
+
+
+
+ A debugging string.
+
+
+
+ (allow-none): A GstStructure with details
+
+
+
+
+
+ This message is posted when an element has a new local #GstContext.
+
+ The new have-context message.
+
+MT safe.
+
+
+
+
+ The object originating the message.
+
+
+
+ the context
+
+
+
+
+
+ Create a new info message. The message will make copies of @error and
+@debug.
+
+ the new info message.
+
+MT safe.
+
+
+
+
+ The object originating the message.
+
+
+
+ The GError for this message.
+
+
+
+ A debugging string.
+
+
+
+
+
+ Create a new info message. The message will make copies of @error and
+@debug.
+
+ the new warning message.
+
+
+
+
+ The object originating the message.
+
+
+
+ The GError for this message.
+
+
+
+ A debugging string.
+
+
+
+ (allow-none): A GstStructure with details
+
+
+
+
+
+ This message can be posted by elements when their latency requirements have
+changed.
+
+ The new latency message.
+
+MT safe.
+
+
+
+
+ The object originating the message.
+
+
+
+
+
+ This message is posted when an element needs a specific #GstContext.
+
+ The new need-context message.
+
+MT safe.
+
+
+
+
+ The object originating the message.
+
+
+
+ The context type that is needed
+
+
+
+
+
+ Create a new clock message. This message is posted whenever the
+pipeline selects a new clock for the pipeline.
+
+ The new new clock message.
+
+MT safe.
+
+
+
+
+ The object originating the message.
+
+
+
+ the new selected clock
+
+
+
+
+
+ Progress messages are posted by elements when they use an asynchronous task
+to perform actions triggered by a state change.
+
+@code contains a well defined string describing the action.
+@text should contain a user visible string detailing the current action.
+
+ The new qos message.
+
+
+
+
+ The object originating the message.
+
+
+
+ a #GstProgressType
+
+
+
+ a progress code
+
+
+
+ free, user visible text describing the progress
+
+
+
+
+
+
+ a newly allocated #GstMessage
+
+
+
+
+ The #GstObject whose property changed (may or may not be a #GstElement)
+
+
+
+ name of the property that changed
+
+
+
+ new property value, or %NULL
+
+
+
+
+
+ A QOS message is posted on the bus whenever an element decides to drop a
+buffer because of QoS reasons or whenever it changes its processing strategy
+because of QoS reasons (quality adjustments such as processing at lower
+accuracy).
+
+This message can be posted by an element that performs synchronisation against the
+clock (live) or it could be dropped by an element that performs QoS because of QOS
+events received from a downstream element (!live).
+
+@running_time, @stream_time, @timestamp, @duration should be set to the
+respective running-time, stream-time, timestamp and duration of the (dropped)
+buffer that generated the QoS event. Values can be left to
+GST_CLOCK_TIME_NONE when unknown.
+
+ The new qos message.
+
+MT safe.
+
+
+
+
+ The object originating the message.
+
+
+
+ if the message was generated by a live element
+
+
+
+ the running time of the buffer that generated the message
+
+
+
+ the stream time of the buffer that generated the message
+
+
+
+ the timestamps of the buffer that generated the message
+
+
+
+ the duration of the buffer that generated the message
+
+
+
+
+
+ Creates a new redirect message and adds a new entry to it. Redirect messages
+are posted when an element detects that the actual data has to be retrieved
+from a different location. This is useful if such a redirection cannot be
+handled inside a source element, for example when HTTP 302/303 redirects
+return a non-HTTP URL.
+
+The redirect message can hold multiple entries. The first one is added
+when the redirect message is created, with the given location, tag_list,
+entry_struct arguments. Use gst_message_add_redirect_entry() to add more
+entries.
+
+Each entry has a location, a tag list, and a structure. All of these are
+optional. The tag list and structure are useful for additional metadata,
+such as bitrate statistics for the given location.
+
+By default, message recipients should treat entries in the order they are
+stored. The recipient should therefore try entry #0 first, and if this
+entry is not acceptable or working, try entry #1 etc. Senders must make
+sure that they add entries in this order. However, recipients are free to
+ignore the order and pick an entry that is "best" for them. One example
+would be a recipient that scans the entries for the one with the highest
+bitrate tag.
+
+The specified location string is copied. However, ownership over the tag
+list and structure are transferred to the message.
+
+ a newly allocated #GstMessage
+
+
+
+
+ The #GstObject whose property changed (may or may not be a #GstElement)
+
+
+
+ location string for the new entry
+
+
+
+ tag list for the new entry
+
+
+
+ structure for the new entry
+
+
+
+
+
+ This message can be posted by elements when they want to have their state
+changed. A typical use case would be an audio server that wants to pause the
+pipeline because a higher priority stream is being played.
+
+ the new request state message.
+
+MT safe.
+
+
+
+
+ The object originating the message.
+
+
+
+ The new requested state
+
+
+
+
+
+ This message is posted when the pipeline running-time should be reset to
+@running_time, like after a flushing seek.
+
+ The new reset_time message.
+
+MT safe.
+
+
+
+
+ The object originating the message.
+
+
+
+ the requested running-time
+
+
+
+
+
+ Create a new segment done message. This message is posted by elements that
+finish playback of a segment as a result of a segment seek. This message
+is received by the application after all elements that posted a segment_start
+have posted the segment_done.
+
+ the new segment done message.
+
+MT safe.
+
+
+
+
+ The object originating the message.
+
+
+
+ The format of the position being done
+
+
+
+ The position of the segment being done
+
+
+
+
+
+ Create a new segment message. This message is posted by elements that
+start playback of a segment as a result of a segment seek. This message
+is not received by the application but is used for maintenance reasons in
+container elements.
+
+ the new segment start message.
+
+MT safe.
+
+
+
+
+ The object originating the message.
+
+
+
+ The format of the position being played
+
+
+
+ The position of the segment being played
+
+
+
+
+
+ Create a state change message. This message is posted whenever an element
+changed its state.
+
+ the new state change message.
+
+MT safe.
+
+
+
+
+ The object originating the message.
+
+
+
+ the previous state
+
+
+
+ the new (current) state
+
+
+
+ the pending (target) state
+
+
+
+
+
+ Create a state dirty message. This message is posted whenever an element
+changed its state asynchronously and is used internally to update the
+states of container objects.
+
+ the new state dirty message.
+
+MT safe.
+
+
+
+
+ The object originating the message
+
+
+
+
+
+ This message is posted by elements when they complete a part, when @intermediate set
+to %TRUE, or a complete step operation.
+
+@duration will contain the amount of time (in GST_FORMAT_TIME) of the stepped
+@amount of media in format @format.
+
+ the new step_done message.
+
+MT safe.
+
+
+
+
+ The object originating the message.
+
+
+
+ the format of @amount
+
+
+
+ the amount of stepped data
+
+
+
+ the rate of the stepped amount
+
+
+
+ is this an flushing step
+
+
+
+ is this an intermediate step
+
+
+
+ the duration of the data
+
+
+
+ the step caused EOS
+
+
+
+
+
+ This message is posted by elements when they accept or activate a new step
+event for @amount in @format.
+
+@active is set to %FALSE when the element accepted the new step event and has
+queued it for execution in the streaming threads.
+
+@active is set to %TRUE when the element has activated the step operation and
+is now ready to start executing the step in the streaming thread. After this
+message is emitted, the application can queue a new step operation in the
+element.
+
+ The new step_start message.
+
+MT safe.
+
+
+
+
+ The object originating the message.
+
+
+
+ if the step is active or queued
+
+
+
+ the format of @amount
+
+
+
+ the amount of stepped data
+
+
+
+ the rate of the stepped amount
+
+
+
+ is this an flushing step
+
+
+
+ is this an intermediate step
+
+
+
+
+
+ Creates a new stream-collection message. The message is used to announce new
+#GstStreamCollection
+
+ a newly allocated #GstMessage
+
+
+
+
+ The #GstObject that created the message
+
+
+
+ The #GstStreamCollection
+
+
+
+
+
+ Create a new stream_start message. This message is generated and posted in
+the sink elements of a GstBin. The bin will only forward the STREAM_START
+message to the application if all sinks have posted an STREAM_START message.
+
+ The new stream_start message.
+
+MT safe.
+
+
+
+
+ The object originating the message.
+
+
+
+
+
+ Create a new stream status message. This message is posted when a streaming
+thread is created/destroyed or when the state changed.
+
+ the new stream status message.
+
+MT safe.
+
+
+
+
+ The object originating the message.
+
+
+
+ The stream status type.
+
+
+
+ the owner element of @src.
+
+
+
+
+
+ Creates a new steams-selected message. The message is used to announce
+that an array of streams has been selected. This is generally in response
+to a #GST_EVENT_SELECT_STREAMS event, or when an element (such as decodebin3)
+makes an initial selection of streams.
+
+The message also contains the #GstStreamCollection to which the various streams
+belong to.
+
+Users of gst_message_new_streams_selected() can add the selected streams with
+gst_message_streams_selected_add().
+
+ a newly allocated #GstMessage
+
+
+
+
+ The #GstObject that created the message
+
+
+
+ The #GstStreamCollection
+
+
+
+
+
+ Create a new structure change message. This message is posted when the
+structure of a pipeline is in the process of being changed, for example
+when pads are linked or unlinked.
+
+@src should be the sinkpad that unlinked or linked.
+
+ the new structure change message.
+
+MT safe.
+
+
+
+
+ The object originating the message.
+
+
+
+ The change type.
+
+
+
+ The owner element of @src.
+
+
+
+ Whether the structure change is busy.
+
+
+
+
+
+ Create a new tag message. The message will take ownership of the tag list.
+The message is posted by elements that discovered a new taglist.
+
+ the new tag message.
+
+MT safe.
+
+
+
+
+ The object originating the message.
+
+
+
+ the tag list for the message.
+
+
+
+
+
+ Create a new TOC message. The message is posted by elements
+that discovered or updated a TOC.
+
+ a new TOC message.
+
+MT safe.
+
+
+
+
+ the object originating the message.
+
+
+
+ #GstToc structure for the message.
+
+
+
+ whether TOC was updated or not.
+
+
+
+
+
+ Create a new warning message. The message will make copies of @error and
+@debug.
+
+ the new warning message.
+
+MT safe.
+
+
+
+
+ The object originating the message.
+
+
+
+ The GError for this message.
+
+
+
+ A debugging string.
+
+
+
+
+
+ Create a new warning message. The message will make copies of @error and
+@debug.
+
+ the new warning message.
+
+
+
+
+ The object originating the message.
+
+
+
+ The GError for this message.
+
+
+
+ A debugging string.
+
+
+
+ (allow-none): A GstStructure with details
+
+
+
+
+
+ Creates and appends a new entry.
+
+The specified location string is copied. However, ownership over the tag
+list and structure are transferred to the message.
+
+
+
+
+
+ a #GstMessage of type %GST_MESSAGE_REDIRECT
+
+
+
+ location string for the new entry
+
+
+
+ tag list for the new entry
+
+
+
+ structure for the new entry
+
+
+
+
+
+
+ the number of entries stored in the message
+
+
+
+
+ a #GstMessage of type %GST_MESSAGE_REDIRECT
+
+
+
+
+
+ Retrieve the sequence number of a message.
+
+Messages have ever-incrementing sequence numbers, which may also be set
+explicitly via gst_message_set_seqnum(). Sequence numbers are typically used
+to indicate that a message corresponds to some other set of messages or
+events, for example a SEGMENT_DONE message corresponding to a SEEK event. It
+is considered good practice to make this correspondence when possible, though
+it is not required.
+
+Note that events and messages share the same sequence number incrementor;
+two events or messages will never have the same sequence number unless
+that correspondence was made explicitly.
+
+ The message's sequence number.
+
+MT safe.
+
+
+
+
+ A #GstMessage.
+
+
+
+
+
+ Extracts the object managing the streaming thread from @message.
+
+ a GValue containing the object that manages the
+streaming thread. This object is usually of type GstTask but other types can
+be added in the future. The object remains valid as long as @message is
+valid.
+
+
+
+
+ A valid #GstMessage of type GST_MESSAGE_STREAM_STATUS.
+
+
+
+
+
+ Access the structure of the message.
+
+ The structure of the message. The
+structure is still owned by the message, which means that you should not
+free it and that the pointer becomes invalid when you free the message.
+
+MT safe.
+
+
+
+
+ The #GstMessage.
+
+
+
+
+
+ Checks if @message has the given @name. This function is usually used to
+check the name of a custom message.
+
+ %TRUE if @name matches the name of the message structure.
+
+
+
+
+ The #GstMessage.
+
+
+
+ name to check
+
+
+
+
+
+ Extract the running_time from the async_done message.
+
+MT safe.
+
+
+
+
+
+ A valid #GstMessage of type GST_MESSAGE_ASYNC_DONE.
+
+
+
+ Result location for the running_time or %NULL
+
+
+
+
+
+ Extracts the buffering percent from the GstMessage. see also
+gst_message_new_buffering().
+
+MT safe.
+
+
+
+
+
+ A valid #GstMessage of type GST_MESSAGE_BUFFERING.
+
+
+
+ Return location for the percent.
+
+
+
+
+
+ Extracts the buffering stats values from @message.
+
+
+
+
+
+ A valid #GstMessage of type GST_MESSAGE_BUFFERING.
+
+
+
+ a buffering mode, or %NULL
+
+
+
+ the average input rate, or %NULL
+
+
+
+ the average output rate, or %NULL
+
+
+
+ amount of buffering time left in
+ milliseconds, or %NULL
+
+
+
+
+
+ Extracts the lost clock from the GstMessage.
+The clock object returned remains valid until the message is freed.
+
+MT safe.
+
+
+
+
+
+ A valid #GstMessage of type GST_MESSAGE_CLOCK_LOST.
+
+
+
+ a pointer to hold the lost clock
+
+
+
+
+
+ Extracts the clock and ready flag from the GstMessage.
+The clock object returned remains valid until the message is freed.
+
+MT safe.
+
+
+
+
+
+ A valid #GstMessage of type GST_MESSAGE_CLOCK_PROVIDE.
+
+
+
+ a pointer to hold a clock
+ object, or %NULL
+
+
+
+ a pointer to hold the ready flag, or %NULL
+
+
+
+
+
+ Parse a context type from an existing GST_MESSAGE_NEED_CONTEXT message.
+
+ a #gboolean indicating if the parsing succeeded.
+
+
+
+
+ a GST_MESSAGE_NEED_CONTEXT type message
+
+
+
+ the context type, or %NULL
+
+
+
+
+
+ Parses a device-added message. The device-added message is produced by
+#GstDeviceProvider or a #GstDeviceMonitor. It announces the appearance
+of monitored devices.
+
+
+
+
+
+ a #GstMessage of type %GST_MESSAGE_DEVICE_ADDED
+
+
+
+ A location where to store a
+ pointer to the new #GstDevice, or %NULL
+
+
+
+
+
+ Parses a device-removed message. The device-removed message is produced by
+#GstDeviceProvider or a #GstDeviceMonitor. It announces the
+disappearance of monitored devices.
+
+
+
+
+
+ a #GstMessage of type %GST_MESSAGE_DEVICE_REMOVED
+
+
+
+ A location where to store a
+ pointer to the removed #GstDevice, or %NULL
+
+
+
+
+
+ Extracts the GError and debug string from the GstMessage. The values returned
+in the output arguments are copies; the caller must free them when done.
+
+Typical usage of this function might be:
+|[<!-- language="C" -->
+ ...
+ switch (GST_MESSAGE_TYPE (msg)) {
+ case GST_MESSAGE_ERROR: {
+ GError *err = NULL;
+ gchar *dbg_info = NULL;
+
+ gst_message_parse_error (msg, &err, &dbg_info);
+ g_printerr ("ERROR from element %s: %s\n",
+ GST_OBJECT_NAME (msg->src), err->message);
+ g_printerr ("Debugging info: %s\n", (dbg_info) ? dbg_info : "none");
+ g_error_free (err);
+ g_free (dbg_info);
+ break;
+ }
+ ...
+ }
+ ...
+]|
+
+MT safe.
+
+
+
+
+
+ A valid #GstMessage of type GST_MESSAGE_ERROR.
+
+
+
+ location for the GError
+
+
+
+ location for the debug message,
+ or %NULL
+
+
+
+
+
+ Returns the optional details structure, may be NULL if none.
+The returned structure must not be freed.
+
+
+
+
+
+ The message object
+
+
+
+ A pointer to the returned details
+
+
+
+
+
+ Extract the group from the STREAM_START message.
+
+ %TRUE if the message had a group id set, %FALSE otherwise
+
+MT safe.
+
+
+
+
+ A valid #GstMessage of type GST_MESSAGE_STREAM_START.
+
+
+
+ Result location for the group id or
+ %NULL
+
+
+
+
+
+ Extract the context from the HAVE_CONTEXT message.
+
+MT safe.
+
+
+
+
+
+ A valid #GstMessage of type GST_MESSAGE_HAVE_CONTEXT.
+
+
+
+ Result location for the
+ context or %NULL
+
+
+
+
+
+ Extracts the GError and debug string from the GstMessage. The values returned
+in the output arguments are copies; the caller must free them when done.
+
+MT safe.
+
+
+
+
+
+ A valid #GstMessage of type GST_MESSAGE_INFO.
+
+
+
+ location for the GError
+
+
+
+ location for the debug message,
+ or %NULL
+
+
+
+
+
+ Returns the optional details structure, may be NULL if none
+The returned structure must not be freed.
+
+
+
+
+
+ The message object
+
+
+
+ A pointer to the returned details structure
+
+
+
+
+
+ Extracts the new clock from the GstMessage.
+The clock object returned remains valid until the message is freed.
+
+MT safe.
+
+
+
+
+
+ A valid #GstMessage of type GST_MESSAGE_NEW_CLOCK.
+
+
+
+ a pointer to hold the selected
+ new clock
+
+
+
+
+
+ Parses the progress @type, @code and @text.
+
+
+
+
+
+ A valid #GstMessage of type GST_MESSAGE_PROGRESS.
+
+
+
+ location for the type
+
+
+
+ location for the code
+
+
+
+ location for the text
+
+
+
+
+
+ Parses a property-notify message. These will be posted on the bus only
+when set up with gst_element_add_property_notify_watch() or
+gst_element_add_property_deep_notify_watch().
+
+
+
+
+
+ a #GstMessage of type %GST_MESSAGE_PROPERTY_NOTIFY
+
+
+
+ location where to store a
+ pointer to the object whose property got changed, or %NULL
+
+
+
+ return location for
+ the name of the property that got changed, or %NULL
+
+
+
+ return location for
+ the new value of the property that got changed, or %NULL. This will
+ only be set if the property notify watch was told to include the value
+ when it was set up
+
+
+
+
+
+ Extract the timestamps and live status from the QoS message.
+
+The returned values give the running_time, stream_time, timestamp and
+duration of the dropped buffer. Values of GST_CLOCK_TIME_NONE mean unknown
+values.
+
+MT safe.
+
+
+
+
+
+ A valid #GstMessage of type GST_MESSAGE_QOS.
+
+
+
+ if the message was generated by a live element
+
+
+
+ the running time of the buffer that
+ generated the message
+
+
+
+ the stream time of the buffer that
+ generated the message
+
+
+
+ the timestamps of the buffer that
+ generated the message
+
+
+
+ the duration of the buffer that
+ generated the message
+
+
+
+
+
+ Extract the QoS stats representing the history of the current continuous
+pipeline playback period.
+
+When @format is @GST_FORMAT_UNDEFINED both @dropped and @processed are
+invalid. Values of -1 for either @processed or @dropped mean unknown values.
+
+MT safe.
+
+
+
+
+
+ A valid #GstMessage of type GST_MESSAGE_QOS.
+
+
+
+ Units of the 'processed' and 'dropped' fields.
+ Video sinks and video filters will use GST_FORMAT_BUFFERS (frames).
+ Audio sinks and audio filters will likely use GST_FORMAT_DEFAULT
+ (samples).
+
+
+
+ Total number of units correctly processed
+ since the last state change to READY or a flushing operation.
+
+
+
+ Total number of units dropped since the last
+ state change to READY or a flushing operation.
+
+
+
+
+
+ Extract the QoS values that have been calculated/analysed from the QoS data
+
+MT safe.
+
+
+
+
+
+ A valid #GstMessage of type GST_MESSAGE_QOS.
+
+
+
+ The difference of the running-time against
+ the deadline.
+
+
+
+ Long term prediction of the ideal rate
+ relative to normal rate to get optimal quality.
+
+
+
+ An element dependent integer value that
+ specifies the current quality level of the element. The default
+ maximum quality is 1000000.
+
+
+
+
+
+ Parses the location and/or structure from the entry with the given index.
+The index must be between 0 and gst_message_get_num_redirect_entries() - 1.
+Returned pointers are valid for as long as this message exists.
+
+
+
+
+
+ a #GstMessage of type %GST_MESSAGE_REDIRECT
+
+
+
+ index of the entry to parse
+
+
+
+ return location for
+ the pointer to the entry's location string, or %NULL
+
+
+
+ return location for
+ the pointer to the entry's tag list, or %NULL
+
+
+
+ return location
+ for the pointer to the entry's structure, or %NULL
+
+
+
+
+
+ Extract the requested state from the request_state message.
+
+MT safe.
+
+
+
+
+
+ A valid #GstMessage of type GST_MESSAGE_REQUEST_STATE.
+
+
+
+ Result location for the requested state or %NULL
+
+
+
+
+
+ Extract the running-time from the RESET_TIME message.
+
+MT safe.
+
+
+
+
+
+ A valid #GstMessage of type GST_MESSAGE_RESET_TIME.
+
+
+
+ Result location for the running_time or
+ %NULL
+
+
+
+
+
+ Extracts the position and format from the segment done message.
+
+MT safe.
+
+
+
+
+
+ A valid #GstMessage of type GST_MESSAGE_SEGMENT_DONE.
+
+
+
+ Result location for the format, or %NULL
+
+
+
+ Result location for the position, or %NULL
+
+
+
+
+
+ Extracts the position and format from the segment start message.
+
+MT safe.
+
+
+
+
+
+ A valid #GstMessage of type GST_MESSAGE_SEGMENT_START.
+
+
+
+ Result location for the format, or %NULL
+
+
+
+ Result location for the position, or %NULL
+
+
+
+
+
+ Extracts the old and new states from the GstMessage.
+
+Typical usage of this function might be:
+|[<!-- language="C" -->
+ ...
+ switch (GST_MESSAGE_TYPE (msg)) {
+ case GST_MESSAGE_STATE_CHANGED: {
+ GstState old_state, new_state;
+
+ gst_message_parse_state_changed (msg, &old_state, &new_state, NULL);
+ g_print ("Element %s changed state from %s to %s.\n",
+ GST_OBJECT_NAME (msg->src),
+ gst_element_state_get_name (old_state),
+ gst_element_state_get_name (new_state));
+ break;
+ }
+ ...
+ }
+ ...
+]|
+
+MT safe.
+
+
+
+
+
+ a valid #GstMessage of type GST_MESSAGE_STATE_CHANGED
+
+
+
+ the previous state, or %NULL
+
+
+
+ the new (current) state, or %NULL
+
+
+
+ the pending (target) state, or %NULL
+
+
+
+
+
+ Extract the values the step_done message.
+
+MT safe.
+
+
+
+
+
+ A valid #GstMessage of type GST_MESSAGE_STEP_DONE.
+
+
+
+ result location for the format
+
+
+
+ result location for the amount
+
+
+
+ result location for the rate
+
+
+
+ result location for the flush flag
+
+
+
+ result location for the intermediate flag
+
+
+
+ result location for the duration
+
+
+
+ result location for the EOS flag
+
+
+
+
+
+ Extract the values from step_start message.
+
+MT safe.
+
+
+
+
+
+ A valid #GstMessage of type GST_MESSAGE_STEP_DONE.
+
+
+
+ result location for the active flag
+
+
+
+ result location for the format
+
+
+
+ result location for the amount
+
+
+
+ result location for the rate
+
+
+
+ result location for the flush flag
+
+
+
+ result location for the intermediate flag
+
+
+
+
+
+ Parses a stream-collection message.
+
+
+
+
+
+ a #GstMessage of type %GST_MESSAGE_STREAM_COLLECTION
+
+
+
+ A location where to store a
+ pointer to the #GstStreamCollection, or %NULL
+
+
+
+
+
+ Extracts the stream status type and owner the GstMessage. The returned
+owner remains valid for as long as the reference to @message is valid and
+should thus not be unreffed.
+
+MT safe.
+
+
+
+
+
+ A valid #GstMessage of type GST_MESSAGE_STREAM_STATUS.
+
+
+
+ A pointer to hold the status type
+
+
+
+ The owner element of the message source
+
+
+
+
+
+ Parses a streams-selected message.
+
+
+
+
+
+ a #GstMessage of type %GST_MESSAGE_STREAMS_SELECTED
+
+
+
+ A location where to store a
+ pointer to the #GstStreamCollection, or %NULL
+
+
+
+
+
+ Extracts the change type and completion status from the GstMessage.
+
+MT safe.
+
+
+
+
+
+ A valid #GstMessage of type GST_MESSAGE_STRUCTURE_CHANGE.
+
+
+
+ A pointer to hold the change type
+
+
+
+ The owner element of the
+ message source
+
+
+
+ a pointer to hold whether the change is in
+ progress or has been completed
+
+
+
+
+
+ Extracts the tag list from the GstMessage. The tag list returned in the
+output argument is a copy; the caller must free it when done.
+
+Typical usage of this function might be:
+|[<!-- language="C" -->
+ ...
+ switch (GST_MESSAGE_TYPE (msg)) {
+ case GST_MESSAGE_TAG: {
+ GstTagList *tags = NULL;
+
+ gst_message_parse_tag (msg, &tags);
+ g_print ("Got tags from element %s\n", GST_OBJECT_NAME (msg->src));
+ handle_tags (tags);
+ gst_tag_list_unref (tags);
+ break;
+ }
+ ...
+ }
+ ...
+]|
+
+MT safe.
+
+
+
+
+
+ A valid #GstMessage of type GST_MESSAGE_TAG.
+
+
+
+ return location for the tag-list.
+
+
+
+
+
+ Extract the TOC from the #GstMessage. The TOC returned in the
+output argument is a copy; the caller must free it with
+gst_toc_unref() when done.
+
+MT safe.
+
+
+
+
+
+ a valid #GstMessage of type GST_MESSAGE_TOC.
+
+
+
+ return location for the TOC.
+
+
+
+ return location for the updated flag.
+
+
+
+
+
+ Extracts the GError and debug string from the GstMessage. The values returned
+in the output arguments are copies; the caller must free them when done.
+
+MT safe.
+
+
+
+
+
+ A valid #GstMessage of type GST_MESSAGE_WARNING.
+
+
+
+ location for the GError
+
+
+
+ location for the debug message,
+ or %NULL
+
+
+
+
+
+ Returns the optional details structure, may be NULL if none
+The returned structure must not be freed.
+
+
+
+
+
+ The message object
+
+
+
+ A pointer to the returned details structure
+
+
+
+
+
+ Configures the buffering stats values in @message.
+
+
+
+
+
+ A valid #GstMessage of type GST_MESSAGE_BUFFERING.
+
+
+
+ a buffering mode
+
+
+
+ the average input rate
+
+
+
+ the average output rate
+
+
+
+ amount of buffering time left in milliseconds
+
+
+
+
+
+ Sets the group id on the stream-start message.
+
+All streams that have the same group id are supposed to be played
+together, i.e. all streams inside a container file should have the
+same group id but different stream ids. The group id should change
+each time the stream is started, resulting in different group ids
+each time a file is played for example.
+
+MT safe.
+
+
+
+
+
+ the message
+
+
+
+ the group id
+
+
+
+
+
+ Set the QoS stats representing the history of the current continuous pipeline
+playback period.
+
+When @format is @GST_FORMAT_UNDEFINED both @dropped and @processed are
+invalid. Values of -1 for either @processed or @dropped mean unknown values.
+
+MT safe.
+
+
+
+
+
+ A valid #GstMessage of type GST_MESSAGE_QOS.
+
+
+
+ Units of the 'processed' and 'dropped' fields. Video sinks and video
+filters will use GST_FORMAT_BUFFERS (frames). Audio sinks and audio filters
+will likely use GST_FORMAT_DEFAULT (samples).
+
+
+
+ Total number of units correctly processed since the last state
+change to READY or a flushing operation.
+
+
+
+ Total number of units dropped since the last state change to READY
+or a flushing operation.
+
+
+
+
+
+ Set the QoS values that have been calculated/analysed from the QoS data
+
+MT safe.
+
+
+
+
+
+ A valid #GstMessage of type GST_MESSAGE_QOS.
+
+
+
+ The difference of the running-time against the deadline.
+
+
+
+ Long term prediction of the ideal rate relative to normal rate
+to get optimal quality.
+
+
+
+ An element dependent integer value that specifies the current
+quality level of the element. The default maximum quality is 1000000.
+
+
+
+
+
+ Set the sequence number of a message.
+
+This function might be called by the creator of a message to indicate that
+the message relates to other messages or events. See gst_message_get_seqnum()
+for more information.
+
+MT safe.
+
+
+
+
+
+ A #GstMessage.
+
+
+
+ A sequence number.
+
+
+
+
+
+ Configures the object handling the streaming thread. This is usually a
+GstTask object but other objects might be added in the future.
+
+
+
+
+
+ A valid #GstMessage of type GST_MESSAGE_STREAM_STATUS.
+
+
+
+ the object controlling the streaming
+
+
+
+
+
+ Adds the @stream to the @message.
+
+
+
+
+
+ a #GstMessage of type %GST_MESSAGE_STREAMS_SELECTED
+
+
+
+ a #GstStream to add to @message
+
+
+
+
+
+ Returns the number of streams contained in the @message.
+
+ The number of streams contained within.
+
+
+
+
+ a #GstMessage of type %GST_MESSAGE_STREAMS_SELECTED
+
+
+
+
+
+ Retrieves the #GstStream with index @index from the @message.
+
+ A #GstStream
+
+
+
+
+ a #GstMessage of type %GST_MESSAGE_STREAMS_SELECTED
+
+
+
+ Index of the stream to retrieve
+
+
+
+
+
+ Get a writable version of the structure.
+
+ The structure of the message. The structure
+is still owned by the message, which means that you should not free
+it and that the pointer becomes invalid when you free the message.
+This function checks if @message is writable and will never return
+%NULL.
+
+MT safe.
+
+
+
+
+ The #GstMessage.
+
+
+
+
+
+
+ The different message types that are available.
+
+ an undefined message
+
+
+ end-of-stream reached in a pipeline. The application will
+only receive this message in the PLAYING state and every time it sets a
+pipeline to PLAYING that is in the EOS state. The application can perform a
+flushing seek in the pipeline, which will undo the EOS state again.
+
+
+ an error occurred. When the application receives an error
+message it should stop playback of the pipeline and not assume that more
+data will be played.
+
+
+ a warning occurred.
+
+
+ an info message occurred
+
+
+ a tag was found.
+
+
+ the pipeline is buffering. When the application
+receives a buffering message in the PLAYING state for a non-live pipeline it
+must PAUSE the pipeline until the buffering completes, when the percentage
+field in the message is 100%. For live pipelines, no action must be
+performed and the buffering percentage can be used to inform the user about
+the progress.
+
+
+ a state change happened
+
+
+ an element changed state in a streaming thread.
+This message is deprecated.
+
+
+ a stepping operation finished.
+
+
+ an element notifies its capability of providing
+ a clock. This message is used internally and
+ never forwarded to the application.
+
+
+ The current clock as selected by the pipeline became
+ unusable. The pipeline will select a new clock on
+ the next PLAYING state change. The application
+ should set the pipeline to PAUSED and back to
+ PLAYING when this message is received.
+
+
+ a new clock was selected in the pipeline.
+
+
+ the structure of the pipeline changed. This
+message is used internally and never forwarded to the application.
+
+
+ status about a stream, emitted when it starts,
+ stops, errors, etc..
+
+
+ message posted by the application, possibly
+ via an application-specific element.
+
+
+ element-specific message, see the specific element's
+ documentation
+
+
+ pipeline started playback of a segment. This
+message is used internally and never forwarded to the application.
+
+
+ pipeline completed playback of a segment. This
+message is forwarded to the application after all elements that posted
+@GST_MESSAGE_SEGMENT_START posted a GST_MESSAGE_SEGMENT_DONE message.
+
+
+ The duration of a pipeline changed. The
+application can get the new duration with a duration query.
+
+
+ Posted by elements when their latency changes. The
+application should recalculate and distribute a new latency.
+
+
+ Posted by elements when they start an ASYNC
+#GstStateChange. This message is not forwarded to the application but is used
+internally.
+
+
+ Posted by elements when they complete an ASYNC
+#GstStateChange. The application will only receive this message from the toplevel
+pipeline.
+
+
+ Posted by elements when they want the pipeline to
+change state. This message is a suggestion to the application which can
+decide to perform the state change on (part of) the pipeline.
+
+
+ A stepping operation was started.
+
+
+ A buffer was dropped or an element changed its processing
+strategy for Quality of Service reasons.
+
+
+ A progress message.
+
+
+ A new table of contents (TOC) was found or previously found TOC
+was updated.
+
+
+ Message to request resetting the pipeline's
+ running time from the pipeline. This is an internal message which
+ applications will likely never receive.
+
+
+ Message indicating start of a new stream. Useful
+ e.g. when using playbin in gapless playback mode, to get notified when
+ the next title actually starts playing (which will be some time after
+ the URI for the next title has been set).
+
+
+ Message indicating that an element wants a specific context (Since 1.2)
+
+
+ Message indicating that an element created a context (Since 1.2)
+
+
+ Message is an extended message type (see below).
+ These extended message IDs can't be used directly with mask-based API
+ like gst_bus_poll() or gst_bus_timed_pop_filtered(), but you can still
+ filter for GST_MESSAGE_EXTENDED and then check the result for the
+ specific type. (Since 1.4)
+
+
+ Message indicating a #GstDevice was added to
+ a #GstDeviceProvider (Since 1.4)
+
+
+ Message indicating a #GstDevice was removed
+ from a #GstDeviceProvider (Since 1.4)
+
+
+ Message indicating a #GObject property has
+ changed (Since 1.10)
+
+
+ Message indicating a new #GstStreamCollection
+ is available (Since 1.10)
+
+
+ Message indicating the active selection of
+ #GstStreams has changed (Since 1.10)
+
+
+ Message indicating to request the application to
+ try to play the given URL(s). Useful if for example a HTTP 302/303
+ response is received with a non-HTTP URL inside. (Since 1.10)
+
+
+ mask for all of the above messages.
+
+
+ Get a printable name for the given message type. Do not modify or free.
+
+ a reference to the static name of the message.
+
+
+
+
+ the message type
+
+
+
+
+
+ Get the unique quark for the given message type.
+
+ the quark associated with the message type
+
+
+
+
+ the message type
+
+
+
+
+
+
+ The #GstMeta structure should be included as the first member of a #GstBuffer
+metadata structure. The structure defines the API of the metadata and should
+be accessible to all elements using the metadata.
+
+A metadata API is registered with gst_meta_api_type_register() which takes a
+name for the metadata API and some tags associated with the metadata.
+With gst_meta_api_type_has_tag() one can check if a certain metadata API
+contains a given tag.
+
+Multiple implementations of a metadata API can be registered.
+To implement a metadata API, gst_meta_register() should be used. This
+function takes all parameters needed to create, free and transform metadata
+along with the size of the metadata. The function returns a #GstMetaInfo
+structure that contains the information for the implementation of the API.
+
+A specific implementation can be retrieved by name with gst_meta_get_info().
+
+See #GstBuffer for how the metadata can be added, retrieved and removed from
+buffers.
+
+ extra flags for the metadata
+
+
+
+ pointer to the #GstMetaInfo
+
+
+
+
+ an array of tags as strings.
+
+
+
+
+
+
+ an API
+
+
+
+
+
+ Check if @api was registered with @tag.
+
+ %TRUE if @api was registered with @tag.
+
+
+
+
+ an API
+
+
+
+ the tag to check
+
+
+
+
+
+ Register and return a GType for the @api and associate it with
+@tags.
+
+ a unique GType for @api.
+
+
+
+
+ an API to register
+
+
+
+ tags for @api
+
+
+
+
+
+ Lookup a previously registered meta info structure by its implementation name
+@impl.
+
+ a #GstMetaInfo with @impl, or
+%NULL when no such metainfo exists.
+
+
+
+
+ the name
+
+
+
+
+
+ Register a new #GstMeta implementation.
+
+The same @info can be retrieved later with gst_meta_get_info() by using
+@impl as the key.
+
+ a #GstMetaInfo that can be used to
+access metadata.
+
+
+
+
+ the type of the #GstMeta API
+
+
+
+ the name of the #GstMeta implementation
+
+
+
+ the size of the #GstMeta structure
+
+
+
+ a #GstMetaInitFunction
+
+
+
+ a #GstMetaFreeFunction
+
+
+
+ a #GstMetaTransformFunction
+
+
+
+
+
+
+ Extra metadata flags.
+
+ no flags
+
+
+ metadata should not be modified
+
+
+ metadata is managed by a bufferpool
+
+
+ metadata should not be removed
+
+
+ additional flags can be added starting from this flag.
+
+
+
+ Function called when @meta is freed in @buffer.
+
+
+
+
+
+ a #GstMeta
+
+
+
+ a #GstBuffer
+
+
+
+
+
+ The #GstMetaInfo provides information about a specific metadata
+structure.
+
+ tag identifying the metadata structure and api
+
+
+
+ type identifying the implementor of the api
+
+
+
+ size of the metadata
+
+
+
+ function for initializing the metadata
+
+
+
+ function for freeing the metadata
+
+
+
+ function for transforming the metadata
+
+
+
+
+ Function called when @meta is initialized in @buffer.
+
+
+
+
+
+ a #GstMeta
+
+
+
+ parameters passed to the init function
+
+
+
+ a #GstBuffer
+
+
+
+
+
+ Extra data passed to a "gst-copy" transform #GstMetaTransformFunction.
+
+ %TRUE if only region is copied
+
+
+
+ the offset to copy, 0 if @region is %FALSE, otherwise > 0
+
+
+
+ the size to copy, -1 or the buffer size when @region is %FALSE
+
+
+
+
+ Function called for each @meta in @buffer as a result of performing a
+transformation on @transbuf. Additional @type specific transform data
+is passed to the function as @data.
+
+Implementations should check the @type of the transform and parse
+additional type specific fields in @data that should be used to update
+the metadata on @transbuf.
+
+ %TRUE if the transform could be performed
+
+
+
+
+ a #GstBuffer
+
+
+
+ a #GstMeta
+
+
+
+ a #GstBuffer
+
+
+
+ the transform type
+
+
+
+ transform specific data.
+
+
+
+
+
+ #GstMiniObject is a simple structure that can be used to implement refcounted
+types.
+
+Subclasses will include #GstMiniObject as the first member in their structure
+and then call gst_mini_object_init() to initialize the #GstMiniObject fields.
+
+gst_mini_object_ref() and gst_mini_object_unref() increment and decrement the
+refcount respectively. When the refcount of a mini-object reaches 0, the
+dispose function is called first and when this returns %TRUE, the free
+function of the miniobject is called.
+
+A copy can be made with gst_mini_object_copy().
+
+gst_mini_object_is_writable() will return %TRUE when the refcount of the
+object is exactly 1, meaning the current caller has the only reference to the
+object. gst_mini_object_make_writable() will return a writable version of the
+object, which might be a new copy when the refcount was not 1.
+
+Opaque data can be associated with a #GstMiniObject with
+gst_mini_object_set_qdata() and gst_mini_object_get_qdata(). The data is
+meant to be specific to the particular object and is not automatically copied
+with gst_mini_object_copy() or similar methods.
+
+A weak reference can be added and remove with gst_mini_object_weak_ref()
+and gst_mini_object_weak_unref() respectively.
+
+ the GType of the object
+
+
+
+ atomic refcount
+
+
+
+ atomic state of the locks
+
+
+
+ extra flags.
+
+
+
+ a copy function
+
+
+
+ a dispose function
+
+
+
+ the free function
+
+
+
+
+
+
+
+
+
+ Creates a copy of the mini-object.
+
+MT safe
+
+ the new mini-object if copying is
+possible, %NULL otherwise.
+
+
+
+
+ the mini-object to copy
+
+
+
+
+
+ This function gets back user data pointers stored via
+gst_mini_object_set_qdata().
+
+ The user data pointer set, or
+%NULL
+
+
+
+
+ The GstMiniObject to get a stored user data pointer from
+
+
+
+ A #GQuark, naming the user data pointer
+
+
+
+
+
+ Initializes a mini-object with the desired type and copy/dispose/free
+functions.
+
+
+
+
+
+ a #GstMiniObject
+
+
+
+ initial #GstMiniObjectFlags
+
+
+
+ the #GType of the mini-object to create
+
+
+
+ the copy function, or %NULL
+
+
+
+ the dispose function, or %NULL
+
+
+
+ the free function or %NULL
+
+
+
+
+
+ If @mini_object has the LOCKABLE flag set, check if the current EXCLUSIVE
+lock on @object is the only one, this means that changes to the object will
+not be visible to any other object.
+
+If the LOCKABLE flag is not set, check if the refcount of @mini_object is
+exactly 1, meaning that no other reference exists to the object and that the
+object is therefore writable.
+
+Modification of a mini-object should only be done after verifying that it
+is writable.
+
+ %TRUE if the object is writable.
+
+
+
+
+ the mini-object to check
+
+
+
+
+
+ Lock the mini-object with the specified access mode in @flags.
+
+ %TRUE if @object could be locked.
+
+
+
+
+ the mini-object to lock
+
+
+
+ #GstLockFlags
+
+
+
+
+
+ Checks if a mini-object is writable. If not, a writable copy is made and
+returned. This gives away the reference to the original mini object,
+and returns a reference to the new object.
+
+MT safe
+
+ a mini-object (possibly the same pointer) that
+ is writable.
+
+
+
+
+ the mini-object to make writable
+
+
+
+
+
+ Increase the reference count of the mini-object.
+
+Note that the refcount affects the writability
+of @mini-object, see gst_mini_object_is_writable(). It is
+important to note that keeping additional references to
+GstMiniObject instances can potentially increase the number
+of memcpy operations in a pipeline, especially if the miniobject
+is a #GstBuffer.
+
+ the mini-object.
+
+
+
+
+ the mini-object
+
+
+
+
+
+ This sets an opaque, named pointer on a miniobject.
+The name is specified through a #GQuark (retrieved e.g. via
+g_quark_from_static_string()), and the pointer
+can be gotten back from the @object with gst_mini_object_get_qdata()
+until the @object is disposed.
+Setting a previously set user data pointer, overrides (frees)
+the old pointer set, using %NULL as pointer essentially
+removes the data stored.
+
+@destroy may be specified which is called with @data as argument
+when the @object is disposed, or the data is being overwritten by
+a call to gst_mini_object_set_qdata() with the same @quark.
+
+
+
+
+
+ a #GstMiniObject
+
+
+
+ A #GQuark, naming the user data pointer
+
+
+
+ An opaque user data pointer
+
+
+
+ Function to invoke with @data as argument, when @data
+ needs to be freed
+
+
+
+
+
+ This function gets back user data pointers stored via gst_mini_object_set_qdata()
+and removes the data from @object without invoking its destroy() function (if
+any was set).
+
+ The user data pointer set, or
+%NULL
+
+
+
+
+ The GstMiniObject to get a stored user data pointer from
+
+
+
+ A #GQuark, naming the user data pointer
+
+
+
+
+
+ Unlock the mini-object with the specified access mode in @flags.
+
+
+
+
+
+ the mini-object to unlock
+
+
+
+ #GstLockFlags
+
+
+
+
+
+ Decreases the reference count of the mini-object, possibly freeing
+the mini-object.
+
+
+
+
+
+ the mini-object
+
+
+
+
+
+ Adds a weak reference callback to a mini object. Weak references are
+used for notification when a mini object is finalized. They are called
+"weak references" because they allow you to safely hold a pointer
+to the mini object without calling gst_mini_object_ref()
+(gst_mini_object_ref() adds a strong reference, that is, forces the object
+to stay alive).
+
+
+
+
+
+ #GstMiniObject to reference weakly
+
+
+
+ callback to invoke before the mini object is freed
+
+
+
+ extra data to pass to notify
+
+
+
+
+
+ Removes a weak reference callback from a mini object.
+
+
+
+
+
+ #GstMiniObject to remove a weak reference from
+
+
+
+ callback to search for
+
+
+
+ data to search for
+
+
+
+
+
+ Atomically modifies a pointer to point to a new mini-object.
+The reference count of @olddata is decreased and the reference count of
+@newdata is increased.
+
+Either @newdata and the value pointed to by @olddata may be %NULL.
+
+ %TRUE if @newdata was different from @olddata
+
+
+
+
+ pointer to a pointer to a
+ mini-object to be replaced
+
+
+
+ pointer to new mini-object
+
+
+
+
+
+ Replace the current #GstMiniObject pointer to by @olddata with %NULL and
+return the old value.
+
+ the #GstMiniObject at @oldata
+
+
+
+
+ pointer to a pointer to a mini-object to
+ be stolen
+
+
+
+
+
+ Modifies a pointer to point to a new mini-object. The modification
+is done atomically. This version is similar to gst_mini_object_replace()
+except that it does not increase the refcount of @newdata and thus
+takes ownership of @newdata.
+
+Either @newdata and the value pointed to by @olddata may be %NULL.
+
+ %TRUE if @newdata was different from @olddata
+
+
+
+
+ pointer to a pointer to a mini-object to
+ be replaced
+
+
+
+ pointer to new mini-object
+
+
+
+
+
+
+ Function prototype for methods to create copies of instances.
+
+ reference to cloned instance.
+
+
+
+
+ MiniObject to copy
+
+
+
+
+
+ Function prototype for when a miniobject has lost its last refcount.
+Implementation of the mini object are allowed to revive the
+passed object by doing a gst_mini_object_ref(). If the object is not
+revived after the dispose function, the function should return %TRUE
+and the memory associated with the object is freed.
+
+ %TRUE if the object should be cleaned up.
+
+
+
+
+ MiniObject to dispose
+
+
+
+
+
+ Flags for the mini object
+
+ the object can be locked and unlocked with
+gst_mini_object_lock() and gst_mini_object_unlock().
+
+
+ the object is permanently locked in
+READONLY mode. Only read locks can be performed on the object.
+
+
+ the object is expected to stay alive
+even after gst_deinit() has been called and so should be ignored by leak
+detection tools. (Since 1.10)
+
+
+ first flag that can be used by subclasses.
+
+
+
+ Virtual function prototype for methods to free resources used by
+mini-objects.
+
+
+
+
+
+ MiniObject to free
+
+
+
+
+
+ A #GstMiniObjectNotify function can be added to a mini object as a
+callback that gets triggered when gst_mini_object_unref() drops the
+last ref and @obj is about to be freed.
+
+
+
+
+
+ data that was provided when the notify was added
+
+
+
+ the mini object
+
+
+
+
+
+ Constant that defines one GStreamer nanosecond
+
+
+
+ #GstObject provides a root for the object hierarchy tree filed in by the
+GStreamer library. It is currently a thin wrapper on top of
+#GInitiallyUnowned. It is an abstract class that is not very usable on its own.
+
+#GstObject gives us basic refcounting, parenting functionality and locking.
+Most of the functions are just extended for special GStreamer needs and can be
+found under the same name in the base class of #GstObject which is #GObject
+(e.g. g_object_ref() becomes gst_object_ref()).
+
+Since #GstObject derives from #GInitiallyUnowned, it also inherits the
+floating reference. Be aware that functions such as gst_bin_add() and
+gst_element_add_pad() take ownership of the floating reference.
+
+In contrast to #GObject instances, #GstObject adds a name property. The functions
+gst_object_set_name() and gst_object_get_name() are used to set/get the name
+of the object.
+
+## controlled properties
+
+Controlled properties offers a lightweight way to adjust gobject properties
+over stream-time. It works by using time-stamped value pairs that are queued
+for element-properties. At run-time the elements continuously pull value
+changes for the current stream-time.
+
+What needs to be changed in a #GstElement?
+Very little - it is just two steps to make a plugin controllable!
+
+ * mark gobject-properties paramspecs that make sense to be controlled,
+ by GST_PARAM_CONTROLLABLE.
+
+ * when processing data (get, chain, loop function) at the beginning call
+ gst_object_sync_values(element,timestamp).
+ This will make the controller update all GObject properties that are
+ under its control with the current values based on the timestamp.
+
+What needs to be done in applications? Again it's not a lot to change.
+
+ * create a #GstControlSource.
+ csource = gst_interpolation_control_source_new ();
+ g_object_set (csource, "mode", GST_INTERPOLATION_MODE_LINEAR, NULL);
+
+ * Attach the #GstControlSource on the controller to a property.
+ gst_object_add_control_binding (object, gst_direct_control_binding_new (object, "prop1", csource));
+
+ * Set the control values
+ gst_timed_value_control_source_set ((GstTimedValueControlSource *)csource,0 * GST_SECOND, value1);
+ gst_timed_value_control_source_set ((GstTimedValueControlSource *)csource,1 * GST_SECOND, value2);
+
+ * start your pipeline
+
+ Checks to see if there is any object named @name in @list. This function
+does not do any locking of any kind. You might want to protect the
+provided list with the lock of the owner of the list. This function
+will lock each #GstObject in the list to compare the name, so be
+careful when passing a list with a locked object.
+
+ %TRUE if a #GstObject named @name does not appear in @list,
+%FALSE if it does.
+
+MT safe. Grabs and releases the LOCK of each object in the list.
+
+
+
+
+ a list of #GstObject to
+ check through
+
+
+
+
+
+ the name to search for
+
+
+
+
+
+ A default deep_notify signal callback for an object. The user data
+should contain a pointer to an array of strings that should be excluded
+from the notify. The default handler will print the new value of the property
+using g_print.
+
+MT safe. This function grabs and releases @object's LOCK for getting its
+ path string.
+
+
+
+
+
+ the #GObject that signalled the notify.
+
+
+
+ a #GstObject that initiated the notify.
+
+
+
+ a #GParamSpec of the property.
+
+
+
+
+ a set of user-specified properties to exclude or %NULL to show
+ all changes.
+
+
+
+
+
+
+
+ Increase the reference count of @object, and possibly remove the floating
+reference, if @object has a floating reference.
+
+In other words, if the object is floating, then this call "assumes ownership"
+of the floating reference, converting it to a normal reference by clearing
+the floating flag while leaving the reference count unchanged. If the object
+is not floating, then this call adds a new normal reference increasing the
+reference count by one.
+
+For more background on "floating references" please see the #GObject
+documentation.
+
+
+
+
+
+ a #GstObject to sink
+
+
+
+
+
+ Atomically modifies a pointer to point to a new object.
+The reference count of @oldobj is decreased and the reference count of
+@newobj is increased.
+
+Either @newobj and the value pointed to by @oldobj may be %NULL.
+
+ %TRUE if @newobj was different from @oldobj
+
+
+
+
+ pointer to a place of
+ a #GstObject to replace
+
+
+
+ a new #GstObject
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Attach the #GstControlBinding to the object. If there already was a
+#GstControlBinding for this property it will be replaced.
+
+The object's reference count will be incremented, and any floating
+reference will be removed (see gst_object_ref_sink())
+
+ %FALSE if the given @binding has not been setup for this object or
+has been setup for a non suitable property, %TRUE otherwise.
+
+
+
+
+ the controller object
+
+
+
+ the #GstControlBinding that should be used
+
+
+
+
+
+ A default error function that uses g_printerr() to display the error message
+and the optional debug sting..
+
+The default handler will simply print the error string using g_print.
+
+
+
+
+
+ the #GstObject that initiated the error.
+
+
+
+ the GError.
+
+
+
+ an additional debug information string, or %NULL
+
+
+
+
+
+ Gets the corresponding #GstControlBinding for the property. This should be
+unreferenced again after use.
+
+ the #GstControlBinding for
+@property_name or %NULL if the property is not controlled.
+
+
+
+
+ the object
+
+
+
+ name of the property
+
+
+
+
+
+ Obtain the control-rate for this @object. Audio processing #GstElement
+objects will use this rate to sub-divide their processing loop and call
+gst_object_sync_values() inbetween. The length of the processing segment
+should be up to @control-rate nanoseconds.
+
+If the @object is not under property control, this will return
+%GST_CLOCK_TIME_NONE. This allows the element to avoid the sub-dividing.
+
+The control-rate is not expected to change if the element is in
+%GST_STATE_PAUSED or %GST_STATE_PLAYING.
+
+ the control rate in nanoseconds
+
+
+
+
+ the object that has controlled properties
+
+
+
+
+
+ Gets a number of #GValues for the given controlled property starting at the
+requested time. The array @values need to hold enough space for @n_values of
+#GValue.
+
+This function is useful if one wants to e.g. draw a graph of the control
+curve or apply a control curve sample by sample.
+
+ %TRUE if the given array could be filled, %FALSE otherwise
+
+
+
+
+ the object that has controlled properties
+
+
+
+ the name of the property to get
+
+
+
+ the time that should be processed
+
+
+
+ the time spacing between subsequent values
+
+
+
+ the number of values
+
+
+
+ array to put control-values in
+
+
+
+
+
+
+
+ Returns a copy of the name of @object.
+Caller should g_free() the return value after usage.
+For a nameless object, this returns %NULL, which you can safely g_free()
+as well.
+
+Free-function: g_free
+
+ the name of @object. g_free()
+after usage.
+
+MT safe. This function grabs and releases @object's LOCK.
+
+
+
+
+ a #GstObject
+
+
+
+
+
+ Returns the parent of @object. This function increases the refcount
+of the parent object so you should gst_object_unref() it after usage.
+
+ parent of @object, this can be
+ %NULL if @object has no parent. unref after usage.
+
+MT safe. Grabs and releases @object's LOCK.
+
+
+
+
+ a #GstObject
+
+
+
+
+
+ Generates a string describing the path of @object in
+the object hierarchy. Only useful (or used) for debugging.
+
+Free-function: g_free
+
+ a string describing the path of @object. You must
+ g_free() the string after usage.
+
+MT safe. Grabs and releases the #GstObject's LOCK for all objects
+ in the hierarchy.
+
+
+
+
+ a #GstObject
+
+
+
+
+
+ Gets the value for the given controlled property at the requested time.
+
+ the GValue of the property at the given time,
+or %NULL if the property isn't controlled.
+
+
+
+
+ the object that has controlled properties
+
+
+
+ the name of the property to get
+
+
+
+ the time the control-change should be read from
+
+
+
+
+
+ Gets a number of values for the given controlled property starting at the
+requested time. The array @values need to hold enough space for @n_values of
+the same type as the objects property's type.
+
+This function is useful if one wants to e.g. draw a graph of the control
+curve or apply a control curve sample by sample.
+
+The values are unboxed and ready to be used. The similar function
+gst_object_get_g_value_array() returns the array as #GValues and is
+better suites for bindings.
+
+ %TRUE if the given array could be filled, %FALSE otherwise
+
+
+
+
+ the object that has controlled properties
+
+
+
+ the name of the property to get
+
+
+
+ the time that should be processed
+
+
+
+ the time spacing between subsequent values
+
+
+
+ the number of values
+
+
+
+ array to put control-values in
+
+
+
+
+
+ Check if the @object has active controlled properties.
+
+ %TRUE if the object has active controlled properties
+
+
+
+
+ the object that has controlled properties
+
+
+
+
+
+ Check if @object has an ancestor @ancestor somewhere up in
+the hierarchy. One can e.g. check if a #GstElement is inside a #GstPipeline.
+ Use gst_object_has_as_ancestor() instead.
+
+MT safe. Grabs and releases @object's locks.
+
+ %TRUE if @ancestor is an ancestor of @object.
+
+
+
+
+ a #GstObject to check
+
+
+
+ a #GstObject to check as ancestor
+
+
+
+
+
+ Check if @object has an ancestor @ancestor somewhere up in
+the hierarchy. One can e.g. check if a #GstElement is inside a #GstPipeline.
+
+ %TRUE if @ancestor is an ancestor of @object.
+
+MT safe. Grabs and releases @object's locks.
+
+
+
+
+ a #GstObject to check
+
+
+
+ a #GstObject to check as ancestor
+
+
+
+
+
+ Check if @parent is the parent of @object.
+E.g. a #GstElement can check if it owns a given #GstPad.
+
+ %FALSE if either @object or @parent is %NULL. %TRUE if @parent is
+ the parent of @object. Otherwise %FALSE.
+
+MT safe. Grabs and releases @object's locks.
+
+
+
+
+ a #GstObject to check
+
+
+
+ a #GstObject to check as parent
+
+
+
+
+
+ Increments the reference count on @object. This function
+does not take the lock on @object because it relies on
+atomic refcounting.
+
+This object returns the input parameter to ease writing
+constructs like :
+ result = gst_object_ref (object->parent);
+
+ A pointer to @object
+
+
+
+
+ a #GstObject to reference
+
+
+
+
+
+ Removes the corresponding #GstControlBinding. If it was the
+last ref of the binding, it will be disposed.
+
+ %TRUE if the binding could be removed.
+
+
+
+
+ the object
+
+
+
+ the binding
+
+
+
+
+
+ This function is used to disable the control bindings on a property for
+some time, i.e. gst_object_sync_values() will do nothing for the
+property.
+
+
+
+
+
+ the object that has controlled properties
+
+
+
+ property to disable
+
+
+
+ boolean that specifies whether to disable the controller
+or not.
+
+
+
+
+
+ This function is used to disable all controlled properties of the @object for
+some time, i.e. gst_object_sync_values() will do nothing.
+
+
+
+
+
+ the object that has controlled properties
+
+
+
+ boolean that specifies whether to disable the controller
+or not.
+
+
+
+
+
+ Change the control-rate for this @object. Audio processing #GstElement
+objects will use this rate to sub-divide their processing loop and call
+gst_object_sync_values() inbetween. The length of the processing segment
+should be up to @control-rate nanoseconds.
+
+The control-rate should not change if the element is in %GST_STATE_PAUSED or
+%GST_STATE_PLAYING.
+
+
+
+
+
+ the object that has controlled properties
+
+
+
+ the new control-rate in nanoseconds.
+
+
+
+
+
+ Sets the name of @object, or gives @object a guaranteed unique
+name (if @name is %NULL).
+This function makes a copy of the provided name, so the caller
+retains ownership of the name it sent.
+
+ %TRUE if the name could be set. Since Objects that have
+a parent cannot be renamed, this function returns %FALSE in those
+cases.
+
+MT safe. This function grabs and releases @object's LOCK.
+
+
+
+
+ a #GstObject
+
+
+
+ new name of object
+
+
+
+
+
+ Sets the parent of @object to @parent. The object's reference count will
+be incremented, and any floating reference will be removed (see gst_object_ref_sink()).
+
+ %TRUE if @parent could be set or %FALSE when @object
+already had a parent or @object and @parent are the same.
+
+MT safe. Grabs and releases @object's LOCK.
+
+
+
+
+ a #GstObject
+
+
+
+ new parent of object
+
+
+
+
+
+ Returns a suggestion for timestamps where buffers should be split
+to get best controller results.
+
+ Returns the suggested timestamp or %GST_CLOCK_TIME_NONE
+if no control-rate was set.
+
+
+
+
+ the object that has controlled properties
+
+
+
+
+
+ Sets the properties of the object, according to the #GstControlSources that
+(maybe) handle them and for the given timestamp.
+
+If this function fails, it is most likely the application developers fault.
+Most probably the control sources are not setup correctly.
+
+ %TRUE if the controller values could be applied to the object
+properties, %FALSE otherwise
+
+
+
+
+ the object that has controlled properties
+
+
+
+ the time that should be processed
+
+
+
+
+
+ Clear the parent of @object, removing the associated reference.
+This function decreases the refcount of @object.
+
+MT safe. Grabs and releases @object's lock.
+
+
+
+
+
+ a #GstObject to unparent
+
+
+
+
+
+ Decrements the reference count on @object. If reference count hits
+zero, destroy @object. This function does not take the lock
+on @object as it relies on atomic refcounting.
+
+The unref method should never be called with the LOCK held since
+this might deadlock the dispose function.
+
+
+
+
+
+ a #GstObject to unreference
+
+
+
+
+
+
+
+
+ The parent of the object. Please note, that when changing the 'parent'
+property, we don't emit #GObject::notify and #GstObject::deep-notify
+signals due to locking issues. In some cases one can use
+#GstBin::element-added or #GstBin::element-removed signals on the parent to
+achieve a similar effect.
+
+
+
+
+
+
+ object LOCK
+
+
+
+ The name of the object
+
+
+
+ this object's parent, weak ref
+
+
+
+ flags for this object
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ The deep notify signal is used to be notified of property changes. It is
+typically attached to the toplevel bin to receive notifications from all
+the elements contained in that bin.
+
+
+
+
+
+ the object that originated the signal
+
+
+
+ the property that changed
+
+
+
+
+
+
+ GStreamer base object class.
+
+ parent
+
+
+
+ separator used by gst_object_get_path_string()
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ The standard flags that an gstobject may have.
+
+ the object is expected to stay alive even
+after gst_deinit() has been called and so should be ignored by leak
+detection tools. (Since 1.10)
+
+
+ subclasses can add additional flags starting from this flag
+
+
+
+ Use this flag on GObject properties to signal they can make sense to be.
+controlled over time. This hint is used by the GstController.
+
+
+
+ Use this flag on GObject properties of GstElements to indicate that
+they can be changed when the element is in the PAUSED or lower state.
+This flag implies GST_PARAM_MUTABLE_READY.
+
+
+
+ Use this flag on GObject properties of GstElements to indicate that
+they can be changed when the element is in the PLAYING or lower state.
+This flag implies GST_PARAM_MUTABLE_PAUSED.
+
+
+
+ Use this flag on GObject properties of GstElements to indicate that
+they can be changed when the element is in the READY or lower state.
+
+
+
+ Bits based on GST_PARAM_USER_SHIFT can be used by 3rd party applications.
+
+
+
+ The field name in a GstCaps that is used to signal the UUID of the protection
+system.
+
+
+
+ printf format type used to debug GStreamer types. You can use this in
+combination with GStreamer's debug logging system as well as the functions
+gst_info_vasprintf(), gst_info_strdup_vprintf() and gst_info_strdup_printf()
+to pretty-print the following types: #GstCaps, #GstStructure,
+#GstCapsFeatures, #GstTagList, #GstDateTime, #GstBuffer, #GstBufferList,
+#GstMessage, #GstEvent, #GstQuery, #GstContext, #GstPad, #GstObject. All
+#GObject types will be printed as typename plus pointer, and everything
+else will simply be printed as pointer address.
+
+This can only be used on types whose size is >= sizeof(gpointer).
+
+
+
+ A #GstElement is linked to other elements via "pads", which are extremely
+light-weight generic link points.
+
+Pads have a #GstPadDirection, source pads produce data, sink pads consume
+data.
+
+Pads are typically created from a #GstPadTemplate with
+gst_pad_new_from_template() and are then added to a #GstElement. This usually
+happens when the element is created but it can also happen dynamically based
+on the data that the element is processing or based on the pads that the
+application requests.
+
+Pads without pad templates can be created with gst_pad_new(),
+which takes a direction and a name as an argument. If the name is %NULL,
+then a guaranteed unique name will be assigned to it.
+
+A #GstElement creating a pad will typically use the various
+gst_pad_set_*_function() calls to register callbacks for events, queries or
+dataflow on the pads.
+
+gst_pad_get_parent() will retrieve the #GstElement that owns the pad.
+
+After two pads are retrieved from an element by gst_element_get_static_pad(),
+the pads can be linked with gst_pad_link(). (For quick links,
+you can also use gst_element_link(), which will make the obvious
+link for you if it's straightforward.). Pads can be unlinked again with
+gst_pad_unlink(). gst_pad_get_peer() can be used to check what the pad is
+linked to.
+
+Before dataflow is possible on the pads, they need to be activated with
+gst_pad_set_active().
+
+gst_pad_query() and gst_pad_peer_query() can be used to query various
+properties of the pad and the stream.
+
+To send a #GstEvent on a pad, use gst_pad_send_event() and
+gst_pad_push_event(). Some events will be sticky on the pad, meaning that
+after they pass on the pad they can be queried later with
+gst_pad_get_sticky_event() and gst_pad_sticky_events_foreach().
+gst_pad_get_current_caps() and gst_pad_has_current_caps() are convenience
+functions to query the current sticky CAPS event on a pad.
+
+GstElements will use gst_pad_push() and gst_pad_pull_range() to push out
+or pull in a buffer.
+
+The dataflow, events and queries that happen on a pad can be monitored with
+probes that can be installed with gst_pad_add_probe(). gst_pad_is_blocked()
+can be used to check if a block probe is installed on the pad.
+gst_pad_is_blocking() checks if the blocking probe is currently blocking the
+pad. gst_pad_remove_probe() is used to remove a previously installed probe
+and unblock blocking probes if any.
+
+Pad have an offset that can be retrieved with gst_pad_get_offset(). This
+offset will be applied to the running_time of all data passing over the pad.
+gst_pad_set_offset() can be used to change the offset.
+
+Convenience functions exist to start, pause and stop the task on a pad with
+gst_pad_start_task(), gst_pad_pause_task() and gst_pad_stop_task()
+respectively.
+
+ Creates a new pad with the given name in the given direction.
+If name is %NULL, a guaranteed unique name (across all pads)
+will be assigned.
+This function makes a copy of the name so you can safely free the name.
+
+ a new #GstPad, or %NULL in
+case of an error.
+
+MT safe.
+
+
+
+
+ the name of the new pad.
+
+
+
+ the #GstPadDirection of the pad.
+
+
+
+
+
+ Creates a new pad with the given name from the given static template.
+If name is %NULL, a guaranteed unique name (across all pads)
+will be assigned.
+This function makes a copy of the name so you can safely free the name.
+
+ a new #GstPad, or %NULL in
+case of an error.
+
+
+
+
+ the #GstStaticPadTemplate to use
+
+
+
+ the name of the pad
+
+
+
+
+
+ Creates a new pad with the given name from the given template.
+If name is %NULL, a guaranteed unique name (across all pads)
+will be assigned.
+This function makes a copy of the name so you can safely free the name.
+
+ a new #GstPad, or %NULL in
+case of an error.
+
+
+
+
+ the pad template to use
+
+
+
+ the name of the pad
+
+
+
+
+
+ Gets a string representing the given pad-link return.
+
+ a static string with the name of the pad-link return.
+
+
+
+
+ a #GstPadLinkReturn to get the name of.
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Activates or deactivates the given pad in @mode via dispatching to the
+pad's activatemodefunc. For use from within pad activation functions only.
+
+If you don't know what this is, you probably don't want to call it.
+
+ %TRUE if the operation was successful.
+
+MT safe.
+
+
+
+
+ the #GstPad to activate or deactivate.
+
+
+
+ the requested activation mode
+
+
+
+ whether or not the pad should be active.
+
+
+
+
+
+ Be notified of different states of pads. The provided callback is called for
+every state that matches @mask.
+
+Probes are called in groups: First GST_PAD_PROBE_TYPE_BLOCK probes are
+called, then others, then finally GST_PAD_PROBE_TYPE_IDLE. The only
+exception here are GST_PAD_PROBE_TYPE_IDLE probes that are called
+immediately if the pad is already idle while calling gst_pad_add_probe().
+In each of the groups, probes are called in the order in which they were
+added.
+
+ an id or 0 if no probe is pending. The id can be used to remove the
+probe with gst_pad_remove_probe(). When using GST_PAD_PROBE_TYPE_IDLE it can
+happen that the probe can be run immediately and if the probe returns
+GST_PAD_PROBE_REMOVE this functions returns 0.
+
+MT safe.
+
+
+
+
+ the #GstPad to add the probe to
+
+
+
+ the probe mask
+
+
+
+ #GstPadProbeCallback that will be called with notifications of
+ the pad state
+
+
+
+ user data passed to the callback
+
+
+
+ #GDestroyNotify for user_data
+
+
+
+
+
+ Checks if the source pad and the sink pad are compatible so they can be
+linked.
+
+ %TRUE if the pads can be linked.
+
+
+
+
+ the source #GstPad.
+
+
+
+ the sink #GstPad.
+
+
+
+
+
+ Chain a buffer to @pad.
+
+The function returns #GST_FLOW_FLUSHING if the pad was flushing.
+
+If the buffer type is not acceptable for @pad (as negotiated with a
+preceding GST_EVENT_CAPS event), this function returns
+#GST_FLOW_NOT_NEGOTIATED.
+
+The function proceeds calling the chain function installed on @pad (see
+gst_pad_set_chain_function()) and the return value of that function is
+returned to the caller. #GST_FLOW_NOT_SUPPORTED is returned if @pad has no
+chain function.
+
+In all cases, success or failure, the caller loses its reference to @buffer
+after calling this function.
+
+ a #GstFlowReturn from the pad.
+
+MT safe.
+
+
+
+
+ a sink #GstPad, returns GST_FLOW_ERROR if not.
+
+
+
+ the #GstBuffer to send, return GST_FLOW_ERROR
+ if not.
+
+
+
+
+
+ Chain a bufferlist to @pad.
+
+The function returns #GST_FLOW_FLUSHING if the pad was flushing.
+
+If @pad was not negotiated properly with a CAPS event, this function
+returns #GST_FLOW_NOT_NEGOTIATED.
+
+The function proceeds calling the chainlist function installed on @pad (see
+gst_pad_set_chain_list_function()) and the return value of that function is
+returned to the caller. #GST_FLOW_NOT_SUPPORTED is returned if @pad has no
+chainlist function.
+
+In all cases, success or failure, the caller loses its reference to @list
+after calling this function.
+
+MT safe.
+
+ a #GstFlowReturn from the pad.
+
+
+
+
+ a sink #GstPad, returns GST_FLOW_ERROR if not.
+
+
+
+ the #GstBufferList to send, return GST_FLOW_ERROR
+ if not.
+
+
+
+
+
+ Check and clear the #GST_PAD_FLAG_NEED_RECONFIGURE flag on @pad and return %TRUE
+if the flag was set.
+
+ %TRUE is the GST_PAD_FLAG_NEED_RECONFIGURE flag was set on @pad.
+
+
+
+
+ the #GstPad to check
+
+
+
+
+
+ Creates a stream-id for the source #GstPad @pad by combining the
+upstream information with the optional @stream_id of the stream
+of @pad. @pad must have a parent #GstElement and which must have zero
+or one sinkpad. @stream_id can only be %NULL if the parent element
+of @pad has only a single source pad.
+
+This function generates an unique stream-id by getting the upstream
+stream-start event stream ID and appending @stream_id to it. If the
+element has no sinkpad it will generate an upstream stream-id by
+doing an URI query on the element and in the worst case just uses
+a random number. Source elements that don't implement the URI
+handler interface should ideally generate a unique, deterministic
+stream-id manually instead.
+
+Since stream IDs are sorted alphabetically, any numbers in the
+stream ID should be printed with a fixed number of characters,
+preceded by 0's, such as by using the format \%03u instead of \%u.
+
+ A stream-id for @pad. g_free() after usage.
+
+
+
+
+ A source #GstPad
+
+
+
+ Parent #GstElement of @pad
+
+
+
+ The stream-id
+
+
+
+
+
+ Creates a stream-id for the source #GstPad @pad by combining the
+upstream information with the optional @stream_id of the stream
+of @pad. @pad must have a parent #GstElement and which must have zero
+or one sinkpad. @stream_id can only be %NULL if the parent element
+of @pad has only a single source pad.
+
+This function generates an unique stream-id by getting the upstream
+stream-start event stream ID and appending @stream_id to it. If the
+element has no sinkpad it will generate an upstream stream-id by
+doing an URI query on the element and in the worst case just uses
+a random number. Source elements that don't implement the URI
+handler interface should ideally generate a unique, deterministic
+stream-id manually instead.
+
+ A stream-id for @pad. g_free() after usage.
+
+
+
+
+ A source #GstPad
+
+
+
+ Parent #GstElement of @pad
+
+
+
+ The stream-id
+
+
+
+ parameters for the @stream_id format string
+
+
+
+
+
+ Creates a stream-id for the source #GstPad @pad by combining the
+upstream information with the optional @stream_id of the stream
+of @pad. @pad must have a parent #GstElement and which must have zero
+or one sinkpad. @stream_id can only be %NULL if the parent element
+of @pad has only a single source pad.
+
+This function generates an unique stream-id by getting the upstream
+stream-start event stream ID and appending @stream_id to it. If the
+element has no sinkpad it will generate an upstream stream-id by
+doing an URI query on the element and in the worst case just uses
+a random number. Source elements that don't implement the URI
+handler interface should ideally generate a unique, deterministic
+stream-id manually instead.
+
+ A stream-id for @pad. g_free() after usage.
+
+
+
+
+ A source #GstPad
+
+
+
+ Parent #GstElement of @pad
+
+
+
+ The stream-id
+
+
+
+ parameters for the @stream_id format string
+
+
+
+
+
+ Invokes the default event handler for the given pad.
+
+The EOS event will pause the task associated with @pad before it is forwarded
+to all internally linked pads,
+
+The event is sent to all pads internally linked to @pad. This function
+takes ownership of @event.
+
+ %TRUE if the event was sent successfully.
+
+
+
+
+ a #GstPad to call the default event handler on.
+
+
+
+ the parent of @pad or %NULL
+
+
+
+ the #GstEvent to handle.
+
+
+
+
+
+ Calls @forward for all internally linked pads of @pad. This function deals with
+dynamically changing internal pads and will make sure that the @forward
+function is only called once for each pad.
+
+When @forward returns %TRUE, no further pads will be processed.
+
+ %TRUE if one of the dispatcher functions returned %TRUE.
+
+
+
+
+ a #GstPad
+
+
+
+ a #GstPadForwardFunction
+
+
+
+ user data passed to @forward
+
+
+
+
+
+ Gets the capabilities of the allowed media types that can flow through
+@pad and its peer.
+
+The allowed capabilities is calculated as the intersection of the results of
+calling gst_pad_query_caps() on @pad and its peer. The caller owns a reference
+on the resulting caps.
+
+ the allowed #GstCaps of the
+ pad link. Unref the caps when you no longer need it. This
+ function returns %NULL when @pad has no peer.
+
+MT safe.
+
+
+
+
+ a #GstPad.
+
+
+
+
+
+ Gets the capabilities currently configured on @pad with the last
+#GST_EVENT_CAPS event.
+
+ the current caps of the pad with
+incremented ref-count or %NULL when pad has no caps. Unref after usage.
+
+
+
+
+ a #GstPad to get the current capabilities of.
+
+
+
+
+
+ Gets the direction of the pad. The direction of the pad is
+decided at construction time so this function does not take
+the LOCK.
+
+ the #GstPadDirection of the pad.
+
+MT safe.
+
+
+
+
+ a #GstPad to get the direction of.
+
+
+
+
+
+ Gets the private data of a pad.
+No locking is performed in this function.
+
+ a #gpointer to the private data.
+
+
+
+
+ the #GstPad to get the private data of.
+
+
+
+
+
+ Gets the #GstFlowReturn return from the last data passed by this pad.
+
+
+
+
+
+ the #GstPad
+
+
+
+
+
+ Get the offset applied to the running time of @pad. @pad has to be a source
+pad.
+
+ the offset.
+
+
+
+
+ a #GstPad
+
+
+
+
+
+ Gets the template for @pad.
+
+ the #GstPadTemplate from which
+ this pad was instantiated, or %NULL if this pad has no
+ template. Unref after usage.
+
+
+
+
+ a #GstPad.
+
+
+
+
+
+ Gets the capabilities for @pad's template.
+
+ the #GstCaps of this pad template.
+Unref after usage.
+
+
+
+
+ a #GstPad to get the template capabilities from.
+
+
+
+
+
+ Gets the parent of @pad, cast to a #GstElement. If a @pad has no parent or
+its parent is not an element, return %NULL.
+
+ the parent of the pad. The
+caller has a reference on the parent, so unref when you're finished
+with it.
+
+MT safe.
+
+
+
+
+ a pad
+
+
+
+
+
+ Gets the peer of @pad. This function refs the peer pad so
+you need to unref it after use.
+
+ the peer #GstPad. Unref after usage.
+
+MT safe.
+
+
+
+
+ a #GstPad to get the peer of.
+
+
+
+
+
+ When @pad is flushing this function returns #GST_FLOW_FLUSHING
+immediately and @buffer is %NULL.
+
+Calls the getrange function of @pad, see #GstPadGetRangeFunction for a
+description of a getrange function. If @pad has no getrange function
+installed (see gst_pad_set_getrange_function()) this function returns
+#GST_FLOW_NOT_SUPPORTED.
+
+If @buffer points to a variable holding %NULL, a valid new #GstBuffer will be
+placed in @buffer when this function returns #GST_FLOW_OK. The new buffer
+must be freed with gst_buffer_unref() after usage.
+
+When @buffer points to a variable that points to a valid #GstBuffer, the
+buffer will be filled with the result data when this function returns
+#GST_FLOW_OK. If the provided buffer is larger than @size, only
+@size bytes will be filled in the result buffer and its size will be updated
+accordingly.
+
+Note that less than @size bytes can be returned in @buffer when, for example,
+an EOS condition is near or when @buffer is not large enough to hold @size
+bytes. The caller should check the result buffer size to get the result size.
+
+When this function returns any other result value than #GST_FLOW_OK, @buffer
+will be unchanged.
+
+This is a lowlevel function. Usually gst_pad_pull_range() is used.
+
+ a #GstFlowReturn from the pad.
+
+MT safe.
+
+
+
+
+ a src #GstPad, returns #GST_FLOW_ERROR if not.
+
+
+
+ The start offset of the buffer
+
+
+
+ The length of the buffer
+
+
+
+ a pointer to hold the #GstBuffer,
+ returns #GST_FLOW_ERROR if %NULL.
+
+
+
+
+
+ Returns a new reference of the sticky event of type @event_type
+from the event.
+
+ a #GstEvent of type
+@event_type or %NULL when no event of @event_type was on
+@pad. Unref after usage.
+
+
+
+
+ the #GstPad to get the event from.
+
+
+
+ the #GstEventType that should be retrieved.
+
+
+
+ the index of the event
+
+
+
+
+
+ Returns the current #GstStream for the @pad, or %NULL if none has been
+set yet, i.e. the pad has not received a stream-start event yet.
+
+This is a convenience wrapper around gst_pad_get_sticky_event() and
+gst_event_parse_stream().
+
+ the current #GstStream for @pad, or %NULL.
+ unref the returned stream when no longer needed.
+
+
+
+
+ A source #GstPad
+
+
+
+
+
+ Returns the current stream-id for the @pad, or %NULL if none has been
+set yet, i.e. the pad has not received a stream-start event yet.
+
+This is a convenience wrapper around gst_pad_get_sticky_event() and
+gst_event_parse_stream_start().
+
+The returned stream-id string should be treated as an opaque string, its
+contents should not be interpreted.
+
+ a newly-allocated copy of the stream-id for
+ @pad, or %NULL. g_free() the returned string when no longer
+ needed.
+
+
+
+
+ A source #GstPad
+
+
+
+
+
+ Get @pad task state. If no task is currently
+set, #GST_TASK_STOPPED is returned.
+
+ The current state of @pad's task.
+
+
+
+
+ the #GstPad to get task state from
+
+
+
+
+
+ Check if @pad has caps set on it with a #GST_EVENT_CAPS event.
+
+ %TRUE when @pad has caps associated with it.
+
+
+
+
+ a #GstPad to check
+
+
+
+
+
+ Query if a pad is active
+
+ %TRUE if the pad is active.
+
+MT safe.
+
+
+
+
+ the #GstPad to query
+
+
+
+
+
+ Checks if the pad is blocked or not. This function returns the
+last requested state of the pad. It is not certain that the pad
+is actually blocking at this point (see gst_pad_is_blocking()).
+
+ %TRUE if the pad is blocked.
+
+MT safe.
+
+
+
+
+ the #GstPad to query
+
+
+
+
+
+ Checks if the pad is blocking or not. This is a guaranteed state
+of whether the pad is actually blocking on a #GstBuffer or a #GstEvent.
+
+ %TRUE if the pad is blocking.
+
+MT safe.
+
+
+
+
+ the #GstPad to query
+
+
+
+
+
+ Checks if a @pad is linked to another pad or not.
+
+ %TRUE if the pad is linked, %FALSE otherwise.
+
+MT safe.
+
+
+
+
+ pad to check
+
+
+
+
+
+ Gets an iterator for the pads to which the given pad is linked to inside
+of the parent element.
+
+Each #GstPad element yielded by the iterator will have its refcount increased,
+so unref after use.
+
+Free-function: gst_iterator_free
+
+ a new #GstIterator of #GstPad
+ or %NULL when the pad does not have an iterator function
+ configured. Use gst_iterator_free() after usage.
+
+
+
+
+ the GstPad to get the internal links of.
+
+
+
+
+
+ Iterate the list of pads to which the given pad is linked to inside of
+the parent element.
+This is the default handler, and thus returns an iterator of all of the
+pads inside the parent element with opposite direction.
+
+The caller must free this iterator after use with gst_iterator_free().
+
+ a #GstIterator of #GstPad, or %NULL if @pad
+has no parent. Unref each returned pad with gst_object_unref().
+
+
+
+
+ the #GstPad to get the internal links of.
+
+
+
+ the parent of @pad or %NULL
+
+
+
+
+
+ Links the source pad and the sink pad.
+
+ A result code indicating if the connection worked or
+ what went wrong.
+
+MT Safe.
+
+
+
+
+ the source #GstPad to link.
+
+
+
+ the sink #GstPad to link.
+
+
+
+
+
+ Links the source pad and the sink pad.
+
+This variant of #gst_pad_link provides a more granular control on the
+checks being done when linking. While providing some considerable speedups
+the caller of this method must be aware that wrong usage of those flags
+can cause severe issues. Refer to the documentation of #GstPadLinkCheck
+for more information.
+
+MT Safe.
+
+ A result code indicating if the connection worked or
+ what went wrong.
+
+
+
+
+ the source #GstPad to link.
+
+
+
+ the sink #GstPad to link.
+
+
+
+ the checks to validate when linking
+
+
+
+
+
+ Links @src to @sink, creating any #GstGhostPad's in between as necessary.
+
+This is a convenience function to save having to create and add intermediate
+#GstGhostPad's as required for linking across #GstBin boundaries.
+
+If @src or @sink pads don't have parent elements or do not share a common
+ancestor, the link will fail.
+
+ whether the link succeeded.
+
+
+
+
+ a #GstPad
+
+
+
+ a #GstPad
+
+
+
+
+
+ Links @src to @sink, creating any #GstGhostPad's in between as necessary.
+
+This is a convenience function to save having to create and add intermediate
+#GstGhostPad's as required for linking across #GstBin boundaries.
+
+If @src or @sink pads don't have parent elements or do not share a common
+ancestor, the link will fail.
+
+Calling gst_pad_link_maybe_ghosting_full() with
+@flags == %GST_PAD_LINK_CHECK_DEFAULT is the recommended way of linking
+pads with safety checks applied.
+
+ whether the link succeeded.
+
+
+
+
+ a #GstPad
+
+
+
+ a #GstPad
+
+
+
+ some #GstPadLinkCheck flags
+
+
+
+
+
+ Mark a pad for needing reconfiguration. The next call to
+gst_pad_check_reconfigure() will return %TRUE after this call.
+
+
+
+
+
+ the #GstPad to mark
+
+
+
+
+
+ Check the #GST_PAD_FLAG_NEED_RECONFIGURE flag on @pad and return %TRUE
+if the flag was set.
+
+ %TRUE is the GST_PAD_FLAG_NEED_RECONFIGURE flag is set on @pad.
+
+
+
+
+ the #GstPad to check
+
+
+
+
+
+ Pause the task of @pad. This function will also wait until the
+function executed by the task is finished if this function is not
+called from the task function.
+
+ a %TRUE if the task could be paused or %FALSE when the pad
+has no task.
+
+
+
+
+ the #GstPad to pause the task of
+
+
+
+
+
+ Performs gst_pad_query() on the peer of @pad.
+
+The caller is responsible for both the allocation and deallocation of
+the query structure.
+
+ %TRUE if the query could be performed. This function returns %FALSE
+if @pad has no peer.
+
+
+
+
+ a #GstPad to invoke the peer query on.
+
+
+
+ the #GstQuery to perform.
+
+
+
+
+
+ Check if the peer of @pad accepts @caps. If @pad has no peer, this function
+returns %TRUE.
+
+ %TRUE if the peer of @pad can accept the caps or @pad has no peer.
+
+
+
+
+ a #GstPad to check the peer of
+
+
+
+ a #GstCaps to check on the pad
+
+
+
+
+
+ Gets the capabilities of the peer connected to this pad. Similar to
+gst_pad_query_caps().
+
+When called on srcpads @filter contains the caps that
+upstream could produce in the order preferred by upstream. When
+called on sinkpads @filter contains the caps accepted by
+downstream in the preferred order. @filter might be %NULL but
+if it is not %NULL the returned caps will be a subset of @filter.
+
+ the caps of the peer pad with incremented
+ref-count. When there is no peer pad, this function returns @filter or,
+when @filter is %NULL, ANY caps.
+
+
+
+
+ a #GstPad to get the capabilities of.
+
+
+
+ a #GstCaps filter, or %NULL.
+
+
+
+
+
+ Queries the peer pad of a given sink pad to convert @src_val in @src_format
+to @dest_format.
+
+ %TRUE if the query could be performed.
+
+
+
+
+ a #GstPad, on whose peer pad to invoke the convert query on.
+ Must be a sink pad.
+
+
+
+ a #GstFormat to convert from.
+
+
+
+ a value to convert.
+
+
+
+ the #GstFormat to convert to.
+
+
+
+ a pointer to the result.
+
+
+
+
+
+ Queries the peer pad of a given sink pad for the total stream duration.
+
+ %TRUE if the query could be performed.
+
+
+
+
+ a #GstPad on whose peer pad to invoke the duration query on.
+ Must be a sink pad.
+
+
+
+ the #GstFormat requested
+
+
+
+ a location in which to store the total
+ duration, or %NULL.
+
+
+
+
+
+ Queries the peer of a given sink pad for the stream position.
+
+ %TRUE if the query could be performed.
+
+
+
+
+ a #GstPad on whose peer to invoke the position query on.
+ Must be a sink pad.
+
+
+
+ the #GstFormat requested
+
+
+
+ a location in which to store the current
+ position, or %NULL.
+
+
+
+
+
+ Checks if all internally linked pads of @pad accepts the caps in @query and
+returns the intersection of the results.
+
+This function is useful as a default accept caps query function for an element
+that can handle any stream format, but requires caps that are acceptable for
+all opposite pads.
+
+ %TRUE if @query could be executed
+
+
+
+
+ a #GstPad to proxy.
+
+
+
+ an ACCEPT_CAPS #GstQuery.
+
+
+
+
+
+ Calls gst_pad_query_caps() for all internally linked pads of @pad and returns
+the intersection of the results.
+
+This function is useful as a default caps query function for an element
+that can handle any stream format, but requires all its pads to have
+the same caps. Two such elements are tee and adder.
+
+ %TRUE if @query could be executed
+
+
+
+
+ a #GstPad to proxy.
+
+
+
+ a CAPS #GstQuery.
+
+
+
+
+
+ Pulls a @buffer from the peer pad or fills up a provided buffer.
+
+This function will first trigger the pad block signal if it was
+installed.
+
+When @pad is not linked #GST_FLOW_NOT_LINKED is returned else this
+function returns the result of gst_pad_get_range() on the peer pad.
+See gst_pad_get_range() for a list of return values and for the
+semantics of the arguments of this function.
+
+If @buffer points to a variable holding %NULL, a valid new #GstBuffer will be
+placed in @buffer when this function returns #GST_FLOW_OK. The new buffer
+must be freed with gst_buffer_unref() after usage. When this function
+returns any other result value, @buffer will still point to %NULL.
+
+When @buffer points to a variable that points to a valid #GstBuffer, the
+buffer will be filled with the result data when this function returns
+#GST_FLOW_OK. When this function returns any other result value,
+@buffer will be unchanged. If the provided buffer is larger than @size, only
+@size bytes will be filled in the result buffer and its size will be updated
+accordingly.
+
+Note that less than @size bytes can be returned in @buffer when, for example,
+an EOS condition is near or when @buffer is not large enough to hold @size
+bytes. The caller should check the result buffer size to get the result size.
+
+ a #GstFlowReturn from the peer pad.
+
+MT safe.
+
+
+
+
+ a sink #GstPad, returns GST_FLOW_ERROR if not.
+
+
+
+ The start offset of the buffer
+
+
+
+ The length of the buffer
+
+
+
+ a pointer to hold the #GstBuffer, returns
+ GST_FLOW_ERROR if %NULL.
+
+
+
+
+
+ Pushes a buffer to the peer of @pad.
+
+This function will call installed block probes before triggering any
+installed data probes.
+
+The function proceeds calling gst_pad_chain() on the peer pad and returns
+the value from that function. If @pad has no peer, #GST_FLOW_NOT_LINKED will
+be returned.
+
+In all cases, success or failure, the caller loses its reference to @buffer
+after calling this function.
+
+ a #GstFlowReturn from the peer pad.
+
+MT safe.
+
+
+
+
+ a source #GstPad, returns #GST_FLOW_ERROR if not.
+
+
+
+ the #GstBuffer to push returns GST_FLOW_ERROR
+ if not.
+
+
+
+
+
+ Sends the event to the peer of the given pad. This function is
+mainly used by elements to send events to their peer
+elements.
+
+This function takes ownership of the provided event so you should
+gst_event_ref() it if you want to reuse the event after this call.
+
+ %TRUE if the event was handled.
+
+MT safe.
+
+
+
+
+ a #GstPad to push the event to.
+
+
+
+ the #GstEvent to send to the pad.
+
+
+
+
+
+ Pushes a buffer list to the peer of @pad.
+
+This function will call installed block probes before triggering any
+installed data probes.
+
+The function proceeds calling the chain function on the peer pad and returns
+the value from that function. If @pad has no peer, #GST_FLOW_NOT_LINKED will
+be returned. If the peer pad does not have any installed chainlist function
+every group buffer of the list will be merged into a normal #GstBuffer and
+chained via gst_pad_chain().
+
+In all cases, success or failure, the caller loses its reference to @list
+after calling this function.
+
+ a #GstFlowReturn from the peer pad.
+
+MT safe.
+
+
+
+
+ a source #GstPad, returns #GST_FLOW_ERROR if not.
+
+
+
+ the #GstBufferList to push returns GST_FLOW_ERROR
+ if not.
+
+
+
+
+
+ Dispatches a query to a pad. The query should have been allocated by the
+caller via one of the type-specific allocation functions. The element that
+the pad belongs to is responsible for filling the query with an appropriate
+response, which should then be parsed with a type-specific query parsing
+function.
+
+Again, the caller is responsible for both the allocation and deallocation of
+the query structure.
+
+Please also note that some queries might need a running pipeline to work.
+
+ %TRUE if the query could be performed.
+
+
+
+
+ a #GstPad to invoke the default query on.
+
+
+
+ the #GstQuery to perform.
+
+
+
+
+
+ Check if the given pad accepts the caps.
+
+ %TRUE if the pad can accept the caps.
+
+
+
+
+ a #GstPad to check
+
+
+
+ a #GstCaps to check on the pad
+
+
+
+
+
+ Gets the capabilities this pad can produce or consume.
+Note that this method doesn't necessarily return the caps set by sending a
+gst_event_new_caps() - use gst_pad_get_current_caps() for that instead.
+gst_pad_query_caps returns all possible caps a pad can operate with, using
+the pad's CAPS query function, If the query fails, this function will return
+@filter, if not %NULL, otherwise ANY.
+
+When called on sinkpads @filter contains the caps that
+upstream could produce in the order preferred by upstream. When
+called on srcpads @filter contains the caps accepted by
+downstream in the preferred order. @filter might be %NULL but
+if it is not %NULL the returned caps will be a subset of @filter.
+
+Note that this function does not return writable #GstCaps, use
+gst_caps_make_writable() before modifying the caps.
+
+ the caps of the pad with incremented ref-count.
+
+
+
+
+ a #GstPad to get the capabilities of.
+
+
+
+ suggested #GstCaps, or %NULL
+
+
+
+
+
+ Queries a pad to convert @src_val in @src_format to @dest_format.
+
+ %TRUE if the query could be performed.
+
+
+
+
+ a #GstPad to invoke the convert query on.
+
+
+
+ a #GstFormat to convert from.
+
+
+
+ a value to convert.
+
+
+
+ the #GstFormat to convert to.
+
+
+
+ a pointer to the result.
+
+
+
+
+
+ Invokes the default query handler for the given pad.
+The query is sent to all pads internally linked to @pad. Note that
+if there are many possible sink pads that are internally linked to
+@pad, only one will be sent the query.
+Multi-sinkpad elements should implement custom query handlers.
+
+ %TRUE if the query was performed successfully.
+
+
+
+
+ a #GstPad to call the default query handler on.
+
+
+
+ the parent of @pad or %NULL
+
+
+
+ the #GstQuery to handle.
+
+
+
+
+
+ Queries a pad for the total stream duration.
+
+ %TRUE if the query could be performed.
+
+
+
+
+ a #GstPad to invoke the duration query on.
+
+
+
+ the #GstFormat requested
+
+
+
+ a location in which to store the total
+ duration, or %NULL.
+
+
+
+
+
+ Queries a pad for the stream position.
+
+ %TRUE if the query could be performed.
+
+
+
+
+ a #GstPad to invoke the position query on.
+
+
+
+ the #GstFormat requested
+
+
+
+ A location in which to store the current position, or %NULL.
+
+
+
+
+
+ Remove the probe with @id from @pad.
+
+MT safe.
+
+
+
+
+
+ the #GstPad with the probe
+
+
+
+ the probe id to remove
+
+
+
+
+
+ Sends the event to the pad. This function can be used
+by applications to send events in the pipeline.
+
+If @pad is a source pad, @event should be an upstream event. If @pad is a
+sink pad, @event should be a downstream event. For example, you would not
+send a #GST_EVENT_EOS on a src pad; EOS events only propagate downstream.
+Furthermore, some downstream events have to be serialized with data flow,
+like EOS, while some can travel out-of-band, like #GST_EVENT_FLUSH_START. If
+the event needs to be serialized with data flow, this function will take the
+pad's stream lock while calling its event function.
+
+To find out whether an event type is upstream, downstream, or downstream and
+serialized, see #GstEventTypeFlags, gst_event_type_get_flags(),
+#GST_EVENT_IS_UPSTREAM, #GST_EVENT_IS_DOWNSTREAM, and
+#GST_EVENT_IS_SERIALIZED. Note that in practice that an application or
+plugin doesn't need to bother itself with this information; the core handles
+all necessary locks and checks.
+
+This function takes ownership of the provided event so you should
+gst_event_ref() it if you want to reuse the event after this call.
+
+ %TRUE if the event was handled.
+
+
+
+
+ a #GstPad to send the event to.
+
+
+
+ the #GstEvent to send to the pad.
+
+
+
+
+
+ Sets the given activate function for @pad. The activate function will
+dispatch to gst_pad_activate_mode() to perform the actual activation.
+Only makes sense to set on sink pads.
+
+Call this function if your sink pad can start a pull-based task.
+
+
+
+
+
+ a #GstPad.
+
+
+
+ the #GstPadActivateFunction to set.
+
+
+
+ user_data passed to @notify
+
+
+
+ notify called when @activate will not be used anymore.
+
+
+
+
+
+ Sets the given activate_mode function for the pad. An activate_mode function
+prepares the element for data passing.
+
+
+
+
+
+ a #GstPad.
+
+
+
+ the #GstPadActivateModeFunction to set.
+
+
+
+ user_data passed to @notify
+
+
+
+ notify called when @activatemode will not be used anymore.
+
+
+
+
+
+ Activates or deactivates the given pad.
+Normally called from within core state change functions.
+
+If @active, makes sure the pad is active. If it is already active, either in
+push or pull mode, just return. Otherwise dispatches to the pad's activate
+function to perform the actual activation.
+
+If not @active, calls gst_pad_activate_mode() with the pad's current mode
+and a %FALSE argument.
+
+ %TRUE if the operation was successful.
+
+MT safe.
+
+
+
+
+ the #GstPad to activate or deactivate.
+
+
+
+ whether or not the pad should be active.
+
+
+
+
+
+ Sets the given chain function for the pad. The chain function is called to
+process a #GstBuffer input buffer. see #GstPadChainFunction for more details.
+
+
+
+
+
+ a sink #GstPad.
+
+
+
+ the #GstPadChainFunction to set.
+
+
+
+ user_data passed to @notify
+
+
+
+ notify called when @chain will not be used anymore.
+
+
+
+
+
+ Sets the given chain list function for the pad. The chainlist function is
+called to process a #GstBufferList input buffer list. See
+#GstPadChainListFunction for more details.
+
+
+
+
+
+ a sink #GstPad.
+
+
+
+ the #GstPadChainListFunction to set.
+
+
+
+ user_data passed to @notify
+
+
+
+ notify called when @chainlist will not be used anymore.
+
+
+
+
+
+ Set the given private data gpointer on the pad.
+This function can only be used by the element that owns the pad.
+No locking is performed in this function.
+
+
+
+
+
+ the #GstPad to set the private data of.
+
+
+
+ The private data to attach to the pad.
+
+
+
+
+
+ Sets the given event handler for the pad.
+
+
+
+
+
+ a #GstPad of either direction.
+
+
+
+ the #GstPadEventFullFunction to set.
+
+
+
+ user_data passed to @notify
+
+
+
+ notify called when @event will not be used anymore.
+
+
+
+
+
+ Sets the given event handler for the pad.
+
+
+
+
+
+ a #GstPad of either direction.
+
+
+
+ the #GstPadEventFunction to set.
+
+
+
+ user_data passed to @notify
+
+
+
+ notify called when @event will not be used anymore.
+
+
+
+
+
+ Sets the given getrange function for the pad. The getrange function is
+called to produce a new #GstBuffer to start the processing pipeline. see
+#GstPadGetRangeFunction for a description of the getrange function.
+
+
+
+
+
+ a source #GstPad.
+
+
+
+ the #GstPadGetRangeFunction to set.
+
+
+
+ user_data passed to @notify
+
+
+
+ notify called when @get will not be used anymore.
+
+
+
+
+
+ Sets the given internal link iterator function for the pad.
+
+
+
+
+
+ a #GstPad of either direction.
+
+
+
+ the #GstPadIterIntLinkFunction to set.
+
+
+
+ user_data passed to @notify
+
+
+
+ notify called when @iterintlink will not be used anymore.
+
+
+
+
+
+ Sets the given link function for the pad. It will be called when
+the pad is linked with another pad.
+
+The return value #GST_PAD_LINK_OK should be used when the connection can be
+made.
+
+The return value #GST_PAD_LINK_REFUSED should be used when the connection
+cannot be made for some reason.
+
+If @link is installed on a source pad, it should call the #GstPadLinkFunction
+of the peer sink pad, if present.
+
+
+
+
+
+ a #GstPad.
+
+
+
+ the #GstPadLinkFunction to set.
+
+
+
+ user_data passed to @notify
+
+
+
+ notify called when @link will not be used anymore.
+
+
+
+
+
+ Set the offset that will be applied to the running time of @pad.
+
+
+
+
+
+ a #GstPad
+
+
+
+ the offset
+
+
+
+
+
+ Set the given query function for the pad.
+
+
+
+
+
+ a #GstPad of either direction.
+
+
+
+ the #GstPadQueryFunction to set.
+
+
+
+ user_data passed to @notify
+
+
+
+ notify called when @query will not be used anymore.
+
+
+
+
+
+ Sets the given unlink function for the pad. It will be called
+when the pad is unlinked.
+
+
+
+
+
+ a #GstPad.
+
+
+
+ the #GstPadUnlinkFunction to set.
+
+
+
+ user_data passed to @notify
+
+
+
+ notify called when @unlink will not be used anymore.
+
+
+
+
+
+ Starts a task that repeatedly calls @func with @user_data. This function
+is mostly used in pad activation functions to start the dataflow.
+The #GST_PAD_STREAM_LOCK of @pad will automatically be acquired
+before @func is called.
+
+ a %TRUE if the task could be started.
+
+
+
+
+ the #GstPad to start the task of
+
+
+
+ the task function to call
+
+
+
+ user data passed to the task function
+
+
+
+ called when @user_data is no longer referenced
+
+
+
+
+
+ Iterates all sticky events on @pad and calls @foreach_func for every
+event. If @foreach_func returns %FALSE the iteration is immediately stopped.
+
+
+
+
+
+ the #GstPad that should be used for iteration.
+
+
+
+ the #GstPadStickyEventsForeachFunction that
+ should be called for every event.
+
+
+
+ the optional user data.
+
+
+
+
+
+ Stop the task of @pad. This function will also make sure that the
+function executed by the task will effectively stop if not called
+from the GstTaskFunction.
+
+This function will deadlock if called from the GstTaskFunction of
+the task. Use gst_task_pause() instead.
+
+Regardless of whether the pad has a task, the stream lock is acquired and
+released so as to ensure that streaming through this pad has finished.
+
+ a %TRUE if the task could be stopped or %FALSE on error.
+
+
+
+
+ the #GstPad to stop the task of
+
+
+
+
+
+ Store the sticky @event on @pad
+
+ #GST_FLOW_OK on success, #GST_FLOW_FLUSHING when the pad
+was flushing or #GST_FLOW_EOS when the pad was EOS.
+
+
+
+
+ a #GstPad
+
+
+
+ a #GstEvent
+
+
+
+
+
+ Unlinks the source pad from the sink pad. Will emit the #GstPad::unlinked
+signal on both pads.
+
+ %TRUE if the pads were unlinked. This function returns %FALSE if
+the pads were not linked together.
+
+MT safe.
+
+
+
+
+ the source #GstPad to unlink.
+
+
+
+ the sink #GstPad to unlink.
+
+
+
+
+
+ A helper function you can use that sets the FIXED_CAPS flag
+This way the default CAPS query will always return the negotiated caps
+or in case the pad is not negotiated, the padtemplate caps.
+
+The negotiated caps are the caps of the last CAPS event that passed on the
+pad. Use this function on a pad that, once it negotiated to a CAPS, cannot
+be renegotiated to something else.
+
+
+
+
+
+ the pad to use
+
+
+
+
+
+
+
+
+
+
+
+ The offset that will be applied to the running time of the pad.
+
+
+
+
+
+
+
+
+
+ private data owned by the parent element
+
+
+
+ padtemplate for this pad
+
+
+
+ the direction of the pad, cannot change after creating
+ the pad.
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Signals that a pad has been linked to the peer pad.
+
+
+
+
+
+ the peer pad that has been connected
+
+
+
+
+
+ Signals that a pad has been unlinked from the peer pad.
+
+
+
+
+
+ the peer pad that has been disconnected
+
+
+
+
+
+
+ This function is called when the pad is activated during the element
+READY to PAUSED state change. By default this function will call the
+activate function that puts the pad in push mode but elements can
+override this function to activate the pad in pull mode if they wish.
+
+ %TRUE if the pad could be activated.
+
+
+
+
+ a #GstPad
+
+
+
+ the parent of @pad
+
+
+
+
+
+ The prototype of the push and pull activate functions.
+
+ %TRUE if the pad could be activated or deactivated.
+
+
+
+
+ a #GstPad
+
+
+
+ the parent of @pad
+
+
+
+ the requested activation mode of @pad
+
+
+
+ activate or deactivate the pad.
+
+
+
+
+
+ A function that will be called on sinkpads when chaining buffers.
+The function typically processes the data contained in the buffer and
+either consumes the data or passes it on to the internally linked pad(s).
+
+The implementer of this function receives a refcount to @buffer and should
+gst_buffer_unref() when the buffer is no longer needed.
+
+When a chain function detects an error in the data stream, it must post an
+error on the bus and return an appropriate #GstFlowReturn value.
+
+ #GST_FLOW_OK for success
+
+
+
+
+ the sink #GstPad that performed the chain.
+
+
+
+ the parent of @pad. If the #GST_PAD_FLAG_NEED_PARENT
+ flag is set, @parent is guaranteed to be not-%NULL and remain valid
+ during the execution of this function.
+
+
+
+ the #GstBuffer that is chained, not %NULL.
+
+
+
+
+
+ A function that will be called on sinkpads when chaining buffer lists.
+The function typically processes the data contained in the buffer list and
+either consumes the data or passes it on to the internally linked pad(s).
+
+The implementer of this function receives a refcount to @list and
+should gst_buffer_list_unref() when the list is no longer needed.
+
+When a chainlist function detects an error in the data stream, it must
+post an error on the bus and return an appropriate #GstFlowReturn value.
+
+ #GST_FLOW_OK for success
+
+
+
+
+ the sink #GstPad that performed the chain.
+
+
+
+ the parent of @pad. If the #GST_PAD_FLAG_NEED_PARENT
+ flag is set, @parent is guaranteed to be not-%NULL and remain valid
+ during the execution of this function.
+
+
+
+ the #GstBufferList that is chained, not %NULL.
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ The direction of a pad.
+
+ direction is unknown.
+
+
+ the pad is a source pad.
+
+
+ the pad is a sink pad.
+
+
+
+ Function signature to handle an event for the pad.
+
+This variant is for specific elements that will take into account the
+last downstream flow return (from a pad push), in which case they can
+return it.
+
+ %GST_FLOW_OK if the event was handled properly, or any other
+#GstFlowReturn dependent on downstream state.
+
+
+
+
+ the #GstPad to handle the event.
+
+
+
+ the parent of @pad. If the #GST_PAD_FLAG_NEED_PARENT
+ flag is set, @parent is guaranteed to be not-%NULL and remain valid
+ during the execution of this function.
+
+
+
+ the #GstEvent to handle.
+
+
+
+
+
+ Function signature to handle an event for the pad.
+
+ %TRUE if the pad could handle the event.
+
+
+
+
+ the #GstPad to handle the event.
+
+
+
+ the parent of @pad. If the #GST_PAD_FLAG_NEED_PARENT
+ flag is set, @parent is guaranteed to be not-%NULL and remain valid
+ during the execution of this function.
+
+
+
+ the #GstEvent to handle.
+
+
+
+
+
+ Pad state flags
+
+ is dataflow on a pad blocked
+
+
+ is pad flushing
+
+
+ is pad in EOS state
+
+
+ is pad currently blocking on a buffer or event
+
+
+ ensure that there is a parent object before calling
+ into the pad callbacks.
+
+
+ the pad should be reconfigured/renegotiated.
+ The flag has to be unset manually after
+ reconfiguration happened.
+
+
+ the pad has pending events
+
+
+ the pad is using fixed caps. This means that
+ once the caps are set on the pad, the default caps query function
+ will only return those caps.
+
+
+ the default event and query handler will forward
+ all events and queries to the internally linked pads
+ instead of discarding them.
+
+
+ the default query handler will forward
+ allocation queries to the internally linked pads
+ instead of discarding them.
+
+
+ the default query handler will forward
+ scheduling queries to the internally linked pads
+ instead of discarding them.
+
+
+ the default accept-caps handler will check
+ it the caps intersect the query-caps result instead
+ of checking for a subset. This is interesting for
+ parsers that can accept incompletely specified caps.
+
+
+ the default accept-caps handler will use
+ the template pad caps instead of query caps to
+ compare with the accept caps. Use this in combination
+ with %GST_PAD_FLAG_ACCEPT_INTERSECT. (Since 1.6)
+
+
+ offset to define more flags
+
+
+
+ A forward function is called for all internally linked pads, see
+gst_pad_forward().
+
+ %TRUE if the dispatching procedure has to be stopped.
+
+
+
+
+ the #GstPad that is forwarded.
+
+
+
+ the gpointer to optional user data.
+
+
+
+
+
+ This function will be called on source pads when a peer element
+request a buffer at the specified @offset and @length. If this function
+returns #GST_FLOW_OK, the result buffer will be stored in @buffer. The
+contents of @buffer is invalid for any other return value.
+
+This function is installed on a source pad with
+gst_pad_set_getrange_function() and can only be called on source pads after
+they are successfully activated with gst_pad_activate_mode() with the
+#GST_PAD_MODE_PULL.
+
+@offset and @length are always given in byte units. @offset must normally be a value
+between 0 and the length in bytes of the data available on @pad. The
+length (duration in bytes) can be retrieved with a #GST_QUERY_DURATION or with a
+#GST_QUERY_SEEKING.
+
+Any @offset larger or equal than the length will make the function return
+#GST_FLOW_EOS, which corresponds to EOS. In this case @buffer does not
+contain a valid buffer.
+
+The buffer size of @buffer will only be smaller than @length when @offset is
+near the end of the stream. In all other cases, the size of @buffer must be
+exactly the requested size.
+
+It is allowed to call this function with a 0 @length and valid @offset, in
+which case @buffer will contain a 0-sized buffer and the function returns
+#GST_FLOW_OK.
+
+When this function is called with a -1 @offset, the sequentially next buffer
+of length @length in the stream is returned.
+
+When this function is called with a -1 @length, a buffer with a default
+optimal length is returned in @buffer. The length might depend on the value
+of @offset.
+
+ #GST_FLOW_OK for success and a valid buffer in @buffer. Any other
+return value leaves @buffer undefined.
+
+
+
+
+ the src #GstPad to perform the getrange on.
+
+
+
+ the parent of @pad. If the #GST_PAD_FLAG_NEED_PARENT
+ flag is set, @parent is guaranteed to be not-%NULL and remain valid
+ during the execution of this function.
+
+
+
+ the offset of the range
+
+
+
+ the length of the range
+
+
+
+ a memory location to hold the result buffer, cannot be %NULL.
+
+
+
+
+
+ The signature of the internal pad link iterator function.
+
+ a new #GstIterator that will iterate over all pads that are
+linked to the given pad on the inside of the parent element.
+
+the caller must call gst_iterator_free() after usage.
+
+
+
+
+ The #GstPad to query.
+
+
+
+ the parent of @pad. If the #GST_PAD_FLAG_NEED_PARENT
+ flag is set, @parent is guaranteed to be not-%NULL and remain valid
+ during the execution of this function.
+
+
+
+
+
+ The amount of checking to be done when linking pads. @GST_PAD_LINK_CHECK_CAPS
+and @GST_PAD_LINK_CHECK_TEMPLATE_CAPS are mutually exclusive. If both are
+specified, expensive but safe @GST_PAD_LINK_CHECK_CAPS are performed.
+
+> Only disable some of the checks if you are 100% certain you know the link
+> will not fail because of hierarchy/caps compatibility failures. If uncertain,
+> use the default checks (%GST_PAD_LINK_CHECK_DEFAULT) or the regular methods
+> for linking the pads.
+
+ Don't check hierarchy or caps compatibility.
+
+
+ Check the pads have same parents/grandparents.
+ Could be omitted if it is already known that the two elements that own the
+ pads are in the same bin.
+
+
+ Check if the pads are compatible by using
+ their template caps. This is much faster than @GST_PAD_LINK_CHECK_CAPS, but
+ would be unsafe e.g. if one pad has %GST_CAPS_ANY.
+
+
+ Check if the pads are compatible by comparing the
+ caps returned by gst_pad_query_caps().
+
+
+ Disables pushing a reconfigure event when pads are
+ linked.
+
+
+ The default checks done when linking
+ pads (i.e. the ones used by gst_pad_link()).
+
+
+
+ Function signature to handle a new link on the pad.
+
+ the result of the link with the specified peer.
+
+
+
+
+ the #GstPad that is linked.
+
+
+
+ the parent of @pad. If the #GST_PAD_FLAG_NEED_PARENT
+ flag is set, @parent is guaranteed to be not-%NULL and remain valid
+ during the execution of this function.
+
+
+
+ the peer #GstPad of the link
+
+
+
+
+
+ Result values from gst_pad_link and friends.
+
+ link succeeded
+
+
+ pads have no common grandparent
+
+
+ pad was already linked
+
+
+ pads have wrong direction
+
+
+ pads do not have common format
+
+
+ pads cannot cooperate in scheduling
+
+
+ refused for some reason
+
+
+
+ The status of a GstPad. After activating a pad, which usually happens when the
+parent element goes from READY to PAUSED, the GstPadMode defines if the
+pad operates in push or pull mode.
+
+ Pad will not handle dataflow
+
+
+ Pad handles dataflow in downstream push mode
+
+
+ Pad handles dataflow in upstream pull mode
+
+
+ Return the name of a pad mode, for use in debug messages mostly.
+
+ short mnemonic for pad mode @mode
+
+
+
+
+ the pad mode
+
+
+
+
+
+
+ Indicates when this pad will become available.
+
+ the pad is always available
+
+
+ the pad will become available depending on the media stream
+
+
+ the pad is only available on request with
+ gst_element_request_pad().
+
+
+
+
+
+ Callback used by gst_pad_add_probe(). Gets called to notify about the current
+blocking type.
+
+The callback is allowed to modify the data pointer in @info.
+
+ a #GstPadProbeReturn
+
+
+
+
+ the #GstPad that is blocked
+
+
+
+ #GstPadProbeInfo
+
+
+
+ the gpointer to optional user data.
+
+
+
+
+
+ Info passed in the #GstPadProbeCallback.
+
+ the current probe type
+
+
+
+ the id of the probe
+
+
+
+ type specific data, check the @type field to know the
+ datatype. This field can be %NULL.
+
+
+
+ offset of pull probe, this field is valid when @type contains
+ #GST_PAD_PROBE_TYPE_PULL
+
+
+
+ size of pull probe, this field is valid when @type contains
+ #GST_PAD_PROBE_TYPE_PULL
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ The #GstBuffer from the probe
+
+
+
+
+ a #GstPadProbeInfo
+
+
+
+
+
+
+ The #GstBufferList from the probe
+
+
+
+
+ a #GstPadProbeInfo
+
+
+
+
+
+
+ The #GstEvent from the probe
+
+
+
+
+ a #GstPadProbeInfo
+
+
+
+
+
+
+ The #GstQuery from the probe
+
+
+
+
+ a #GstPadProbeInfo
+
+
+
+
+
+
+ Different return values for the #GstPadProbeCallback.
+
+ drop data in data probes. For push mode this means that
+ the data item is not sent downstream. For pull mode, it means that
+ the data item is not passed upstream. In both cases, no other probes
+ are called for this item and %GST_FLOW_OK or %TRUE is returned to the
+ caller.
+
+
+ normal probe return value. This leaves the probe in
+ place, and defers decisions about dropping or passing data to other
+ probes, if any. If there are no other probes, the default behaviour
+ for the probe type applies ('block' for blocking probes,
+ and 'pass' for non-blocking probes).
+
+
+ remove this probe.
+
+
+ pass the data item in the block probe and block on the
+ next item.
+
+
+ Data has been handled in the probe and will not be
+ forwarded further. For events and buffers this is the same behaviour as
+ %GST_PAD_PROBE_DROP (except that in this case you need to unref the buffer
+ or event yourself). For queries it will also return %TRUE to the caller.
+ The probe can also modify the #GstFlowReturn value by using the
+ #GST_PAD_PROBE_INFO_FLOW_RETURN() accessor.
+ Note that the resulting query must contain valid entries.
+ Since: 1.6
+
+
+
+ The different probing types that can occur. When either one of
+@GST_PAD_PROBE_TYPE_IDLE or @GST_PAD_PROBE_TYPE_BLOCK is used, the probe will be a
+blocking probe.
+
+ invalid probe type
+
+
+ probe idle pads and block while the callback is called
+
+
+ probe and block pads
+
+
+ probe buffers
+
+
+ probe buffer lists
+
+
+ probe downstream events
+
+
+ probe upstream events
+
+
+ probe flush events. This probe has to be
+ explicitly enabled and is not included in the
+ @@GST_PAD_PROBE_TYPE_EVENT_DOWNSTREAM or
+ @@GST_PAD_PROBE_TYPE_EVENT_UPSTREAM probe types.
+
+
+ probe downstream queries
+
+
+ probe upstream queries
+
+
+ probe push
+
+
+ probe pull
+
+
+ probe and block at the next opportunity, at data flow or when idle
+
+
+ probe downstream data (buffers, buffer lists, and events)
+
+
+ probe upstream data (events)
+
+
+ probe upstream and downstream data (buffers, buffer lists, and events)
+
+
+ probe and block downstream data (buffers, buffer lists, and events)
+
+
+ probe and block upstream data (events)
+
+
+ probe upstream and downstream events
+
+
+ probe upstream and downstream queries
+
+
+ probe upstream events and queries and downstream buffers, buffer lists, events and queries
+
+
+ probe push and pull
+
+
+
+ The signature of the query function.
+
+ %TRUE if the query could be performed.
+
+
+
+
+ the #GstPad to query.
+
+
+
+ the parent of @pad. If the #GST_PAD_FLAG_NEED_PARENT
+ flag is set, @parent is guaranteed to be not-%NULL and remain valid
+ during the execution of this function.
+
+
+
+ the #GstQuery object to execute
+
+
+
+
+
+ Callback used by gst_pad_sticky_events_foreach().
+
+When this function returns %TRUE, the next event will be
+returned. When %FALSE is returned, gst_pad_sticky_events_foreach() will return.
+
+When @event is set to %NULL, the item will be removed from the list of sticky events.
+@event can be replaced by assigning a new reference to it.
+This function is responsible for unreffing the old event when
+removing or modifying.
+
+ %TRUE if the iteration should continue
+
+
+
+
+ the #GstPad.
+
+
+
+ a sticky #GstEvent.
+
+
+
+ the #gpointer to optional user data.
+
+
+
+
+
+ Padtemplates describe the possible media types a pad or an elementfactory can
+handle. This allows for both inspection of handled types before loading the
+element plugin as well as identifying pads on elements that are not yet
+created (request or sometimes pads).
+
+Pad and PadTemplates have #GstCaps attached to it to describe the media type
+they are capable of dealing with. gst_pad_template_get_caps() or
+GST_PAD_TEMPLATE_CAPS() are used to get the caps of a padtemplate. It's not
+possible to modify the caps of a padtemplate after creation.
+
+PadTemplates have a #GstPadPresence property which identifies the lifetime
+of the pad and that can be retrieved with GST_PAD_TEMPLATE_PRESENCE(). Also
+the direction of the pad can be retrieved from the #GstPadTemplate with
+GST_PAD_TEMPLATE_DIRECTION().
+
+The GST_PAD_TEMPLATE_NAME_TEMPLATE () is important for GST_PAD_REQUEST pads
+because it has to be used as the name in the gst_element_get_request_pad()
+call to instantiate a pad from this template.
+
+Padtemplates can be created with gst_pad_template_new() or with
+gst_static_pad_template_get (), which creates a #GstPadTemplate from a
+#GstStaticPadTemplate that can be filled with the
+convenient GST_STATIC_PAD_TEMPLATE() macro.
+
+A padtemplate can be used to create a pad (see gst_pad_new_from_template()
+or gst_pad_new_from_static_template ()) or to add to an element class
+(see gst_element_class_add_static_pad_template ()).
+
+The following code example shows the code to create a pad from a padtemplate.
+|[<!-- language="C" -->
+ GstStaticPadTemplate my_template =
+ GST_STATIC_PAD_TEMPLATE (
+ "sink", // the name of the pad
+ GST_PAD_SINK, // the direction of the pad
+ GST_PAD_ALWAYS, // when this pad will be present
+ GST_STATIC_CAPS ( // the capabilities of the padtemplate
+ "audio/x-raw, "
+ "channels = (int) [ 1, 6 ]"
+ )
+ );
+ void
+ my_method (void)
+ {
+ GstPad *pad;
+ pad = gst_pad_new_from_static_template (&my_template, "sink");
+ ...
+ }
+]|
+
+The following example shows you how to add the padtemplate to an
+element class, this is usually done in the class_init of the class:
+|[<!-- language="C" -->
+ static void
+ my_element_class_init (GstMyElementClass *klass)
+ {
+ GstElementClass *gstelement_class = GST_ELEMENT_CLASS (klass);
+
+ gst_element_class_add_static_pad_template (gstelement_class, &my_template);
+ }
+]|
+
+ Creates a new pad template with a name according to the given template
+and with the given arguments.
+
+ a new #GstPadTemplate.
+
+
+
+
+ the name template.
+
+
+
+ the #GstPadDirection of the template.
+
+
+
+ the #GstPadPresence of the pad.
+
+
+
+ a #GstCaps set for the template.
+
+
+
+
+
+ Converts a #GstStaticPadTemplate into a #GstPadTemplate with a type.
+
+ a new #GstPadTemplate.
+
+
+
+
+ the static pad template
+
+
+
+ The #GType of the pad to create
+
+
+
+
+
+ Creates a new pad template with a name according to the given template
+and with the given arguments.
+
+ a new #GstPadTemplate.
+
+
+
+
+ the name template.
+
+
+
+ the #GstPadDirection of the template.
+
+
+
+ the #GstPadPresence of the pad.
+
+
+
+ a #GstCaps set for the template.
+
+
+
+ The #GType of the pad to create
+
+
+
+
+
+ Emit the pad-created signal for this template when created by this pad.
+
+
+
+
+
+ a #GstPadTemplate that has been created
+
+
+
+ the #GstPad that created it
+
+
+
+
+
+ Gets the capabilities of the pad template.
+
+ the #GstCaps of the pad template.
+Unref after usage.
+
+
+
+
+ a #GstPadTemplate to get capabilities of.
+
+
+
+
+
+ Emit the pad-created signal for this template when created by this pad.
+
+
+
+
+
+ a #GstPadTemplate that has been created
+
+
+
+ the #GstPad that created it
+
+
+
+
+
+ The capabilities of the pad described by the pad template.
+
+
+
+ The direction of the pad described by the pad template.
+
+
+
+ The type of the pad described by the pad template.
+
+
+
+ The name template of the pad template.
+
+
+
+ When the pad described by the pad template will become available.
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ This signal is fired when an element creates a pad from this template.
+
+
+
+
+
+ the pad that was created.
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ a #GstPadTemplate that has been created
+
+
+
+ the #GstPad that created it
+
+
+
+
+
+
+
+
+
+
+
+
+ Flags for the padtemplate
+
+ first flag that can be used by subclasses.
+
+
+
+ Function signature to handle a unlinking the pad prom its peer.
+
+
+
+
+
+ the #GstPad that is linked.
+
+
+
+ the parent of @pad. If the #GST_PAD_FLAG_NEED_PARENT
+ flag is set, @parent is guaranteed to be not-%NULL and remain valid
+ during the execution of this function.
+
+
+
+
+
+
+
+
+
+ A GParamSpec derived structure that contains the meta data for fractional
+properties.
+
+ super class
+
+
+
+
+
+
+
+ A GParamSpec derived structure that contains the meta data for fractional
+properties.
+
+ super class
+
+
+
+ minimal numerator
+
+
+
+ minimal denominator
+
+
+
+ maximal numerator
+
+
+
+ maximal denominator
+
+
+
+ default numerator
+
+
+
+ default denominator
+
+
+
+
+ The #GstParentBufferMeta is a #GstMeta which can be attached to a #GstBuffer
+to hold a reference to another buffer that is only released when the child
+#GstBuffer is released.
+
+Typically, #GstParentBufferMeta is used when the child buffer is directly
+using the #GstMemory of the parent buffer, and wants to prevent the parent
+buffer from being returned to a buffer pool until the #GstMemory is available
+for re-use.
+
+ the parent #GstMeta structure
+
+
+
+ the #GstBuffer on which a reference is being held.
+
+
+
+ Get the global #GstMetaInfo describing the #GstParentBufferMeta meta.
+
+ The #GstMetaInfo
+
+
+
+
+
+ Opaque structure.
+
+ Allocates a parse context for use with gst_parse_launch_full() or
+gst_parse_launchv_full().
+
+Free-function: gst_parse_context_free
+
+ a newly-allocated parse context. Free
+ with gst_parse_context_free() when no longer needed.
+
+
+
+
+ Copies the @context.
+
+ A copied #GstParseContext
+
+
+
+
+ a #GstParseContext
+
+
+
+
+
+ Frees a parse context previously allocated with gst_parse_context_new().
+
+
+
+
+
+ a #GstParseContext
+
+
+
+
+
+ Retrieve missing elements from a previous run of gst_parse_launch_full()
+or gst_parse_launchv_full(). Will only return results if an error code
+of %GST_PARSE_ERROR_NO_SUCH_ELEMENT was returned.
+
+ a
+ %NULL-terminated array of element factory name strings of missing
+ elements. Free with g_strfreev() when no longer needed.
+
+
+
+
+
+
+ a #GstParseContext
+
+
+
+
+
+
+ The different parsing errors that can occur.
+
+ A syntax error occurred.
+
+
+ The description contained an unknown element
+
+
+ An element did not have a specified property
+
+
+ There was an error linking two pads.
+
+
+ There was an error setting a property
+
+
+ An empty bin was specified.
+
+
+ An empty description was specified
+
+
+ A delayed link did not get resolved.
+
+
+ Get the error quark used by the parsing subsystem.
+
+ the quark of the parse errors.
+
+
+
+
+
+ Parsing options.
+
+ Do not use any special parsing options.
+
+
+ Always return %NULL when an error occurs
+ (default behaviour is to return partially constructed bins or elements
+ in some cases)
+
+
+ If a bin only has a single element,
+ just return the element.
+
+
+ If more than one toplevel element is described
+ by the pipeline description string, put them in a #GstBin instead of a
+ #GstPipeline. (Since 1.10)
+
+
+
+ A #GstPipeline is a special #GstBin used as the toplevel container for
+the filter graph. The #GstPipeline will manage the selection and
+distribution of a global #GstClock as well as provide a #GstBus to the
+application.
+
+gst_pipeline_new() is used to create a pipeline. when you are done with
+the pipeline, use gst_object_unref() to free its resources including all
+added #GstElement objects (if not otherwise referenced).
+
+Elements are added and removed from the pipeline using the #GstBin
+methods like gst_bin_add() and gst_bin_remove() (see #GstBin).
+
+Before changing the state of the #GstPipeline (see #GstElement) a #GstBus
+can be retrieved with gst_pipeline_get_bus(). This bus can then be
+used to receive #GstMessage from the elements in the pipeline.
+
+By default, a #GstPipeline will automatically flush the pending #GstBus
+messages when going to the NULL state to ensure that no circular
+references exist when no messages are read from the #GstBus. This
+behaviour can be changed with gst_pipeline_set_auto_flush_bus().
+
+When the #GstPipeline performs the PAUSED to PLAYING state change it will
+select a clock for the elements. The clock selection algorithm will by
+default select a clock provided by an element that is most upstream
+(closest to the source). For live pipelines (ones that return
+#GST_STATE_CHANGE_NO_PREROLL from the gst_element_set_state() call) this
+will select the clock provided by the live source. For normal pipelines
+this will select a clock provided by the sinks (most likely the audio
+sink). If no element provides a clock, a default #GstSystemClock is used.
+
+The clock selection can be controlled with the gst_pipeline_use_clock()
+method, which will enforce a given clock on the pipeline. With
+gst_pipeline_auto_clock() the default clock selection algorithm can be
+restored.
+
+A #GstPipeline maintains a running time for the elements. The running
+time is defined as the difference between the current clock time and
+the base time. When the pipeline goes to READY or a flushing seek is
+performed on it, the running time is reset to 0. When the pipeline is
+set from PLAYING to PAUSED, the current clock time is sampled and used to
+configure the base time for the elements when the pipeline is set
+to PLAYING again. The effect is that the running time (as the difference
+between the clock time and the base time) will count how much time was spent
+in the PLAYING state. This default behaviour can be changed with the
+gst_element_set_start_time() method.
+
+
+ Create a new pipeline with the given name.
+
+ newly created GstPipeline
+
+MT safe.
+
+
+
+
+ name of new pipeline
+
+
+
+
+
+ Let @pipeline select a clock automatically. This is the default
+behaviour.
+
+Use this function if you previous forced a fixed clock with
+gst_pipeline_use_clock() and want to restore the default
+pipeline clock selection algorithm.
+
+MT safe.
+
+
+
+
+
+ a #GstPipeline
+
+
+
+
+
+ Check if @pipeline will automatically flush messages when going to
+the NULL state.
+
+ whether the pipeline will automatically flush its bus when
+going from READY to NULL state or not.
+
+MT safe.
+
+
+
+
+ a #GstPipeline
+
+
+
+
+
+ Gets the #GstBus of @pipeline. The bus allows applications to receive
+#GstMessage packets.
+
+ a #GstBus, unref after usage.
+
+MT safe.
+
+
+
+
+ a #GstPipeline
+
+
+
+
+
+ Gets the current clock used by @pipeline. Users of object
+oriented languages should use gst_pipeline_get_pipeline_clock()
+to avoid confusion with gst_element_get_clock() which has a different behavior.
+
+Unlike gst_element_get_clock(), this function will always return a
+clock, even if the pipeline is not in the PLAYING state.
+
+ a #GstClock, unref after usage.
+
+
+
+
+ a #GstPipeline
+
+
+
+
+
+ Get the configured delay (see gst_pipeline_set_delay()).
+
+ The configured delay.
+
+MT safe.
+
+
+
+
+ a #GstPipeline
+
+
+
+
+
+ Gets the latency that should be configured on the pipeline. See
+gst_pipeline_set_latency().
+
+ Latency to configure on the pipeline or GST_CLOCK_TIME_NONE
+
+
+
+
+ a #GstPipeline
+
+
+
+
+
+ Gets the current clock used by @pipeline.
+
+Unlike gst_element_get_clock(), this function will always return a
+clock, even if the pipeline is not in the PLAYING state.
+
+ a #GstClock, unref after usage.
+
+
+
+
+ a #GstPipeline
+
+
+
+
+
+ Usually, when a pipeline goes from READY to NULL state, it automatically
+flushes all pending messages on the bus, which is done for refcounting
+purposes, to break circular references.
+
+This means that applications that update state using (async) bus messages
+(e.g. do certain things when a pipeline goes from PAUSED to READY) might
+not get to see messages when the pipeline is shut down, because they might
+be flushed before they can be dispatched in the main thread. This behaviour
+can be disabled using this function.
+
+It is important that all messages on the bus are handled when the
+automatic flushing is disabled else memory leaks will be introduced.
+
+MT safe.
+
+
+
+
+
+ a #GstPipeline
+
+
+
+ whether or not to automatically flush the bus when
+the pipeline goes from READY to NULL state
+
+
+
+
+
+ Set the clock for @pipeline. The clock will be distributed
+to all the elements managed by the pipeline.
+
+ %TRUE if the clock could be set on the pipeline. %FALSE if
+ some element did not accept the clock.
+
+MT safe.
+
+
+
+
+ a #GstPipeline
+
+
+
+ the clock to set
+
+
+
+
+
+ Set the expected delay needed for all elements to perform the
+PAUSED to PLAYING state change. @delay will be added to the
+base time of the elements so that they wait an additional @delay
+amount of time before starting to process buffers and cannot be
+#GST_CLOCK_TIME_NONE.
+
+This option is used for tuning purposes and should normally not be
+used.
+
+MT safe.
+
+
+
+
+
+ a #GstPipeline
+
+
+
+ the delay
+
+
+
+
+
+ Sets the latency that should be configured on the pipeline. Setting
+GST_CLOCK_TIME_NONE will restore the default behaviour of using the minimum
+latency from the LATENCY query. Setting this is usually not required and
+the pipeline will figure out an appropriate latency automatically.
+
+Setting a too low latency, especially lower than the minimum latency from
+the LATENCY query, will most likely cause the pipeline to fail.
+
+
+
+
+
+ a #GstPipeline
+
+
+
+ latency to configure
+
+
+
+
+
+ Force @pipeline to use the given @clock. The pipeline will
+always use the given clock even if new clock providers are added
+to this pipeline.
+
+If @clock is %NULL all clocking will be disabled which will make
+the pipeline run as fast as possible.
+
+MT safe.
+
+
+
+
+
+ a #GstPipeline
+
+
+
+ the clock to use
+
+
+
+
+
+ Whether or not to automatically flush all messages on the
+pipeline's bus when going from READY to NULL state. Please see
+gst_pipeline_set_auto_flush_bus() for more information on this option.
+
+
+
+ The expected delay needed for elements to spin up to the
+PLAYING state expressed in nanoseconds.
+see gst_pipeline_set_delay() for more information on this option.
+
+
+
+ Latency to configure on the pipeline. See gst_pipeline_set_latency().
+
+
+
+
+
+
+ The fixed clock of the pipeline, used when
+ GST_PIPELINE_FLAG_FIXED_CLOCK is set.
+
+
+
+ The stream time of the pipeline. A better name for this
+ property would be the running_time, the total time spent in the
+ PLAYING state without being flushed. (deprecated, use the start_time
+ on GstElement).
+
+
+
+ Extra delay added to base_time to compensate for computing delays
+ when setting elements to PLAYING.
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Pipeline flags
+
+ this pipeline works with a fixed clock
+
+
+ offset to define more flags
+
+
+
+
+
+ GStreamer is extensible, so #GstElement instances can be loaded at runtime.
+A plugin system can provide one or more of the basic
+<application>GStreamer</application> #GstPluginFeature subclasses.
+
+A plugin should export a symbol <symbol>gst_plugin_desc</symbol> that is a
+struct of type #GstPluginDesc.
+the plugin loader will check the version of the core library the plugin was
+linked against and will create a new #GstPlugin. It will then call the
+#GstPluginInitFunc function that was provided in the
+<symbol>gst_plugin_desc</symbol>.
+
+Once you have a handle to a #GstPlugin (e.g. from the #GstRegistry), you
+can add any object that subclasses #GstPluginFeature.
+
+Usually plugins are always automatically loaded so you don't need to call
+gst_plugin_load() explicitly to bring it into memory. There are options to
+statically link plugins to an app or even use GStreamer without a plugin
+repository in which case gst_plugin_load() can be needed to bring the plugin
+into memory.
+
+ Unrefs each member of @list, then frees the list.
+
+
+
+
+
+ list of #GstPlugin
+
+
+
+
+
+
+
+ Load the named plugin. Refs the plugin.
+
+ a reference to a loaded plugin, or
+%NULL on error.
+
+
+
+
+ name of plugin to load
+
+
+
+
+
+ Loads the given plugin and refs it. Caller needs to unref after use.
+
+ a reference to the existing loaded GstPlugin, a
+reference to the newly-loaded GstPlugin, or %NULL if an error occurred.
+
+
+
+
+ the plugin filename to load
+
+
+
+
+
+ Registers a static plugin, ie. a plugin which is private to an application
+or library and contained within the application or library (as opposed to
+being shipped as a separate module file).
+
+You must make sure that GStreamer has been initialised (with gst_init() or
+via gst_init_get_option_group()) before calling this function.
+
+ %TRUE if the plugin was registered correctly, otherwise %FALSE.
+
+
+
+
+ the major version number of the GStreamer core that the
+ plugin was compiled for, you can just use GST_VERSION_MAJOR here
+
+
+
+ the minor version number of the GStreamer core that the
+ plugin was compiled for, you can just use GST_VERSION_MINOR here
+
+
+
+ a unique name of the plugin (ideally prefixed with an application- or
+ library-specific namespace prefix in order to avoid name conflicts in
+ case a similar plugin with the same name ever gets added to GStreamer)
+
+
+
+ description of the plugin
+
+
+
+ pointer to the init function of this plugin.
+
+
+
+ version string of the plugin
+
+
+
+ effective license of plugin. Must be one of the approved licenses
+ (see #GstPluginDesc above) or the plugin will not be registered.
+
+
+
+ source module plugin belongs to
+
+
+
+ shipped package plugin belongs to
+
+
+
+ URL to provider of plugin
+
+
+
+
+
+ Registers a static plugin, ie. a plugin which is private to an application
+or library and contained within the application or library (as opposed to
+being shipped as a separate module file) with a #GstPluginInitFullFunc
+which allows user data to be passed to the callback function (useful
+for bindings).
+
+You must make sure that GStreamer has been initialised (with gst_init() or
+via gst_init_get_option_group()) before calling this function.
+
+ %TRUE if the plugin was registered correctly, otherwise %FALSE.
+
+
+
+
+ the major version number of the GStreamer core that the
+ plugin was compiled for, you can just use GST_VERSION_MAJOR here
+
+
+
+ the minor version number of the GStreamer core that the
+ plugin was compiled for, you can just use GST_VERSION_MINOR here
+
+
+
+ a unique name of the plugin (ideally prefixed with an application- or
+ library-specific namespace prefix in order to avoid name conflicts in
+ case a similar plugin with the same name ever gets added to GStreamer)
+
+
+
+ description of the plugin
+
+
+
+ pointer to the init function with user data
+ of this plugin.
+
+
+
+ version string of the plugin
+
+
+
+ effective license of plugin. Must be one of the approved licenses
+ (see #GstPluginDesc above) or the plugin will not be registered.
+
+
+
+ source module plugin belongs to
+
+
+
+ shipped package plugin belongs to
+
+
+
+ URL to provider of plugin
+
+
+
+ gpointer to user data
+
+
+
+
+
+ Make GStreamer aware of external dependencies which affect the feature
+set of this plugin (ie. the elements or typefinders associated with it).
+
+GStreamer will re-inspect plugins with external dependencies whenever any
+of the external dependencies change. This is useful for plugins which wrap
+other plugin systems, e.g. a plugin which wraps a plugin-based visualisation
+library and makes visualisations available as GStreamer elements, or a
+codec loader which exposes elements and/or caps dependent on what external
+codec libraries are currently installed.
+
+
+
+
+
+ a #GstPlugin
+
+
+
+ %NULL-terminated array of environment variables affecting the
+ feature set of the plugin (e.g. an environment variable containing
+ paths where to look for additional modules/plugins of a library),
+ or %NULL. Environment variable names may be followed by a path component
+ which will be added to the content of the environment variable, e.g.
+ "HOME/.mystuff/plugins".
+
+
+
+
+
+ %NULL-terminated array of directories/paths where dependent files
+ may be, or %NULL.
+
+
+
+
+
+ %NULL-terminated array of file names (or file name suffixes,
+ depending on @flags) to be used in combination with the paths from
+ @paths and/or the paths extracted from the environment variables in
+ @env_vars, or %NULL.
+
+
+
+
+
+ optional flags, or #GST_PLUGIN_DEPENDENCY_FLAG_NONE
+
+
+
+
+
+ Make GStreamer aware of external dependencies which affect the feature
+set of this plugin (ie. the elements or typefinders associated with it).
+
+GStreamer will re-inspect plugins with external dependencies whenever any
+of the external dependencies change. This is useful for plugins which wrap
+other plugin systems, e.g. a plugin which wraps a plugin-based visualisation
+library and makes visualisations available as GStreamer elements, or a
+codec loader which exposes elements and/or caps dependent on what external
+codec libraries are currently installed.
+
+Convenience wrapper function for gst_plugin_add_dependency() which
+takes simple strings as arguments instead of string arrays, with multiple
+arguments separated by predefined delimiters (see above).
+
+
+
+
+
+ the #GstPlugin
+
+
+
+ one or more environment variables (separated by ':', ';' or ','),
+ or %NULL. Environment variable names may be followed by a path component
+ which will be added to the content of the environment variable, e.g.
+ "HOME/.mystuff/plugins:MYSTUFF_PLUGINS_PATH"
+
+
+
+ one ore more directory paths (separated by ':' or ';' or ','),
+ or %NULL. Example: "/usr/lib/mystuff/plugins"
+
+
+
+ one or more file names or file name suffixes (separated by commas),
+ or %NULL
+
+
+
+ optional flags, or #GST_PLUGIN_DEPENDENCY_FLAG_NONE
+
+
+
+
+
+ Gets the plugin specific data cache. If it is %NULL there is no cached data
+stored. This is the case when the registry is getting rebuilt.
+
+ The cached data as a
+#GstStructure or %NULL.
+
+
+
+
+ a plugin
+
+
+
+
+
+ Get the long descriptive name of the plugin
+
+ the long name of the plugin
+
+
+
+
+ plugin to get long name of
+
+
+
+
+
+ get the filename of the plugin
+
+ the filename of the plugin
+
+
+
+
+ plugin to get the filename of
+
+
+
+
+
+ get the license of the plugin
+
+ the license of the plugin
+
+
+
+
+ plugin to get the license of
+
+
+
+
+
+ Get the short name of the plugin
+
+ the name of the plugin
+
+
+
+
+ plugin to get the name of
+
+
+
+
+
+ get the URL where the plugin comes from
+
+ the origin of the plugin
+
+
+
+
+ plugin to get the origin of
+
+
+
+
+
+ get the package the plugin belongs to.
+
+ the package of the plugin
+
+
+
+
+ plugin to get the package of
+
+
+
+
+
+ Get the release date (and possibly time) in form of a string, if available.
+
+For normal GStreamer plugin releases this will usually just be a date in
+the form of "YYYY-MM-DD", while pre-releases and builds from git may contain
+a time component after the date as well, in which case the string will be
+formatted like "YYYY-MM-DDTHH:MMZ" (e.g. "2012-04-30T09:30Z").
+
+There may be plugins that do not have a valid release date set on them.
+
+ the date string of the plugin, or %NULL if not
+available.
+
+
+
+
+ plugin to get the release date of
+
+
+
+
+
+ get the source module the plugin belongs to.
+
+ the source of the plugin
+
+
+
+
+ plugin to get the source of
+
+
+
+
+
+ get the version of the plugin
+
+ the version of the plugin
+
+
+
+
+ plugin to get the version of
+
+
+
+
+
+ queries if the plugin is loaded into memory
+
+ %TRUE is loaded, %FALSE otherwise
+
+
+
+
+ plugin to query
+
+
+
+
+
+ Loads @plugin. Note that the *return value* is the loaded plugin; @plugin is
+untouched. The normal use pattern of this function goes like this:
+
+|[
+GstPlugin *loaded_plugin;
+loaded_plugin = gst_plugin_load (plugin);
+// presumably, we're no longer interested in the potentially-unloaded plugin
+gst_object_unref (plugin);
+plugin = loaded_plugin;
+]|
+
+ a reference to a loaded plugin, or
+%NULL on error.
+
+
+
+
+ plugin to load
+
+
+
+
+
+ Adds plugin specific data to cache. Passes the ownership of the structure to
+the @plugin.
+
+The cache is flushed every time the registry is rebuilt.
+
+
+
+
+
+ a plugin
+
+
+
+ a structure containing the data to cache
+
+
+
+
+
+
+
+
+ Flags used in connection with gst_plugin_add_dependency().
+
+ no special flags
+
+
+ recurse into subdirectories
+
+
+ use paths
+ argument only if none of the environment variables is set
+
+
+ interpret
+ filename argument as filter suffix and check all matching files in
+ the directory
+
+
+ interpret
+ filename argument as filter prefix and check all matching files in
+ the directory. Since 1.8.
+
+
+ interpret
+ non-absolute paths as relative to the main executable directory. Since
+ 1.14.
+
+
+
+ A plugin should export a variable of this type called plugin_desc. The plugin
+loader will use the data provided there to initialize the plugin.
+
+The @licence parameter must be one of: LGPL, GPL, QPL, GPL/QPL, MPL,
+BSD, MIT/X11, Proprietary, unknown.
+
+ the major version number of core that plugin was compiled for
+
+
+
+ the minor version number of core that plugin was compiled for
+
+
+
+ a unique name of the plugin
+
+
+
+ description of plugin
+
+
+
+ pointer to the init function of this plugin.
+
+
+
+ version of the plugin
+
+
+
+ effective license of plugin
+
+
+
+ source module plugin belongs to
+
+
+
+ shipped package plugin belongs to
+
+
+
+ URL to provider of plugin
+
+
+
+ date time string in ISO 8601
+ format (or rather, a subset thereof), or %NULL. Allowed are the
+ following formats: "YYYY-MM-DD" and "YYY-MM-DDTHH:MMZ" (with
+ 'T' a separator and 'Z' indicating UTC/Zulu time). This field
+ should be set via the GST_PACKAGE_RELEASE_DATETIME
+ preprocessor macro.
+
+
+
+
+
+
+
+
+
+ The plugin loading errors
+
+ The plugin could not be loaded
+
+
+ The plugin has unresolved dependencies
+
+
+ The plugin has already be loaded from a different file
+
+
+ Get the error quark.
+
+ The error quark used in GError messages
+
+
+
+
+
+ This is a base class for anything that can be added to a #GstPlugin.
+
+ Copies the list of features. Caller should call @gst_plugin_feature_list_free
+when done with the list.
+
+ a copy of @list,
+ with each feature's reference count incremented.
+
+
+
+
+
+
+ list
+ of #GstPluginFeature
+
+
+
+
+
+
+
+ Debug the plugin feature names in @list.
+
+
+
+
+
+ a #GList of
+ plugin features
+
+
+
+
+
+
+
+ Unrefs each member of @list, then frees the list.
+
+
+
+
+
+ list
+ of #GstPluginFeature
+
+
+
+
+
+
+
+ Compares the two given #GstPluginFeature instances. This function can be
+used as a #GCompareFunc when sorting by rank and then by name.
+
+ negative value if the rank of p1 > the rank of p2 or the ranks are
+equal but the name of p1 comes before the name of p2; zero if the rank
+and names are equal; positive value if the rank of p1 < the rank of p2 or the
+ranks are equal but the name of p2 comes before the name of p1
+
+
+
+
+ a #GstPluginFeature
+
+
+
+ a #GstPluginFeature
+
+
+
+
+
+ Checks whether the given plugin feature is at least
+ the required version
+
+ %TRUE if the plugin feature has at least
+ the required version, otherwise %FALSE.
+
+
+
+
+ a feature
+
+
+
+ minimum required major version
+
+
+
+ minimum required minor version
+
+
+
+ minimum required micro version
+
+
+
+
+
+ Get the plugin that provides this feature.
+
+ the plugin that provides this
+ feature, or %NULL. Unref with gst_object_unref() when no
+ longer needed.
+
+
+
+
+ a feature
+
+
+
+
+
+ Get the name of the plugin that provides this feature.
+
+ the name of the plugin that provides this
+ feature, or %NULL if the feature is not associated with a
+ plugin.
+
+
+
+
+ a feature
+
+
+
+
+
+ Gets the rank of a plugin feature.
+
+ The rank of the feature
+
+
+
+
+ a feature
+
+
+
+
+
+ Loads the plugin containing @feature if it's not already loaded. @feature is
+unaffected; use the return value instead.
+
+Normally this function is used like this:
+|[<!-- language="C" -->
+GstPluginFeature *loaded_feature;
+
+loaded_feature = gst_plugin_feature_load (feature);
+// presumably, we're no longer interested in the potentially-unloaded feature
+gst_object_unref (feature);
+feature = loaded_feature;
+]|
+
+ a reference to the loaded
+feature, or %NULL on error
+
+
+
+
+ the plugin feature to check
+
+
+
+
+
+ Specifies a rank for a plugin feature, so that autoplugging uses
+the most appropriate feature.
+
+
+
+
+
+ feature to rank
+
+
+
+ rank value - higher number means more priority rank
+
+
+
+
+
+
+
+
+ A function that can be used with e.g. gst_registry_feature_filter()
+to get a list of pluginfeature that match certain criteria.
+
+ %TRUE for a positive match, %FALSE otherwise
+
+
+
+
+ the pluginfeature to check
+
+
+
+ the user_data that has been passed on e.g.
+ gst_registry_feature_filter()
+
+
+
+
+
+ A function that can be used with e.g. gst_registry_plugin_filter()
+to get a list of plugins that match certain criteria.
+
+ %TRUE for a positive match, %FALSE otherwise
+
+
+
+
+ the plugin to check
+
+
+
+ the user_data that has been passed on e.g. gst_registry_plugin_filter()
+
+
+
+
+
+ The plugin loading state
+
+ Temporarily loaded plugins
+
+
+ The plugin won't be scanned (again)
+
+
+
+ A plugin should provide a pointer to a function of either #GstPluginInitFunc
+or this type in the plugin_desc struct.
+The function will be called by the loader at startup. One would then
+register each #GstPluginFeature. This version allows
+user data to be passed to init function (useful for bindings).
+
+ %TRUE if plugin initialised successfully
+
+
+
+
+ The plugin object
+
+
+
+ extra data
+
+
+
+
+
+ A plugin should provide a pointer to a function of this type in the
+plugin_desc struct.
+This function will be called by the loader at startup. One would then
+register each #GstPluginFeature.
+
+ %TRUE if plugin initialised successfully
+
+
+
+
+ The plugin object
+
+
+
+
+
+ A #GstPoll keeps track of file descriptors much like fd_set (used with
+select()) or a struct pollfd array (used with poll()). Once created with
+gst_poll_new(), the set can be used to wait for file descriptors to be
+readable and/or writable. It is possible to make this wait be controlled
+by specifying %TRUE for the @controllable flag when creating the set (or
+later calling gst_poll_set_controllable()).
+
+New file descriptors are added to the set using gst_poll_add_fd(), and
+removed using gst_poll_remove_fd(). Controlling which file descriptors
+should be waited for to become readable and/or writable are done using
+gst_poll_fd_ctl_read() and gst_poll_fd_ctl_write().
+
+Use gst_poll_wait() to wait for the file descriptors to actually become
+readable and/or writable, or to timeout if no file descriptor is available
+in time. The wait can be controlled by calling gst_poll_restart() and
+gst_poll_set_flushing().
+
+Once the file descriptor set has been waited for, one can use
+gst_poll_fd_has_closed() to see if the file descriptor has been closed,
+gst_poll_fd_has_error() to see if it has generated an error,
+gst_poll_fd_can_read() to see if it is possible to read from the file
+descriptor, and gst_poll_fd_can_write() to see if it is possible to
+write to it.
+
+ Add a file descriptor to the file descriptor set.
+
+ %TRUE if the file descriptor was successfully added to the set.
+
+
+
+
+ a file descriptor set.
+
+
+
+ a file descriptor.
+
+
+
+
+
+ Check if @fd in @set has data to be read.
+
+ %TRUE if the descriptor has data to be read.
+
+
+
+
+ a file descriptor set.
+
+
+
+ a file descriptor.
+
+
+
+
+
+ Check if @fd in @set can be used for writing.
+
+ %TRUE if the descriptor can be used for writing.
+
+
+
+
+ a file descriptor set.
+
+
+
+ a file descriptor.
+
+
+
+
+
+ Control whether the descriptor @fd in @set will be monitored for
+readability.
+
+ %TRUE if the descriptor was successfully updated.
+
+
+
+
+ a file descriptor set.
+
+
+
+ a file descriptor.
+
+
+
+ a new status.
+
+
+
+
+
+ Control whether the descriptor @fd in @set will be monitored for
+writability.
+
+ %TRUE if the descriptor was successfully updated.
+
+
+
+
+ a file descriptor set.
+
+
+
+ a file descriptor.
+
+
+
+ a new status.
+
+
+
+
+
+ Check if @fd in @set has closed the connection.
+
+ %TRUE if the connection was closed.
+
+
+
+
+ a file descriptor set.
+
+
+
+ a file descriptor.
+
+
+
+
+
+ Check if @fd in @set has an error.
+
+ %TRUE if the descriptor has an error.
+
+
+
+
+ a file descriptor set.
+
+
+
+ a file descriptor.
+
+
+
+
+
+ Mark @fd as ignored so that the next call to gst_poll_wait() will yield
+the same result for @fd as last time. This function must be called if no
+operation (read/write/recv/send/etc.) will be performed on @fd before
+the next call to gst_poll_wait().
+
+The reason why this is needed is because the underlying implementation
+might not allow querying the fd more than once between calls to one of
+the re-enabling operations.
+
+
+
+
+
+ a file descriptor set.
+
+
+
+ a file descriptor.
+
+
+
+
+
+ Free a file descriptor set.
+
+
+
+
+
+ a file descriptor set.
+
+
+
+
+
+ Get a GPollFD for the reading part of the control socket. This is useful when
+integrating with a GSource and GMainLoop.
+
+
+
+
+
+ a #GstPoll
+
+
+
+ a #GPollFD
+
+
+
+
+
+ Read a byte from the control socket of the controllable @set.
+
+This function only works for timer #GstPoll objects created with
+gst_poll_new_timer().
+
+ %TRUE on success. %FALSE when when there was no byte to read or
+reading the byte failed. If there was no byte to read, and only then, errno
+will contain EWOULDBLOCK or EAGAIN. For all other values of errno this always signals a
+critical error.
+
+
+
+
+ a #GstPoll.
+
+
+
+
+
+ Remove a file descriptor from the file descriptor set.
+
+ %TRUE if the file descriptor was successfully removed from the set.
+
+
+
+
+ a file descriptor set.
+
+
+
+ a file descriptor.
+
+
+
+
+
+ Restart any gst_poll_wait() that is in progress. This function is typically
+used after adding or removing descriptors to @set.
+
+If @set is not controllable, then this call will have no effect.
+
+This function only works for non-timer #GstPoll objects created with
+gst_poll_new().
+
+
+
+
+
+ a #GstPoll.
+
+
+
+
+
+ When @controllable is %TRUE, this function ensures that future calls to
+gst_poll_wait() will be affected by gst_poll_restart() and
+gst_poll_set_flushing().
+
+This function only works for non-timer #GstPoll objects created with
+gst_poll_new().
+
+ %TRUE if the controllability of @set could be updated.
+
+
+
+
+ a #GstPoll.
+
+
+
+ new controllable state.
+
+
+
+
+
+ When @flushing is %TRUE, this function ensures that current and future calls
+to gst_poll_wait() will return -1, with errno set to EBUSY.
+
+Unsetting the flushing state will restore normal operation of @set.
+
+This function only works for non-timer #GstPoll objects created with
+gst_poll_new().
+
+
+
+
+
+ a #GstPoll.
+
+
+
+ new flushing state.
+
+
+
+
+
+ Wait for activity on the file descriptors in @set. This function waits up to
+the specified @timeout. A timeout of #GST_CLOCK_TIME_NONE waits forever.
+
+For #GstPoll objects created with gst_poll_new(), this function can only be
+called from a single thread at a time. If called from multiple threads,
+-1 will be returned with errno set to EPERM.
+
+This is not true for timer #GstPoll objects created with
+gst_poll_new_timer(), where it is allowed to have multiple threads waiting
+simultaneously.
+
+ The number of #GstPollFD in @set that have activity or 0 when no
+activity was detected after @timeout. If an error occurs, -1 is returned
+and errno is set.
+
+
+
+
+ a #GstPoll.
+
+
+
+ a timeout in nanoseconds.
+
+
+
+
+
+ Write a byte to the control socket of the controllable @set.
+This function is mostly useful for timer #GstPoll objects created with
+gst_poll_new_timer().
+
+It will make any current and future gst_poll_wait() function return with
+1, meaning the control socket is set. After an equal amount of calls to
+gst_poll_read_control() have been performed, calls to gst_poll_wait() will
+block again until their timeout expired.
+
+This function only works for timer #GstPoll objects created with
+gst_poll_new_timer().
+
+ %TRUE on success. %FALSE when when the byte could not be written.
+errno contains the detailed error code but will never be EAGAIN, EINTR or
+EWOULDBLOCK. %FALSE always signals a critical error.
+
+
+
+
+ a #GstPoll.
+
+
+
+
+
+ Create a new file descriptor set. If @controllable, it
+is possible to restart or flush a call to gst_poll_wait() with
+gst_poll_restart() and gst_poll_set_flushing() respectively.
+
+Free-function: gst_poll_free
+
+ a new #GstPoll, or %NULL in
+ case of an error. Free with gst_poll_free().
+
+
+
+
+ whether it should be possible to control a wait.
+
+
+
+
+
+ Create a new poll object that can be used for scheduling cancellable
+timeouts.
+
+A timeout is performed with gst_poll_wait(). Multiple timeouts can be
+performed from different threads.
+
+Free-function: gst_poll_free
+
+ a new #GstPoll, or %NULL in
+ case of an error. Free with gst_poll_free().
+
+
+
+
+
+ A file descriptor object.
+
+ a file descriptor
+
+
+
+
+
+
+ Initializes @fd. Alternatively you can initialize it with
+#GST_POLL_FD_INIT.
+
+
+
+
+
+ a #GstPollFD
+
+
+
+
+
+
+ This interface offers methods to query and manipulate parameter preset sets.
+A preset is a bunch of property settings, together with meta data and a name.
+The name of a preset serves as key for subsequent method calls to manipulate
+single presets.
+All instances of one type will share the list of presets. The list is created
+on demand, if presets are not used, the list is not created.
+
+The interface comes with a default implementation that serves most plugins.
+Wrapper plugins will override most methods to implement support for the
+native preset format of those wrapped plugins.
+One method that is useful to be overridden is gst_preset_get_property_names().
+With that one can control which properties are saved and in which order.
+When implementing support for read-only presets, one should set the vmethods
+for gst_preset_save_preset() and gst_preset_delete_preset() to %NULL.
+Applications can use gst_preset_is_editable() to check for that.
+
+The default implementation supports presets located in a system directory,
+application specific directory and in the users home directory. When getting
+a list of presets individual presets are read and overlaid in 1) system,
+2) application and 3) user order. Whenever an earlier entry is newer, the
+later entries will be updated. Since 1.8 you can also provide extra paths
+where to find presets through the GST_PRESET_PATH environment variable.
+Presets found in those paths will be concidered as "app presets".
+
+ Gets the directory for application specific presets if set by the
+application.
+
+ the directory or %NULL, don't free or modify
+the string
+
+
+
+
+ Sets an extra directory as an absolute path that should be considered when
+looking for presets. Any presets in the application dir will shadow the
+system presets.
+
+ %TRUE for success, %FALSE if the dir already has been set
+
+
+
+
+ the application specific preset dir
+
+
+
+
+
+ Delete the given preset.
+
+ %TRUE for success, %FALSE if e.g. there is no preset with that @name
+
+
+
+
+ a #GObject that implements #GstPreset
+
+
+
+ preset name to remove
+
+
+
+
+
+ Gets the @value for an existing meta data @tag. Meta data @tag names can be
+something like e.g. "comment". Returned values need to be released when done.
+
+ %TRUE for success, %FALSE if e.g. there is no preset with that @name
+or no value for the given @tag
+
+
+
+
+ a #GObject that implements #GstPreset
+
+
+
+ preset name
+
+
+
+ meta data item name
+
+
+
+ value
+
+
+
+
+
+ Get a copy of preset names as a %NULL terminated string array.
+
+
+ list with names, use g_strfreev() after usage.
+
+
+
+
+
+
+ a #GObject that implements #GstPreset
+
+
+
+
+
+ Get a the names of the GObject properties that can be used for presets.
+
+ an
+ array of property names which should be freed with g_strfreev() after use.
+
+
+
+
+
+
+ a #GObject that implements #GstPreset
+
+
+
+
+
+ Load the given preset.
+
+ %TRUE for success, %FALSE if e.g. there is no preset with that @name
+
+
+
+
+ a #GObject that implements #GstPreset
+
+
+
+ preset name to load
+
+
+
+
+
+ Renames a preset. If there is already a preset by the @new_name it will be
+overwritten.
+
+ %TRUE for success, %FALSE if e.g. there is no preset with @old_name
+
+
+
+
+ a #GObject that implements #GstPreset
+
+
+
+ current preset name
+
+
+
+ new preset name
+
+
+
+
+
+ Save the current object settings as a preset under the given name. If there
+is already a preset by this @name it will be overwritten.
+
+ %TRUE for success, %FALSE
+
+
+
+
+ a #GObject that implements #GstPreset
+
+
+
+ preset name to save
+
+
+
+
+
+ Sets a new @value for an existing meta data item or adds a new item. Meta
+data @tag names can be something like e.g. "comment". Supplying %NULL for the
+@value will unset an existing value.
+
+ %TRUE for success, %FALSE if e.g. there is no preset with that @name
+
+
+
+
+ a #GObject that implements #GstPreset
+
+
+
+ preset name
+
+
+
+ meta data item name
+
+
+
+ new value
+
+
+
+
+
+ Delete the given preset.
+
+ %TRUE for success, %FALSE if e.g. there is no preset with that @name
+
+
+
+
+ a #GObject that implements #GstPreset
+
+
+
+ preset name to remove
+
+
+
+
+
+ Gets the @value for an existing meta data @tag. Meta data @tag names can be
+something like e.g. "comment". Returned values need to be released when done.
+
+ %TRUE for success, %FALSE if e.g. there is no preset with that @name
+or no value for the given @tag
+
+
+
+
+ a #GObject that implements #GstPreset
+
+
+
+ preset name
+
+
+
+ meta data item name
+
+
+
+ value
+
+
+
+
+
+ Get a copy of preset names as a %NULL terminated string array.
+
+
+ list with names, use g_strfreev() after usage.
+
+
+
+
+
+
+ a #GObject that implements #GstPreset
+
+
+
+
+
+ Get a the names of the GObject properties that can be used for presets.
+
+ an
+ array of property names which should be freed with g_strfreev() after use.
+
+
+
+
+
+
+ a #GObject that implements #GstPreset
+
+
+
+
+
+ Check if one can add new presets, change existing ones and remove presets.
+
+ %TRUE if presets are editable or %FALSE if they are static
+
+
+
+
+ a #GObject that implements #GstPreset
+
+
+
+
+
+ Load the given preset.
+
+ %TRUE for success, %FALSE if e.g. there is no preset with that @name
+
+
+
+
+ a #GObject that implements #GstPreset
+
+
+
+ preset name to load
+
+
+
+
+
+ Renames a preset. If there is already a preset by the @new_name it will be
+overwritten.
+
+ %TRUE for success, %FALSE if e.g. there is no preset with @old_name
+
+
+
+
+ a #GObject that implements #GstPreset
+
+
+
+ current preset name
+
+
+
+ new preset name
+
+
+
+
+
+ Save the current object settings as a preset under the given name. If there
+is already a preset by this @name it will be overwritten.
+
+ %TRUE for success, %FALSE
+
+
+
+
+ a #GObject that implements #GstPreset
+
+
+
+ preset name to save
+
+
+
+
+
+ Sets a new @value for an existing meta data item or adds a new item. Meta
+data @tag names can be something like e.g. "comment". Supplying %NULL for the
+@value will unset an existing value.
+
+ %TRUE for success, %FALSE if e.g. there is no preset with that @name
+
+
+
+
+ a #GObject that implements #GstPreset
+
+
+
+ preset name
+
+
+
+ meta data item name
+
+
+
+ new value
+
+
+
+
+
+
+ #GstPreset interface.
+
+ parent interface type.
+
+
+
+
+
+
+ list with names, use g_strfreev() after usage.
+
+
+
+
+
+
+ a #GObject that implements #GstPreset
+
+
+
+
+
+
+
+
+ an
+ array of property names which should be freed with g_strfreev() after use.
+
+
+
+
+
+
+ a #GObject that implements #GstPreset
+
+
+
+
+
+
+
+
+ %TRUE for success, %FALSE if e.g. there is no preset with that @name
+
+
+
+
+ a #GObject that implements #GstPreset
+
+
+
+ preset name to load
+
+
+
+
+
+
+
+
+ %TRUE for success, %FALSE
+
+
+
+
+ a #GObject that implements #GstPreset
+
+
+
+ preset name to save
+
+
+
+
+
+
+
+
+ %TRUE for success, %FALSE if e.g. there is no preset with @old_name
+
+
+
+
+ a #GObject that implements #GstPreset
+
+
+
+ current preset name
+
+
+
+ new preset name
+
+
+
+
+
+
+
+
+ %TRUE for success, %FALSE if e.g. there is no preset with that @name
+
+
+
+
+ a #GObject that implements #GstPreset
+
+
+
+ preset name to remove
+
+
+
+
+
+
+
+
+ %TRUE for success, %FALSE if e.g. there is no preset with that @name
+
+
+
+
+ a #GObject that implements #GstPreset
+
+
+
+ preset name
+
+
+
+ meta data item name
+
+
+
+ new value
+
+
+
+
+
+
+
+
+ %TRUE for success, %FALSE if e.g. there is no preset with that @name
+or no value for the given @tag
+
+
+
+
+ a #GObject that implements #GstPreset
+
+
+
+ preset name
+
+
+
+ meta data item name
+
+
+
+ value
+
+
+
+
+
+
+
+
+
+
+
+
+ The type of a %GST_MESSAGE_PROGRESS. The progress messages inform the
+application of the status of asynchronous tasks.
+
+ A new task started.
+
+
+ A task completed and a new one continues.
+
+
+ A task completed.
+
+
+ A task was canceled.
+
+
+ A task caused an error. An error message is also
+ posted on the bus.
+
+
+
+ The #GstPromise object implements the container for values that may
+be available later. i.e. a Future or a Promise in
+<ulink url="https://en.wikipedia.org/wiki/Futures_and_promises">https://en.wikipedia.org/wiki/Futures_and_promises</ulink>
+As with all Future/Promise-like functionality, there is the concept of the
+producer of the value and the consumer of the value.
+
+A #GstPromise is created with gst_promise_new() by the consumer and passed
+to the producer to avoid thread safety issues with the change callback.
+A #GstPromise can be replied to with a value (or an error) by the producer
+with gst_promise_reply(). gst_promise_interrupt() is for the consumer to
+indicate to the producer that the value is not needed anymore and producing
+that value can stop. The @GST_PROMISE_RESULT_EXPIRED state set by a call
+to gst_promise_expire() indicates to the consumer that a value will never
+be produced and is intended to be called by a third party that implements
+some notion of message handling such as #GstBus.
+A callback can also be installed at #GstPromise creation for
+result changes with gst_promise_new_with_change_func().
+The change callback can be used to chain #GstPromises's together as in the
+following example.
+|[<!-- language="C" -->
+const GstStructure *reply;
+GstPromise *p;
+if (gst_promise_wait (promise) != GST_PROMISE_RESULT_REPLIED)
+ return; // interrupted or expired value
+reply = gst_promise_get_reply (promise);
+if (error in reply)
+ return; // propagate error
+p = gst_promise_new_with_change_func (another_promise_change_func, user_data, notify);
+pass p to promise-using API
+]|
+
+Each #GstPromise starts out with a #GstPromiseResult of
+%GST_PROMISE_RESULT_PENDING and only ever transitions once
+into one of the other #GstPromiseResult's.
+
+In order to support multi-threaded code, gst_promise_reply(),
+gst_promise_interrupt() and gst_promise_expire() may all be from
+different threads with some restrictions and the final result of the promise
+is whichever call is made first. There are two restrictions on ordering:
+
+1. That gst_promise_reply() and gst_promise_interrupt() cannot be called
+after gst_promise_expire()
+2. That gst_promise_reply() and gst_promise_interrupt()
+cannot be called twice.
+
+The change function set with gst_promise_new_with_change_func() is
+called directly from either the gst_promise_reply(),
+gst_promise_interrupt() or gst_promise_expire() and can be called
+from an arbitrary thread. #GstPromise using APIs can restrict this to
+a single thread or a subset of threads but that is entirely up to the API
+that uses #GstPromise.
+
+ parent #GstMiniObject
+
+
+
+
+ a new #GstPromise
+
+
+
+
+ @func will be called exactly once when transitioning out of
+%GST_PROMISE_RESULT_PENDING into any of the other #GstPromiseResult
+states.
+
+ a new #GstPromise
+
+
+
+
+ a #GstPromiseChangeFunc to call
+
+
+
+ argument to call @func with
+
+
+
+ notification function that @user_data is no longer needed
+
+
+
+
+
+ Expire a @promise. This will wake up any waiters with
+%GST_PROMISE_RESULT_EXPIRED. Called by a message loop when the parent
+message is handled and/or destroyed (possibly unanswered).
+
+
+
+
+
+ a #GstPromise
+
+
+
+
+
+ Retrieve the reply set on @promise. @promise must be in
+%GST_PROMISE_RESULT_REPLIED and the returned structure is owned by @promise
+
+ The reply set on @promise
+
+
+
+
+ a #GstPromise
+
+
+
+
+
+ Interrupt waiting for a @promise. This will wake up any waiters with
+%GST_PROMISE_RESULT_INTERRUPTED. Called when the consumer does not want
+the value produced anymore.
+
+
+
+
+
+ a #GstPromise
+
+
+
+
+
+ Set a reply on @promise. This will wake up any waiters with
+%GST_PROMISE_RESULT_REPLIED. Called by the producer of the value to
+indicate success (or failure).
+
+If @promise has already been interrupted by the consumer, then this reply
+is not visible to the consumer.
+
+
+
+
+
+ a #GstPromise
+
+
+
+ a #GstStructure with the the reply contents
+
+
+
+
+
+ Wait for @promise to move out of the %GST_PROMISE_RESULT_PENDING state.
+If @promise is not in %GST_PROMISE_RESULT_PENDING then it will return
+immediately with the current result.
+
+ the result of the promise
+
+
+
+
+ a #GstPromise
+
+
+
+
+
+
+
+
+
+
+
+ a #GstPromise
+
+
+
+ user data
+
+
+
+
+
+ The result of a #GstPromise
+
+ Initial state. Waiting for transition to any
+ other state.
+
+
+ Interrupted by the consumer as it doesn't
+ want the value anymore.
+
+
+ A producer marked a reply
+
+
+ The promise expired (the carrying object
+ lost all refs) and the promise will never be fulfilled.
+
+
+
+ Metadata type that holds information about a sample from a protection-protected
+track, including the information needed to decrypt it (if it is encrypted).
+
+ the parent #GstMeta.
+
+
+
+ the cryptographic information needed to decrypt the sample.
+
+
+
+
+
+
+
+
+
+
+ Invoke the default chain function of the proxy pad.
+
+ a #GstFlowReturn from the pad.
+
+
+
+
+ a sink #GstPad, returns GST_FLOW_ERROR if not.
+
+
+
+ the parent of @pad or %NULL
+
+
+
+ the #GstBuffer to send, return GST_FLOW_ERROR
+ if not.
+
+
+
+
+
+ Invoke the default chain list function of the proxy pad.
+
+ a #GstFlowReturn from the pad.
+
+
+
+
+ a sink #GstPad, returns GST_FLOW_ERROR if not.
+
+
+
+ the parent of @pad or %NULL
+
+
+
+ the #GstBufferList to send, return GST_FLOW_ERROR
+ if not.
+
+
+
+
+
+ Invoke the default getrange function of the proxy pad.
+
+ a #GstFlowReturn from the pad.
+
+
+
+
+ a src #GstPad, returns #GST_FLOW_ERROR if not.
+
+
+
+ the parent of @pad
+
+
+
+ The start offset of the buffer
+
+
+
+ The length of the buffer
+
+
+
+ a pointer to hold the #GstBuffer,
+ returns #GST_FLOW_ERROR if %NULL.
+
+
+
+
+
+ Invoke the default iterate internal links function of the proxy pad.
+
+ a #GstIterator of #GstPad, or %NULL if @pad
+has no parent. Unref each returned pad with gst_object_unref().
+
+
+
+
+ the #GstPad to get the internal links of.
+
+
+
+ the parent of @pad or %NULL
+
+
+
+
+
+ Get the internal pad of @pad. Unref target pad after usage.
+
+The internal pad of a #GstGhostPad is the internally used
+pad of opposite direction, which is used to link to the target.
+
+ the target #GstProxyPad, can
+be %NULL. Unref target pad after usage.
+
+
+
+
+ the #GstProxyPad
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ The different types of QoS events that can be given to the
+gst_event_new_qos() method.
+
+ The QoS event type that is produced when upstream
+ elements are producing data too quickly and the element can't keep up
+ processing the data. Upstream should reduce their production rate. This
+ type is also used when buffers arrive early or in time.
+
+
+ The QoS event type that is produced when upstream
+ elements are producing data too slowly and need to speed up their
+ production rate.
+
+
+ The QoS event type that is produced when the
+ application enabled throttling to limit the data rate.
+
+
+
+
+
+
+ The same thing as #GST_QUERY_TYPE_UPSTREAM | #GST_QUERY_TYPE_DOWNSTREAM.
+
+
+
+ Queries can be performed on pads (gst_pad_query()) and elements
+(gst_element_query()). Please note that some queries might need a running
+pipeline to work.
+
+Queries can be created using the gst_query_new_*() functions.
+Query values can be set using gst_query_set_*(), and parsed using
+gst_query_parse_*() helpers.
+
+The following example shows how to query the duration of a pipeline:
+|[<!-- language="C" -->
+ GstQuery *query;
+ gboolean res;
+ query = gst_query_new_duration (GST_FORMAT_TIME);
+ res = gst_element_query (pipeline, query);
+ if (res) {
+ gint64 duration;
+ gst_query_parse_duration (query, NULL, &duration);
+ g_print ("duration = %"GST_TIME_FORMAT, GST_TIME_ARGS (duration));
+ } else {
+ g_print ("duration query failed...");
+ }
+ gst_query_unref (query);
+]|
+
+ The parent #GstMiniObject type
+
+
+
+ the #GstQueryType
+
+
+
+ Constructs a new query object for querying if @caps are accepted.
+
+Free-function: gst_query_unref()
+
+ a new #GstQuery
+
+
+
+
+ a fixed #GstCaps
+
+
+
+
+
+ Constructs a new query object for querying the allocation properties.
+
+Free-function: gst_query_unref()
+
+ a new #GstQuery
+
+
+
+
+ the negotiated caps
+
+
+
+ return a pool
+
+
+
+
+
+ Constructs a new query object for querying the buffering status of
+a stream.
+
+Free-function: gst_query_unref()
+
+ a new #GstQuery
+
+
+
+
+ the default #GstFormat for the new query
+
+
+
+
+
+ Constructs a new query object for querying the caps.
+
+The CAPS query should return the allowable caps for a pad in the context
+of the element's state, its link to other elements, and the devices or files
+it has opened. These caps must be a subset of the pad template caps. In the
+NULL state with no links, the CAPS query should ideally return the same caps
+as the pad template. In rare circumstances, an object property can affect
+the caps returned by the CAPS query, but this is discouraged.
+
+For most filters, the caps returned by CAPS query is directly affected by the
+allowed caps on other pads. For demuxers and decoders, the caps returned by
+the srcpad's getcaps function is directly related to the stream data. Again,
+the CAPS query should return the most specific caps it reasonably can, since this
+helps with autoplugging.
+
+The @filter is used to restrict the result caps, only the caps matching
+@filter should be returned from the CAPS query. Specifying a filter might
+greatly reduce the amount of processing an element needs to do.
+
+Free-function: gst_query_unref()
+
+ a new #GstQuery
+
+
+
+
+ a filter
+
+
+
+
+
+ Constructs a new query object for querying the pipeline-local context.
+
+Free-function: gst_query_unref()
+
+ a new #GstQuery
+
+
+
+
+ Context type to query
+
+
+
+
+
+ Constructs a new convert query object. Use gst_query_unref()
+when done with it. A convert query is used to ask for a conversion between
+one format and another.
+
+Free-function: gst_query_unref()
+
+ a #GstQuery
+
+
+
+
+ the source #GstFormat for the new query
+
+
+
+ the value to convert
+
+
+
+ the target #GstFormat
+
+
+
+
+
+ Constructs a new custom query object. Use gst_query_unref()
+when done with it.
+
+Free-function: gst_query_unref()
+
+ a new #GstQuery
+
+
+
+
+ the query type
+
+
+
+ a structure for the query
+
+
+
+
+
+ Constructs a new query object for querying the drain state.
+
+Free-function: gst_query_unref()
+
+ a new #GstQuery
+
+
+
+
+ Constructs a new stream duration query object to query in the given format.
+Use gst_query_unref() when done with it. A duration query will give the
+total length of the stream.
+
+Free-function: gst_query_unref()
+
+ a new #GstQuery
+
+
+
+
+ the #GstFormat for this duration query
+
+
+
+
+
+ Constructs a new query object for querying formats of
+the stream.
+
+Free-function: gst_query_unref()
+
+ a new #GstQuery
+
+
+
+
+ Constructs a new latency query object.
+Use gst_query_unref() when done with it. A latency query is usually performed
+by sinks to compensate for additional latency introduced by elements in the
+pipeline.
+
+Free-function: gst_query_unref()
+
+ a #GstQuery
+
+
+
+
+ Constructs a new query stream position query object. Use gst_query_unref()
+when done with it. A position query is used to query the current position
+of playback in the streams, in some format.
+
+Free-function: gst_query_unref()
+
+ a new #GstQuery
+
+
+
+
+ the default #GstFormat for the new query
+
+
+
+
+
+ Constructs a new query object for querying the scheduling properties.
+
+Free-function: gst_query_unref()
+
+ a new #GstQuery
+
+
+
+
+ Constructs a new query object for querying seeking properties of
+the stream.
+
+Free-function: gst_query_unref()
+
+ a new #GstQuery
+
+
+
+
+ the default #GstFormat for the new query
+
+
+
+
+
+ Constructs a new segment query object. Use gst_query_unref()
+when done with it. A segment query is used to discover information about the
+currently configured segment for playback.
+
+Free-function: gst_query_unref()
+
+ a new #GstQuery
+
+
+
+
+ the #GstFormat for the new query
+
+
+
+
+
+ Constructs a new query URI query object. Use gst_query_unref()
+when done with it. An URI query is used to query the current URI
+that is used by the source or sink.
+
+Free-function: gst_query_unref()
+
+ a new #GstQuery
+
+
+
+
+ Add @api with @params as one of the supported metadata API to @query.
+
+
+
+
+
+ a GST_QUERY_ALLOCATION type query #GstQuery
+
+
+
+ the metadata API
+
+
+
+ API specific parameters
+
+
+
+
+
+ Add @allocator and its @params as a supported memory allocator.
+
+
+
+
+
+ a GST_QUERY_ALLOCATION type query #GstQuery
+
+
+
+ the memory allocator
+
+
+
+ a #GstAllocationParams
+
+
+
+
+
+ Set the pool parameters in @query.
+
+
+
+
+
+ A valid #GstQuery of type GST_QUERY_ALLOCATION.
+
+
+
+ the #GstBufferPool
+
+
+
+ the buffer size
+
+
+
+ the min buffers
+
+
+
+ the max buffers
+
+
+
+
+
+ Set the buffering-ranges array field in @query. The current last
+start position of the array should be inferior to @start.
+
+ a #gboolean indicating if the range was added or not.
+
+
+
+
+ a GST_QUERY_BUFFERING type query #GstQuery
+
+
+
+ start position of the range
+
+
+
+ stop position of the range
+
+
+
+
+
+ Add @mode as one of the supported scheduling modes to @query.
+
+
+
+
+
+ a GST_QUERY_SCHEDULING type query #GstQuery
+
+
+
+ a #GstPadMode
+
+
+
+
+
+ Check if @query has metadata @api set. When this function returns %TRUE,
+@index will contain the index where the requested API and the parameters
+can be found.
+
+ %TRUE when @api is in the list of metadata.
+
+
+
+
+ a GST_QUERY_ALLOCATION type query #GstQuery
+
+
+
+ the metadata API
+
+
+
+ the index
+
+
+
+
+
+ Retrieve the number of values currently stored in the
+meta API array of the query's structure.
+
+ the metadata API array size as a #guint.
+
+
+
+
+ a GST_QUERY_ALLOCATION type query #GstQuery
+
+
+
+
+
+ Retrieve the number of values currently stored in the
+allocator params array of the query's structure.
+
+If no memory allocator is specified, the downstream element can handle
+the default memory allocator. The first memory allocator in the query
+should be generic and allow mapping to system memory, all following
+allocators should be ordered by preference with the preferred one first.
+
+ the allocator array size as a #guint.
+
+
+
+
+ a GST_QUERY_ALLOCATION type query #GstQuery
+
+
+
+
+
+ Retrieve the number of values currently stored in the
+pool array of the query's structure.
+
+ the pool array size as a #guint.
+
+
+
+
+ a GST_QUERY_ALLOCATION type query #GstQuery
+
+
+
+
+
+ Retrieve the number of values currently stored in the
+buffered-ranges array of the query's structure.
+
+ the range array size as a #guint.
+
+
+
+
+ a GST_QUERY_BUFFERING type query #GstQuery
+
+
+
+
+
+ Retrieve the number of values currently stored in the
+scheduling mode array of the query's structure.
+
+ the scheduling mode array size as a #guint.
+
+
+
+
+ a GST_QUERY_SCHEDULING type query #GstQuery
+
+
+
+
+
+ Get the structure of a query.
+
+ the #GstStructure of the query. The
+ structure is still owned by the query and will therefore be freed when the
+ query is unreffed.
+
+
+
+
+ a #GstQuery
+
+
+
+
+
+ Check if @query has scheduling mode set.
+
+> When checking if upstream supports pull mode, it is usually not
+> enough to just check for GST_PAD_MODE_PULL with this function, you
+> also want to check whether the scheduling flags returned by
+> gst_query_parse_scheduling() have the seeking flag set (meaning
+> random access is supported, not only sequential pulls).
+
+ %TRUE when @mode is in the list of scheduling modes.
+
+
+
+
+ a GST_QUERY_SCHEDULING type query #GstQuery
+
+
+
+ the scheduling mode
+
+
+
+
+
+ Check if @query has scheduling mode set and @flags is set in
+query scheduling flags.
+
+ %TRUE when @mode is in the list of scheduling modes
+ and @flags are compatible with query flags.
+
+
+
+
+ a GST_QUERY_SCHEDULING type query #GstQuery
+
+
+
+ the scheduling mode
+
+
+
+ #GstSchedulingFlags
+
+
+
+
+
+ Get the caps from @query. The caps remains valid as long as @query remains
+valid.
+
+
+
+
+
+ The query to parse
+
+
+
+ A pointer to the caps
+
+
+
+
+
+ Parse the result from @query and store in @result.
+
+
+
+
+
+ a GST_QUERY_ACCEPT_CAPS type query #GstQuery
+
+
+
+ location for the result
+
+
+
+
+
+ Parse an allocation query, writing the requested caps in @caps and
+whether a pool is needed in @need_pool, if the respective parameters
+are non-%NULL.
+
+Pool details can be retrieved using gst_query_get_n_allocation_pools() and
+gst_query_parse_nth_allocation_pool().
+
+
+
+
+
+ a #GstQuery
+
+
+
+ The #GstCaps
+
+
+
+ Whether a #GstBufferPool is needed
+
+
+
+
+
+ Get the percentage of buffered data. This is a value between 0 and 100.
+The @busy indicator is %TRUE when the buffering is in progress.
+
+
+
+
+
+ A valid #GstQuery of type GST_QUERY_BUFFERING.
+
+
+
+ if buffering is busy, or %NULL
+
+
+
+ a buffering percent, or %NULL
+
+
+
+
+
+ Parse an available query, writing the format into @format, and
+other results into the passed parameters, if the respective parameters
+are non-%NULL
+
+
+
+
+
+ a GST_QUERY_BUFFERING type query #GstQuery
+
+
+
+ the format to set for the @segment_start
+ and @segment_end values, or %NULL
+
+
+
+ the start to set, or %NULL
+
+
+
+ the stop to set, or %NULL
+
+
+
+ estimated total amount of download
+ time remaining in milliseconds, or %NULL
+
+
+
+
+
+ Extracts the buffering stats values from @query.
+
+
+
+
+
+ A valid #GstQuery of type GST_QUERY_BUFFERING.
+
+
+
+ a buffering mode, or %NULL
+
+
+
+ the average input rate, or %NULL
+
+
+
+ the average output rat, or %NULL
+
+
+
+ amount of buffering time left in
+ milliseconds, or %NULL
+
+
+
+
+
+ Get the filter from the caps @query. The caps remains valid as long as
+@query remains valid.
+
+
+
+
+
+ The query to parse
+
+
+
+ A pointer to the caps filter
+
+
+
+
+
+ Get the caps result from @query. The caps remains valid as long as
+@query remains valid.
+
+
+
+
+
+ The query to parse
+
+
+
+ A pointer to the caps
+
+
+
+
+
+ Get the context from the context @query. The context remains valid as long as
+@query remains valid.
+
+
+
+
+
+ The query to parse
+
+
+
+ A pointer to store the #GstContext
+
+
+
+
+
+ Parse a context type from an existing GST_QUERY_CONTEXT query.
+
+ a #gboolean indicating if the parsing succeeded.
+
+
+
+
+ a GST_QUERY_CONTEXT type query
+
+
+
+ the context type, or %NULL
+
+
+
+
+
+ Parse a convert query answer. Any of @src_format, @src_value, @dest_format,
+and @dest_value may be %NULL, in which case that value is omitted.
+
+
+
+
+
+ a #GstQuery
+
+
+
+ the storage for the #GstFormat of the
+ source value, or %NULL
+
+
+
+ the storage for the source value, or %NULL
+
+
+
+ the storage for the #GstFormat of the
+ destination value, or %NULL
+
+
+
+ the storage for the destination value,
+ or %NULL
+
+
+
+
+
+ Parse a duration query answer. Write the format of the duration into @format,
+and the value into @duration, if the respective variables are non-%NULL.
+
+
+
+
+
+ a #GstQuery
+
+
+
+ the storage for the #GstFormat of the duration
+ value, or %NULL.
+
+
+
+ the storage for the total duration, or %NULL.
+
+
+
+
+
+ Parse a latency query answer.
+
+
+
+
+
+ a #GstQuery
+
+
+
+ storage for live or %NULL
+
+
+
+ the storage for the min latency or %NULL
+
+
+
+ the storage for the max latency or %NULL
+
+
+
+
+
+ Parse the number of formats in the formats @query.
+
+
+
+
+
+ a #GstQuery
+
+
+
+ the number of formats in this query.
+
+
+
+
+
+ Parse an available query and get the metadata API
+at @index of the metadata API array.
+
+ a #GType of the metadata API at @index.
+
+
+
+
+ a GST_QUERY_ALLOCATION type query #GstQuery
+
+
+
+ position in the metadata API array to read
+
+
+
+ API specific parameters
+
+
+
+
+
+ Parse an available query and get the allocator and its params
+at @index of the allocator array.
+
+
+
+
+
+ a GST_QUERY_ALLOCATION type query #GstQuery
+
+
+
+ position in the allocator array to read
+
+
+
+ variable to hold the result
+
+
+
+ parameters for the allocator
+
+
+
+
+
+ Get the pool parameters in @query.
+
+Unref @pool with gst_object_unref() when it's not needed any more.
+
+
+
+
+
+ A valid #GstQuery of type GST_QUERY_ALLOCATION.
+
+
+
+ index to parse
+
+
+
+ the #GstBufferPool
+
+
+
+ the buffer size
+
+
+
+ the min buffers
+
+
+
+ the max buffers
+
+
+
+
+
+ Parse an available query and get the start and stop values stored
+at the @index of the buffered ranges array.
+
+ a #gboolean indicating if the parsing succeeded.
+
+
+
+
+ a GST_QUERY_BUFFERING type query #GstQuery
+
+
+
+ position in the buffered-ranges array to read
+
+
+
+ the start position to set, or %NULL
+
+
+
+ the stop position to set, or %NULL
+
+
+
+
+
+ Parse the format query and retrieve the @nth format from it into
+@format. If the list contains less elements than @nth, @format will be
+set to GST_FORMAT_UNDEFINED.
+
+
+
+
+
+ a #GstQuery
+
+
+
+ the nth format to retrieve.
+
+
+
+ a pointer to store the nth format
+
+
+
+
+
+ Parse an available query and get the scheduling mode
+at @index of the scheduling modes array.
+
+ a #GstPadMode of the scheduling mode at @index.
+
+
+
+
+ a GST_QUERY_SCHEDULING type query #GstQuery
+
+
+
+ position in the scheduling modes array to read
+
+
+
+
+
+ Parse a position query, writing the format into @format, and the position
+into @cur, if the respective parameters are non-%NULL.
+
+
+
+
+
+ a #GstQuery
+
+
+
+ the storage for the #GstFormat of the
+ position values (may be %NULL)
+
+
+
+ the storage for the current position (may be %NULL)
+
+
+
+
+
+ Set the scheduling properties.
+
+
+
+
+
+ A valid #GstQuery of type GST_QUERY_SCHEDULING.
+
+
+
+ #GstSchedulingFlags
+
+
+
+ the suggested minimum size of pull requests
+
+
+
+ the suggested maximum size of pull requests:
+
+
+
+ the suggested alignment of pull requests
+
+
+
+
+
+ Parse a seeking query, writing the format into @format, and
+other results into the passed parameters, if the respective parameters
+are non-%NULL
+
+
+
+
+
+ a GST_QUERY_SEEKING type query #GstQuery
+
+
+
+ the format to set for the @segment_start
+ and @segment_end values, or %NULL
+
+
+
+ the seekable flag to set, or %NULL
+
+
+
+ the segment_start to set, or %NULL
+
+
+
+ the segment_end to set, or %NULL
+
+
+
+
+
+ Parse a segment query answer. Any of @rate, @format, @start_value, and
+@stop_value may be %NULL, which will cause this value to be omitted.
+
+See gst_query_set_segment() for an explanation of the function arguments.
+
+
+
+
+
+ a #GstQuery
+
+
+
+ the storage for the rate of the segment, or %NULL
+
+
+
+ the storage for the #GstFormat of the values,
+ or %NULL
+
+
+
+ the storage for the start value, or %NULL
+
+
+
+ the storage for the stop value, or %NULL
+
+
+
+
+
+ Parse an URI query, writing the URI into @uri as a newly
+allocated string, if the respective parameters are non-%NULL.
+Free the string with g_free() after usage.
+
+
+
+
+
+ a #GstQuery
+
+
+
+ the storage for the current URI
+ (may be %NULL)
+
+
+
+
+
+ Parse an URI query, writing the URI into @uri as a newly
+allocated string, if the respective parameters are non-%NULL.
+Free the string with g_free() after usage.
+
+
+
+
+
+ a #GstQuery
+
+
+
+ the storage for the redirect URI
+ (may be %NULL)
+
+
+
+
+
+ Parse an URI query, and set @permanent to %TRUE if there is a redirection
+and it should be considered permanent. If a redirection is permanent,
+applications should update their internal storage of the URI, otherwise
+they should make all future requests to the original URI.
+
+
+
+
+
+ a #GstQuery
+
+
+
+ if the URI redirection is permanent
+ (may be %NULL)
+
+
+
+
+
+ Remove the metadata API at @index of the metadata API array.
+
+
+
+
+
+ a GST_QUERY_ALLOCATION type query #GstQuery
+
+
+
+ position in the metadata API array to remove
+
+
+
+
+
+ Remove the allocation param at @index of the allocation param array.
+
+
+
+
+
+ a GST_QUERY_ALLOCATION type query #GstQuery
+
+
+
+ position in the allocation param array to remove
+
+
+
+
+
+ Remove the allocation pool at @index of the allocation pool array.
+
+
+
+
+
+ a GST_QUERY_ALLOCATION type query #GstQuery
+
+
+
+ position in the allocation pool array to remove
+
+
+
+
+
+ Set @result as the result for the @query.
+
+
+
+
+
+ a GST_QUERY_ACCEPT_CAPS type query #GstQuery
+
+
+
+ the result to set
+
+
+
+
+
+ Set the percentage of buffered data. This is a value between 0 and 100.
+The @busy indicator is %TRUE when the buffering is in progress.
+
+
+
+
+
+ A valid #GstQuery of type GST_QUERY_BUFFERING.
+
+
+
+ if buffering is busy
+
+
+
+ a buffering percent
+
+
+
+
+
+ Set the available query result fields in @query.
+
+
+
+
+
+ a #GstQuery
+
+
+
+ the format to set for the @start and @stop values
+
+
+
+ the start to set
+
+
+
+ the stop to set
+
+
+
+ estimated total amount of download time remaining in
+ milliseconds
+
+
+
+
+
+ Configures the buffering stats values in @query.
+
+
+
+
+
+ A valid #GstQuery of type GST_QUERY_BUFFERING.
+
+
+
+ a buffering mode
+
+
+
+ the average input rate
+
+
+
+ the average output rate
+
+
+
+ amount of buffering time left in milliseconds
+
+
+
+
+
+ Set the @caps result in @query.
+
+
+
+
+
+ The query to use
+
+
+
+ A pointer to the caps
+
+
+
+
+
+ Answer a context query by setting the requested context.
+
+
+
+
+
+ a #GstQuery with query type GST_QUERY_CONTEXT
+
+
+
+ the requested #GstContext
+
+
+
+
+
+ Answer a convert query by setting the requested values.
+
+
+
+
+
+ a #GstQuery
+
+
+
+ the source #GstFormat
+
+
+
+ the source value
+
+
+
+ the destination #GstFormat
+
+
+
+ the destination value
+
+
+
+
+
+ Answer a duration query by setting the requested value in the given format.
+
+
+
+
+
+ a #GstQuery
+
+
+
+ the #GstFormat for the duration
+
+
+
+ the duration of the stream
+
+
+
+
+
+ Set the formats query result fields in @query. The number of formats passed
+must be equal to @n_formats.
+
+
+
+
+
+ a #GstQuery
+
+
+
+ the number of formats to set.
+
+
+
+ A number of @GstFormats equal to @n_formats.
+
+
+
+
+
+ Set the formats query result fields in @query. The number of formats passed
+in the @formats array must be equal to @n_formats.
+
+
+
+
+
+ a #GstQuery
+
+
+
+ the number of formats to set.
+
+
+
+ an array containing @n_formats
+ @GstFormat values.
+
+
+
+
+
+
+
+ Answer a latency query by setting the requested values in the given format.
+
+
+
+
+
+ a #GstQuery
+
+
+
+ if there is a live element upstream
+
+
+
+ the minimal latency of the upstream elements
+
+
+
+ the maximal latency of the upstream elements
+
+
+
+
+
+ Parse an available query and get the allocator and its params
+at @index of the allocator array.
+
+
+
+
+
+ a GST_QUERY_ALLOCATION type query #GstQuery
+
+
+
+ position in the allocator array to set
+
+
+
+ new allocator to set
+
+
+
+ parameters for the allocator
+
+
+
+
+
+ Set the pool parameters in @query.
+
+
+
+
+
+ A valid #GstQuery of type GST_QUERY_ALLOCATION.
+
+
+
+ index to modify
+
+
+
+ the #GstBufferPool
+
+
+
+ the buffer size
+
+
+
+ the min buffers
+
+
+
+ the max buffers
+
+
+
+
+
+ Answer a position query by setting the requested value in the given format.
+
+
+
+
+
+ a #GstQuery with query type GST_QUERY_POSITION
+
+
+
+ the requested #GstFormat
+
+
+
+ the position to set
+
+
+
+
+
+ Set the scheduling properties.
+
+
+
+
+
+ A valid #GstQuery of type GST_QUERY_SCHEDULING.
+
+
+
+ #GstSchedulingFlags
+
+
+
+ the suggested minimum size of pull requests
+
+
+
+ the suggested maximum size of pull requests
+
+
+
+ the suggested alignment of pull requests
+
+
+
+
+
+ Set the seeking query result fields in @query.
+
+
+
+
+
+ a #GstQuery
+
+
+
+ the format to set for the @segment_start and @segment_end values
+
+
+
+ the seekable flag to set
+
+
+
+ the segment_start to set
+
+
+
+ the segment_end to set
+
+
+
+
+
+ Answer a segment query by setting the requested values. The normal
+playback segment of a pipeline is 0 to duration at the default rate of
+1.0. If a seek was performed on the pipeline to play a different
+segment, this query will return the range specified in the last seek.
+
+@start_value and @stop_value will respectively contain the configured
+playback range start and stop values expressed in @format.
+The values are always between 0 and the duration of the media and
+@start_value <= @stop_value. @rate will contain the playback rate. For
+negative rates, playback will actually happen from @stop_value to
+@start_value.
+
+
+
+
+
+ a #GstQuery
+
+
+
+ the rate of the segment
+
+
+
+ the #GstFormat of the segment values (@start_value and @stop_value)
+
+
+
+ the start value
+
+
+
+ the stop value
+
+
+
+
+
+ Answer a URI query by setting the requested URI.
+
+
+
+
+
+ a #GstQuery with query type GST_QUERY_URI
+
+
+
+ the URI to set
+
+
+
+
+
+ Answer a URI query by setting the requested URI redirection.
+
+
+
+
+
+ a #GstQuery with query type GST_QUERY_URI
+
+
+
+ the URI to set
+
+
+
+
+
+ Answer a URI query by setting the requested URI redirection
+to permanent or not.
+
+
+
+
+
+ a #GstQuery with query type %GST_QUERY_URI
+
+
+
+ whether the redirect is permanent or not
+
+
+
+
+
+ Get the structure of a query. This method should be called with a writable
+@query so that the returned structure is guaranteed to be writable.
+
+ the #GstStructure of the query. The structure is
+ still owned by the query and will therefore be freed when the query
+ is unreffed.
+
+
+
+
+ a #GstQuery
+
+
+
+
+
+
+ Standard predefined Query types
+
+ unknown query type
+
+
+ current position in stream
+
+
+ total duration of the stream
+
+
+ latency of stream
+
+
+ current jitter of stream
+
+
+ current rate of the stream
+
+
+ seeking capabilities
+
+
+ segment start/stop positions
+
+
+ convert values between formats
+
+
+ query supported formats for convert
+
+
+ query available media for efficient seeking.
+
+
+ a custom application or element defined query.
+
+
+ query the URI of the source or sink.
+
+
+ the buffer allocation properties
+
+
+ the scheduling properties
+
+
+ the accept caps query
+
+
+ the caps query
+
+
+ wait till all serialized data is consumed downstream
+
+
+ query the pipeline-local context from
+ downstream or upstream (since 1.2)
+
+
+ Gets the #GstQueryTypeFlags associated with @type.
+
+ a #GstQueryTypeFlags.
+
+
+
+
+ a #GstQueryType
+
+
+
+
+
+ Get a printable name for the given query type. Do not modify or free.
+
+ a reference to the static name of the query.
+
+
+
+
+ the query type
+
+
+
+
+
+ Get the unique quark for the given query type.
+
+ the quark associated with the query type
+
+
+
+
+ the query type
+
+
+
+
+
+
+ #GstQueryTypeFlags indicate the aspects of the different #GstQueryType
+values. You can get the type flags of a #GstQueryType with the
+gst_query_type_get_flags() function.
+
+ Set if the query can travel upstream.
+
+
+ Set if the query can travel downstream.
+
+
+ Set if the query should be serialized with data
+ flow.
+
+
+
+ Element priority ranks. Defines the order in which the autoplugger (or
+similar rank-picking mechanisms, such as e.g. gst_element_make_from_uri())
+will choose this element over an alternative one with the same function.
+
+These constants serve as a rough guidance for defining the rank of a
+#GstPluginFeature. Any value is valid, including values bigger than
+@GST_RANK_PRIMARY.
+
+ will be chosen last or not at all
+
+
+ unlikely to be chosen
+
+
+ likely to be chosen
+
+
+ will be chosen first
+
+
+
+ #GstReferenceTimestampMeta can be used to attach alternative timestamps and
+possibly durations to a #GstBuffer. These are generally not according to
+the pipeline clock and could be e.g. the NTP timestamp when the media was
+captured.
+
+The reference is stored as a #GstCaps in @reference. Examples of valid
+references would be "timestamp/x-drivername-stream" for timestamps that are locally
+generated by some driver named "drivername" when generating the stream,
+e.g. based on a frame counter, or "timestamp/x-ntp, host=pool.ntp.org,
+port=123" for timestamps based on a specific NTP server.
+
+ the parent #GstMeta structure
+
+
+
+ identifier for the timestamp reference.
+
+
+
+ timestamp
+
+
+
+ duration, or %GST_CLOCK_TIME_NONE
+
+
+
+ Get the global #GstMetaInfo describing the #GstReferenceTimestampMeta meta.
+
+ The #GstMetaInfo
+
+
+
+
+
+ One registry holds the metadata of a set of plugins.
+
+<emphasis role="bold">Design:</emphasis>
+
+The #GstRegistry object is a list of plugins and some functions for dealing
+with them. Each #GstPlugin is matched 1-1 with a file on disk, and may or may
+not be loaded at a given time.
+
+The primary source, at all times, of plugin information is each plugin file
+itself. Thus, if an application wants information about a particular plugin,
+or wants to search for a feature that satisfies given criteria, the primary
+means of doing so is to load every plugin and look at the resulting
+information that is gathered in the default registry. Clearly, this is a time
+consuming process, so we cache information in the registry file. The format
+and location of the cache file is internal to gstreamer.
+
+On startup, plugins are searched for in the plugin search path. The following
+locations are checked in this order:
+
+* location from --gst-plugin-path commandline option.
+* the GST_PLUGIN_PATH environment variable.
+* the GST_PLUGIN_SYSTEM_PATH environment variable.
+* default locations (if GST_PLUGIN_SYSTEM_PATH is not set).
+ Those default locations are:
+ `$XDG_DATA_HOME/gstreamer-$GST_API_VERSION/plugins/`
+ and `$prefix/libs/gstreamer-$GST_API_VERSION/`.
+ [$XDG_DATA_HOME](http://standards.freedesktop.org/basedir-spec/basedir-spec-latest.html) defaults to
+ `$HOME/.local/share`.
+
+The registry cache file is loaded from
+`$XDG_CACHE_HOME/gstreamer-$GST_API_VERSION/registry-$ARCH.bin`
+(where $XDG_CACHE_HOME defaults to `$HOME/.cache`) or the file listed in the `GST_REGISTRY`
+env var. One reason to change the registry location is for testing.
+
+For each plugin that is found in the plugin search path, there could be 3
+possibilities for cached information:
+
+ * the cache may not contain information about a given file.
+ * the cache may have stale information.
+ * the cache may have current information.
+
+In the first two cases, the plugin is loaded and the cache updated. In
+addition to these cases, the cache may have entries for plugins that are not
+relevant to the current process. These are marked as not available to the
+current process. If the cache is updated for whatever reason, it is marked
+dirty.
+
+A dirty cache is written out at the end of initialization. Each entry is
+checked to make sure the information is minimally valid. If not, the entry is
+simply dropped.
+
+## Implementation notes:
+
+The "cache" and "registry" are different concepts and can represent
+different sets of plugins. For various reasons, at init time, the cache is
+stored in the default registry, and plugins not relevant to the current
+process are marked with the %GST_PLUGIN_FLAG_CACHED bit. These plugins are
+removed at the end of initialization.
+
+ By default GStreamer will perform scanning and rebuilding of the
+registry file using a helper child process.
+
+Applications might want to disable this behaviour with the
+gst_registry_fork_set_enabled() function, in which case new plugins
+are scanned (and loaded) into the application process.
+
+ %TRUE if GStreamer will use the child helper process when
+rebuilding the registry.
+
+
+
+
+ Applications might want to disable/enable spawning of a child helper process
+when rebuilding the registry. See gst_registry_fork_is_enabled() for more
+information.
+
+
+
+
+
+ whether rebuilding the registry can use a temporary child helper process.
+
+
+
+
+
+ Retrieves the singleton plugin registry. The caller does not own a
+reference on the registry, as it is alive as long as GStreamer is
+initialized.
+
+ the #GstRegistry.
+
+
+
+
+ Add the feature to the registry. The feature-added signal will be emitted.
+
+@feature's reference count will be incremented, and any floating
+reference will be removed (see gst_object_ref_sink())
+
+ %TRUE on success.
+
+MT safe.
+
+
+
+
+ the registry to add the plugin to
+
+
+
+ the feature to add
+
+
+
+
+
+ Add the plugin to the registry. The plugin-added signal will be emitted.
+
+@plugin's reference count will be incremented, and any floating
+reference will be removed (see gst_object_ref_sink())
+
+ %TRUE on success.
+
+MT safe.
+
+
+
+
+ the registry to add the plugin to
+
+
+
+ the plugin to add
+
+
+
+
+
+ Checks whether a plugin feature by the given name exists in
+@registry and whether its version is at least the
+version required.
+
+ %TRUE if the feature could be found and the version is
+the same as the required version or newer, and %FALSE otherwise.
+
+
+
+
+ a #GstRegistry
+
+
+
+ the name of the feature (e.g. "oggdemux")
+
+
+
+ the minimum major version number
+
+
+
+ the minimum minor version number
+
+
+
+ the minimum micro version number
+
+
+
+
+
+ Runs a filter against all features of the plugins in the registry
+and returns a GList with the results.
+If the first flag is set, only the first match is
+returned (as a list with a single object).
+
+ a #GList of
+ #GstPluginFeature. Use gst_plugin_feature_list_free() after usage.
+
+MT safe.
+
+
+
+
+
+
+ registry to query
+
+
+
+ the filter to use
+
+
+
+ only return first match
+
+
+
+ user data passed to the filter function
+
+
+
+
+
+ Find the pluginfeature with the given name and type in the registry.
+
+ the pluginfeature with the
+ given name and type or %NULL if the plugin was not
+ found. gst_object_unref() after usage.
+
+MT safe.
+
+
+
+
+ the registry to search
+
+
+
+ the pluginfeature name to find
+
+
+
+ the pluginfeature type to find
+
+
+
+
+
+ Find the plugin with the given name in the registry.
+The plugin will be reffed; caller is responsible for unreffing.
+
+ the plugin with the given name
+ or %NULL if the plugin was not found. gst_object_unref() after
+ usage.
+
+MT safe.
+
+
+
+
+ the registry to search
+
+
+
+ the plugin name to find
+
+
+
+
+
+ Retrieves a #GList of #GstPluginFeature of @type.
+
+ a #GList of
+ #GstPluginFeature of @type. Use gst_plugin_feature_list_free() after use
+
+MT safe.
+
+
+
+
+
+
+ a #GstRegistry
+
+
+
+ a #GType.
+
+
+
+
+
+ Retrieves a #GList of features of the plugin with name @name.
+
+ a #GList of
+ #GstPluginFeature. Use gst_plugin_feature_list_free() after usage.
+
+
+
+
+
+
+ a #GstRegistry.
+
+
+
+ a plugin name.
+
+
+
+
+
+ Returns the registry's feature list cookie. This changes
+every time a feature is added or removed from the registry.
+
+ the feature list cookie.
+
+
+
+
+ the registry
+
+
+
+
+
+ Get a copy of all plugins registered in the given registry. The refcount
+of each element in the list in incremented.
+
+ a #GList of #GstPlugin.
+ Use gst_plugin_list_free() after usage.
+
+MT safe.
+
+
+
+
+
+
+ the registry to search
+
+
+
+
+
+ Look up a plugin in the given registry with the given filename.
+If found, plugin is reffed.
+
+ the #GstPlugin if found, or
+ %NULL if not. gst_object_unref() after usage.
+
+
+
+
+ the registry to look up in
+
+
+
+ the name of the file to look up
+
+
+
+
+
+ Find a #GstPluginFeature with @name in @registry.
+
+ a #GstPluginFeature with its refcount incremented,
+ use gst_object_unref() after usage.
+
+MT safe.
+
+
+
+
+ a #GstRegistry
+
+
+
+ a #GstPluginFeature name
+
+
+
+
+
+ Runs a filter against all plugins in the registry and returns a #GList with
+the results. If the first flag is set, only the first match is
+returned (as a list with a single object).
+Every plugin is reffed; use gst_plugin_list_free() after use, which
+will unref again.
+
+ a #GList of #GstPlugin.
+ Use gst_plugin_list_free() after usage.
+
+MT safe.
+
+
+
+
+
+
+ registry to query
+
+
+
+ the filter to use
+
+
+
+ only return first match
+
+
+
+ user data passed to the filter function
+
+
+
+
+
+ Remove the feature from the registry.
+
+MT safe.
+
+
+
+
+
+ the registry to remove the feature from
+
+
+
+ the feature to remove
+
+
+
+
+
+ Remove the plugin from the registry.
+
+MT safe.
+
+
+
+
+
+ the registry to remove the plugin from
+
+
+
+ the plugin to remove
+
+
+
+
+
+ Scan the given path for plugins to add to the registry. The syntax of the
+path is specific to the registry.
+
+ %TRUE if registry changed
+
+
+
+
+ the registry to add found plugins to
+
+
+
+ the path to scan
+
+
+
+
+
+
+
+
+
+
+
+ Signals that a feature has been added to the registry (possibly
+replacing a previously-added one by the same name)
+
+
+
+
+
+ the feature that has been added
+
+
+
+
+
+ Signals that a plugin has been added to the registry (possibly
+replacing a previously-added one by the same name)
+
+
+
+
+
+ the plugin that has been added
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Resource errors are for any resource used by an element:
+memory, files, network connections, process space, ...
+They're typically used by source and sink elements.
+
+ a general error which doesn't fit in any other
+category. Make sure you add a custom message to the error call.
+
+
+ do not use this except as a placeholder for
+deciding where to go while developing code.
+
+
+ used when the resource could not be found.
+
+
+ used when resource is busy.
+
+
+ used when resource fails to open for reading.
+
+
+ used when resource fails to open for writing.
+
+
+ used when resource cannot be opened for
+both reading and writing, or either (but unspecified which).
+
+
+ used when the resource can't be closed.
+
+
+ used when the resource can't be read from.
+
+
+ used when the resource can't be written to.
+
+
+ used when a seek on the resource fails.
+
+
+ used when a synchronize on the resource fails.
+
+
+ used when settings can't be manipulated on.
+
+
+ used when the resource has no space left.
+
+
+ used when the resource can't be opened
+ due to missing authorization.
+ (Since 1.2.4)
+
+
+ the number of resource error types.
+
+
+
+
+
+
+
+
+ Constant that defines one GStreamer second.
+
+
+
+ printf format type used to debug GStreamer segments. You can use this in
+combination with GStreamer's debug logging system as well as the functions
+gst_info_vasprintf(), gst_info_strdup_vprintf() and gst_info_strdup_printf()
+to pretty-print #GstSegment structures.
+This can only be used on pointers to GstSegment structures.
+
+
+
+ A value which is guaranteed to never be returned by
+gst_util_seqnum_next().
+
+Can be used as a default value in variables used to store seqnum.
+
+
+
+ A string that can be used in printf-like format strings to display a signed
+#GstClockTimeDiff or #gint64 value in h:m:s format. Use GST_TIME_ARGS() to
+construct the matching arguments.
+
+Example:
+|[
+printf("%" GST_STIME_FORMAT "\n", GST_STIME_ARGS(ts));
+]|
+
+
+
+ A #GstSample is a small object containing data, a type, timing and
+extra arbitrary information.
+
+ Create a new #GstSample with the provided details.
+
+Free-function: gst_sample_unref
+
+ the new #GstSample. gst_sample_unref()
+ after usage.
+
+
+
+
+ a #GstBuffer, or %NULL
+
+
+
+ a #GstCaps, or %NULL
+
+
+
+ a #GstSegment, or %NULL
+
+
+
+ a #GstStructure, or %NULL
+
+
+
+
+
+ Get the buffer associated with @sample
+
+ the buffer of @sample or %NULL
+ when there is no buffer. The buffer remains valid as long as
+ @sample is valid. If you need to hold on to it for longer than
+ that, take a ref to the buffer with gst_buffer_ref().
+
+
+
+
+ a #GstSample
+
+
+
+
+
+ Get the buffer list associated with @sample
+
+ the buffer list of @sample or %NULL
+ when there is no buffer list. The buffer list remains valid as long as
+ @sample is valid. If you need to hold on to it for longer than
+ that, take a ref to the buffer list with gst_mini_object_ref ().
+
+
+
+
+ a #GstSample
+
+
+
+
+
+ Get the caps associated with @sample
+
+ the caps of @sample or %NULL
+ when there is no caps. The caps remain valid as long as @sample is
+ valid. If you need to hold on to the caps for longer than that,
+ take a ref to the caps with gst_caps_ref().
+
+
+
+
+ a #GstSample
+
+
+
+
+
+ Get extra information associated with @sample.
+
+ the extra info of @sample.
+ The info remains valid as long as @sample is valid.
+
+
+
+
+ a #GstSample
+
+
+
+
+
+ Get the segment associated with @sample
+
+ the segment of @sample.
+ The segment remains valid as long as @sample is valid.
+
+
+
+
+ a #GstSample
+
+
+
+
+
+ Set the buffer list associated with @sample
+
+
+
+
+
+ a #GstSample
+
+
+
+ a #GstBufferList
+
+
+
+
+
+
+ The different scheduling flags.
+
+ if seeking is possible
+
+
+ if sequential access is recommended
+
+
+ if bandwidth is limited and buffering possible (since 1.2)
+
+
+
+ The different search modes.
+
+ Only search for exact matches.
+
+
+ Search for an exact match or the element just before.
+
+
+ Search for an exact match or the element just after.
+
+
+
+ Flags to be used with gst_element_seek() or gst_event_new_seek(). All flags
+can be used together.
+
+A non flushing seek might take some time to perform as the currently
+playing data in the pipeline will not be cleared.
+
+An accurate seek might be slower for formats that don't have any indexes
+or timestamp markers in the stream. Specifying this flag might require a
+complete scan of the file in those cases.
+
+When performing a segment seek: after the playback of the segment completes,
+no EOS will be emitted by the element that performed the seek, but a
+%GST_MESSAGE_SEGMENT_DONE message will be posted on the bus by the element.
+When this message is posted, it is possible to send a new seek event to
+continue playback. With this seek method it is possible to perform seamless
+looping or simple linear editing.
+
+When doing fast forward (rate > 1.0) or fast reverse (rate < -1.0) trickmode
+playback, the %GST_SEEK_FLAG_TRICKMODE flag can be used to instruct decoders
+and demuxers to adjust the playback rate by skipping frames. This can improve
+performance and decrease CPU usage because not all frames need to be decoded.
+
+Beyond that, the %GST_SEEK_FLAG_TRICKMODE_KEY_UNITS flag can be used to
+request that decoders skip all frames except key units, and
+%GST_SEEK_FLAG_TRICKMODE_NO_AUDIO flags can be used to request that audio
+decoders do no decoding at all, and simple output silence.
+
+The %GST_SEEK_FLAG_SNAP_BEFORE flag can be used to snap to the previous
+relevant location, and the %GST_SEEK_FLAG_SNAP_AFTER flag can be used to
+select the next relevant location. If %GST_SEEK_FLAG_KEY_UNIT is specified,
+the relevant location is a keyframe. If both flags are specified, the nearest
+of these locations will be selected. If none are specified, the implementation is
+free to select whichever it wants.
+
+The before and after here are in running time, so when playing backwards,
+the next location refers to the one that will played in next, and not the
+one that is located after in the actual source stream.
+
+Also see part-seeking.txt in the GStreamer design documentation for more
+details on the meaning of these flags and the behaviour expected of
+elements that handle them.
+
+ no flag
+
+
+ flush pipeline
+
+
+ accurate position is requested, this might
+ be considerably slower for some formats.
+
+
+ seek to the nearest keyframe. This might be
+ faster but less accurate.
+
+
+ perform a segment seek.
+
+
+ when doing fast forward or fast reverse playback, allow
+ elements to skip frames instead of generating all
+ frames. (Since 1.6)
+
+
+ Deprecated backward compatibility flag, replaced
+ by %GST_SEEK_FLAG_TRICKMODE
+
+
+ go to a location before the requested position,
+ if %GST_SEEK_FLAG_KEY_UNIT this means the keyframe at or before
+ the requested position the one at or before the seek target.
+
+
+ go to a location after the requested position,
+ if %GST_SEEK_FLAG_KEY_UNIT this means the keyframe at of after the
+ requested position.
+
+
+ go to a position near the requested position,
+ if %GST_SEEK_FLAG_KEY_UNIT this means the keyframe closest
+ to the requested position, if both keyframes are at an equal
+ distance, behaves like %GST_SEEK_FLAG_SNAP_BEFORE.
+
+
+ when doing fast forward or fast reverse
+ playback, request that elements only decode keyframes
+ and skip all other content, for formats that have
+ keyframes. (Since 1.6)
+
+
+ when doing fast forward or fast reverse
+ playback, request that audio decoder elements skip
+ decoding and output only gap events or silence. (Since 1.6)
+
+
+
+ The different types of seek events. When constructing a seek event with
+gst_event_new_seek() or when doing gst_segment_do_seek ().
+
+ no change in position is required
+
+
+ absolute position is requested
+
+
+ relative position to duration is requested
+
+
+
+ This helper structure holds the relevant values for tracking the region of
+interest in a media file, called a segment.
+
+The structure can be used for two purposes:
+
+ * performing seeks (handling seek events)
+ * tracking playback regions (handling newsegment events)
+
+The segment is usually configured by the application with a seek event which
+is propagated upstream and eventually handled by an element that performs the seek.
+
+The configured segment is then propagated back downstream with a newsegment event.
+This information is then used to clip media to the segment boundaries.
+
+A segment structure is initialized with gst_segment_init(), which takes a #GstFormat
+that will be used as the format of the segment values. The segment will be configured
+with a start value of 0 and a stop/duration of -1, which is undefined. The default
+rate and applied_rate is 1.0.
+
+The public duration field contains the duration of the segment. When using
+the segment for seeking, the start and time members should normally be left
+to their default 0 value. The stop position is left to -1 unless explicitly
+configured to a different value after a seek event.
+
+The current position in the segment should be set by changing the position
+member in the structure.
+
+For elements that perform seeks, the current segment should be updated with the
+gst_segment_do_seek() and the values from the seek event. This method will update
+all the segment fields. The position field will contain the new playback position.
+If the start_type was different from GST_SEEK_TYPE_NONE, playback continues from
+the position position, possibly with updated flags or rate.
+
+For elements that want to use #GstSegment to track the playback region,
+update the segment fields with the information from the newsegment event.
+The gst_segment_clip() method can be used to check and clip
+the media data to the segment boundaries.
+
+For elements that want to synchronize to the pipeline clock, gst_segment_to_running_time()
+can be used to convert a timestamp to a value that can be used to synchronize
+to the clock. This function takes into account the base as well as
+any rate or applied_rate conversions.
+
+For elements that need to perform operations on media data in stream_time,
+gst_segment_to_stream_time() can be used to convert a timestamp and the segment
+info to stream time (which is always between 0 and the duration of the stream).
+
+ flags for this segment
+
+
+
+ the playback rate of the segment
+
+
+
+ the already applied rate to the segment
+
+
+
+ the format of the segment values
+
+
+
+ the running time (plus elapsed time, see offset) of the segment start
+
+
+
+ the amount (in buffer timestamps) that has already been elapsed in
+ the segment
+
+
+
+ the start of the segment in buffer timestamp time (PTS)
+
+
+
+ the stop of the segment in buffer timestamp time (PTS)
+
+
+
+ the stream time of the segment start
+
+
+
+ the buffer timestamp position in the segment (used internally by
+ elements such as sources, demuxers or parsers to track progress)
+
+
+
+ the duration of the stream
+
+
+
+
+
+
+
+
+ Allocate a new #GstSegment structure and initialize it using
+gst_segment_init().
+
+Free-function: gst_segment_free
+
+ a new #GstSegment, free with gst_segment_free().
+
+
+
+
+ Clip the given @start and @stop values to the segment boundaries given
+in @segment. @start and @stop are compared and clipped to @segment
+start and stop values.
+
+If the function returns %FALSE, @start and @stop are known to fall
+outside of @segment and @clip_start and @clip_stop are not updated.
+
+When the function returns %TRUE, @clip_start and @clip_stop will be
+updated. If @clip_start or @clip_stop are different from @start or @stop
+respectively, the region fell partially in the segment.
+
+Note that when @stop is -1, @clip_stop will be set to the end of the
+segment. Depending on the use case, this may or may not be what you want.
+
+ %TRUE if the given @start and @stop times fall partially or
+ completely in @segment, %FALSE if the values are completely outside
+ of the segment.
+
+
+
+
+ a #GstSegment structure.
+
+
+
+ the format of the segment.
+
+
+
+ the start position in the segment
+
+
+
+ the stop position in the segment
+
+
+
+ the clipped start position in the segment
+
+
+
+ the clipped stop position in the segment
+
+
+
+
+
+ Create a copy of given @segment.
+
+Free-function: gst_segment_free
+
+ a new #GstSegment, free with gst_segment_free().
+
+
+
+
+ a #GstSegment
+
+
+
+
+
+ Copy the contents of @src into @dest.
+
+
+
+
+
+ a #GstSegment
+
+
+
+ a #GstSegment
+
+
+
+
+
+ Update the segment structure with the field values of a seek event (see
+gst_event_new_seek()).
+
+After calling this method, the segment field position and time will
+contain the requested new position in the segment. The new requested
+position in the segment depends on @rate and @start_type and @stop_type.
+
+For positive @rate, the new position in the segment is the new @segment
+start field when it was updated with a @start_type different from
+#GST_SEEK_TYPE_NONE. If no update was performed on @segment start position
+(#GST_SEEK_TYPE_NONE), @start is ignored and @segment position is
+unmodified.
+
+For negative @rate, the new position in the segment is the new @segment
+stop field when it was updated with a @stop_type different from
+#GST_SEEK_TYPE_NONE. If no stop was previously configured in the segment, the
+duration of the segment will be used to update the stop position.
+If no update was performed on @segment stop position (#GST_SEEK_TYPE_NONE),
+@stop is ignored and @segment position is unmodified.
+
+The applied rate of the segment will be set to 1.0 by default.
+If the caller can apply a rate change, it should update @segment
+rate and applied_rate after calling this function.
+
+@update will be set to %TRUE if a seek should be performed to the segment
+position field. This field can be %FALSE if, for example, only the @rate
+has been changed but not the playback position.
+
+ %TRUE if the seek could be performed.
+
+
+
+
+ a #GstSegment structure.
+
+
+
+ the rate of the segment.
+
+
+
+ the format of the segment.
+
+
+
+ the segment flags for the segment
+
+
+
+ the seek method
+
+
+
+ the seek start value
+
+
+
+ the seek method
+
+
+
+ the seek stop value
+
+
+
+ boolean holding whether position was updated.
+
+
+
+
+
+ Free the allocated segment @segment.
+
+
+
+
+
+ a #GstSegment
+
+
+
+
+
+ The start/position fields are set to 0 and the stop/duration
+fields are set to -1 (unknown). The default rate of 1.0 and no
+flags are set.
+
+Initialize @segment to its default values.
+
+
+
+
+
+ a #GstSegment structure.
+
+
+
+ the format of the segment.
+
+
+
+
+
+ Checks for two segments being equal. Equality here is defined
+as perfect equality, including floating point values.
+
+ %TRUE if the segments are equal, %FALSE otherwise.
+
+
+
+
+ a #GstSegment structure.
+
+
+
+ a #GstSegment structure.
+
+
+
+
+
+ Adjust the values in @segment so that @offset is applied to all
+future running-time calculations.
+
+ %TRUE if the segment could be updated successfully. If %FALSE is
+returned, @offset is not in @segment.
+
+
+
+
+ a #GstSegment structure.
+
+
+
+ the format of the segment.
+
+
+
+ the offset to apply in the segment
+
+
+
+
+
+ Convert @running_time into a position in the segment so that
+gst_segment_to_running_time() with that position returns @running_time.
+
+ the position in the segment for @running_time. This function returns
+-1 when @running_time is -1 or when it is not inside @segment.
+
+
+
+
+ a #GstSegment structure.
+
+
+
+ the format of the segment.
+
+
+
+ the running_time in the segment
+
+
+
+
+
+ Translate @running_time to the segment position using the currently configured
+segment. Compared to gst_segment_position_from_running_time() this function can
+return negative segment position.
+
+This function is typically used by elements that need to synchronize buffers
+against the clock or each other.
+
+@running_time can be any value and the result of this function for values
+outside of the segment is extrapolated.
+
+When 1 is returned, @running_time resulted in a positive position returned
+in @position.
+
+When this function returns -1, the returned @position should be negated
+to get the real negative segment position.
+
+ a 1 or -1 on success, 0 on failure.
+
+
+
+
+ a #GstSegment structure.
+
+
+
+ the format of the segment.
+
+
+
+ the running-time
+
+
+
+ the resulting position in the segment
+
+
+
+
+
+ Convert @stream_time into a position in the segment so that
+gst_segment_to_stream_time() with that position returns @stream_time.
+
+ the position in the segment for @stream_time. This function returns
+-1 when @stream_time is -1 or when it is not inside @segment.
+
+
+
+
+ a #GstSegment structure.
+
+
+
+ the format of the segment.
+
+
+
+ the stream_time in the segment
+
+
+
+
+
+ Translate @stream_time to the segment position using the currently configured
+segment. Compared to gst_segment_position_from_stream_time() this function can
+return negative segment position.
+
+This function is typically used by elements that need to synchronize buffers
+against the clock or each other.
+
+@stream_time can be any value and the result of this function for values outside
+of the segment is extrapolated.
+
+When 1 is returned, @stream_time resulted in a positive position returned
+in @position.
+
+When this function returns -1, the returned @position should be negated
+to get the real negative segment position.
+
+ a 1 or -1 on success, 0 on failure.
+
+
+
+
+ a #GstSegment structure.
+
+
+
+ the format of the segment.
+
+
+
+ the stream-time
+
+
+
+ the resulting position in the segment
+
+
+
+
+
+ Adjust the start/stop and base values of @segment such that the next valid
+buffer will be one with @running_time.
+
+ %TRUE if the segment could be updated successfully. If %FALSE is
+returned, @running_time is -1 or not in @segment.
+
+
+
+
+ a #GstSegment structure.
+
+
+
+ the format of the segment.
+
+
+
+ the running_time in the segment
+
+
+
+
+
+ Convert @running_time into a position in the segment so that
+gst_segment_to_running_time() with that position returns @running_time.
+ Use gst_segment_position_from_running_time() instead.
+
+ the position in the segment for @running_time. This function returns
+-1 when @running_time is -1 or when it is not inside @segment.
+
+
+
+
+ a #GstSegment structure.
+
+
+
+ the format of the segment.
+
+
+
+ the running_time in the segment
+
+
+
+
+
+ Translate @position to the total running time using the currently configured
+segment. Position is a value between @segment start and stop time.
+
+This function is typically used by elements that need to synchronize to the
+global clock in a pipeline. The running time is a constantly increasing value
+starting from 0. When gst_segment_init() is called, this value will reset to
+0.
+
+This function returns -1 if the position is outside of @segment start and stop.
+
+ the position as the total running time or -1 when an invalid position
+was given.
+
+
+
+
+ a #GstSegment structure.
+
+
+
+ the format of the segment.
+
+
+
+ the position in the segment
+
+
+
+
+
+ Translate @position to the total running time using the currently configured
+segment. Compared to gst_segment_to_running_time() this function can return
+negative running-time.
+
+This function is typically used by elements that need to synchronize buffers
+against the clock or eachother.
+
+@position can be any value and the result of this function for values outside
+of the segment is extrapolated.
+
+When 1 is returned, @position resulted in a positive running-time returned
+in @running_time.
+
+When this function returns -1, the returned @running_time should be negated
+to get the real negative running time.
+
+ a 1 or -1 on success, 0 on failure.
+
+
+
+
+ a #GstSegment structure.
+
+
+
+ the format of the segment.
+
+
+
+ the position in the segment
+
+
+
+ result running-time
+
+
+
+
+
+ Translate @position to stream time using the currently configured
+segment. The @position value must be between @segment start and
+stop value.
+
+This function is typically used by elements that need to operate on
+the stream time of the buffers it receives, such as effect plugins.
+In those use cases, @position is typically the buffer timestamp or
+clock time that one wants to convert to the stream time.
+The stream time is always between 0 and the total duration of the
+media stream.
+
+ the position in stream_time or -1 when an invalid position
+was given.
+
+
+
+
+ a #GstSegment structure.
+
+
+
+ the format of the segment.
+
+
+
+ the position in the segment
+
+
+
+
+
+ Translate @position to the total stream time using the currently configured
+segment. Compared to gst_segment_to_stream_time() this function can return
+negative stream-time.
+
+This function is typically used by elements that need to synchronize buffers
+against the clock or eachother.
+
+@position can be any value and the result of this function for values outside
+of the segment is extrapolated.
+
+When 1 is returned, @position resulted in a positive stream-time returned
+in @stream_time.
+
+When this function returns -1, the returned @stream_time should be negated
+to get the real negative stream time.
+
+ a 1 or -1 on success, 0 on failure.
+
+
+
+
+ a #GstSegment structure.
+
+
+
+ the format of the segment.
+
+
+
+ the position in the segment
+
+
+
+ result stream-time
+
+
+
+
+
+
+ Flags for the GstSegment structure. Currently mapped to the corresponding
+values of the seek flags.
+
+ no flags
+
+
+ reset the pipeline running_time to the segment
+ running_time
+
+
+ perform skip playback (Since 1.6)
+
+
+ Deprecated backward compatibility flag, replaced
+ by @GST_SEGMENT_FLAG_TRICKMODE
+
+
+ send SEGMENT_DONE instead of EOS
+
+
+ Decode only keyframes, where
+ possible (Since 1.6)
+
+
+ Do not decode any audio, where
+ possible (Since 1.6)
+
+
+
+
+ Try to retrieve as much information as
+ possible when getting the stack trace
+
+
+
+ The possible states an element can be in. States can be changed using
+gst_element_set_state() and checked using gst_element_get_state().
+
+ no pending state.
+
+
+ the NULL state or initial state of an element.
+
+
+ the element is ready to go to PAUSED.
+
+
+ the element is PAUSED, it is ready to accept and
+ process data. Sink elements however only accept one
+ buffer and then block.
+
+
+ the element is PLAYING, the #GstClock is running and
+ the data is flowing.
+
+
+
+ These are the different state changes an element goes through.
+%GST_STATE_NULL ⇒ %GST_STATE_PLAYING is called an upwards state change
+and %GST_STATE_PLAYING ⇒ %GST_STATE_NULL a downwards state change.
+
+ state change from NULL to READY.
+ * The element must check if the resources it needs are available. Device
+ sinks and -sources typically try to probe the device to constrain their
+ caps.
+ * The element opens the device (in case feature need to be probed).
+
+
+ state change from READY to PAUSED.
+ * The element pads are activated in order to receive data in PAUSED.
+ Streaming threads are started.
+ * Some elements might need to return %GST_STATE_CHANGE_ASYNC and complete
+ the state change when they have enough information. It is a requirement
+ for sinks to return %GST_STATE_CHANGE_ASYNC and complete the state change
+ when they receive the first buffer or %GST_EVENT_EOS (preroll).
+ Sinks also block the dataflow when in PAUSED.
+ * A pipeline resets the running_time to 0.
+ * Live sources return %GST_STATE_CHANGE_NO_PREROLL and don't generate data.
+
+
+ state change from PAUSED to PLAYING.
+ * Most elements ignore this state change.
+ * The pipeline selects a #GstClock and distributes this to all the children
+ before setting them to PLAYING. This means that it is only allowed to
+ synchronize on the #GstClock in the PLAYING state.
+ * The pipeline uses the #GstClock and the running_time to calculate the
+ base_time. The base_time is distributed to all children when performing
+ the state change.
+ * Sink elements stop blocking on the preroll buffer or event and start
+ rendering the data.
+ * Sinks can post %GST_MESSAGE_EOS in the PLAYING state. It is not allowed
+ to post %GST_MESSAGE_EOS when not in the PLAYING state.
+ * While streaming in PAUSED or PLAYING elements can create and remove
+ sometimes pads.
+ * Live sources start generating data and return %GST_STATE_CHANGE_SUCCESS.
+
+
+ state change from PLAYING to PAUSED.
+ * Most elements ignore this state change.
+ * The pipeline calculates the running_time based on the last selected
+ #GstClock and the base_time. It stores this information to continue
+ playback when going back to the PLAYING state.
+ * Sinks unblock any #GstClock wait calls.
+ * When a sink does not have a pending buffer to play, it returns
+ #GST_STATE_CHANGE_ASYNC from this state change and completes the state
+ change when it receives a new buffer or an %GST_EVENT_EOS.
+ * Any queued %GST_MESSAGE_EOS items are removed since they will be reposted
+ when going back to the PLAYING state. The EOS messages are queued in
+ #GstBin containers.
+ * Live sources stop generating data and return %GST_STATE_CHANGE_NO_PREROLL.
+
+
+ state change from PAUSED to READY.
+ * Sinks unblock any waits in the preroll.
+ * Elements unblock any waits on devices
+ * Chain or get_range functions return %GST_FLOW_FLUSHING.
+ * The element pads are deactivated so that streaming becomes impossible and
+ all streaming threads are stopped.
+ * The sink forgets all negotiated formats
+ * Elements remove all sometimes pads
+
+
+ state change from READY to NULL.
+ * Elements close devices
+ * Elements reset any internal state.
+
+
+ state change from NULL to NULL. (Since 1.14)
+
+
+ state change from READY to READY,
+This might happen when going to PAUSED asynchronously failed, in that case
+elements should make sure they are in a proper, coherent READY state. (Since 1.14)
+
+
+ state change from PAUSED to PAUSED.
+This might happen when elements were in PLAYING state and 'lost state',
+they should make sure to go back to real 'PAUSED' state (prerolling for example). (Since 1.14)
+
+
+ state change from PLAYING to PLAYING. (Since 1.14)
+
+
+ Gets a string representing the given state transition.
+
+ a string with the name of the state
+ result.
+
+
+
+
+ a #GstStateChange to get the name of.
+
+
+
+
+
+
+ The possible return values from a state change function such as
+gst_element_set_state(). Only @GST_STATE_CHANGE_FAILURE is a real failure.
+
+ the state change failed
+
+
+ the state change succeeded
+
+
+ the state change will happen asynchronously
+
+
+ the state change succeeded but the element
+ cannot produce data in %GST_STATE_PAUSED.
+ This typically happens with live sources.
+
+
+
+ Datastructure to initialize #GstCaps from a string description usually
+used in conjunction with GST_STATIC_CAPS() and gst_static_caps_get() to
+instantiate a #GstCaps.
+
+ the cached #GstCaps
+
+
+
+ a string describing a caps
+
+
+
+
+
+
+
+
+ Clean up the cached caps contained in @static_caps.
+
+
+
+
+
+ the #GstStaticCaps to clean
+
+
+
+
+
+ Converts a #GstStaticCaps to a #GstCaps.
+
+ a pointer to the #GstCaps. Unref
+ after usage. Since the core holds an additional ref to the
+ returned caps, use gst_caps_make_writable() on the returned caps
+ to modify it.
+
+
+
+
+ the #GstStaticCaps to convert
+
+
+
+
+
+
+ Structure describing the #GstStaticPadTemplate.
+
+ the name of the template
+
+
+
+ the direction of the template
+
+
+
+ the presence of the template
+
+
+
+ the caps of the template.
+
+
+
+ Converts a #GstStaticPadTemplate into a #GstPadTemplate.
+
+ a new #GstPadTemplate.
+
+
+
+
+ the static pad template
+
+
+
+
+
+ Gets the capabilities of the static pad template.
+
+ the #GstCaps of the static pad template.
+Unref after usage. Since the core holds an additional
+ref to the returned caps, use gst_caps_make_writable()
+on the returned caps to modify it.
+
+
+
+
+ a #GstStaticPadTemplate to get capabilities of.
+
+
+
+
+
+
+ A high-level object representing a single stream. It might be backed, or
+not, by an actual flow of data in a pipeline (#GstPad).
+
+A #GstStream does not care about data changes (such as decoding, encoding,
+parsing,...) as long as the underlying data flow corresponds to the same
+high-level flow (ex: a certain audio track).
+
+A #GstStream contains all the information pertinent to a stream, such as
+stream-id, tags, caps, type, ...
+
+Elements can subclass a #GstStream for internal usage (to contain information
+pertinent to streams of data).
+
+ Create a new #GstStream for the given @stream_id, @caps, @type
+and @flags
+
+ The new #GstStream
+
+
+
+
+ the id for the new stream. If %NULL,
+a new one will be automatically generated
+
+
+
+ the #GstCaps of the stream
+
+
+
+ the #GstStreamType of the stream
+
+
+
+ the #GstStreamFlags of the stream
+
+
+
+
+
+ Retrieve the caps for @stream, if any
+
+ The #GstCaps for @stream
+
+
+
+
+ a #GstStream
+
+
+
+
+
+ Retrieve the current stream flags for @stream
+
+ The #GstStreamFlags for @stream
+
+
+
+
+ a #GstStream
+
+
+
+
+
+ Returns the stream ID of @stream.
+
+ the stream ID of @stream. Only valid
+during the lifetime of @stream.
+
+
+
+
+ a #GstStream
+
+
+
+
+
+ Retrieve the stream type for @stream
+
+ The #GstStreamType for @stream
+
+
+
+
+ a #GstStream
+
+
+
+
+
+ Retrieve the tags for @stream, if any
+
+ The #GstTagList for @stream
+
+
+
+
+ a #GstStream
+
+
+
+
+
+ Set the caps for the #GstStream
+
+
+
+
+
+ a #GstStream
+
+
+
+ a #GstCaps
+
+
+
+
+
+ Set the @flags for the @stream.
+
+
+
+
+
+ a #GstStream
+
+
+
+ the flags to set on @stream
+
+
+
+
+
+ Set the stream type of @stream
+
+
+
+
+
+ a #GstStream
+
+
+
+ the type to set on @stream
+
+
+
+
+
+ Set the tags for the #GstStream
+
+
+
+
+
+ a #GstStream
+
+
+
+ a #GstTagList
+
+
+
+
+
+ The #GstCaps of the #GstStream.
+
+
+
+
+
+
+ The unique identifier of the #GstStream. Can only be set at construction
+time.
+
+
+
+ The #GstStreamType of the #GstStream. Can only be set at construction time.
+
+
+
+ The #GstTagList of the #GstStream.
+
+
+
+
+
+
+ The Stream Identifier for this #GstStream
+
+
+
+
+
+
+
+
+
+
+
+
+ GstStream class structure
+
+ the parent class structure
+
+
+
+
+
+
+
+
+
+ A collection of #GstStream that are available.
+
+A #GstStreamCollection will be provided by elements that can make those
+streams available. Applications can use the collection to show the user
+what streams are available by using %gst_stream_collection_get_stream()
+
+Once posted, a #GstStreamCollection is immutable. Updates are made by sending
+a new #GstStreamCollection message, which may or may not share some of
+the #GstStream objects from the collection it replaces. The receiver can check
+the sender of a stream collection message to know which collection is
+obsoleted.
+
+Several elements in a pipeline can provide #GstStreamCollection.
+
+Applications can activate streams from a collection by using the
+#GST_EVENT_SELECT_STREAMS event on a pipeline, bin or element.
+
+ Create a new #GstStreamCollection.
+
+ The new #GstStreamCollection.
+
+
+
+
+ The stream id of the parent stream
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Add the given @stream to the @collection.
+
+ %TRUE if the @stream was properly added, else %FALSE
+
+
+
+
+ a #GstStreamCollection
+
+
+
+ the #GstStream to add
+
+
+
+
+
+ Get the number of streams this collection contains
+
+ The number of streams that @collection contains
+
+
+
+
+ a #GstStreamCollection
+
+
+
+
+
+ Retrieve the #GstStream with index @index from the collection.
+
+The caller should not modify the returned #GstStream
+
+ A #GstStream
+
+
+
+
+ a #GstStreamCollection
+
+
+
+ Index of the stream to retrieve
+
+
+
+
+
+ Returns the upstream id of the @collection.
+
+ The upstream id
+
+
+
+
+ a #GstStreamCollection
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ GstStreamCollection class structure
+
+ the parent class structure
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Stream errors are for anything related to the stream being processed:
+format errors, media type errors, ...
+They're typically used by decoders, demuxers, converters, ...
+
+ a general error which doesn't fit in any other
+category. Make sure you add a custom message to the error call.
+
+
+ do not use this except as a placeholder for
+deciding where to go while developing code.
+
+
+ use this when you do not want to implement
+this functionality yet.
+
+
+ used when the element doesn't know the
+stream's type.
+
+
+ used when the element doesn't handle this type
+of stream.
+
+
+ used when there's no codec to handle the
+stream's type.
+
+
+ used when decoding fails.
+
+
+ used when encoding fails.
+
+
+ used when demuxing fails.
+
+
+ used when muxing fails.
+
+
+ used when the stream is of the wrong format
+(for example, wrong caps).
+
+
+ used when the stream is encrypted and can't be
+decrypted because this is not supported by the element.
+
+
+ used when the stream is encrypted and
+can't be decrypted because no suitable key is available.
+
+
+ the number of stream error types.
+
+
+
+
+
+
+
+
+
+ This stream has no special attributes
+
+
+ This stream is a sparse stream (e.g. a subtitle
+ stream), data may flow only in irregular intervals with large gaps in
+ between.
+
+
+ This stream should be selected by default. This
+ flag may be used by demuxers to signal that a stream should be selected
+ by default in a playback scenario.
+
+
+ This stream should not be selected by default.
+ This flag may be used by demuxers to signal that a stream should not
+ be selected by default in a playback scenario, but only if explicitly
+ selected by the user (e.g. an audio track for the hard of hearing or
+ a director's commentary track).
+
+
+
+
+
+ The type of a %GST_MESSAGE_STREAM_STATUS. The stream status messages inform the
+application of new streaming threads and their status.
+
+ A new thread need to be created.
+
+
+ a thread entered its loop function
+
+
+ a thread left its loop function
+
+
+ a thread is destroyed
+
+
+ a thread is started
+
+
+ a thread is paused
+
+
+ a thread is stopped
+
+
+
+ #GstStreamType describes a high level classification set for
+flows of data in #GstStream objects.
+
+Note that this is a flag, and therefore users should not assume it
+will be a single value. Do not use the equality operator for checking
+whether a stream is of a certain type.
+
+ The stream is of unknown (unclassified) type.
+
+
+ The stream is of audio data
+
+
+ The stream carries video data
+
+
+ The stream is a muxed container type
+
+
+ The stream contains subtitle / subpicture data.
+
+
+ Get a descriptive string for a given #GstStreamType
+
+ A string describing the stream type
+
+
+
+
+ a #GstStreamType
+
+
+
+
+
+
+ A #GstStructure is a collection of key/value pairs. The keys are expressed
+as GQuarks and the values can be of any GType.
+
+In addition to the key/value pairs, a #GstStructure also has a name. The name
+starts with a letter and can be filled by letters, numbers and any of "/-_.:".
+
+#GstStructure is used by various GStreamer subsystems to store information
+in a flexible and extensible way. A #GstStructure does not have a refcount
+because it usually is part of a higher level object such as #GstCaps,
+#GstMessage, #GstEvent, #GstQuery. It provides a means to enforce mutability
+using the refcount of the parent with the gst_structure_set_parent_refcount()
+method.
+
+A #GstStructure can be created with gst_structure_new_empty() or
+gst_structure_new(), which both take a name and an optional set of
+key/value pairs along with the types of the values.
+
+Field values can be changed with gst_structure_set_value() or
+gst_structure_set().
+
+Field values can be retrieved with gst_structure_get_value() or the more
+convenient gst_structure_get_*() functions.
+
+Fields can be removed with gst_structure_remove_field() or
+gst_structure_remove_fields().
+
+Strings in structures must be ASCII or UTF-8 encoded. Other encodings are
+not allowed. Strings may be %NULL however.
+
+Be aware that the current #GstCaps / #GstStructure serialization into string
+has limited support for nested #GstCaps / #GstStructure fields. It can only
+support one level of nesting. Using more levels will lead to unexpected
+behavior when using serialization features, such as gst_caps_to_string() or
+gst_value_serialize() and their counterparts.
+
+ the GType of a structure
+
+
+
+
+
+
+ Creates a new #GstStructure with the given name. Parses the
+list of variable arguments and sets fields to the values listed.
+Variable arguments should be passed as field name, field type,
+and value. Last variable argument should be %NULL.
+
+Free-function: gst_structure_free
+
+ a new #GstStructure
+
+
+
+
+ name of new structure
+
+
+
+ name of first field to set
+
+
+
+ additional arguments
+
+
+
+
+
+ Creates a new, empty #GstStructure with the given @name.
+
+See gst_structure_set_name() for constraints on the @name parameter.
+
+Free-function: gst_structure_free
+
+ a new, empty #GstStructure
+
+
+
+
+ name of new structure
+
+
+
+
+
+ Creates a #GstStructure from a string representation.
+If end is not %NULL, a pointer to the place inside the given string
+where parsing ended will be returned.
+
+The current implementation of serialization will lead to unexpected results
+when there are nested #GstCaps / #GstStructure deeper than one level.
+
+Free-function: gst_structure_free
+
+ a new #GstStructure or %NULL
+ when the string could not be parsed. Free with
+ gst_structure_free() after use.
+
+
+
+
+ a string representation of a #GstStructure
+
+
+
+
+
+ Creates a new #GstStructure with the given name as a GQuark, followed by
+fieldname quark, GType, argument(s) "triplets" in the same format as
+gst_structure_id_set(). Basically a convenience wrapper around
+gst_structure_new_id_empty() and gst_structure_id_set().
+
+The last variable argument must be %NULL (or 0).
+
+Free-function: gst_structure_free
+
+ a new #GstStructure
+
+
+
+
+ name of new structure
+
+
+
+ the GQuark for the name of the field to set
+
+
+
+ variable arguments
+
+
+
+
+
+ Creates a new, empty #GstStructure with the given name as a GQuark.
+
+Free-function: gst_structure_free
+
+ a new, empty #GstStructure
+
+
+
+
+ name of new structure
+
+
+
+
+
+ Creates a new #GstStructure with the given @name. Structure fields
+are set according to the varargs in a manner similar to
+gst_structure_new().
+
+See gst_structure_set_name() for constraints on the @name parameter.
+
+Free-function: gst_structure_free
+
+ a new #GstStructure
+
+
+
+
+ name of new structure
+
+
+
+ name of first field to set
+
+
+
+ variable argument list
+
+
+
+
+
+ Tries intersecting @struct1 and @struct2 and reports whether the result
+would not be empty.
+
+ %TRUE if intersection would not be empty
+
+
+
+
+ a #GstStructure
+
+
+
+ a #GstStructure
+
+
+
+
+
+ Duplicates a #GstStructure and all its fields and values.
+
+Free-function: gst_structure_free
+
+ a new #GstStructure.
+
+
+
+
+ a #GstStructure to duplicate
+
+
+
+
+
+ Calls the provided function once for each field in the #GstStructure. In
+contrast to gst_structure_foreach(), the function may modify the fields.
+In contrast to gst_structure_map_in_place(), the field is removed from
+the structure if %FALSE is returned from the function.
+The structure must be mutable.
+
+
+
+
+
+ a #GstStructure
+
+
+
+ a function to call for each field
+
+
+
+ private data
+
+
+
+
+
+ Fixate all values in @structure using gst_value_fixate().
+@structure will be modified in-place and should be writable.
+
+
+
+
+
+ a #GstStructure
+
+
+
+
+
+ Fixates a #GstStructure by changing the given field with its fixated value.
+
+ %TRUE if the structure field could be fixated
+
+
+
+
+ a #GstStructure
+
+
+
+ a field in @structure
+
+
+
+
+
+ Fixates a #GstStructure by changing the given @field_name field to the given
+@target boolean if that field is not fixed yet.
+
+ %TRUE if the structure could be fixated
+
+
+
+
+ a #GstStructure
+
+
+
+ a field in @structure
+
+
+
+ the target value of the fixation
+
+
+
+
+
+ Fixates a #GstStructure by changing the given field to the nearest
+double to @target that is a subset of the existing field.
+
+ %TRUE if the structure could be fixated
+
+
+
+
+ a #GstStructure
+
+
+
+ a field in @structure
+
+
+
+ the target value of the fixation
+
+
+
+
+
+ Fixates a #GstStructure by changing the given field to the nearest
+fraction to @target_numerator/@target_denominator that is a subset
+of the existing field.
+
+ %TRUE if the structure could be fixated
+
+
+
+
+ a #GstStructure
+
+
+
+ a field in @structure
+
+
+
+ The numerator of the target value of the fixation
+
+
+
+ The denominator of the target value of the fixation
+
+
+
+
+
+ Fixates a #GstStructure by changing the given field to the nearest
+integer to @target that is a subset of the existing field.
+
+ %TRUE if the structure could be fixated
+
+
+
+
+ a #GstStructure
+
+
+
+ a field in @structure
+
+
+
+ the target value of the fixation
+
+
+
+
+
+ Fixates a #GstStructure by changing the given @field_name field to the given
+@target string if that field is not fixed yet.
+
+ %TRUE if the structure could be fixated
+
+
+
+
+ a #GstStructure
+
+
+
+ a field in @structure
+
+
+
+ the target value of the fixation
+
+
+
+
+
+ Calls the provided function once for each field in the #GstStructure. The
+function must not modify the fields. Also see gst_structure_map_in_place()
+and gst_structure_filter_and_map_in_place().
+
+ %TRUE if the supplied function returns %TRUE For each of the fields,
+%FALSE otherwise.
+
+
+
+
+ a #GstStructure
+
+
+
+ a function to call for each field
+
+
+
+ private data
+
+
+
+
+
+ Frees a #GstStructure and all its fields and values. The structure must not
+have a parent when this function is called.
+
+
+
+
+
+ the #GstStructure to free
+
+
+
+
+
+ Parses the variable arguments and reads fields from @structure accordingly.
+Variable arguments should be in the form field name, field type
+(as a GType), pointer(s) to a variable(s) to hold the return value(s).
+The last variable argument should be %NULL.
+
+For refcounted (mini)objects you will receive a new reference which
+you must release with a suitable _unref() when no longer needed. For
+strings and boxed types you will receive a copy which you will need to
+release with either g_free() or the suitable function for the boxed type.
+
+ %FALSE if there was a problem reading any of the fields (e.g.
+ because the field requested did not exist, or was of a type other
+ than the type specified), otherwise %TRUE.
+
+
+
+
+ a #GstStructure
+
+
+
+ the name of the first field to read
+
+
+
+ variable arguments
+
+
+
+
+
+ This is useful in language bindings where unknown #GValue types are not
+supported. This function will convert the %GST_TYPE_ARRAY and
+%GST_TYPE_LIST into a newly allocated #GValueArray and return it through
+@array. Be aware that this is slower then getting the #GValue directly.
+
+ %TRUE if the value could be set correctly. If there was no field
+with @fieldname or the existing field did not contain an int, this function
+returns %FALSE.
+
+
+
+
+ a #GstStructure
+
+
+
+ the name of a field
+
+
+
+ a pointer to a #GValueArray
+
+
+
+
+
+ Sets the boolean pointed to by @value corresponding to the value of the
+given field. Caller is responsible for making sure the field exists
+and has the correct type.
+
+ %TRUE if the value could be set correctly. If there was no field
+with @fieldname or the existing field did not contain a boolean, this
+function returns %FALSE.
+
+
+
+
+ a #GstStructure
+
+
+
+ the name of a field
+
+
+
+ a pointer to a #gboolean to set
+
+
+
+
+
+ Sets the clock time pointed to by @value corresponding to the clock time
+of the given field. Caller is responsible for making sure the field exists
+and has the correct type.
+
+ %TRUE if the value could be set correctly. If there was no field
+with @fieldname or the existing field did not contain a #GstClockTime, this
+function returns %FALSE.
+
+
+
+
+ a #GstStructure
+
+
+
+ the name of a field
+
+
+
+ a pointer to a #GstClockTime to set
+
+
+
+
+
+ Sets the date pointed to by @value corresponding to the date of the
+given field. Caller is responsible for making sure the field exists
+and has the correct type.
+
+On success @value will point to a newly-allocated copy of the date which
+should be freed with g_date_free() when no longer needed (note: this is
+inconsistent with e.g. gst_structure_get_string() which doesn't return a
+copy of the string).
+
+ %TRUE if the value could be set correctly. If there was no field
+with @fieldname or the existing field did not contain a data, this function
+returns %FALSE.
+
+
+
+
+ a #GstStructure
+
+
+
+ the name of a field
+
+
+
+ a pointer to a #GDate to set
+
+
+
+
+
+ Sets the datetime pointed to by @value corresponding to the datetime of the
+given field. Caller is responsible for making sure the field exists
+and has the correct type.
+
+On success @value will point to a reference of the datetime which
+should be unreffed with gst_date_time_unref() when no longer needed
+(note: this is inconsistent with e.g. gst_structure_get_string()
+which doesn't return a copy of the string).
+
+ %TRUE if the value could be set correctly. If there was no field
+with @fieldname or the existing field did not contain a data, this function
+returns %FALSE.
+
+
+
+
+ a #GstStructure
+
+
+
+ the name of a field
+
+
+
+ a pointer to a #GstDateTime to set
+
+
+
+
+
+ Sets the double pointed to by @value corresponding to the value of the
+given field. Caller is responsible for making sure the field exists
+and has the correct type.
+
+ %TRUE if the value could be set correctly. If there was no field
+with @fieldname or the existing field did not contain a double, this
+function returns %FALSE.
+
+
+
+
+ a #GstStructure
+
+
+
+ the name of a field
+
+
+
+ a pointer to a gdouble to set
+
+
+
+
+
+ Sets the int pointed to by @value corresponding to the value of the
+given field. Caller is responsible for making sure the field exists,
+has the correct type and that the enumtype is correct.
+
+ %TRUE if the value could be set correctly. If there was no field
+with @fieldname or the existing field did not contain an enum of the given
+type, this function returns %FALSE.
+
+
+
+
+ a #GstStructure
+
+
+
+ the name of a field
+
+
+
+ the enum type of a field
+
+
+
+ a pointer to an int to set
+
+
+
+
+
+ Finds the field with the given name, and returns the type of the
+value it contains. If the field is not found, G_TYPE_INVALID is
+returned.
+
+ the #GValue of the field
+
+
+
+
+ a #GstStructure
+
+
+
+ the name of the field
+
+
+
+
+
+ Read the GstFlagSet flags and mask out of the structure into the
+provided pointers.
+
+ %TRUE if the values could be set correctly. If there was no field
+with @fieldname or the existing field did not contain a GstFlagSet, this
+function returns %FALSE.
+
+
+
+
+ a #GstStructure
+
+
+
+ the name of a field
+
+
+
+ a pointer to a guint for the flags field
+
+
+
+ a pointer to a guint for the mask field
+
+
+
+
+
+ Sets the integers pointed to by @value_numerator and @value_denominator
+corresponding to the value of the given field. Caller is responsible
+for making sure the field exists and has the correct type.
+
+ %TRUE if the values could be set correctly. If there was no field
+with @fieldname or the existing field did not contain a GstFraction, this
+function returns %FALSE.
+
+
+
+
+ a #GstStructure
+
+
+
+ the name of a field
+
+
+
+ a pointer to an int to set
+
+
+
+ a pointer to an int to set
+
+
+
+
+
+ Sets the int pointed to by @value corresponding to the value of the
+given field. Caller is responsible for making sure the field exists
+and has the correct type.
+
+ %TRUE if the value could be set correctly. If there was no field
+with @fieldname or the existing field did not contain an int, this function
+returns %FALSE.
+
+
+
+
+ a #GstStructure
+
+
+
+ the name of a field
+
+
+
+ a pointer to an int to set
+
+
+
+
+
+ Sets the #gint64 pointed to by @value corresponding to the value of the
+given field. Caller is responsible for making sure the field exists
+and has the correct type.
+
+ %TRUE if the value could be set correctly. If there was no field
+with @fieldname or the existing field did not contain a #gint64, this function
+returns %FALSE.
+
+
+
+
+ a #GstStructure
+
+
+
+ the name of a field
+
+
+
+ a pointer to a #gint64 to set
+
+
+
+
+
+ This is useful in language bindings where unknown #GValue types are not
+supported. This function will convert the %GST_TYPE_ARRAY and
+%GST_TYPE_LIST into a newly allocated GValueArray and return it through
+@array. Be aware that this is slower then getting the #GValue directly.
+
+ %TRUE if the value could be set correctly. If there was no field
+with @fieldname or the existing field did not contain an int, this function
+returns %FALSE.
+
+Since 1.12
+
+
+
+
+ a #GstStructure
+
+
+
+ the name of a field
+
+
+
+ a pointer to a #GValueArray
+
+
+
+
+
+ Get the name of @structure as a string.
+
+ the name of the structure.
+
+
+
+
+ a #GstStructure
+
+
+
+
+
+ Get the name of @structure as a GQuark.
+
+ the quark representing the name of the structure.
+
+
+
+
+ a #GstStructure
+
+
+
+
+
+ Finds the field corresponding to @fieldname, and returns the string
+contained in the field's value. Caller is responsible for making
+sure the field exists and has the correct type.
+
+The string should not be modified, and remains valid until the next
+call to a gst_structure_*() function with the given structure.
+
+ a pointer to the string or %NULL when the
+field did not exist or did not contain a string.
+
+
+
+
+ a #GstStructure
+
+
+
+ the name of a field
+
+
+
+
+
+ Sets the uint pointed to by @value corresponding to the value of the
+given field. Caller is responsible for making sure the field exists
+and has the correct type.
+
+ %TRUE if the value could be set correctly. If there was no field
+with @fieldname or the existing field did not contain a uint, this function
+returns %FALSE.
+
+
+
+
+ a #GstStructure
+
+
+
+ the name of a field
+
+
+
+ a pointer to a uint to set
+
+
+
+
+
+ Sets the #guint64 pointed to by @value corresponding to the value of the
+given field. Caller is responsible for making sure the field exists
+and has the correct type.
+
+ %TRUE if the value could be set correctly. If there was no field
+with @fieldname or the existing field did not contain a #guint64, this function
+returns %FALSE.
+
+
+
+
+ a #GstStructure
+
+
+
+ the name of a field
+
+
+
+ a pointer to a #guint64 to set
+
+
+
+
+
+ Parses the variable arguments and reads fields from @structure accordingly.
+valist-variant of gst_structure_get(). Look at the documentation of
+gst_structure_get() for more details.
+
+ %TRUE, or %FALSE if there was a problem reading any of the fields
+
+
+
+
+ a #GstStructure
+
+
+
+ the name of the first field to read
+
+
+
+ variable arguments
+
+
+
+
+
+ Get the value of the field with name @fieldname.
+
+ the #GValue corresponding to the field with the given
+name.
+
+
+
+
+ a #GstStructure
+
+
+
+ the name of the field to get
+
+
+
+
+
+ Check if @structure contains a field named @fieldname.
+
+ %TRUE if the structure contains a field with the given name
+
+
+
+
+ a #GstStructure
+
+
+
+ the name of a field
+
+
+
+
+
+ Check if @structure contains a field named @fieldname and with GType @type.
+
+ %TRUE if the structure contains a field with the given name and type
+
+
+
+
+ a #GstStructure
+
+
+
+ the name of a field
+
+
+
+ the type of a value
+
+
+
+
+
+ Checks if the structure has the given name
+
+ %TRUE if @name matches the name of the structure.
+
+
+
+
+ a #GstStructure
+
+
+
+ structure name to check for
+
+
+
+
+
+ Parses the variable arguments and reads fields from @structure accordingly.
+Variable arguments should be in the form field id quark, field type
+(as a GType), pointer(s) to a variable(s) to hold the return value(s).
+The last variable argument should be %NULL (technically it should be a
+0 quark, but we require %NULL so compilers that support it can check for
+the %NULL terminator and warn if it's not there).
+
+This function is just like gst_structure_get() only that it is slightly
+more efficient since it saves the string-to-quark lookup in the global
+quark hashtable.
+
+For refcounted (mini)objects you will receive a new reference which
+you must release with a suitable _unref() when no longer needed. For
+strings and boxed types you will receive a copy which you will need to
+release with either g_free() or the suitable function for the boxed type.
+
+ %FALSE if there was a problem reading any of the fields (e.g.
+ because the field requested did not exist, or was of a type other
+ than the type specified), otherwise %TRUE.
+
+
+
+
+ a #GstStructure
+
+
+
+ the quark of the first field to read
+
+
+
+ variable arguments
+
+
+
+
+
+ Parses the variable arguments and reads fields from @structure accordingly.
+valist-variant of gst_structure_id_get(). Look at the documentation of
+gst_structure_id_get() for more details.
+
+ %TRUE, or %FALSE if there was a problem reading any of the fields
+
+
+
+
+ a #GstStructure
+
+
+
+ the quark of the first field to read
+
+
+
+ variable arguments
+
+
+
+
+
+ Get the value of the field with GQuark @field.
+
+ the #GValue corresponding to the field with the given
+name identifier.
+
+
+
+
+ a #GstStructure
+
+
+
+ the #GQuark of the field to get
+
+
+
+
+
+ Check if @structure contains a field named @field.
+
+ %TRUE if the structure contains a field with the given name
+
+
+
+
+ a #GstStructure
+
+
+
+ #GQuark of the field name
+
+
+
+
+
+ Check if @structure contains a field named @field and with GType @type.
+
+ %TRUE if the structure contains a field with the given name and type
+
+
+
+
+ a #GstStructure
+
+
+
+ #GQuark of the field name
+
+
+
+ the type of a value
+
+
+
+
+
+ Identical to gst_structure_set, except that field names are
+passed using the GQuark for the field name. This allows more efficient
+setting of the structure if the caller already knows the associated
+quark values.
+The last variable argument must be %NULL.
+
+
+
+
+
+ a #GstStructure
+
+
+
+ the GQuark for the name of the field to set
+
+
+
+ variable arguments
+
+
+
+
+
+ va_list form of gst_structure_id_set().
+
+
+
+
+
+ a #GstStructure
+
+
+
+ the name of the field to set
+
+
+
+ variable arguments
+
+
+
+
+
+ Sets the field with the given GQuark @field to @value. If the field
+does not exist, it is created. If the field exists, the previous
+value is replaced and freed.
+
+
+
+
+
+ a #GstStructure
+
+
+
+ a #GQuark representing a field
+
+
+
+ the new value of the field
+
+
+
+
+
+ Sets the field with the given GQuark @field to @value. If the field
+does not exist, it is created. If the field exists, the previous
+value is replaced and freed.
+
+
+
+
+
+ a #GstStructure
+
+
+
+ a #GQuark representing a field
+
+
+
+ the new value of the field
+
+
+
+
+
+ Intersects @struct1 and @struct2 and returns the intersection.
+
+ Intersection of @struct1 and @struct2
+
+
+
+
+ a #GstStructure
+
+
+
+ a #GstStructure
+
+
+
+
+
+ Tests if the two #GstStructure are equal.
+
+ %TRUE if the two structures have the same name and field.
+
+
+
+
+ a #GstStructure.
+
+
+
+ a #GstStructure.
+
+
+
+
+
+ Checks if @subset is a subset of @superset, i.e. has the same
+structure name and for all fields that are existing in @superset,
+@subset has a value that is a subset of the value in @superset.
+
+ %TRUE if @subset is a subset of @superset
+
+
+
+
+ a #GstStructure
+
+
+
+ a potentially greater #GstStructure
+
+
+
+
+
+ Calls the provided function once for each field in the #GstStructure. In
+contrast to gst_structure_foreach(), the function may modify but not delete the
+fields. The structure must be mutable.
+
+ %TRUE if the supplied function returns %TRUE For each of the fields,
+%FALSE otherwise.
+
+
+
+
+ a #GstStructure
+
+
+
+ a function to call for each field
+
+
+
+ private data
+
+
+
+
+
+ Get the number of fields in the structure.
+
+ the number of fields in the structure
+
+
+
+
+ a #GstStructure
+
+
+
+
+
+ Get the name of the given field number, counting from 0 onwards.
+
+ the name of the given field number
+
+
+
+
+ a #GstStructure
+
+
+
+ the index to get the name of
+
+
+
+
+
+ Removes all fields in a GstStructure.
+
+
+
+
+
+ a #GstStructure
+
+
+
+
+
+ Removes the field with the given name. If the field with the given
+name does not exist, the structure is unchanged.
+
+
+
+
+
+ a #GstStructure
+
+
+
+ the name of the field to remove
+
+
+
+
+
+ Removes the fields with the given names. If a field does not exist, the
+argument is ignored.
+
+
+
+
+
+ a #GstStructure
+
+
+
+ the name of the field to remove
+
+
+
+ %NULL-terminated list of more fieldnames to remove
+
+
+
+
+
+ va_list form of gst_structure_remove_fields().
+
+
+
+
+
+ a #GstStructure
+
+
+
+ the name of the field to remove
+
+
+
+ %NULL-terminated list of more fieldnames to remove
+
+
+
+
+
+ Parses the variable arguments and sets fields accordingly. Fields that
+weren't already part of the structure are added as needed.
+Variable arguments should be in the form field name, field type
+(as a GType), value(s). The last variable argument should be %NULL.
+
+
+
+
+
+ a #GstStructure
+
+
+
+ the name of the field to set
+
+
+
+ variable arguments
+
+
+
+
+
+ This is useful in language bindings where unknown GValue types are not
+supported. This function will convert a @array to %GST_TYPE_ARRAY and set
+the field specified by @fieldname. Be aware that this is slower then using
+%GST_TYPE_ARRAY in a #GValue directly.
+
+Since 1.12
+
+
+
+
+
+ a #GstStructure
+
+
+
+ the name of a field
+
+
+
+ a pointer to a #GValueArray
+
+
+
+
+
+ This is useful in language bindings where unknown GValue types are not
+supported. This function will convert a @array to %GST_TYPE_ARRAY and set
+the field specified by @fieldname. Be aware that this is slower then using
+%GST_TYPE_ARRAY in a #GValue directly.
+
+Since 1.12
+
+
+
+
+
+ a #GstStructure
+
+
+
+ the name of a field
+
+
+
+ a pointer to a #GValueArray
+
+
+
+
+
+ Sets the name of the structure to the given @name. The string
+provided is copied before being used. It must not be empty, start with a
+letter and can be followed by letters, numbers and any of "/-_.:".
+
+
+
+
+
+ a #GstStructure
+
+
+
+ the new name of the structure
+
+
+
+
+
+ Sets the parent_refcount field of #GstStructure. This field is used to
+determine whether a structure is mutable or not. This function should only be
+called by code implementing parent objects of #GstStructure, as described in
+the MT Refcounting section of the design documents.
+
+ %TRUE if the parent refcount could be set.
+
+
+
+
+ a #GstStructure
+
+
+
+ a pointer to the parent's refcount
+
+
+
+
+
+ va_list form of gst_structure_set().
+
+
+
+
+
+ a #GstStructure
+
+
+
+ the name of the field to set
+
+
+
+ variable arguments
+
+
+
+
+
+ Sets the field with the given name @field to @value. If the field
+does not exist, it is created. If the field exists, the previous
+value is replaced and freed.
+
+
+
+
+
+ a #GstStructure
+
+
+
+ the name of the field to set
+
+
+
+ the new value of the field
+
+
+
+
+
+ Sets the field with the given name @field to @value. If the field
+does not exist, it is created. If the field exists, the previous
+value is replaced and freed. The function will take ownership of @value.
+
+
+
+
+
+ a #GstStructure
+
+
+
+ the name of the field to set
+
+
+
+ the new value of the field
+
+
+
+
+
+ Converts @structure to a human-readable string representation.
+
+For debugging purposes its easier to do something like this:
+|[<!-- language="C" -->
+GST_LOG ("structure is %" GST_PTR_FORMAT, structure);
+]|
+This prints the structure in human readable form.
+
+The current implementation of serialization will lead to unexpected results
+when there are nested #GstCaps / #GstStructure deeper than one level.
+
+Free-function: g_free
+
+ a pointer to string allocated by g_malloc().
+ g_free() after usage.
+
+
+
+
+ a #GstStructure
+
+
+
+
+
+ Creates a #GstStructure from a string representation.
+If end is not %NULL, a pointer to the place inside the given string
+where parsing ended will be returned.
+
+Free-function: gst_structure_free
+
+ a new #GstStructure or %NULL
+ when the string could not be parsed. Free with
+ gst_structure_free() after use.
+
+
+
+
+ a string representation of a #GstStructure.
+
+
+
+ pointer to store the end of the string in.
+
+
+
+
+
+
+ The type of a %GST_MESSAGE_STRUCTURE_CHANGE.
+
+ Pad linking is starting or done.
+
+
+ Pad unlinking is starting or done.
+
+
+
+ A function that will be called in gst_structure_filter_and_map_in_place().
+The function may modify @value, and the value will be removed from
+the structure if %FALSE is returned.
+
+ %TRUE if the field should be preserved, %FALSE if it
+should be removed.
+
+
+
+
+ the #GQuark of the field name
+
+
+
+ the #GValue of the field
+
+
+
+ user data
+
+
+
+
+
+ A function that will be called in gst_structure_foreach(). The function may
+not modify @value.
+
+ %TRUE if the foreach operation should continue, %FALSE if
+the foreach operation should stop with %FALSE.
+
+
+
+
+ the #GQuark of the field name
+
+
+
+ the #GValue of the field
+
+
+
+ user data
+
+
+
+
+
+ A function that will be called in gst_structure_map_in_place(). The function
+may modify @value.
+
+ %TRUE if the map operation should continue, %FALSE if
+the map operation should stop with %FALSE.
+
+
+
+
+ the #GQuark of the field name
+
+
+
+ the #GValue of the field
+
+
+
+ user data
+
+
+
+
+
+ The GStreamer core provides a GstSystemClock based on the system time.
+Asynchronous callbacks are scheduled from an internal thread.
+
+Clock implementors are encouraged to subclass this systemclock as it
+implements the async notification.
+
+Subclasses can however override all of the important methods for sync and
+async notifications to implement their own callback methods or blocking
+wait operations.
+
+ Get a handle to the default system clock. The refcount of the
+clock will be increased so you need to unref the clock after
+usage.
+
+ the default clock.
+
+MT safe.
+
+
+
+
+ Sets the default system clock that can be obtained with
+gst_system_clock_obtain().
+
+This is mostly used for testing and debugging purposes when you
+want to have control over the time reported by the default system
+clock.
+
+MT safe.
+
+
+
+
+
+ a #GstClock
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ album containing this data (string)
+
+The album name as it should be displayed, e.g. 'The Jazz Guitar'
+
+
+
+ The artist of the entire album, as it should be displayed.
+
+
+
+ The artist of the entire album, as it should be sorted.
+
+
+
+ album gain in db (double)
+
+
+
+ peak of the album (double)
+
+
+
+ album containing this data, as used for sorting (string)
+
+The album name as it should be sorted, e.g. 'Jazz Guitar, The'
+
+
+
+ count of discs inside collection this disc belongs to (unsigned integer)
+
+
+
+ disc number inside a collection (unsigned integer)
+
+
+
+ Arbitrary application data (sample)
+
+Some formats allow applications to add their own arbitrary data
+into files. This data is application dependent.
+
+
+
+ Name of the application used to create the media (string)
+
+
+
+ person(s) responsible for the recording (string)
+
+The artist name as it should be displayed, e.g. 'Jimi Hendrix' or
+'The Guitar Heroes'
+
+
+
+ person(s) responsible for the recording, as used for sorting (string)
+
+The artist name as it should be sorted, e.g. 'Hendrix, Jimi' or
+'Guitar Heroes, The'
+
+
+
+ generic file attachment (sample) (sample taglist should specify the content
+type and if possible set "filename" to the file name of the
+attachment)
+
+
+
+ codec the audio data is stored in (string)
+
+
+
+ number of beats per minute in audio (double)
+
+
+
+ exact or average bitrate in bits/s (unsigned integer)
+
+
+
+ codec the data is stored in (string)
+
+
+
+ free text commenting the data (string)
+
+
+
+ person(s) who composed the recording (string)
+
+
+
+ The composer's name, used for sorting (string)
+
+
+
+ conductor/performer refinement (string)
+
+
+
+ contact information (string)
+
+
+
+ container format the data is stored in (string)
+
+
+
+ copyright notice of the data (string)
+
+
+
+ URI to location where copyright details can be found (string)
+
+
+
+ date the data was created (#GDate structure)
+
+
+
+ date and time the data was created (#GstDateTime structure)
+
+
+
+ short text describing the content of the data (string)
+
+
+
+ Manufacturer of the device used to create the media (string)
+
+
+
+ Model of the device used to create the media (string)
+
+
+
+ length in GStreamer time units (nanoseconds) (unsigned 64-bit integer)
+
+
+
+ name of the person or organisation that encoded the file. May contain a
+copyright message if the person or organisation also holds the copyright
+(string)
+
+Note: do not use this field to describe the encoding application. Use
+#GST_TAG_APPLICATION_NAME or #GST_TAG_COMMENT for that.
+
+
+
+ encoder used to encode this stream (string)
+
+
+
+ version of the encoder used to encode this stream (unsigned integer)
+
+
+
+ key/value text commenting the data (string)
+
+Must be in the form of 'key=comment' or
+'key[lc]=comment' where 'lc' is an ISO-639
+language code.
+
+This tag is used for unknown Vorbis comment tags,
+unknown APE tags and certain ID3v2 comment fields.
+
+
+
+ genre this data belongs to (string)
+
+
+
+ Indicates the direction the device is pointing to when capturing
+a media. It is represented as degrees in floating point representation,
+0 means the geographic north, and increases clockwise (double from 0 to 360)
+
+See also #GST_TAG_GEO_LOCATION_MOVEMENT_DIRECTION
+
+
+
+ The city (english name) where the media has been produced (string).
+
+
+
+ The country (english name) where the media has been produced (string).
+
+
+
+ geo elevation of where the media has been recorded or produced in meters
+according to WGS84 (zero is average sea level) (double).
+
+
+
+ Represents the expected error on the horizontal positioning in
+meters (double).
+
+
+
+ geo latitude location of where the media has been recorded or produced in
+degrees according to WGS84 (zero at the equator, negative values for southern
+latitudes) (double).
+
+
+
+ geo longitude location of where the media has been recorded or produced in
+degrees according to WGS84 (zero at the prime meridian in Greenwich/UK,
+negative values for western longitudes). (double).
+
+
+
+ Indicates the movement direction of the device performing the capture
+of a media. It is represented as degrees in floating point representation,
+0 means the geographic north, and increases clockwise (double from 0 to 360)
+
+See also #GST_TAG_GEO_LOCATION_CAPTURE_DIRECTION
+
+
+
+ Speed of the capturing device when performing the capture.
+Represented in m/s. (double)
+
+See also #GST_TAG_GEO_LOCATION_MOVEMENT_DIRECTION
+
+
+
+ human readable descriptive location of where the media has been recorded or
+produced. (string).
+
+
+
+ A location 'smaller' than GST_TAG_GEO_LOCATION_CITY that specifies better
+where the media has been produced. (e.g. the neighborhood) (string).
+
+This tag has been added as this is how it is handled/named in XMP's
+Iptc4xmpcore schema.
+
+
+
+ Groups together media that are related and spans multiple tracks. An
+example are multiple pieces of a concerto. (string)
+
+
+
+ Homepage for this media (i.e. artist or movie homepage) (string)
+
+
+
+ image (sample) (sample taglist should specify the content type and preferably
+also set "image-type" field as #GstTagImageType)
+
+
+
+ Represents the 'Orientation' tag from EXIF. Defines how the image
+should be rotated and mirrored for display. (string)
+
+This tag has a predefined set of allowed values:
+ "rotate-0"
+ "rotate-90"
+ "rotate-180"
+ "rotate-270"
+ "flip-rotate-0"
+ "flip-rotate-90"
+ "flip-rotate-180"
+ "flip-rotate-270"
+
+The naming is adopted according to a possible transformation to perform
+on the image to fix its orientation, obviously equivalent operations will
+yield the same result.
+
+Rotations indicated by the values are in clockwise direction and
+'flip' means an horizontal mirroring.
+
+
+
+ Information about the people behind a remix and similar
+interpretations of another existing piece (string)
+
+
+
+ International Standard Recording Code - see http://www.ifpi.org/isrc/ (string)
+
+
+
+ comma separated keywords describing the content (string).
+
+
+
+ ISO-639-2 or ISO-639-1 code for the language the content is in (string)
+
+There is utility API in libgsttag in gst-plugins-base to obtain a translated
+language name from the language code: gst_tag_get_language_name()
+
+
+
+ Name of the language the content is in (string)
+
+Free-form name of the language the content is in, if a language code
+is not available. This tag should not be set in addition to a language
+code. It is undefined what language or locale the language name is in.
+
+
+
+ license of data (string)
+
+
+
+ URI to location where license details can be found (string)
+
+
+
+ Origin of media as a URI (location, where the original of the file or stream
+is hosted) (string)
+
+
+
+ The lyrics of the media (string)
+
+
+
+ maximum bitrate in bits/s (unsigned integer)
+
+
+
+ <ulink url="http://en.wikipedia.org/wiki/Note#Note_designation_in_accordance_with_octave_name">Midi note number</ulink>
+of the audio track. This is useful for sample instruments and in particular
+for multi-samples.
+
+
+
+ minimum bitrate in bits/s (unsigned integer)
+
+
+
+ nominal bitrate in bits/s (unsigned integer). The actual bitrate might be
+different from this target bitrate.
+
+
+
+ organization (string)
+
+
+
+ person(s) performing (string)
+
+
+
+ image that is meant for preview purposes, e.g. small icon-sized version
+(sample) (sample taglist should specify the content type)
+
+
+
+ Any private data that may be contained in tags (sample).
+
+It is represented by #GstSample in which #GstBuffer contains the
+binary data and the sample's info #GstStructure may contain any
+extra information that identifies the origin or meaning of the data.
+
+Private frames in ID3v2 tags ('PRIV' frames) will be represented
+using this tag, in which case the GstStructure will be named
+"ID3PrivateFrame" and contain a field named "owner" of type string
+which contains the owner-identification string from the tag.
+
+
+
+ Name of the label or publisher (string)
+
+
+
+ reference level of track and album gain values (double)
+
+
+
+ serial number of track (unsigned integer)
+
+
+
+ Number of the episode within a season/show (unsigned integer)
+
+
+
+ Name of the show, used for displaying (string)
+
+
+
+ Number of the season of a show/series (unsigned integer)
+
+
+
+ Name of the show, used for sorting (string)
+
+
+
+ codec/format the subtitle data is stored in (string)
+
+
+
+ commonly used title (string)
+
+The title as it should be displayed, e.g. 'The Doll House'
+
+
+
+ commonly used title, as used for sorting (string)
+
+The title as it should be sorted, e.g. 'Doll House, The'
+
+
+
+ count of tracks inside collection this track belongs to (unsigned integer)
+
+
+
+ track gain in db (double)
+
+
+
+ track number inside a collection (unsigned integer)
+
+
+
+ peak of the track (double)
+
+
+
+ Rating attributed by a person (likely the application user).
+The higher the value, the more the user likes this media
+(unsigned int from 0 to 100)
+
+
+
+ version of this data (string)
+
+
+
+ codec the video data is stored in (string)
+
+
+
+ A string that can be used in printf-like format strings to display a
+#GstClockTime value in h:m:s format. Use GST_TIME_ARGS() to construct
+the matching arguments.
+
+Example:
+|[<!-- language="C" -->
+printf("%" GST_TIME_FORMAT "\n", GST_TIME_ARGS(ts));
+]|
+
+
+
+ Special value for the repeat_count set in gst_toc_entry_set_loop() or
+returned by gst_toc_entry_set_loop() to indicate infinite looping.
+
+
+
+ Extra tag flags used when registering tags.
+
+ undefined flag
+
+
+ tag is meta data
+
+
+ tag is encoded
+
+
+ tag is decoded
+
+
+ number of tag flags
+
+
+
+ A function that will be called in gst_tag_list_foreach(). The function may
+not modify the tag list.
+
+
+
+
+
+ the #GstTagList
+
+
+
+ a name of a tag in @list
+
+
+
+ user data
+
+
+
+
+
+ List of tags and values used to describe media metadata.
+
+Strings in structures must be ASCII or UTF-8 encoded. Other encodings are
+not allowed. Strings must not be empty or %NULL.
+
+ the parent type
+
+
+
+ Creates a new taglist and appends the values for the given tags. It expects
+tag-value pairs like gst_tag_list_add(), and a %NULL terminator after the
+last pair. The type of the values is implicit and is documented in the API
+reference, but can also be queried at runtime with gst_tag_get_type(). It
+is an error to pass a value of a type not matching the tag type into this
+function. The tag list will make copies of any arguments passed
+(e.g. strings, buffers).
+
+After creation you might also want to set a #GstTagScope on the returned
+taglist to signal if the contained tags are global or stream tags. By
+default stream scope is assumes. See gst_tag_list_set_scope().
+
+Free-function: gst_tag_list_unref
+
+ a new #GstTagList. Free with gst_tag_list_unref()
+ when no longer needed.
+
+
+
+
+ tag
+
+
+
+ %NULL-terminated list of values to set
+
+
+
+
+
+ Creates a new empty GstTagList.
+
+Free-function: gst_tag_list_unref
+
+ An empty tag list
+
+
+
+
+ Deserializes a tag list.
+
+ a new #GstTagList, or %NULL in case of an
+error.
+
+
+
+
+ a string created with gst_tag_list_to_string()
+
+
+
+
+
+ Just like gst_tag_list_new(), only that it takes a va_list argument.
+Useful mostly for language bindings.
+
+Free-function: gst_tag_list_unref
+
+ a new #GstTagList. Free with gst_tag_list_unref()
+ when no longer needed.
+
+
+
+
+ tag / value pairs to set
+
+
+
+
+
+ Sets the values for the given tags using the specified mode.
+
+
+
+
+
+ list to set tags in
+
+
+
+ the mode to use
+
+
+
+ tag
+
+
+
+ %NULL-terminated list of values to set
+
+
+
+
+
+ Sets the values for the given tags using the specified mode.
+
+
+
+
+
+ list to set tags in
+
+
+
+ the mode to use
+
+
+
+ tag
+
+
+
+ tag / value pairs to set
+
+
+
+
+
+ Sets the GValues for the given tags using the specified mode.
+
+
+
+
+
+ list to set tags in
+
+
+
+ the mode to use
+
+
+
+ tag
+
+
+
+ tag / GValue pairs to set
+
+
+
+
+
+ Sets the GValue for a given tag using the specified mode.
+
+
+
+
+
+ list to set tags in
+
+
+
+ the mode to use
+
+
+
+ tag
+
+
+
+ GValue for this tag
+
+
+
+
+
+ Sets the GValues for the given tags using the specified mode.
+
+
+
+
+
+ list to set tags in
+
+
+
+ the mode to use
+
+
+
+ tag
+
+
+
+ GValues to set
+
+
+
+
+
+ Calls the given function for each tag inside the tag list. Note that if there
+is no tag, the function won't be called at all.
+
+
+
+
+
+ list to iterate over
+
+
+
+ function to be called for each tag
+
+
+
+ user specified data
+
+
+
+
+
+ Copies the contents for the given tag into the value, merging multiple values
+into one if multiple values are associated with the tag.
+
+ %TRUE, if a value was copied, %FALSE if the tag didn't exist in the
+ given list.
+
+
+
+
+ a #GstTagList to get the tag from
+
+
+
+ tag to read out
+
+
+
+ location for the result
+
+
+
+
+
+ Gets the value that is at the given index for the given tag in the given
+list.
+
+ %TRUE, if a value was copied, %FALSE if the tag didn't exist in the
+ given list.
+
+
+
+
+ a #GstTagList to get the tag from
+
+
+
+ tag to read out
+
+
+
+ number of entry to read out
+
+
+
+ location for the result
+
+
+
+
+
+ Copies the first date for the given tag in the taglist into the variable
+pointed to by @value. Free the date with g_date_free() when it is no longer
+needed.
+
+Free-function: g_date_free
+
+ %TRUE, if a date was copied, %FALSE if the tag didn't exist in the
+ given list or if it was %NULL.
+
+
+
+
+ a #GstTagList to get the tag from
+
+
+
+ tag to read out
+
+
+
+ address of a GDate pointer
+ variable to store the result into
+
+
+
+
+
+ Gets the date that is at the given index for the given tag in the given
+list and copies it into the variable pointed to by @value. Free the date
+with g_date_free() when it is no longer needed.
+
+Free-function: g_date_free
+
+ %TRUE, if a value was copied, %FALSE if the tag didn't exist in the
+ given list or if it was %NULL.
+
+
+
+
+ a #GstTagList to get the tag from
+
+
+
+ tag to read out
+
+
+
+ number of entry to read out
+
+
+
+ location for the result
+
+
+
+
+
+ Copies the first datetime for the given tag in the taglist into the variable
+pointed to by @value. Unref the date with gst_date_time_unref() when
+it is no longer needed.
+
+Free-function: gst_date_time_unref
+
+ %TRUE, if a datetime was copied, %FALSE if the tag didn't exist in
+ the given list or if it was %NULL.
+
+
+
+
+ a #GstTagList to get the tag from
+
+
+
+ tag to read out
+
+
+
+ address of a #GstDateTime
+ pointer variable to store the result into
+
+
+
+
+
+ Gets the datetime that is at the given index for the given tag in the given
+list and copies it into the variable pointed to by @value. Unref the datetime
+with gst_date_time_unref() when it is no longer needed.
+
+Free-function: gst_date_time_unref
+
+ %TRUE, if a value was copied, %FALSE if the tag didn't exist in the
+ given list or if it was %NULL.
+
+
+
+
+ a #GstTagList to get the tag from
+
+
+
+ tag to read out
+
+
+
+ number of entry to read out
+
+
+
+ location for the result
+
+
+
+
+
+ Copies the contents for the given tag into the value, merging multiple values
+into one if multiple values are associated with the tag.
+
+ %TRUE, if a value was copied, %FALSE if the tag didn't exist in the
+ given list.
+
+
+
+
+ a #GstTagList to get the tag from
+
+
+
+ tag to read out
+
+
+
+ location for the result
+
+
+
+
+
+ Gets the value that is at the given index for the given tag in the given
+list.
+
+ %TRUE, if a value was copied, %FALSE if the tag didn't exist in the
+ given list.
+
+
+
+
+ a #GstTagList to get the tag from
+
+
+
+ tag to read out
+
+
+
+ number of entry to read out
+
+
+
+ location for the result
+
+
+
+
+
+ Copies the contents for the given tag into the value, merging multiple values
+into one if multiple values are associated with the tag.
+
+ %TRUE, if a value was copied, %FALSE if the tag didn't exist in the
+ given list.
+
+
+
+
+ a #GstTagList to get the tag from
+
+
+
+ tag to read out
+
+
+
+ location for the result
+
+
+
+
+
+ Gets the value that is at the given index for the given tag in the given
+list.
+
+ %TRUE, if a value was copied, %FALSE if the tag didn't exist in the
+ given list.
+
+
+
+
+ a #GstTagList to get the tag from
+
+
+
+ tag to read out
+
+
+
+ number of entry to read out
+
+
+
+ location for the result
+
+
+
+
+
+ Copies the contents for the given tag into the value, merging multiple values
+into one if multiple values are associated with the tag.
+
+ %TRUE, if a value was copied, %FALSE if the tag didn't exist in the
+ given list.
+
+
+
+
+ a #GstTagList to get the tag from
+
+
+
+ tag to read out
+
+
+
+ location for the result
+
+
+
+
+
+ Copies the contents for the given tag into the value, merging multiple values
+into one if multiple values are associated with the tag.
+
+ %TRUE, if a value was copied, %FALSE if the tag didn't exist in the
+ given list.
+
+
+
+
+ a #GstTagList to get the tag from
+
+
+
+ tag to read out
+
+
+
+ location for the result
+
+
+
+
+
+ Gets the value that is at the given index for the given tag in the given
+list.
+
+ %TRUE, if a value was copied, %FALSE if the tag didn't exist in the
+ given list.
+
+
+
+
+ a #GstTagList to get the tag from
+
+
+
+ tag to read out
+
+
+
+ number of entry to read out
+
+
+
+ location for the result
+
+
+
+
+
+ Gets the value that is at the given index for the given tag in the given
+list.
+
+ %TRUE, if a value was copied, %FALSE if the tag didn't exist in the
+ given list.
+
+
+
+
+ a #GstTagList to get the tag from
+
+
+
+ tag to read out
+
+
+
+ number of entry to read out
+
+
+
+ location for the result
+
+
+
+
+
+ Copies the contents for the given tag into the value, merging multiple values
+into one if multiple values are associated with the tag.
+
+ %TRUE, if a value was copied, %FALSE if the tag didn't exist in the
+ given list.
+
+
+
+
+ a #GstTagList to get the tag from
+
+
+
+ tag to read out
+
+
+
+ location for the result
+
+
+
+
+
+ Gets the value that is at the given index for the given tag in the given
+list.
+
+ %TRUE, if a value was copied, %FALSE if the tag didn't exist in the
+ given list.
+
+
+
+
+ a #GstTagList to get the tag from
+
+
+
+ tag to read out
+
+
+
+ number of entry to read out
+
+
+
+ location for the result
+
+
+
+
+
+ Copies the first sample for the given tag in the taglist into the variable
+pointed to by @sample. Free the sample with gst_sample_unref() when it is
+no longer needed. You can retrieve the buffer from the sample using
+gst_sample_get_buffer() and the associated caps (if any) with
+gst_sample_get_caps().
+
+Free-function: gst_sample_unref
+
+ %TRUE, if a sample was returned, %FALSE if the tag didn't exist in
+ the given list or if it was %NULL.
+
+
+
+
+ a #GstTagList to get the tag from
+
+
+
+ tag to read out
+
+
+
+ address of a GstSample
+ pointer variable to store the result into
+
+
+
+
+
+ Gets the sample that is at the given index for the given tag in the given
+list and copies it into the variable pointed to by @sample. Free the sample
+with gst_sample_unref() when it is no longer needed. You can retrieve the
+buffer from the sample using gst_sample_get_buffer() and the associated
+caps (if any) with gst_sample_get_caps().
+
+Free-function: gst_sample_unref
+
+ %TRUE, if a sample was copied, %FALSE if the tag didn't exist in the
+ given list or if it was %NULL.
+
+
+
+
+ a #GstTagList to get the tag from
+
+
+
+ tag to read out
+
+
+
+ number of entry to read out
+
+
+
+ address of a GstSample
+ pointer variable to store the result into
+
+
+
+
+
+ Gets the scope of @list.
+
+ The scope of @list
+
+
+
+
+ a #GstTagList
+
+
+
+
+
+ Copies the contents for the given tag into the value, possibly merging
+multiple values into one if multiple values are associated with the tag.
+
+Use gst_tag_list_get_string_index (list, tag, 0, value) if you want
+to retrieve the first string associated with this tag unmodified.
+
+The resulting string in @value will be in UTF-8 encoding and should be
+freed by the caller using g_free when no longer needed. The
+returned string is also guaranteed to be non-%NULL and non-empty.
+
+Free-function: g_free
+
+ %TRUE, if a value was copied, %FALSE if the tag didn't exist in the
+ given list.
+
+
+
+
+ a #GstTagList to get the tag from
+
+
+
+ tag to read out
+
+
+
+ location for the result
+
+
+
+
+
+ Gets the value that is at the given index for the given tag in the given
+list.
+
+The resulting string in @value will be in UTF-8 encoding and should be
+freed by the caller using g_free when no longer needed. The
+returned string is also guaranteed to be non-%NULL and non-empty.
+
+Free-function: g_free
+
+ %TRUE, if a value was copied, %FALSE if the tag didn't exist in the
+ given list.
+
+
+
+
+ a #GstTagList to get the tag from
+
+
+
+ tag to read out
+
+
+
+ number of entry to read out
+
+
+
+ location for the result
+
+
+
+
+
+ Checks how many value are stored in this tag list for the given tag.
+
+ The number of tags stored
+
+
+
+
+ a taglist
+
+
+
+ the tag to query
+
+
+
+
+
+ Copies the contents for the given tag into the value, merging multiple values
+into one if multiple values are associated with the tag.
+
+ %TRUE, if a value was copied, %FALSE if the tag didn't exist in the
+ given list.
+
+
+
+
+ a #GstTagList to get the tag from
+
+
+
+ tag to read out
+
+
+
+ location for the result
+
+
+
+
+
+ Copies the contents for the given tag into the value, merging multiple values
+into one if multiple values are associated with the tag.
+
+ %TRUE, if a value was copied, %FALSE if the tag didn't exist in the
+ given list.
+
+
+
+
+ a #GstTagList to get the tag from
+
+
+
+ tag to read out
+
+
+
+ location for the result
+
+
+
+
+
+ Gets the value that is at the given index for the given tag in the given
+list.
+
+ %TRUE, if a value was copied, %FALSE if the tag didn't exist in the
+ given list.
+
+
+
+
+ a #GstTagList to get the tag from
+
+
+
+ tag to read out
+
+
+
+ number of entry to read out
+
+
+
+ location for the result
+
+
+
+
+
+ Gets the value that is at the given index for the given tag in the given
+list.
+
+ %TRUE, if a value was copied, %FALSE if the tag didn't exist in the
+ given list.
+
+
+
+
+ a #GstTagList to get the tag from
+
+
+
+ tag to read out
+
+
+
+ number of entry to read out
+
+
+
+ location for the result
+
+
+
+
+
+ Gets the value that is at the given index for the given tag in the given
+list.
+
+ The GValue for the specified
+ entry or %NULL if the tag wasn't available or the tag
+ doesn't have as many entries
+
+
+
+
+ a #GstTagList
+
+
+
+ tag to read out
+
+
+
+ number of entry to read out
+
+
+
+
+
+ Inserts the tags of the @from list into the first list using the given mode.
+
+
+
+
+
+ list to merge into
+
+
+
+ list to merge from
+
+
+
+ the mode to use
+
+
+
+
+
+ Checks if the given taglist is empty.
+
+ %TRUE if the taglist is empty, otherwise %FALSE.
+
+
+
+
+ A #GstTagList.
+
+
+
+
+
+ Checks if the two given taglists are equal.
+
+ %TRUE if the taglists are equal, otherwise %FALSE
+
+
+
+
+ a #GstTagList.
+
+
+
+ a #GstTagList.
+
+
+
+
+
+ Merges the two given lists into a new list. If one of the lists is %NULL, a
+copy of the other is returned. If both lists are %NULL, %NULL is returned.
+
+Free-function: gst_tag_list_unref
+
+ the new list
+
+
+
+
+ first list to merge
+
+
+
+ second list to merge
+
+
+
+ the mode to use
+
+
+
+
+
+ Get the number of tags in @list.
+
+ The number of tags in @list.
+
+
+
+
+ A #GstTagList.
+
+
+
+
+
+ Get the name of the tag in @list at @index.
+
+ The name of the tag at @index.
+
+
+
+
+ A #GstTagList.
+
+
+
+ the index
+
+
+
+
+
+ Peeks at the value that is at the given index for the given tag in the given
+list.
+
+The resulting string in @value will be in UTF-8 encoding and doesn't need
+to be freed by the caller. The returned string is also guaranteed to
+be non-%NULL and non-empty.
+
+ %TRUE, if a value was set, %FALSE if the tag didn't exist in the
+ given list.
+
+
+
+
+ a #GstTagList to get the tag from
+
+
+
+ tag to read out
+
+
+
+ number of entry to read out
+
+
+
+ location for the result
+
+
+
+
+
+ Removes the given tag from the taglist.
+
+
+
+
+
+ list to remove tag from
+
+
+
+ tag to remove
+
+
+
+
+
+ Sets the scope of @list to @scope. By default the scope
+of a taglist is stream scope.
+
+
+
+
+
+ a #GstTagList
+
+
+
+ new scope for @list
+
+
+
+
+
+ Serializes a tag list to a string.
+
+ a newly-allocated string, or %NULL in case of
+ an error. The string must be freed with g_free() when no longer
+ needed.
+
+
+
+
+ a #GstTagList
+
+
+
+
+
+ Copies the contents for the given tag into the value,
+merging multiple values into one if multiple values are associated
+with the tag.
+You must g_value_unset() the value after use.
+
+ %TRUE, if a value was copied, %FALSE if the tag didn't exist in the
+ given list.
+
+
+
+
+ uninitialized #GValue to copy into
+
+
+
+ list to get the tag from
+
+
+
+ tag to read out
+
+
+
+
+
+
+ A function for merging multiple values of a tag used when registering
+tags.
+
+
+
+
+
+ the destination #GValue
+
+
+
+ the source #GValue
+
+
+
+
+
+ The different tag merging modes are basically replace, overwrite and append,
+but they can be seen from two directions. Given two taglists: (A) the tags
+already in the element and (B) the ones that are supplied to the element (
+e.g. via gst_tag_setter_merge_tags() / gst_tag_setter_add_tags() or a
+%GST_EVENT_TAG), how are these tags merged?
+In the table below this is shown for the cases that a tag exists in the list
+(A) or does not exists (!A) and combinations thereof.
+
+<table frame="all" colsep="1" rowsep="1">
+ <title>merge mode</title>
+ <tgroup cols='5' align='left'>
+ <thead>
+ <row>
+ <entry>merge mode</entry>
+ <entry>A + B</entry>
+ <entry>A + !B</entry>
+ <entry>!A + B</entry>
+ <entry>!A + !B</entry>
+ </row>
+ </thead>
+ <tbody>
+ <row>
+ <entry>REPLACE_ALL</entry>
+ <entry>B</entry>
+ <entry>-</entry>
+ <entry>B</entry>
+ <entry>-</entry>
+ </row>
+ <row>
+ <entry>REPLACE</entry>
+ <entry>B</entry>
+ <entry>A</entry>
+ <entry>B</entry>
+ <entry>-</entry>
+ </row>
+ <row>
+ <entry>APPEND</entry>
+ <entry>A, B</entry>
+ <entry>A</entry>
+ <entry>B</entry>
+ <entry>-</entry>
+ </row>
+ <row>
+ <entry>PREPEND</entry>
+ <entry>B, A</entry>
+ <entry>A</entry>
+ <entry>B</entry>
+ <entry>-</entry>
+ </row>
+ <row>
+ <entry>KEEP</entry>
+ <entry>A</entry>
+ <entry>A</entry>
+ <entry>B</entry>
+ <entry>-</entry>
+ </row>
+ <row>
+ <entry>KEEP_ALL</entry>
+ <entry>A</entry>
+ <entry>A</entry>
+ <entry>-</entry>
+ <entry>-</entry>
+ </row>
+ </tbody>
+ </tgroup>
+</table>
+
+ undefined merge mode
+
+
+ replace all tags (clear list and append)
+
+
+ replace tags
+
+
+ append tags
+
+
+ prepend tags
+
+
+ keep existing tags
+
+
+ keep all existing tags
+
+
+ the number of merge modes
+
+
+
+ GstTagScope specifies if a taglist applies to the complete
+medium or only to one single stream.
+
+ tags specific to this single stream
+
+
+ global tags for the complete medium
+
+
+
+ Element interface that allows setting of media metadata.
+
+Elements that support changing a stream's metadata will implement this
+interface. Examples of such elements are 'vorbisenc', 'theoraenc' and
+'id3v2mux'.
+
+If you just want to retrieve metadata in your application then all you
+need to do is watch for tag messages on your pipeline's bus. This
+interface is only for setting metadata, not for extracting it. To set tags
+from the application, find tagsetter elements and set tags using e.g.
+gst_tag_setter_merge_tags() or gst_tag_setter_add_tags(). Also consider
+setting the #GstTagMergeMode that is used for tag events that arrive at the
+tagsetter element (default mode is to keep existing tags).
+The application should do that before the element goes to %GST_STATE_PAUSED.
+
+Elements implementing the #GstTagSetter interface often have to merge
+any tags received from upstream and the tags set by the application via
+the interface. This can be done like this:
+
+|[<!-- language="C" -->
+GstTagMergeMode merge_mode;
+const GstTagList *application_tags;
+const GstTagList *event_tags;
+GstTagSetter *tagsetter;
+GstTagList *result;
+
+tagsetter = GST_TAG_SETTER (element);
+
+merge_mode = gst_tag_setter_get_tag_merge_mode (tagsetter);
+application_tags = gst_tag_setter_get_tag_list (tagsetter);
+event_tags = (const GstTagList *) element->event_tags;
+
+GST_LOG_OBJECT (tagsetter, "merging tags, merge mode = %d", merge_mode);
+GST_LOG_OBJECT (tagsetter, "event tags: %" GST_PTR_FORMAT, event_tags);
+GST_LOG_OBJECT (tagsetter, "set tags: %" GST_PTR_FORMAT, application_tags);
+
+result = gst_tag_list_merge (application_tags, event_tags, merge_mode);
+
+GST_LOG_OBJECT (tagsetter, "final tags: %" GST_PTR_FORMAT, result);
+]|
+
+
+ Adds the given tag / value pairs on the setter using the given merge mode.
+The list must be terminated with %NULL.
+
+
+
+
+
+ a #GstTagSetter
+
+
+
+ the mode to use
+
+
+
+ tag to set
+
+
+
+ tag / value pairs to set
+
+
+
+
+
+ Adds the given tag / GValue pairs on the setter using the given merge mode.
+The list must be terminated with %NULL.
+
+
+
+
+
+ a #GstTagSetter
+
+
+
+ the mode to use
+
+
+
+ tag to set
+
+
+
+ tag / GValue pairs to set
+
+
+
+
+
+ Adds the given tag / GValue pair on the setter using the given merge mode.
+
+
+
+
+
+ a #GstTagSetter
+
+
+
+ the mode to use
+
+
+
+ tag to set
+
+
+
+ GValue to set for the tag
+
+
+
+
+
+ Adds the given tag / GValue pairs on the setter using the given merge mode.
+The list must be terminated with %NULL.
+
+
+
+
+
+ a #GstTagSetter
+
+
+
+ the mode to use
+
+
+
+ tag to set
+
+
+
+ more tag / GValue pairs to set
+
+
+
+
+
+ Adds the given tag / value pairs on the setter using the given merge mode.
+The list must be terminated with %NULL.
+
+
+
+
+
+ a #GstTagSetter
+
+
+
+ the mode to use
+
+
+
+ tag to set
+
+
+
+ more tag / value pairs to set
+
+
+
+
+
+ Returns the current list of tags the setter uses. The list should not be
+modified or freed.
+
+This function is not thread-safe.
+
+ a current snapshot of the
+ taglist used in the setter or %NULL if none is used.
+
+
+
+
+ a #GstTagSetter
+
+
+
+
+
+ Queries the mode by which tags inside the setter are overwritten by tags
+from events
+
+ the merge mode used inside the element.
+
+
+
+
+ a #GstTagSetter
+
+
+
+
+
+ Merges the given list into the setter's list using the given mode.
+
+
+
+
+
+ a #GstTagSetter
+
+
+
+ a tag list to merge from
+
+
+
+ the mode to merge with
+
+
+
+
+
+ Reset the internal taglist. Elements should call this from within the
+state-change handler.
+
+
+
+
+
+ a #GstTagSetter
+
+
+
+
+
+ Sets the given merge mode that is used for adding tags from events to tags
+specified by this interface. The default is #GST_TAG_MERGE_KEEP, which keeps
+the tags set with this interface and discards tags from events.
+
+
+
+
+
+ a #GstTagSetter
+
+
+
+ The mode with which tags are added
+
+
+
+
+
+
+ #GstTagSetterInterface interface.
+
+ parent interface type.
+
+
+
+
+ #GstTask is used by #GstElement and #GstPad to provide the data passing
+threads in a #GstPipeline.
+
+A #GstPad will typically start a #GstTask to push or pull data to/from the
+peer pads. Most source elements start a #GstTask to push data. In some cases
+a demuxer element can start a #GstTask to pull data from a peer element. This
+is typically done when the demuxer can perform random access on the upstream
+peer element for improved performance.
+
+Although convenience functions exist on #GstPad to start/pause/stop tasks, it
+might sometimes be needed to create a #GstTask manually if it is not related to
+a #GstPad.
+
+Before the #GstTask can be run, it needs a #GRecMutex that can be set with
+gst_task_set_lock().
+
+The task can be started, paused and stopped with gst_task_start(), gst_task_pause()
+and gst_task_stop() respectively or with the gst_task_set_state() function.
+
+A #GstTask will repeatedly call the #GstTaskFunction with the user data
+that was provided when creating the task with gst_task_new(). While calling
+the function it will acquire the provided lock. The provided lock is released
+when the task pauses or stops.
+
+Stopping a task with gst_task_stop() will not immediately make sure the task is
+not running anymore. Use gst_task_join() to make sure the task is completely
+stopped and the thread is stopped.
+
+After creating a #GstTask, use gst_object_unref() to free its resources. This can
+only be done when the task is not running anymore.
+
+Task functions can send a #GstMessage to send out-of-band data to the
+application. The application can receive messages from the #GstBus in its
+mainloop.
+
+For debugging purposes, the task will configure its object name as the thread
+name on Linux. Please note that the object name should be configured before the
+task is started; changing the object name after the task has been started, has
+no effect on the thread name.
+
+ Create a new Task that will repeatedly call the provided @func
+with @user_data as a parameter. Typically the task will run in
+a new thread.
+
+The function cannot be changed after the task has been created. You
+must create a new #GstTask to change the function.
+
+This function will not yet create and start a thread. Use gst_task_start() or
+gst_task_pause() to create and start the GThread.
+
+Before the task can be used, a #GRecMutex must be configured using the
+gst_task_set_lock() function. This lock will always be acquired while
+@func is called.
+
+ A new #GstTask.
+
+MT safe.
+
+
+
+
+ The #GstTaskFunction to use
+
+
+
+ User data to pass to @func
+
+
+
+ the function to call when @user_data is no longer needed.
+
+
+
+
+
+ Wait for all tasks to be stopped. This is mainly used internally
+to ensure proper cleanup of internal data structures in test suites.
+
+MT safe.
+
+
+
+
+
+ Get the #GstTaskPool that this task will use for its streaming
+threads.
+
+MT safe.
+
+ the #GstTaskPool used by @task. gst_object_unref()
+after usage.
+
+
+
+
+ a #GstTask
+
+
+
+
+
+ Get the current state of the task.
+
+ The #GstTaskState of the task
+
+MT safe.
+
+
+
+
+ The #GstTask to query
+
+
+
+
+
+ Joins @task. After this call, it is safe to unref the task
+and clean up the lock set with gst_task_set_lock().
+
+The task will automatically be stopped with this call.
+
+This function cannot be called from within a task function as this
+would cause a deadlock. The function will detect this and print a
+g_warning.
+
+ %TRUE if the task could be joined.
+
+MT safe.
+
+
+
+
+ The #GstTask to join
+
+
+
+
+
+ Pauses @task. This method can also be called on a task in the
+stopped state, in which case a thread will be started and will remain
+in the paused state. This function does not wait for the task to complete
+the paused state.
+
+ %TRUE if the task could be paused.
+
+MT safe.
+
+
+
+
+ The #GstTask to pause
+
+
+
+
+
+ Call @enter_func when the task function of @task is entered. @user_data will
+be passed to @enter_func and @notify will be called when @user_data is no
+longer referenced.
+
+
+
+
+
+ The #GstTask to use
+
+
+
+ a #GstTaskThreadFunc
+
+
+
+ user data passed to @enter_func
+
+
+
+ called when @user_data is no longer referenced
+
+
+
+
+
+ Call @leave_func when the task function of @task is left. @user_data will
+be passed to @leave_func and @notify will be called when @user_data is no
+longer referenced.
+
+
+
+
+
+ The #GstTask to use
+
+
+
+ a #GstTaskThreadFunc
+
+
+
+ user data passed to @leave_func
+
+
+
+ called when @user_data is no longer referenced
+
+
+
+
+
+ Set the mutex used by the task. The mutex will be acquired before
+calling the #GstTaskFunction.
+
+This function has to be called before calling gst_task_pause() or
+gst_task_start().
+
+MT safe.
+
+
+
+
+
+ The #GstTask to use
+
+
+
+ The #GRecMutex to use
+
+
+
+
+
+ Set @pool as the new GstTaskPool for @task. Any new streaming threads that
+will be created by @task will now use @pool.
+
+MT safe.
+
+
+
+
+
+ a #GstTask
+
+
+
+ a #GstTaskPool
+
+
+
+
+
+ Sets the state of @task to @state.
+
+The @task must have a lock associated with it using
+gst_task_set_lock() when going to GST_TASK_STARTED or GST_TASK_PAUSED or
+this function will return %FALSE.
+
+MT safe.
+
+ %TRUE if the state could be changed.
+
+
+
+
+ a #GstTask
+
+
+
+ the new task state
+
+
+
+
+
+ Starts @task. The @task must have a lock associated with it using
+gst_task_set_lock() or this function will return %FALSE.
+
+ %TRUE if the task could be started.
+
+MT safe.
+
+
+
+
+ The #GstTask to start
+
+
+
+
+
+ Stops @task. This method merely schedules the task to stop and
+will not wait for the task to have completely stopped. Use
+gst_task_join() to stop and wait for completion.
+
+ %TRUE if the task could be stopped.
+
+MT safe.
+
+
+
+
+ The #GstTask to stop
+
+
+
+
+
+
+
+
+ the state of the task
+
+
+
+ used to pause/resume the task
+
+
+
+ The lock taken when iterating the task function
+
+
+
+ the function executed by this task
+
+
+
+ user_data passed to the task function
+
+
+
+ GDestroyNotify for @user_data
+
+
+
+ a flag indicating that the task is running
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ A function that will repeatedly be called in the thread created by
+a #GstTask.
+
+
+
+
+
+ user data passed to the function
+
+
+
+
+
+ This object provides an abstraction for creating threads. The default
+implementation uses a regular GThreadPool to start tasks.
+
+Subclasses can be made to create custom threads.
+
+ Create a new default task pool. The default task pool will use a regular
+GThreadPool for threads.
+
+ a new #GstTaskPool. gst_object_unref() after usage.
+
+
+
+
+ Wait for all tasks to be stopped. This is mainly used internally
+to ensure proper cleanup of internal data structures in test suites.
+
+MT safe.
+
+
+
+
+
+ a #GstTaskPool
+
+
+
+
+
+ Join a task and/or return it to the pool. @id is the id obtained from
+gst_task_pool_push().
+
+
+
+
+
+ a #GstTaskPool
+
+
+
+ the id
+
+
+
+
+
+ Prepare the taskpool for accepting gst_task_pool_push() operations.
+
+MT safe.
+
+
+
+
+
+ a #GstTaskPool
+
+
+
+
+
+ Start the execution of a new thread from @pool.
+
+ a pointer that should be used
+for the gst_task_pool_join function. This pointer can be %NULL, you
+must check @error to detect errors.
+
+
+
+
+ a #GstTaskPool
+
+
+
+ the function to call
+
+
+
+ data to pass to @func
+
+
+
+
+
+ Wait for all tasks to be stopped. This is mainly used internally
+to ensure proper cleanup of internal data structures in test suites.
+
+MT safe.
+
+
+
+
+
+ a #GstTaskPool
+
+
+
+
+
+ Join a task and/or return it to the pool. @id is the id obtained from
+gst_task_pool_push().
+
+
+
+
+
+ a #GstTaskPool
+
+
+
+ the id
+
+
+
+
+
+ Prepare the taskpool for accepting gst_task_pool_push() operations.
+
+MT safe.
+
+
+
+
+
+ a #GstTaskPool
+
+
+
+
+
+ Start the execution of a new thread from @pool.
+
+ a pointer that should be used
+for the gst_task_pool_join function. This pointer can be %NULL, you
+must check @error to detect errors.
+
+
+
+
+ a #GstTaskPool
+
+
+
+ the function to call
+
+
+
+ data to pass to @func
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ The #GstTaskPoolClass object.
+
+ the parent class structure
+
+
+
+
+
+
+
+
+
+ a #GstTaskPool
+
+
+
+
+
+
+
+
+
+
+
+
+ a #GstTaskPool
+
+
+
+
+
+
+
+
+ a pointer that should be used
+for the gst_task_pool_join function. This pointer can be %NULL, you
+must check @error to detect errors.
+
+
+
+
+ a #GstTaskPool
+
+
+
+ the function to call
+
+
+
+ data to pass to @func
+
+
+
+
+
+
+
+
+
+
+
+
+ a #GstTaskPool
+
+
+
+ the id
+
+
+
+
+
+
+
+
+
+
+
+
+ Task function, see gst_task_pool_push().
+
+
+
+
+
+ user data for the task function
+
+
+
+
+
+
+
+ The different states a task can be in
+
+ the task is started and running
+
+
+ the task is stopped
+
+
+ the task is paused
+
+
+
+ Custom GstTask thread callback functions that can be installed.
+
+
+
+
+
+ The #GstTask
+
+
+
+ The #GThread
+
+
+
+ user data
+
+
+
+
+
+ Structure for saving a timestamp and a value.
+
+ timestamp of the value change
+
+
+
+ the corresponding value
+
+
+
+
+ #GstToc functions are used to create/free #GstToc and #GstTocEntry structures.
+Also they are used to convert #GstToc into #GstStructure and vice versa.
+
+#GstToc lets you to inform other elements in pipeline or application that playing
+source has some kind of table of contents (TOC). These may be chapters, editions,
+angles or other types. For example: DVD chapters, Matroska chapters or cue sheet
+TOC. Such TOC will be useful for applications to display instead of just a
+playlist.
+
+Using TOC is very easy. Firstly, create #GstToc structure which represents root
+contents of the source. You can also attach TOC-specific tags to it. Then fill
+it with #GstTocEntry entries by appending them to the #GstToc using
+gst_toc_append_entry(), and appending subentries to a #GstTocEntry using
+gst_toc_entry_append_sub_entry().
+
+Note that root level of the TOC can contain only either editions or chapters. You
+should not mix them together at the same level. Otherwise you will get serialization
+/deserialization errors. Make sure that no one of the entries has negative start and
+ stop values.
+
+Use gst_event_new_toc() to create a new TOC #GstEvent, and gst_event_parse_toc() to
+parse received TOC event. Use gst_event_new_toc_select() to create a new TOC select #GstEvent,
+and gst_event_parse_toc_select() to parse received TOC select event. The same rule for
+the #GstMessage: gst_message_new_toc() to create new TOC #GstMessage, and
+gst_message_parse_toc() to parse received TOC message.
+
+TOCs can have global scope or current scope. Global scope TOCs contain
+all entries that can possibly be selected using a toc select event, and
+are what an application is usually interested in. TOCs with current scope
+only contain the parts of the TOC relevant to the currently selected/playing
+stream; the current scope TOC is used by downstream elements such as muxers
+to write correct TOC entries when transcoding files, for example. When
+playing a DVD, the global TOC would contain a hierarchy of all titles,
+chapters and angles, for example, while the current TOC would only contain
+the chapters for the currently playing title if playback of a specific
+title was requested.
+
+Applications and plugins should not rely on TOCs having a certain kind of
+structure, but should allow for different alternatives. For example, a
+simple CUE sheet embedded in a file may be presented as a flat list of
+track entries, or could have a top-level edition node (or some other
+alternative type entry) with track entries underneath that node; or even
+multiple top-level edition nodes (or some other alternative type entries)
+each with track entries underneath, in case the source file has extracted
+a track listing from different sources).
+
+ Create a new #GstToc structure.
+
+ newly allocated #GstToc structure, free it
+ with gst_toc_unref().
+
+
+
+
+ scope of this TOC
+
+
+
+
+
+ Appends the #GstTocEntry @entry to @toc.
+
+
+
+
+
+ A #GstToc instance
+
+
+
+ A #GstTocEntry
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Find #GstTocEntry with given @uid in the @toc.
+
+ #GstTocEntry with specified
+@uid from the @toc, or %NULL if not found.
+
+
+
+
+ #GstToc to search in.
+
+
+
+ UID to find #GstTocEntry with.
+
+
+
+
+
+ Gets the list of #GstTocEntry of @toc.
+
+ A #GList of #GstTocEntry for @entry
+
+
+
+
+
+
+ A #GstToc instance
+
+
+
+
+
+
+ scope of @toc
+
+
+
+
+ a #GstToc instance
+
+
+
+
+
+ Gets the tags for @toc.
+
+ A #GstTagList for @entry
+
+
+
+
+ A #GstToc instance
+
+
+
+
+
+ Merge @tags into the existing tags of @toc using @mode.
+
+
+
+
+
+ A #GstToc instance
+
+
+
+ A #GstTagList or %NULL
+
+
+
+ A #GstTagMergeMode
+
+
+
+
+
+ Set a #GstTagList with tags for the complete @toc.
+
+
+
+
+
+ A #GstToc instance
+
+
+
+ A #GstTagList or %NULL
+
+
+
+
+
+
+
+ Create new #GstTocEntry structure.
+
+ newly allocated #GstTocEntry structure, free it with gst_toc_entry_unref().
+
+
+
+
+ entry type.
+
+
+
+ unique ID (UID) in the whole TOC.
+
+
+
+
+
+ Appends the #GstTocEntry @subentry to @entry.
+
+
+
+
+
+ A #GstTocEntry instance
+
+
+
+ A #GstTocEntry
+
+
+
+
+
+
+ @entry's entry type
+
+
+
+
+ a #GstTocEntry
+
+
+
+
+
+ Get @loop_type and @repeat_count values from the @entry and write them into
+appropriate storages. Loops are e.g. used by sampled instruments. GStreamer
+is not automatically applying the loop. The application can process this
+meta data and use it e.g. to send a seek-event to loop a section.
+
+ %TRUE if all non-%NULL storage pointers were filled with appropriate
+values, %FALSE otherwise.
+
+
+
+
+ #GstTocEntry to get values from.
+
+
+
+ the storage for the loop_type
+ value, leave %NULL if not need.
+
+
+
+ the storage for the repeat_count
+ value, leave %NULL if not need.
+
+
+
+
+
+ Gets the parent #GstTocEntry of @entry.
+
+ The parent #GstTocEntry of @entry
+
+
+
+
+ A #GstTocEntry instance
+
+
+
+
+
+ Get @start and @stop values from the @entry and write them into appropriate
+storages.
+
+ %TRUE if all non-%NULL storage pointers were filled with appropriate
+values, %FALSE otherwise.
+
+
+
+
+ #GstTocEntry to get values from.
+
+
+
+ the storage for the start value, leave
+ %NULL if not need.
+
+
+
+ the storage for the stop value, leave
+ %NULL if not need.
+
+
+
+
+
+ Gets the sub-entries of @entry.
+
+ A #GList of #GstTocEntry of @entry
+
+
+
+
+
+
+ A #GstTocEntry instance
+
+
+
+
+
+ Gets the tags for @entry.
+
+ A #GstTagList for @entry
+
+
+
+
+ A #GstTocEntry instance
+
+
+
+
+
+ Gets the parent #GstToc of @entry.
+
+ The parent #GstToc of @entry
+
+
+
+
+ A #GstTocEntry instance
+
+
+
+
+
+ Gets the UID of @entry.
+
+ The UID of @entry
+
+
+
+
+ A #GstTocEntry instance
+
+
+
+
+
+
+ %TRUE if @entry's type is an alternative type, otherwise %FALSE
+
+
+
+
+ a #GstTocEntry
+
+
+
+
+
+
+ %TRUE if @entry's type is a sequence type, otherwise %FALSE
+
+
+
+
+ a #GstTocEntry
+
+
+
+
+
+ Merge @tags into the existing tags of @entry using @mode.
+
+
+
+
+
+ A #GstTocEntry instance
+
+
+
+ A #GstTagList or %NULL
+
+
+
+ A #GstTagMergeMode
+
+
+
+
+
+ Set @loop_type and @repeat_count values for the @entry.
+
+
+
+
+
+ #GstTocEntry to set values.
+
+
+
+ loop_type value to set.
+
+
+
+ repeat_count value to set.
+
+
+
+
+
+ Set @start and @stop values for the @entry.
+
+
+
+
+
+ #GstTocEntry to set values.
+
+
+
+ start value to set.
+
+
+
+ stop value to set.
+
+
+
+
+
+ Set a #GstTagList with tags for the complete @entry.
+
+
+
+
+
+ A #GstTocEntry instance
+
+
+
+ A #GstTagList or %NULL
+
+
+
+
+
+
+ The different types of TOC entries (see #GstTocEntry).
+
+There are two types of TOC entries: alternatives or parts in a sequence.
+
+ entry is an angle (i.e. an alternative)
+
+
+ entry is a version (i.e. alternative)
+
+
+ entry is an edition (i.e. alternative)
+
+
+ invalid entry type value
+
+
+ entry is a title (i.e. a part of a sequence)
+
+
+ entry is a track (i.e. a part of a sequence)
+
+
+ entry is a chapter (i.e. a part of a sequence)
+
+
+ Converts @type to a string representation.
+
+ Returns a human-readable string for @type. This string is
+ only for debugging purpose and should not be displayed in a user
+ interface.
+
+
+
+
+ a #GstTocEntryType.
+
+
+
+
+
+
+ How a #GstTocEntry should be repeated. By default, entries are played a
+single time.
+
+ single forward playback
+
+
+ repeat forward
+
+
+ repeat backward
+
+
+ repeat forward and backward
+
+
+
+ The scope of a TOC.
+
+ global TOC representing all selectable options
+ (this is what applications are usually interested in)
+
+
+ TOC for the currently active/selected stream
+ (this is a TOC representing the current stream from start to EOS,
+ and is what a TOC writer / muxer is usually interested in; it will
+ usually be a subset of the global TOC, e.g. just the chapters of
+ the current title, or the chapters selected for playback from the
+ current title)
+
+
+
+ Element interface that allows setting of the TOC.
+
+Elements that support some kind of chapters or editions (or tracks like in
+the FLAC cue sheet) will implement this interface.
+
+If you just want to retrieve the TOC in your application then all you
+need to do is watch for TOC messages on your pipeline's bus (or you can
+perform TOC query). This interface is only for setting TOC data, not for
+extracting it. To set TOC from the application, find proper tocsetter element
+and set TOC using gst_toc_setter_set_toc().
+
+Elements implementing the #GstTocSetter interface can extend existing TOC
+by getting extend UID for that (you can use gst_toc_find_entry() to retrieve it)
+with any TOC entries received from downstream.
+
+
+ Return current TOC the setter uses. The TOC should not be
+modified without making it writable first.
+
+ TOC set, or %NULL. Unref with
+ gst_toc_unref() when no longer needed
+
+
+
+
+ a #GstTocSetter.
+
+
+
+
+
+ Reset the internal TOC. Elements should call this from within the
+state-change handler.
+
+
+
+
+
+ a #GstTocSetter.
+
+
+
+
+
+ Set the given TOC on the setter. Previously set TOC will be
+unreffed before setting a new one.
+
+
+
+
+
+ a #GstTocSetter.
+
+
+
+ a #GstToc to set.
+
+
+
+
+
+
+ #GstTocSetterInterface interface.
+
+ parent interface type.
+
+
+
+
+ Tracing modules will subclass #GstTracer and register through
+gst_tracer_register(). Modules can attach to various hook-types - see
+gst_tracing_register_hook(). When invoked they receive hook specific
+contextual data, which they must not modify.
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Use gst_tracer_factory_get_list() to get a list of tracer factories known to
+GStreamer.
+
+ Gets the list of all registered tracer factories. You must free the
+list using gst_plugin_feature_list_free().
+
+The returned factories are sorted by factory name.
+
+Free-function: gst_plugin_feature_list_free
+
+ the list of all
+ registered #GstTracerFactory.
+
+
+
+
+
+
+ Get the #GType for elements managed by this factory. The type can
+only be retrieved if the element factory is loaded, which can be
+assured with gst_plugin_feature_load().
+
+ the #GType for tracers managed by this factory or 0 if
+the factory is not loaded.
+
+
+
+
+ factory to get managed #GType from
+
+
+
+
+
+
+
+
+
+
+ Tracing modules will create instances of this class to announce the data they
+will log and create a log formatter.
+
+
+
+
+ Flag that describe the value. These flags help applications processing the
+logs to understand the values.
+
+ no flags
+
+
+ the value is optional. When using this flag
+ one need to have an additional boolean arg before this value in the
+ var-args list passed to gst_tracer_record_log().
+
+
+ the value is a combined figure, since the
+ start of tracing. Examples are averages or timestamps.
+
+
+
+ Tracing record will contain fields that contain a meassured value or extra
+meta-data. One such meta data are values that tell where a measurement was
+taken. This enumerating declares to which scope such a meta data field
+relates to. If it is e.g. %GST_TRACER_VALUE_SCOPE_PAD, then each of the log
+events may contain values for different #GstPads.
+
+ the value is related to the process
+
+
+ the value is related to a thread
+
+
+ the value is related to an #GstElement
+
+
+ the value is related to a #GstPad
+
+
+
+ The following functions allow you to detect the media type of an unknown
+stream.
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ The data used by the caller of the typefinding function.
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Get the length of the data stream.
+
+ The length of the data stream, or 0 if it is not available.
+
+
+
+
+ The #GstTypeFind the function was called with
+
+
+
+
+
+ Returns the @size bytes of the stream to identify beginning at offset. If
+offset is a positive number, the offset is relative to the beginning of the
+stream, if offset is a negative number the offset is relative to the end of
+the stream. The returned memory is valid until the typefinding function
+returns and must not be freed.
+
+ the
+ requested data, or %NULL if that data is not available.
+
+
+
+
+
+
+ The #GstTypeFind object the function was called with
+
+
+
+ The offset
+
+
+
+ The number of bytes to return
+
+
+
+
+
+ If a #GstTypeFindFunction calls this function it suggests the caps with the
+given probability. A #GstTypeFindFunction may supply different suggestions
+in one call.
+It is up to the caller of the #GstTypeFindFunction to interpret these values.
+
+
+
+
+
+ The #GstTypeFind object the function was called with
+
+
+
+ The probability in percent that the suggestion is right
+
+
+
+ The fixed #GstCaps to suggest
+
+
+
+
+
+ If a #GstTypeFindFunction calls this function it suggests the caps with the
+given probability. A #GstTypeFindFunction may supply different suggestions
+in one call. It is up to the caller of the #GstTypeFindFunction to interpret
+these values.
+
+This function is similar to gst_type_find_suggest(), only that instead of
+passing a #GstCaps argument you can create the caps on the fly in the same
+way as you can with gst_caps_new_simple().
+
+Make sure you terminate the list of arguments with a %NULL argument and that
+the values passed have the correct type (in terms of width in bytes when
+passed to the vararg function - this applies particularly to gdouble and
+guint64 arguments).
+
+
+
+
+
+ The #GstTypeFind object the function was called with
+
+
+
+ The probability in percent that the suggestion is right
+
+
+
+ the media type of the suggested caps
+
+
+
+ first field of the suggested caps, or %NULL
+
+
+
+ additional arguments to the suggested caps in the same format as the
+ arguments passed to gst_structure_new() (ie. triplets of field name,
+ field GType and field value)
+
+
+
+
+
+ Registers a new typefind function to be used for typefinding. After
+registering this function will be available for typefinding.
+This function is typically called during an element's plugin initialization.
+
+ %TRUE on success, %FALSE otherwise
+
+
+
+
+ A #GstPlugin, or %NULL for a static typefind function
+
+
+
+ The name for registering
+
+
+
+ The rank (or importance) of this typefind function
+
+
+
+ The #GstTypeFindFunction to use
+
+
+
+ Optional comma-separated list of extensions
+ that could belong to this type
+
+
+
+ Optionally the caps that could be returned when typefinding
+ succeeds
+
+
+
+ Optional user data. This user data must be available until the plugin
+ is unloaded.
+
+
+
+ a #GDestroyNotify that will be called on @data when the plugin
+ is unloaded.
+
+
+
+
+
+
+ These functions allow querying informations about registered typefind
+functions. How to create and register these functions is described in
+the section <link linkend="gstreamer-Writing-typefind-functions">
+"Writing typefind functions"</link>.
+
+The following example shows how to write a very simple typefinder that
+identifies the given data. You can get quite a bit more complicated than
+that though.
+|[<!-- language="C" -->
+ typedef struct {
+ guint8 *data;
+ guint size;
+ guint probability;
+ GstCaps *data;
+ } MyTypeFind;
+ static void
+ my_peek (gpointer data, gint64 offset, guint size)
+ {
+ MyTypeFind *find = (MyTypeFind *) data;
+ if (offset >= 0 && offset + size <= find->size) {
+ return find->data + offset;
+ }
+ return NULL;
+ }
+ static void
+ my_suggest (gpointer data, guint probability, GstCaps *caps)
+ {
+ MyTypeFind *find = (MyTypeFind *) data;
+ if (probability > find->probability) {
+ find->probability = probability;
+ gst_caps_replace (&find->caps, caps);
+ }
+ }
+ static GstCaps *
+ find_type (guint8 *data, guint size)
+ {
+ GList *walk, *type_list;
+ MyTypeFind find = {data, size, 0, NULL};
+ GstTypeFind gst_find = {my_peek, my_suggest, &find, };
+ walk = type_list = gst_type_find_factory_get_list ();
+ while (walk) {
+ GstTypeFindFactory *factory = GST_TYPE_FIND_FACTORY (walk->data);
+ walk = g_list_next (walk)
+ gst_type_find_factory_call_function (factory, &gst_find);
+ }
+ g_list_free (type_list);
+ return find.caps;
+ };
+]|
+
+ Gets the list of all registered typefind factories. You must free the
+list using gst_plugin_feature_list_free().
+
+The returned factories are sorted by highest rank first, and then by
+factory name.
+
+Free-function: gst_plugin_feature_list_free
+
+ the list of all
+ registered #GstTypeFindFactory.
+
+
+
+
+
+
+ Calls the #GstTypeFindFunction associated with this factory.
+
+
+
+
+
+ A #GstTypeFindFactory
+
+
+
+ a properly setup #GstTypeFind entry. The get_data
+ and suggest_type members must be set.
+
+
+
+
+
+ Gets the #GstCaps associated with a typefind factory.
+
+ the #GstCaps associated with this factory
+
+
+
+
+ A #GstTypeFindFactory
+
+
+
+
+
+ Gets the extensions associated with a #GstTypeFindFactory. The returned
+array should not be changed. If you need to change stuff in it, you should
+copy it using g_strdupv(). This function may return %NULL to indicate
+a 0-length list.
+
+
+ a %NULL-terminated array of extensions associated with this factory
+
+
+
+
+
+
+ A #GstTypeFindFactory
+
+
+
+
+
+ Check whether the factory has a typefind function. Typefind factories
+without typefind functions are a last-effort fallback mechanism to
+e.g. assume a certain media type based on the file extension.
+
+ %TRUE if the factory has a typefind functions set, otherwise %FALSE
+
+
+
+
+ A #GstTypeFindFactory
+
+
+
+
+
+
+
+
+ A function that will be called by typefinding.
+
+
+
+
+
+ A #GstTypeFind structure
+
+
+
+ optional data to pass to the function
+
+
+
+
+
+ The probability of the typefind function. Higher values have more certainty
+in doing a reliable typefind.
+
+ type undetected.
+
+
+ unlikely typefind.
+
+
+ possible type detected.
+
+
+ likely a type was detected.
+
+
+ nearly certain that a type was detected.
+
+
+ very certain a type was detected.
+
+
+
+ Different URI-related errors that can occur.
+
+ The protocol is not supported
+
+
+ There was a problem with the URI
+
+
+ Could not set or change the URI because the
+ URI handler was in a state where that is not possible or not permitted
+
+
+ There was a problem with the entity that
+ the URI references
+
+
+
+
+
+
+
+
+ The #GstURIHandler is an interface that is implemented by Source and Sink
+#GstElement to unify handling of URI.
+
+An application can use the following functions to quickly get an element
+that handles the given URI for reading or writing
+(gst_element_make_from_uri()).
+
+Source and Sink plugins should implement this interface when possible.
+
+ Gets the currently handled URI.
+
+ the URI currently handled by
+ the @handler. Returns %NULL if there are no URI currently
+ handled. The returned string must be freed with g_free() when no
+ longer needed.
+
+
+
+
+ A #GstURIHandler
+
+
+
+
+
+ Tries to set the URI of the given handler.
+
+ %TRUE if the URI was set successfully, else %FALSE.
+
+
+
+
+ A #GstURIHandler
+
+
+
+ URI to set
+
+
+
+
+
+ Gets the list of protocols supported by @handler. This list may not be
+modified.
+
+ the
+ supported protocols. Returns %NULL if the @handler isn't
+ implemented properly, or the @handler doesn't support any
+ protocols.
+
+
+
+
+
+
+ A #GstURIHandler.
+
+
+
+
+
+ Gets the currently handled URI.
+
+ the URI currently handled by
+ the @handler. Returns %NULL if there are no URI currently
+ handled. The returned string must be freed with g_free() when no
+ longer needed.
+
+
+
+
+ A #GstURIHandler
+
+
+
+
+
+ Gets the type of the given URI handler
+
+ the #GstURIType of the URI handler.
+Returns #GST_URI_UNKNOWN if the @handler isn't implemented correctly.
+
+
+
+
+ A #GstURIHandler.
+
+
+
+
+
+ Tries to set the URI of the given handler.
+
+ %TRUE if the URI was set successfully, else %FALSE.
+
+
+
+
+ A #GstURIHandler
+
+
+
+ URI to set
+
+
+
+
+
+
+ Any #GstElement using this interface should implement these methods.
+
+ The parent interface type
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ the URI currently handled by
+ the @handler. Returns %NULL if there are no URI currently
+ handled. The returned string must be freed with g_free() when no
+ longer needed.
+
+
+
+
+ A #GstURIHandler
+
+
+
+
+
+
+
+
+ %TRUE if the URI was set successfully, else %FALSE.
+
+
+
+
+ A #GstURIHandler
+
+
+
+ URI to set
+
+
+
+
+
+
+
+ The different types of URI direction.
+
+ The URI direction is unknown
+
+
+ The URI is a consumer.
+
+
+ The URI is a producer.
+
+
+
+ Value for #GstUri<!-- -->.port to indicate no port number.
+
+
+
+ Constant that defines one GStreamer microsecond.
+
+
+
+ A #GstUri object can be used to parse and split a URI string into its
+constituant parts. Two #GstUri objects can be joined to make a new #GstUri
+using the algorithm described in RFC3986.
+
+ Creates a new #GstUri object with the given URI parts. The path and query
+strings will be broken down into their elements. All strings should not be
+escaped except where indicated.
+
+ A new #GstUri object.
+
+
+
+
+ The scheme for the new URI.
+
+
+
+ The user-info for the new URI.
+
+
+
+ The host name for the new URI.
+
+
+
+ The port number for the new URI or %GST_URI_NO_PORT.
+
+
+
+ The path for the new URI with '/' separating path
+ elements.
+
+
+
+ The query string for the new URI with '&' separating
+ query elements. Elements containing '&' characters
+ should encode them as "%26".
+
+
+
+ The fragment name for the new URI.
+
+
+
+
+
+ Append a path onto the end of the path in the URI. The path is not
+normalized, call #gst_uri_normalize() to normalize the path.
+
+ %TRUE if the path was appended successfully.
+
+
+
+
+ The #GstUri to modify.
+
+
+
+ Relative path to append to the end of the current path.
+
+
+
+
+
+ Append a single path segment onto the end of the URI path.
+
+ %TRUE if the path was appended successfully.
+
+
+
+
+ The #GstUri to modify.
+
+
+
+ The path segment string to append to the URI path.
+
+
+
+
+
+ Compares two #GstUri objects to see if they represent the same normalized
+URI.
+
+ %TRUE if the normalized versions of the two URI's would be equal.
+
+
+
+
+ First #GstUri to compare.
+
+
+
+ Second #GstUri to compare.
+
+
+
+
+
+ Like gst_uri_from_string() but also joins with a base URI.
+
+ A new #GstUri object.
+
+
+
+
+ The base URI to join the new URI with.
+
+
+
+ The URI string to parse.
+
+
+
+
+
+ Get the fragment name from the URI or %NULL if it doesn't exist.
+If @uri is %NULL then returns %NULL.
+
+ The host name from the #GstUri object or %NULL.
+
+
+
+
+ This #GstUri object.
+
+
+
+
+
+ Get the host name from the URI or %NULL if it doesn't exist.
+If @uri is %NULL then returns %NULL.
+
+ The host name from the #GstUri object or %NULL.
+
+
+
+
+ This #GstUri object.
+
+
+
+
+
+ Get the media fragment table from the URI, as defined by "Media Fragments URI 1.0".
+Hash table returned by this API is a list of "key-value" pairs, and the each
+pair is generated by splitting "URI fragment" per "&" sub-delims, then "key"
+and "value" are splitted by "=" sub-delims. The "key" returned by this API may
+be undefined keyword by standard.
+A value may be %NULL to indicate that the key should appear in the fragment
+string in the URI, but does not have a value. Free the returned #GHashTable
+with #g_hash_table_unref() when it is no longer required.
+Modifying this hash table does not affect the fragment in the URI.
+
+See more about Media Fragments URI 1.0 (W3C) at https://www.w3.org/TR/media-frags/
+
+ The
+ fragment hash table from the URI.
+
+
+
+
+
+
+
+ The #GstUri to get the fragment table from.
+
+
+
+
+
+ Extract the path string from the URI object.
+
+ (nullable): The path from the URI. Once finished
+ with the string should be g_free()'d.
+
+
+
+
+ The #GstUri to get the path from.
+
+
+
+
+
+ Get a list of path segments from the URI.
+
+ A #GList of path segment
+ strings or %NULL if no path segments are available. Free the list
+ when no longer needed with g_list_free_full(list, g_free).
+
+
+
+
+
+
+ The #GstUri to get the path from.
+
+
+
+
+
+ Extract the path string from the URI object as a percent encoded URI path.
+
+ The path from the URI. Once finished
+ with the string should be g_free()'d.
+
+
+
+
+ The #GstUri to get the path from.
+
+
+
+
+
+ Get the port number from the URI or %GST_URI_NO_PORT if it doesn't exist.
+If @uri is %NULL then returns %GST_URI_NO_PORT.
+
+ The port number from the #GstUri object or %GST_URI_NO_PORT.
+
+
+
+
+ This #GstUri object.
+
+
+
+
+
+ Get a list of the query keys from the URI.
+
+ A list of keys from
+ the URI query. Free the list with g_list_free().
+
+
+
+
+
+
+ The #GstUri to examine.
+
+
+
+
+
+ Get a percent encoded URI query string from the @uri.
+
+ A percent encoded query string. Use
+ g_free() when no longer needed.
+
+
+
+
+ The #GstUri to get the query string from.
+
+
+
+
+
+ Get the query table from the URI. Keys and values in the table are freed
+with g_free when they are deleted. A value may be %NULL to indicate that
+the key should appear in the query string in the URI, but does not have a
+value. Free the returned #GHashTable with #g_hash_table_unref() when it is
+no longer required. Modifying this hash table will modify the query in the
+URI.
+
+ The query
+ hash table from the URI.
+
+
+
+
+
+
+
+ The #GstUri to get the query table from.
+
+
+
+
+
+ Get the value associated with the @query_key key. Will return %NULL if the
+key has no value or if the key does not exist in the URI query table. Because
+%NULL is returned for both missing keys and keys with no value, you should
+use gst_uri_query_has_key() to determine if a key is present in the URI
+query.
+
+ The value for the given key, or %NULL if not found.
+
+
+
+
+ The #GstUri to examine.
+
+
+
+ The key to lookup.
+
+
+
+
+
+ Get the scheme name from the URI or %NULL if it doesn't exist.
+If @uri is %NULL then returns %NULL.
+
+ The scheme from the #GstUri object or %NULL.
+
+
+
+
+ This #GstUri object.
+
+
+
+
+
+ Get the userinfo (usually in the form "username:password") from the URI
+or %NULL if it doesn't exist. If @uri is %NULL then returns %NULL.
+
+ The userinfo from the #GstUri object or %NULL.
+
+
+
+
+ This #GstUri object.
+
+
+
+
+
+ Tests the @uri to see if it is normalized. A %NULL @uri is considered to be
+normalized.
+
+ TRUE if the URI is normalized or is %NULL.
+
+
+
+
+ The #GstUri to test to see if it is normalized.
+
+
+
+
+
+ Check if it is safe to write to this #GstUri.
+
+Check if the refcount of @uri is exactly 1, meaning that no other
+reference exists to the #GstUri and that the #GstUri is therefore writable.
+
+Modification of a #GstUri should only be done after verifying that it is
+writable.
+
+ %TRUE if it is safe to write to the object.
+
+
+
+
+ The #GstUri object to test.
+
+
+
+
+
+ Join a reference URI onto a base URI using the method from RFC 3986.
+If either URI is %NULL then the other URI will be returned with the ref count
+increased.
+
+ A #GstUri which represents the base
+ with the reference URI joined on.
+
+
+
+
+ The base URI to join another to.
+
+
+
+ The reference URI to join onto the
+ base URI.
+
+
+
+
+
+ Make the #GstUri writable.
+
+Checks if @uri is writable, and if so the original object is returned. If
+not, then a writable copy is made and returned. This gives away the
+reference to @uri and returns a reference to the new #GstUri.
+If @uri is %NULL then %NULL is returned.
+
+ A writable version of @uri.
+
+
+
+
+ The #GstUri object to make writable.
+
+
+
+
+
+ Like gst_uri_new(), but joins the new URI onto a base URI.
+
+ The new URI joined onto @base.
+
+
+
+
+ The base URI to join the new URI to.
+
+
+
+ The scheme for the new URI.
+
+
+
+ The user-info for the new URI.
+
+
+
+ The host name for the new URI.
+
+
+
+ The port number for the new URI or %GST_URI_NO_PORT.
+
+
+
+ The path for the new URI with '/' separating path
+ elements.
+
+
+
+ The query string for the new URI with '&' separating
+ query elements. Elements containing '&' characters
+ should encode them as "%26".
+
+
+
+ The fragment name for the new URI.
+
+
+
+
+
+ Normalization will remove extra path segments ("." and "..") from the URI. It
+will also convert the scheme and host name to lower case and any
+percent-encoded values to uppercase.
+
+The #GstUri object must be writable. Check with gst_uri_is_writable() or use
+gst_uri_make_writable() first.
+
+ TRUE if the URI was modified.
+
+
+
+
+ The #GstUri to normalize.
+
+
+
+
+
+ Check if there is a query table entry for the @query_key key.
+
+ %TRUE if @query_key exists in the URI query table.
+
+
+
+
+ The #GstUri to examine.
+
+
+
+ The key to lookup.
+
+
+
+
+
+ Remove an entry from the query table by key.
+
+ %TRUE if the key existed in the table and was removed.
+
+
+
+
+ The #GstUri to modify.
+
+
+
+ The key to remove.
+
+
+
+
+
+ Sets the fragment string in the URI. Use a value of %NULL in @fragment to
+unset the fragment string.
+
+ %TRUE if the fragment was set/unset successfully.
+
+
+
+
+ The #GstUri to modify.
+
+
+
+ The fragment string to set.
+
+
+
+
+
+ Set or unset the host for the URI.
+
+ %TRUE if the host was set/unset successfully.
+
+
+
+
+ The #GstUri to modify.
+
+
+
+ The new host string to set or %NULL to unset.
+
+
+
+
+
+ Sets or unsets the path in the URI.
+
+ %TRUE if the path was set successfully.
+
+
+
+
+ The #GstUri to modify.
+
+
+
+ The new path to set with path segments separated by '/', or use %NULL
+ to unset the path.
+
+
+
+
+
+ Replace the path segments list in the URI.
+
+ %TRUE if the path segments were set successfully.
+
+
+
+
+ The #GstUri to modify.
+
+
+
+ The new
+ path list to set.
+
+
+
+
+
+
+
+ Sets or unsets the path in the URI.
+
+ %TRUE if the path was set successfully.
+
+
+
+
+ The #GstUri to modify.
+
+
+
+ The new percent encoded path to set with path segments separated by
+'/', or use %NULL to unset the path.
+
+
+
+
+
+ Set or unset the port number for the URI.
+
+ %TRUE if the port number was set/unset successfully.
+
+
+
+
+ The #GstUri to modify.
+
+
+
+ The new port number to set or %GST_URI_NO_PORT to unset.
+
+
+
+
+
+ Sets or unsets the query table in the URI.
+
+ %TRUE if the query table was set successfully.
+
+
+
+
+ The #GstUri to modify.
+
+
+
+ The new percent encoded query string to use to populate the query
+ table, or use %NULL to unset the query table.
+
+
+
+
+
+ Set the query table to use in the URI. The old table is unreferenced and a
+reference to the new one is used instead. A value if %NULL for @query_table
+will remove the query string from the URI.
+
+ %TRUE if the new table was sucessfully used for the query table.
+
+
+
+
+ The #GstUri to modify.
+
+
+
+ The new
+ query table to use.
+
+
+
+
+
+
+
+
+ This inserts or replaces a key in the query table. A @query_value of %NULL
+indicates that the key has no associated value, but will still be present in
+the query string.
+
+ %TRUE if the query table was sucessfully updated.
+
+
+
+
+ The #GstUri to modify.
+
+
+
+ The key for the query entry.
+
+
+
+ The value for the key.
+
+
+
+
+
+ Set or unset the scheme for the URI.
+
+ %TRUE if the scheme was set/unset successfully.
+
+
+
+
+ The #GstUri to modify.
+
+
+
+ The new scheme to set or %NULL to unset the scheme.
+
+
+
+
+
+ Set or unset the user information for the URI.
+
+ %TRUE if the user information was set/unset successfully.
+
+
+
+
+ The #GstUri to modify.
+
+
+
+ The new user-information string to set or %NULL to unset.
+
+
+
+
+
+ Convert the URI to a string.
+
+Returns the URI as held in this object as a #gchar* nul-terminated string.
+The caller should g_free() the string once they are finished with it.
+The string is put together as described in RFC 3986.
+
+ The string version of the URI.
+
+
+
+
+ This #GstUri to convert to a string.
+
+
+
+
+
+ Constructs a URI for a given valid protocol and location.
+
+Free-function: g_free
+ Use GstURI instead.
+
+ a new string for this URI. Returns %NULL if the
+ given URI protocol is not valid, or the given location is %NULL.
+
+
+
+
+ Protocol for URI
+
+
+
+ Location for URI
+
+
+
+
+
+ Parses a URI string into a new #GstUri object. Will return NULL if the URI
+cannot be parsed.
+
+ A new #GstUri object, or NULL.
+
+
+
+
+ The URI string to parse.
+
+
+
+
+
+ Extracts the location out of a given valid URI, ie. the protocol and "://"
+are stripped from the URI, which means that the location returned includes
+the hostname if one is specified. The returned string must be freed using
+g_free().
+
+Free-function: g_free
+
+ the location for this URI. Returns
+ %NULL if the URI isn't valid. If the URI does not contain a location, an
+ empty string is returned.
+
+
+
+
+ A URI string
+
+
+
+
+
+ Extracts the protocol out of a given valid URI. The returned string must be
+freed using g_free().
+
+ The protocol for this URI.
+
+
+
+
+ A URI string
+
+
+
+
+
+ Checks if the protocol of a given valid URI matches @protocol.
+
+ %TRUE if the protocol matches.
+
+
+
+
+ a URI string
+
+
+
+ a protocol string (e.g. "http")
+
+
+
+
+
+ Tests if the given string is a valid URI identifier. URIs start with a valid
+scheme followed by ":" and maybe a string identifying the location.
+
+ %TRUE if the string is a valid URI
+
+
+
+
+ A URI string
+
+
+
+
+
+ This is a convenience function to join two URI strings and return the result.
+The returned string should be g_free()'d after use.
+
+ A string representing the percent-encoded join of
+ the two URIs.
+
+
+
+
+ The percent-encoded base URI.
+
+
+
+ The percent-encoded reference URI to join to the @base_uri.
+
+
+
+
+
+ Checks if an element exists that supports the given URI protocol. Note
+that a positive return value does not imply that a subsequent call to
+gst_element_make_from_uri() is guaranteed to work.
+
+ %TRUE
+
+
+
+
+ Whether to check for a source or a sink
+
+
+
+ Protocol that should be checked for (e.g. "http" or "smb")
+
+
+
+
+
+ Tests if the given string is a valid protocol identifier. Protocols
+must consist of alphanumeric characters, '+', '-' and '.' and must
+start with a alphabetic character. See RFC 3986 Section 3.1.
+
+ %TRUE if the string is a valid protocol identifier, %FALSE otherwise.
+
+
+
+
+ A string
+
+
+
+
+
+
+ Indicates that the first value provided to a comparison function
+(gst_value_compare()) is equal to the second one.
+
+
+
+ Indicates that the first value provided to a comparison function
+(gst_value_compare()) is greater than the second one.
+
+
+
+ Indicates that the first value provided to a comparison function
+(gst_value_compare()) is lesser than the second one.
+
+
+
+ Indicates that the comparison function (gst_value_compare()) can not
+determine a order for the two provided values.
+
+
+
+ The major version of GStreamer at compile time:
+
+
+
+ The micro version of GStreamer at compile time:
+
+
+
+ The minor version of GStreamer at compile time:
+
+
+
+ The nano version of GStreamer at compile time:
+Actual releases have 0, GIT versions have 1, prerelease versions have 2-...
+
+
+
+
+ Appends @append_value to the GstValueArray in @value.
+
+
+
+
+
+ a #GValue of type #GST_TYPE_ARRAY
+
+
+
+ the value to append
+
+
+
+
+
+ Appends @append_value to the GstValueArray in @value.
+
+
+
+
+
+ a #GValue of type #GST_TYPE_ARRAY
+
+
+
+ the value to append
+
+
+
+
+
+ Gets the number of values contained in @value.
+
+ the number of values
+
+
+
+
+ a #GValue of type #GST_TYPE_ARRAY
+
+
+
+
+
+ Gets the value that is a member of the array contained in @value and
+has the index @index.
+
+ the value at the given index
+
+
+
+
+ a #GValue of type #GST_TYPE_ARRAY
+
+
+
+ index of value to get from the array
+
+
+
+
+
+ Prepends @prepend_value to the GstValueArray in @value.
+
+
+
+
+
+ a #GValue of type #GST_TYPE_ARRAY
+
+
+
+ the value to prepend
+
+
+
+
+
+
+ Used together with gst_value_compare() to compare #GValue items.
+
+ one of GST_VALUE_LESS_THAN, GST_VALUE_EQUAL, GST_VALUE_GREATER_THAN
+or GST_VALUE_UNORDERED
+
+
+
+
+ first value for comparison
+
+
+
+ second value for comparison
+
+
+
+
+
+ Used by gst_value_deserialize() to parse a non-binary form into the #GValue.
+
+ %TRUE for success
+
+
+
+
+ a #GValue
+
+
+
+ a string
+
+
+
+
+
+
+ Appends @append_value to the GstValueList in @value.
+
+
+
+
+
+ a #GValue of type #GST_TYPE_LIST
+
+
+
+ the value to append
+
+
+
+
+
+ Appends @append_value to the GstValueList in @value.
+
+
+
+
+
+ a #GValue of type #GST_TYPE_LIST
+
+
+
+ the value to append
+
+
+
+
+
+ Concatenates copies of @value1 and @value2 into a list. Values that are not
+of type #GST_TYPE_LIST are treated as if they were lists of length 1.
+@dest will be initialized to the type #GST_TYPE_LIST.
+
+
+
+
+
+ an uninitialized #GValue to take the result
+
+
+
+ a #GValue
+
+
+
+ a #GValue
+
+
+
+
+
+ Gets the number of values contained in @value.
+
+ the number of values
+
+
+
+
+ a #GValue of type #GST_TYPE_LIST
+
+
+
+
+
+ Gets the value that is a member of the list contained in @value and
+has the index @index.
+
+ the value at the given index
+
+
+
+
+ a #GValue of type #GST_TYPE_LIST
+
+
+
+ index of value to get from the list
+
+
+
+
+
+ Merges copies of @value1 and @value2. Values that are not
+of type #GST_TYPE_LIST are treated as if they were lists of length 1.
+
+The result will be put into @dest and will either be a list that will not
+contain any duplicates, or a non-list type (if @value1 and @value2
+were equal).
+
+
+
+
+
+ an uninitialized #GValue to take the result
+
+
+
+ a #GValue
+
+
+
+ a #GValue
+
+
+
+
+
+ Prepends @prepend_value to the GstValueList in @value.
+
+
+
+
+
+ a #GValue of type #GST_TYPE_LIST
+
+
+
+ the value to prepend
+
+
+
+
+
+
+ Used by gst_value_serialize() to obtain a non-binary form of the #GValue.
+
+Free-function: g_free
+
+ the string representation of the value
+
+
+
+
+ a #GValue
+
+
+
+
+
+ VTable for the #GValue @type.
+
+ a #GType
+
+
+
+ a #GstValueCompareFunc
+
+
+
+ a #GstValueSerializeFunc
+
+
+
+ a #GstValueDeserializeFunc
+
+
+
+
+
+
+
+
+
+ Get the maximum amount of memory blocks that a buffer can hold. This is a
+compile time constant that can be queried with the function.
+
+When more memory blocks are added, existing memory blocks will be merged
+together to make room for the new block.
+
+ the maximum amount of memory blocks that a buffer can hold.
+
+
+
+
+ Calculates the linear regression of the values @xy and places the
+result in @m_num, @m_denom, @b and @xbase, representing the function
+ y(x) = m_num/m_denom * (x - xbase) + b
+that has the least-square distance from all points @x and @y.
+
+@r_squared will contain the remaining error.
+
+If @temp is not %NULL, it will be used as temporary space for the function,
+in which case the function works without any allocation at all. If @temp is
+%NULL, an allocation will take place. @temp should have at least the same
+amount of memory allocated as @xy, i.e. 2*n*sizeof(GstClockTime).
+
+> This function assumes (x,y) values with reasonable large differences
+> between them. It will not calculate the exact results if the differences
+> between neighbouring values are too small due to not being able to
+> represent sub-integer values during the calculations.
+
+ %TRUE if the linear regression was successfully calculated
+
+
+
+
+ Pairs of (x,y) values
+
+
+
+ Temporary scratch space used by the function
+
+
+
+ number of (x,y) pairs
+
+
+
+ numerator of calculated slope
+
+
+
+ denominator of calculated slope
+
+
+
+ Offset at Y-axis
+
+
+
+ Offset at X-axis
+
+
+
+ R-squared
+
+
+
+
+
+ Creates a #GstCapsFeatures from a string representation.
+
+Free-function: gst_caps_features_free
+
+ a new #GstCapsFeatures or
+ %NULL when the string could not be parsed. Free with
+ gst_caps_features_free() after use.
+
+
+
+
+ a string representation of a #GstCapsFeatures.
+
+
+
+
+
+ Converts @caps from a string representation.
+
+The current implementation of serialization will lead to unexpected results
+when there are nested #GstCaps / #GstStructure deeper than one level.
+
+ a newly allocated #GstCaps
+
+
+
+
+ a string to convert to #GstCaps
+
+
+
+
+
+
+
+
+
+
+ Adds the logging function to the list of logging functions.
+Be sure to use #G_GNUC_NO_INSTRUMENT on that function, it is needed.
+
+
+
+
+
+ the function to use
+
+
+
+ user data
+
+
+
+ called when @user_data is not used anymore
+
+
+
+
+
+ Adds a memory ringbuffer based debug logger that stores up to
+@max_size_per_thread bytes of logs per thread and times out threads after
+@thread_timeout seconds of inactivity.
+
+Logs can be fetched with gst_debug_ring_buffer_logger_get_logs() and the
+logger can be removed again with gst_debug_remove_ring_buffer_logger().
+Only one logger at a time is possible.
+
+
+
+
+
+ Maximum size of log per thread in bytes
+
+
+
+ Timeout for threads in seconds
+
+
+
+
+
+ To aid debugging applications one can use this method to obtain the whole
+network of gstreamer elements that form the pipeline into an dot file.
+This data can be processed with graphviz to get an image.
+
+ a string containing the pipeline in graphviz
+dot format.
+
+
+
+
+ the top-level pipeline that should be analyzed
+
+
+
+
+
+
+
+
+ To aid debugging applications one can use this method to write out the whole
+network of gstreamer elements that form the pipeline into an dot file.
+This file can be processed with graphviz to get an image.
+<informalexample><programlisting>
+ dot -Tpng -oimage.png graph_lowlevel.dot
+</programlisting></informalexample>
+
+
+
+
+
+ the top-level pipeline that should be analyzed
+
+
+
+
+
+
+ output base filename (e.g. "myplayer")
+
+
+
+
+
+ This works like gst_debug_bin_to_dot_file(), but adds the current timestamp
+to the filename, so that it can be used to take multiple snapshots.
+
+
+
+
+
+ the top-level pipeline that should be analyzed
+
+
+
+
+
+
+ output base filename (e.g. "myplayer")
+
+
+
+
+
+ Constructs a string that can be used for getting the desired color in color
+terminals.
+You need to free the string after use.
+
+ a string containing the color
+ definition
+
+
+
+
+ the color info
+
+
+
+
+
+ Constructs an integer that can be used for getting the desired color in
+windows' terminals (cmd.exe). As there is no mean to underline, we simply
+ignore this attribute.
+
+This function returns 0 on non-windows machines.
+
+ an integer containing the color definition
+
+
+
+
+ the color info
+
+
+
+
+
+ Returns a snapshot of a all categories that are currently in use . This list
+may change anytime.
+The caller has to free the list after use.
+
+ the list of
+ debug categories
+
+
+
+
+
+
+ Changes the coloring mode for debug output.
+
+ see @GstDebugColorMode for possible values.
+
+
+
+
+ Returns the default threshold that is used for new categories.
+
+ the default threshold level
+
+
+
+
+
+ a stack trace, if libunwind or glibc backtrace are
+present, else %NULL.
+
+
+
+
+ A set of #GstStackTraceFlags to determine how the stack
+trace should look like. Pass 0 to retrieve a minimal backtrace.
+
+
+
+
+
+ Checks if debugging output is activated.
+
+ %TRUE, if debugging is activated
+
+
+
+
+ Checks if the debugging output should be colored.
+
+ %TRUE, if the debug output should be colored.
+
+
+
+
+ Get the string representation of a debugging level
+
+ the name
+
+
+
+
+ the level to get the name for
+
+
+
+
+
+ Logs the given message using the currently registered debugging handlers.
+
+
+
+
+
+ category to log
+
+
+
+ level of the message is in
+
+
+
+ the file that emitted the message, usually the __FILE__ identifier
+
+
+
+ the function that emitted the message
+
+
+
+ the line from that the message was emitted, usually __LINE__
+
+
+
+ the object this message relates to,
+ or %NULL if none
+
+
+
+ a printf style format string
+
+
+
+ optional arguments for the format
+
+
+
+
+
+ The default logging handler used by GStreamer. Logging functions get called
+whenever a macro like GST_DEBUG or similar is used. By default this function
+is setup to output the message and additional info to stderr (or the log file
+specified via the GST_DEBUG_FILE environment variable) as received via
+@user_data.
+
+You can add other handlers by using gst_debug_add_log_function().
+And you can remove this handler by calling
+gst_debug_remove_log_function(gst_debug_log_default);
+
+
+
+
+
+ category to log
+
+
+
+ level of the message
+
+
+
+ the file that emitted the message, usually the __FILE__ identifier
+
+
+
+ the function that emitted the message
+
+
+
+ the line from that the message was emitted, usually __LINE__
+
+
+
+ the object this message relates to,
+ or %NULL if none
+
+
+
+ the actual message
+
+
+
+ the FILE* to log to
+
+
+
+
+
+ Logs the given message using the currently registered debugging handlers.
+
+
+
+
+
+ category to log
+
+
+
+ level of the message is in
+
+
+
+ the file that emitted the message, usually the __FILE__ identifier
+
+
+
+ the function that emitted the message
+
+
+
+ the line from that the message was emitted, usually __LINE__
+
+
+
+ the object this message relates to,
+ or %NULL if none
+
+
+
+ a printf style format string
+
+
+
+ optional arguments for the format
+
+
+
+
+
+ If libunwind or glibc backtrace are present
+a stack trace is printed.
+
+
+
+
+
+ Removes all registered instances of the given logging functions.
+
+ How many instances of the function were removed
+
+
+
+
+ the log function to remove, or %NULL to
+ remove the default log function
+
+
+
+
+
+ Removes all registered instances of log functions with the given user data.
+
+ How many instances of the function were removed
+
+
+
+
+ user data of the log function to remove
+
+
+
+
+
+ Removes any previously added ring buffer logger with
+gst_debug_add_ring_buffer_logger().
+
+
+
+
+
+ Fetches the current logs per thread from the ring buffer logger. See
+gst_debug_add_ring_buffer_logger() for details.
+
+ NULL-terminated array of
+strings with the debug output per thread
+
+
+
+
+
+
+ If activated, debugging messages are sent to the debugging
+handlers.
+It makes sense to deactivate it for speed issues.
+> This function is not threadsafe. It makes sense to only call it
+during initialization.
+
+
+
+
+
+ Whether to use debugging output or not
+
+
+
+
+
+ Changes the coloring mode for debug output.
+
+This function may be called before gst_init().
+
+
+
+
+
+ The coloring mode for debug output. See @GstDebugColorMode.
+
+
+
+
+
+ Changes the coloring mode for debug output.
+
+This function may be called before gst_init().
+
+
+
+
+
+ The coloring mode for debug output. One of the following:
+"on", "auto", "off", "disable", "unix".
+
+
+
+
+
+ Sets or unsets the use of coloured debugging output.
+Same as gst_debug_set_color_mode () with the argument being
+being GST_DEBUG_COLOR_MODE_ON or GST_DEBUG_COLOR_MODE_OFF.
+
+This function may be called before gst_init().
+
+
+
+
+
+ Whether to use colored output or not
+
+
+
+
+
+ Sets the default threshold to the given level and updates all categories to
+use this threshold.
+
+This function may be called before gst_init().
+
+
+
+
+
+ level to set
+
+
+
+
+
+ Sets all categories which match the given glob style pattern to the given
+level.
+
+
+
+
+
+ name of the categories to set
+
+
+
+ level to set them to
+
+
+
+
+
+ Sets the debug logging wanted in the same form as with the GST_DEBUG
+environment variable. You can use wildcards such as '*', but note that
+the order matters when you use wild cards, e.g. "foosrc:6,*src:3,*:2" sets
+everything to log level 2.
+
+
+
+
+
+ comma-separated list of "category:level" pairs to be used
+ as debug logging levels
+
+
+
+ %TRUE to clear all previously-set debug levels before setting
+ new thresholds
+%FALSE if adding the threshold described by @list to the one already set.
+
+
+
+
+
+ Resets all categories with the given name back to the default level.
+
+
+
+
+
+ name of the categories to set
+
+
+
+
+
+ Clean up any resources created by GStreamer in gst_init().
+
+It is normally not needed to call this function in a normal application
+as the resources will automatically be freed when the program terminates.
+This function is therefore mostly used by testsuites and other memory
+profiling tools.
+
+After this call GStreamer (including this method) should not be used anymore.
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Get a string describing the error message in the current locale.
+
+ a newly allocated string describing
+ the error message (in UTF-8 encoding)
+
+
+
+
+ the GStreamer error domain this error belongs to.
+
+
+
+ the error code belonging to the domain.
+
+
+
+
+
+ Gets the #GstEventTypeFlags associated with @type.
+
+ a #GstEventTypeFlags.
+
+
+
+
+ a #GstEventType
+
+
+
+
+
+ Get a printable name for the given event type. Do not modify or free.
+
+ a reference to the static name of the event.
+
+
+
+
+ the event type
+
+
+
+
+
+ Get the unique quark for the given event type.
+
+ the quark associated with the event type
+
+
+
+
+ the event type
+
+
+
+
+
+ Similar to g_filename_to_uri(), but attempts to handle relative file paths
+as well. Before converting @filename into an URI, it will be prefixed by
+the current working directory if it is a relative path, and then the path
+will be canonicalised so that it doesn't contain any './' or '../' segments.
+
+On Windows #filename should be in UTF-8 encoding.
+
+ newly-allocated URI string, or NULL on error. The caller must
+ free the URI string with g_free() when no longer needed.
+
+
+
+
+ absolute or relative file name path
+
+
+
+
+
+ Gets a string representing the given flow return.
+
+ a static string with the name of the flow return.
+
+
+
+
+ a #GstFlowReturn to get the name of.
+
+
+
+
+
+ Get the unique quark for the given GstFlowReturn.
+
+ the quark associated with the flow return or 0 if an
+invalid return was specified.
+
+
+
+
+ a #GstFlowReturn to get the quark of.
+
+
+
+
+
+ Return the format registered with the given nick.
+
+ The format with @nick or GST_FORMAT_UNDEFINED
+if the format was not registered.
+
+
+
+
+ The nick of the format
+
+
+
+
+
+ Get details about the given format.
+
+ The #GstFormatDefinition for @format or %NULL
+on failure.
+
+MT safe.
+
+
+
+
+ The format to get details of
+
+
+
+
+
+ Get a printable name for the given format. Do not modify or free.
+
+ a reference to the static name of the format
+or %NULL if the format is unknown.
+
+
+
+
+ a #GstFormat
+
+
+
+
+
+ Iterate all the registered formats. The format definition is read
+only.
+
+ a GstIterator of #GstFormatDefinition.
+
+
+
+
+ Create a new GstFormat based on the nick or return an
+already registered format with that nick.
+
+ A new GstFormat or an already registered format
+with the same nick.
+
+MT safe.
+
+
+
+
+ The nick of the new format
+
+
+
+ The description of the new format
+
+
+
+
+
+ Get the unique quark for the given format.
+
+ the quark associated with the format or 0 if the format
+is unknown.
+
+
+
+
+ a #GstFormat
+
+
+
+
+
+ See if the given format is inside the format array.
+
+ %TRUE if the format is found inside the array
+
+
+
+
+ The format array to search
+
+
+
+
+
+ the format to find
+
+
+
+
+
+ This helper is mostly helpful for plugins that need to
+inspect the folder of the main executable to determine
+their set of features.
+
+When a plugin is initialized from the gst-plugin-scanner
+external process, the returned path will be the same as from the
+parent process.
+
+ The path of the executable that
+ initialized GStreamer, or %NULL if it could not be determined.
+
+
+
+
+ Allocates, fills and returns a 0-terminated string from the printf style
+@format string and corresponding arguments.
+
+See gst_info_vasprintf() for when this function is required.
+
+Free with g_free().
+
+ a newly allocated null terminated string or %NULL on any error
+
+
+
+
+ a printf style format string
+
+
+
+ the printf arguments for @format
+
+
+
+
+
+ Allocates, fills and returns a null terminated string from the printf style
+@format string and @args.
+
+See gst_info_vasprintf() for when this function is required.
+
+Free with g_free().
+
+ a newly allocated null terminated string or %NULL on any error
+
+
+
+
+ a printf style format string
+
+
+
+ the va_list of printf arguments for @format
+
+
+
+
+
+ Allocates and fills a string large enough (including the terminating null
+byte) to hold the specified printf style @format and @args.
+
+This function deals with the GStreamer specific printf specifiers
+#GST_PTR_FORMAT and #GST_SEGMENT_FORMAT. If you do not have these specifiers
+in your @format string, you do not need to use this function and can use
+alternatives such as g_vasprintf().
+
+Free @result with g_free().
+
+ the length of the string allocated into @result or -1 on any error
+
+
+
+
+ the resulting string
+
+
+
+ a printf style format string
+
+
+
+ the va_list of printf arguments for @format
+
+
+
+
+
+ Initializes the GStreamer library, setting up internal path lists,
+registering built-in elements, and loading standard plugins.
+
+Unless the plugin registry is disabled at compile time, the registry will be
+loaded. By default this will also check if the registry cache needs to be
+updated and rescan all plugins if needed. See gst_update_registry() for
+details and section
+<link linkend="gst-running">Running GStreamer Applications</link>
+for how to disable automatic registry updates.
+
+> This function will terminate your program if it was unable to initialize
+> GStreamer for some reason. If you want your program to fall back,
+> use gst_init_check() instead.
+
+WARNING: This function does not work in the same way as corresponding
+functions in other glib-style libraries, such as gtk_init\(\). In
+particular, unknown command line options cause this function to
+abort program execution.
+
+
+
+
+
+ pointer to application's argc
+
+
+
+ pointer to application's argv
+
+
+
+
+
+
+
+ Initializes the GStreamer library, setting up internal path lists,
+registering built-in elements, and loading standard plugins.
+
+This function will return %FALSE if GStreamer could not be initialized
+for some reason. If you want your program to fail fatally,
+use gst_init() instead.
+
+ %TRUE if GStreamer could be initialized.
+
+
+
+
+ pointer to application's argc
+
+
+
+ pointer to application's argv
+
+
+
+
+
+
+
+ Returns a #GOptionGroup with GStreamer's argument specifications. The
+group is set up to use standard GOption callbacks, so when using this
+group in combination with GOption parsing methods, all argument parsing
+and initialization is automated.
+
+This function is useful if you want to integrate GStreamer with other
+libraries that use GOption (see g_option_context_add_group() ).
+
+If you use this function, you should make sure you initialise the GLib
+threading system as one of the very first things in your program
+(see the example at the beginning of this section).
+
+ a pointer to GStreamer's option group.
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Use this function to check if GStreamer has been initialized with gst_init()
+or gst_init_check().
+
+ %TRUE if initialization has been done, %FALSE otherwise.
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Get a printable name for the given message type. Do not modify or free.
+
+ a reference to the static name of the message.
+
+
+
+
+ the message type
+
+
+
+
+
+ Get the unique quark for the given message type.
+
+ the quark associated with the message type
+
+
+
+
+ the message type
+
+
+
+
+
+
+ an array of tags as strings.
+
+
+
+
+
+
+ an API
+
+
+
+
+
+ Check if @api was registered with @tag.
+
+ %TRUE if @api was registered with @tag.
+
+
+
+
+ an API
+
+
+
+ the tag to check
+
+
+
+
+
+ Register and return a GType for the @api and associate it with
+@tags.
+
+ a unique GType for @api.
+
+
+
+
+ an API to register
+
+
+
+ tags for @api
+
+
+
+
+
+ Lookup a previously registered meta info structure by its implementation name
+@impl.
+
+ a #GstMetaInfo with @impl, or
+%NULL when no such metainfo exists.
+
+
+
+
+ the name
+
+
+
+
+
+ Register a new #GstMeta implementation.
+
+The same @info can be retrieved later with gst_meta_get_info() by using
+@impl as the key.
+
+ a #GstMetaInfo that can be used to
+access metadata.
+
+
+
+
+ the type of the #GstMeta API
+
+
+
+ the name of the #GstMeta implementation
+
+
+
+ the size of the #GstMeta structure
+
+
+
+ a #GstMetaInitFunction
+
+
+
+ a #GstMetaFreeFunction
+
+
+
+ a #GstMetaTransformFunction
+
+
+
+
+
+ Atomically modifies a pointer to point to a new mini-object.
+The reference count of @olddata is decreased and the reference count of
+@newdata is increased.
+
+Either @newdata and the value pointed to by @olddata may be %NULL.
+
+ %TRUE if @newdata was different from @olddata
+
+
+
+
+ pointer to a pointer to a
+ mini-object to be replaced
+
+
+
+ pointer to new mini-object
+
+
+
+
+
+ Replace the current #GstMiniObject pointer to by @olddata with %NULL and
+return the old value.
+
+ the #GstMiniObject at @oldata
+
+
+
+
+ pointer to a pointer to a mini-object to
+ be stolen
+
+
+
+
+
+ Modifies a pointer to point to a new mini-object. The modification
+is done atomically. This version is similar to gst_mini_object_replace()
+except that it does not increase the refcount of @newdata and thus
+takes ownership of @newdata.
+
+Either @newdata and the value pointed to by @olddata may be %NULL.
+
+ %TRUE if @newdata was different from @olddata
+
+
+
+
+ pointer to a pointer to a mini-object to
+ be replaced
+
+
+
+ pointer to new mini-object
+
+
+
+
+
+ Return the name of a pad mode, for use in debug messages mostly.
+
+ short mnemonic for pad mode @mode
+
+
+
+
+ the pad mode
+
+
+
+
+
+ This function creates a GstArray GParamSpec for use by objects/elements
+that want to expose properties of GstArray type. This function is
+typically * used in connection with g_object_class_install_property() in a
+GObjects's instance_init function.
+
+ a newly created parameter specification
+
+
+
+
+ canonical name of the property specified
+
+
+
+ nick name for the property specified
+
+
+
+ description of the property specified
+
+
+
+ GParamSpec of the array
+
+
+
+ flags for the property specified
+
+
+
+
+
+ This function creates a fraction GParamSpec for use by objects/elements
+that want to expose properties of fraction type. This function is typically
+used in connection with g_object_class_install_property() in a GObjects's
+instance_init function.
+
+ a newly created parameter specification
+
+
+
+
+ canonical name of the property specified
+
+
+
+ nick name for the property specified
+
+
+
+ description of the property specified
+
+
+
+ minimum value (fraction numerator)
+
+
+
+ minimum value (fraction denominator)
+
+
+
+ maximum value (fraction numerator)
+
+
+
+ maximum value (fraction denominator)
+
+
+
+ default value (fraction numerator)
+
+
+
+ default value (fraction denominator)
+
+
+
+ flags for the property specified
+
+
+
+
+
+
+
+
+
+
+ Get the global #GstMetaInfo describing the #GstParentBufferMeta meta.
+
+ The #GstMetaInfo
+
+
+
+
+ This is a convenience wrapper around gst_parse_launch() to create a
+#GstBin from a gst-launch-style pipeline description. See
+gst_parse_launch() and the gst-launch man page for details about the
+syntax. Ghost pads on the bin for unlinked source or sink pads
+within the bin can automatically be created (but only a maximum of
+one ghost pad for each direction will be created; if you expect
+multiple unlinked source pads or multiple unlinked sink pads
+and want them all ghosted, you will have to create the ghost pads
+yourself).
+
+ a
+ newly-created bin, or %NULL if an error occurred.
+
+
+
+
+ command line describing the bin
+
+
+
+ whether to automatically create ghost pads
+ for unlinked source or sink pads within the bin
+
+
+
+
+
+ This is a convenience wrapper around gst_parse_launch() to create a
+#GstBin from a gst-launch-style pipeline description. See
+gst_parse_launch() and the gst-launch man page for details about the
+syntax. Ghost pads on the bin for unlinked source or sink pads
+within the bin can automatically be created (but only a maximum of
+one ghost pad for each direction will be created; if you expect
+multiple unlinked source pads or multiple unlinked sink pads
+and want them all ghosted, you will have to create the ghost pads
+yourself).
+
+ a newly-created
+ element, which is guaranteed to be a bin unless
+ GST_FLAG_NO_SINGLE_ELEMENT_BINS was passed, or %NULL if an error
+ occurred.
+
+
+
+
+ command line describing the bin
+
+
+
+ whether to automatically create ghost pads
+ for unlinked source or sink pads within the bin
+
+
+
+ a parse context allocated with
+ gst_parse_context_new(), or %NULL
+
+
+
+ parsing options, or #GST_PARSE_FLAG_NONE
+
+
+
+
+
+ Get the error quark used by the parsing subsystem.
+
+ the quark of the parse errors.
+
+
+
+
+ Create a new pipeline based on command line syntax.
+Please note that you might get a return value that is not %NULL even though
+the @error is set. In this case there was a recoverable parsing error and you
+can try to play the pipeline.
+
+ a new element on success, %NULL on
+ failure. If more than one toplevel element is specified by the
+ @pipeline_description, all elements are put into a #GstPipeline, which
+ than is returned.
+
+
+
+
+ the command line describing the pipeline
+
+
+
+
+
+ Create a new pipeline based on command line syntax.
+Please note that you might get a return value that is not %NULL even though
+the @error is set. In this case there was a recoverable parsing error and you
+can try to play the pipeline.
+
+ a new element on success, %NULL on
+ failure. If more than one toplevel element is specified by the
+ @pipeline_description, all elements are put into a #GstPipeline, which
+ then is returned (unless the GST_PARSE_FLAG_PLACE_IN_BIN flag is set, in
+ which case they are put in a #GstBin instead).
+
+
+
+
+ the command line describing the pipeline
+
+
+
+ a parse context allocated with
+ gst_parse_context_new(), or %NULL
+
+
+
+ parsing options, or #GST_PARSE_FLAG_NONE
+
+
+
+
+
+ Create a new element based on command line syntax.
+@error will contain an error message if an erroneous pipeline is specified.
+An error does not mean that the pipeline could not be constructed.
+
+ a new element on success and %NULL
+on failure.
+
+
+
+
+ null-terminated array of arguments
+
+
+
+
+
+
+
+ Create a new element based on command line syntax.
+@error will contain an error message if an erroneous pipeline is specified.
+An error does not mean that the pipeline could not be constructed.
+
+ a new element on success; on
+ failure, either %NULL or a partially-constructed bin or element will be
+ returned and @error will be set (unless you passed
+ #GST_PARSE_FLAG_FATAL_ERRORS in @flags, then %NULL will always be returned
+ on failure)
+
+
+
+
+ null-terminated array of arguments
+
+
+
+
+
+ a parse context allocated with
+ gst_parse_context_new(), or %NULL
+
+
+
+ parsing options, or #GST_PARSE_FLAG_NONE
+
+
+
+
+
+ Get the error quark.
+
+ The error quark used in GError messages
+
+
+
+
+ Create a new file descriptor set. If @controllable, it
+is possible to restart or flush a call to gst_poll_wait() with
+gst_poll_restart() and gst_poll_set_flushing() respectively.
+
+Free-function: gst_poll_free
+
+ a new #GstPoll, or %NULL in
+ case of an error. Free with gst_poll_free().
+
+
+
+
+ whether it should be possible to control a wait.
+
+
+
+
+
+ Create a new poll object that can be used for scheduling cancellable
+timeouts.
+
+A timeout is performed with gst_poll_wait(). Multiple timeouts can be
+performed from different threads.
+
+Free-function: gst_poll_free
+
+ a new #GstPoll, or %NULL in
+ case of an error. Free with gst_poll_free().
+
+
+
+
+ Gets the directory for application specific presets if set by the
+application.
+
+ the directory or %NULL, don't free or modify
+the string
+
+
+
+
+ Sets an extra directory as an absolute path that should be considered when
+looking for presets. Any presets in the application dir will shadow the
+system presets.
+
+ %TRUE for success, %FALSE if the dir already has been set
+
+
+
+
+ the application specific preset dir
+
+
+
+
+
+ Outputs a formatted message via the GLib print handler. The default print
+handler simply outputs the message to stdout.
+
+This function will not append a new-line character at the end, unlike
+gst_println() which will.
+
+All strings must be in ASCII or UTF-8 encoding.
+
+This function differs from g_print() in that it supports all the additional
+printf specifiers that are supported by GStreamer's debug logging system,
+such as #GST_PTR_FORMAT and #GST_SEGMENT_FORMAT.
+
+This function is primarily for printing debug output.
+
+
+
+
+
+ a printf style format string
+
+
+
+ the printf arguments for @format
+
+
+
+
+
+ Outputs a formatted message via the GLib error message handler. The default
+handler simply outputs the message to stderr.
+
+This function will not append a new-line character at the end, unlike
+gst_printerrln() which will.
+
+All strings must be in ASCII or UTF-8 encoding.
+
+This function differs from g_printerr() in that it supports the additional
+printf specifiers that are supported by GStreamer's debug logging system,
+such as #GST_PTR_FORMAT and #GST_SEGMENT_FORMAT.
+
+This function is primarily for printing debug output.
+
+
+
+
+
+ a printf style format string
+
+
+
+ the printf arguments for @format
+
+
+
+
+
+ Outputs a formatted message via the GLib error message handler. The default
+handler simply outputs the message to stderr.
+
+This function will append a new-line character at the end, unlike
+gst_printerr() which will not.
+
+All strings must be in ASCII or UTF-8 encoding.
+
+This function differs from g_printerr() in that it supports the additional
+printf specifiers that are supported by GStreamer's debug logging system,
+such as #GST_PTR_FORMAT and #GST_SEGMENT_FORMAT.
+
+This function is primarily for printing debug output.
+
+
+
+
+
+ a printf style format string
+
+
+
+ the printf arguments for @format
+
+
+
+
+
+ Outputs a formatted message via the GLib print handler. The default print
+handler simply outputs the message to stdout.
+
+This function will append a new-line character at the end, unlike
+gst_print() which will not.
+
+All strings must be in ASCII or UTF-8 encoding.
+
+This function differs from g_print() in that it supports all the additional
+printf specifiers that are supported by GStreamer's debug logging system,
+such as #GST_PTR_FORMAT and #GST_SEGMENT_FORMAT.
+
+This function is primarily for printing debug output.
+
+
+
+
+
+ a printf style format string
+
+
+
+ the printf arguments for @format
+
+
+
+
+
+ Iterates the supplied list of UUIDs and checks the GstRegistry for
+all the decryptors supporting one of the supplied UUIDs.
+
+ A null terminated array containing all
+the @system_identifiers supported by the set of available decryptors, or
+%NULL if no matches were found.
+
+
+
+
+
+
+ A null terminated array of strings
+that contains the UUID values of each protection system that is to be
+checked.
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Iterates the supplied list of UUIDs and checks the GstRegistry for
+an element that supports one of the supplied UUIDs. If more than one
+element matches, the system ID of the highest ranked element is selected.
+
+ One of the strings from
+@system_identifiers that indicates the highest ranked element that
+implements the protection system indicated by that system ID, or %NULL if no
+element has been found.
+
+
+
+
+ A null terminated array of strings
+that contains the UUID values of each protection system that is to be
+checked.
+
+
+
+
+
+
+
+ Gets the #GstQueryTypeFlags associated with @type.
+
+ a #GstQueryTypeFlags.
+
+
+
+
+ a #GstQueryType
+
+
+
+
+
+ Get a printable name for the given query type. Do not modify or free.
+
+ a reference to the static name of the query.
+
+
+
+
+ the query type
+
+
+
+
+
+ Get the unique quark for the given query type.
+
+ the quark associated with the query type
+
+
+
+
+ the query type
+
+
+
+
+
+
+
+
+
+
+ Get the global #GstMetaInfo describing the #GstReferenceTimestampMeta meta.
+
+ The #GstMetaInfo
+
+
+
+
+
+
+
+
+
+ Some functions in the GStreamer core might install a custom SIGSEGV handler
+to better catch and report errors to the application. Currently this feature
+is enabled by default when loading plugins.
+
+Applications might want to disable this behaviour with the
+gst_segtrap_set_enabled() function. This is typically done if the application
+wants to install its own handler without GStreamer interfering.
+
+ %TRUE if GStreamer is allowed to install a custom SIGSEGV handler.
+
+
+
+
+ Applications might want to disable/enable the SIGSEGV handling of
+the GStreamer core. See gst_segtrap_is_enabled() for more information.
+
+
+
+
+
+ whether a custom SIGSEGV handler should be installed.
+
+
+
+
+
+ Gets a string representing the given state transition.
+
+ a string with the name of the state
+ result.
+
+
+
+
+ a #GstStateChange to get the name of.
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Get a descriptive string for a given #GstStreamType
+
+ A string describing the stream type
+
+
+
+
+ a #GstStreamType
+
+
+
+
+
+ Creates a #GstStructure from a string representation.
+If end is not %NULL, a pointer to the place inside the given string
+where parsing ended will be returned.
+
+Free-function: gst_structure_free
+
+ a new #GstStructure or %NULL
+ when the string could not be parsed. Free with
+ gst_structure_free() after use.
+
+
+
+
+ a string representation of a #GstStructure.
+
+
+
+ pointer to store the end of the string in.
+
+
+
+
+
+ Checks if the given type is already registered.
+
+ %TRUE if the type is already registered
+
+
+
+
+ name of the tag
+
+
+
+
+
+ Returns the human-readable description of this tag, You must not change or
+free this string.
+
+ the human-readable description of this tag
+
+
+
+
+ the tag
+
+
+
+
+
+ Gets the flag of @tag.
+
+ the flag of this tag.
+
+
+
+
+ the tag
+
+
+
+
+
+ Returns the human-readable name of this tag, You must not change or free
+this string.
+
+ the human-readable name of this tag
+
+
+
+
+ the tag
+
+
+
+
+
+ Gets the #GType used for this tag.
+
+ the #GType of this tag
+
+
+
+
+ the tag
+
+
+
+
+
+ Checks if the given tag is fixed. A fixed tag can only contain one value.
+Unfixed tags can contain lists of values.
+
+ %TRUE, if the given tag is fixed.
+
+
+
+
+ tag to check
+
+
+
+
+
+ Copies the contents for the given tag into the value,
+merging multiple values into one if multiple values are associated
+with the tag.
+You must g_value_unset() the value after use.
+
+ %TRUE, if a value was copied, %FALSE if the tag didn't exist in the
+ given list.
+
+
+
+
+ uninitialized #GValue to copy into
+
+
+
+ list to get the tag from
+
+
+
+ tag to read out
+
+
+
+
+
+ This is a convenience function for the func argument of gst_tag_register().
+It concatenates all given strings using a comma. The tag must be registered
+as a G_TYPE_STRING or this function will fail.
+
+
+
+
+
+ uninitialized GValue to store result in
+
+
+
+ GValue to copy from
+
+
+
+
+
+ This is a convenience function for the func argument of gst_tag_register().
+It creates a copy of the first value from the list.
+
+
+
+
+
+ uninitialized GValue to store result in
+
+
+
+ GValue to copy from
+
+
+
+
+
+ Registers a new tag type for the use with GStreamer's type system. If a type
+with that name is already registered, that one is used.
+The old registration may have used a different type however. So don't rely
+on your supplied values.
+
+Important: if you do not supply a merge function the implication will be
+that there can only be one single value for this tag in a tag list and
+any additional values will silently be discarded when being added (unless
+#GST_TAG_MERGE_REPLACE, #GST_TAG_MERGE_REPLACE_ALL, or
+#GST_TAG_MERGE_PREPEND is used as merge mode, in which case the new
+value will replace the old one in the list).
+
+The merge function will be called from gst_tag_list_copy_value() when
+it is required that one or more values for a tag be condensed into
+one single value. This may happen from gst_tag_list_get_string(),
+gst_tag_list_get_int(), gst_tag_list_get_double() etc. What will happen
+exactly in that case depends on how the tag was registered and if a
+merge function was supplied and if so which one.
+
+Two default merge functions are provided: gst_tag_merge_use_first() and
+gst_tag_merge_strings_with_comma().
+
+
+
+
+
+ the name or identifier string
+
+
+
+ a flag describing the type of tag info
+
+
+
+ the type this data is in
+
+
+
+ human-readable name
+
+
+
+ a human-readable description about this tag
+
+
+
+ function for merging multiple values of this tag, or %NULL
+
+
+
+
+
+ Registers a new tag type for the use with GStreamer's type system.
+
+Same as gst_tag_register(), but @name, @nick, and @blurb must be
+static strings or inlined strings, as they will not be copied. (GStreamer
+plugins will be made resident once loaded, so this function can be used
+even from dynamically loaded plugins.)
+
+
+
+
+
+ the name or identifier string (string constant)
+
+
+
+ a flag describing the type of tag info
+
+
+
+ the type this data is in
+
+
+
+ human-readable name or short description (string constant)
+
+
+
+ a human-readable description for this tag (string constant)
+
+
+
+ function for merging multiple values of this tag, or %NULL
+
+
+
+
+
+ Converts @type to a string representation.
+
+ Returns a human-readable string for @type. This string is
+ only for debugging purpose and should not be displayed in a user
+ interface.
+
+
+
+
+ a #GstTocEntryType.
+
+
+
+
+
+
+
+
+
+
+ Registers a new typefind function to be used for typefinding. After
+registering this function will be available for typefinding.
+This function is typically called during an element's plugin initialization.
+
+ %TRUE on success, %FALSE otherwise
+
+
+
+
+ A #GstPlugin, or %NULL for a static typefind function
+
+
+
+ The name for registering
+
+
+
+ The rank (or importance) of this typefind function
+
+
+
+ The #GstTypeFindFunction to use
+
+
+
+ Optional comma-separated list of extensions
+ that could belong to this type
+
+
+
+ Optionally the caps that could be returned when typefinding
+ succeeds
+
+
+
+ Optional user data. This user data must be available until the plugin
+ is unloaded.
+
+
+
+ a #GDestroyNotify that will be called on @data when the plugin
+ is unloaded.
+
+
+
+
+
+ Forces GStreamer to re-scan its plugin paths and update the default
+plugin registry.
+
+Applications will almost never need to call this function, it is only
+useful if the application knows new plugins have been installed (or old
+ones removed) since the start of the application (or, to be precise, the
+first call to gst_init()) and the application wants to make use of any
+newly-installed plugins without restarting the application.
+
+Applications should assume that the registry update is neither atomic nor
+thread-safe and should therefore not have any dynamic pipelines running
+(including the playbin and decodebin elements) and should also not create
+any elements or access the GStreamer registry while the update is in
+progress.
+
+Note that this function may block for a significant amount of time.
+
+ %TRUE if the registry has been updated successfully (does not
+ imply that there were changes), otherwise %FALSE.
+
+
+
+
+ Constructs a URI for a given valid protocol and location.
+
+Free-function: g_free
+ Use GstURI instead.
+
+ a new string for this URI. Returns %NULL if the
+ given URI protocol is not valid, or the given location is %NULL.
+
+
+
+
+ Protocol for URI
+
+
+
+ Location for URI
+
+
+
+
+
+
+
+
+
+
+ Parses a URI string into a new #GstUri object. Will return NULL if the URI
+cannot be parsed.
+
+ A new #GstUri object, or NULL.
+
+
+
+
+ The URI string to parse.
+
+
+
+
+
+ Extracts the location out of a given valid URI, ie. the protocol and "://"
+are stripped from the URI, which means that the location returned includes
+the hostname if one is specified. The returned string must be freed using
+g_free().
+
+Free-function: g_free
+
+ the location for this URI. Returns
+ %NULL if the URI isn't valid. If the URI does not contain a location, an
+ empty string is returned.
+
+
+
+
+ A URI string
+
+
+
+
+
+ Extracts the protocol out of a given valid URI. The returned string must be
+freed using g_free().
+
+ The protocol for this URI.
+
+
+
+
+ A URI string
+
+
+
+
+
+ Checks if the protocol of a given valid URI matches @protocol.
+
+ %TRUE if the protocol matches.
+
+
+
+
+ a URI string
+
+
+
+ a protocol string (e.g. "http")
+
+
+
+
+
+ Tests if the given string is a valid URI identifier. URIs start with a valid
+scheme followed by ":" and maybe a string identifying the location.
+
+ %TRUE if the string is a valid URI
+
+
+
+
+ A URI string
+
+
+
+
+
+ This is a convenience function to join two URI strings and return the result.
+The returned string should be g_free()'d after use.
+
+ A string representing the percent-encoded join of
+ the two URIs.
+
+
+
+
+ The percent-encoded base URI.
+
+
+
+ The percent-encoded reference URI to join to the @base_uri.
+
+
+
+
+
+ Checks if an element exists that supports the given URI protocol. Note
+that a positive return value does not imply that a subsequent call to
+gst_element_make_from_uri() is guaranteed to work.
+
+ %TRUE
+
+
+
+
+ Whether to check for a source or a sink
+
+
+
+ Protocol that should be checked for (e.g. "http" or "smb")
+
+
+
+
+
+ Tests if the given string is a valid protocol identifier. Protocols
+must consist of alphanumeric characters, '+', '-' and '.' and must
+start with a alphabetic character. See RFC 3986 Section 3.1.
+
+ %TRUE if the string is a valid protocol identifier, %FALSE otherwise.
+
+
+
+
+ A string
+
+
+
+
+
+ Searches inside @array for @search_data by using the comparison function
+@search_func. @array must be sorted ascending.
+
+As @search_data is always passed as second argument to @search_func it's
+not required that @search_data has the same type as the array elements.
+
+The complexity of this search function is O(log (num_elements)).
+
+ The address of the found
+element or %NULL if nothing was found
+
+
+
+
+ the sorted input array
+
+
+
+ number of elements in the array
+
+
+
+ size of every element in bytes
+
+
+
+ function to compare two elements, @search_data will always be passed as second argument
+
+
+
+ search mode that should be used
+
+
+
+ element that should be found
+
+
+
+ data to pass to @search_func
+
+
+
+
+
+ Transforms a #gdouble to a fraction and simplifies
+the result.
+
+
+
+
+
+ #gdouble to transform
+
+
+
+ pointer to a #gint to hold the result numerator
+
+
+
+ pointer to a #gint to hold the result denominator
+
+
+
+
+
+ Dumps the buffer memory into a hex representation. Useful for debugging.
+
+
+
+
+
+ a #GstBuffer whose memory to dump
+
+
+
+
+
+ Dumps the memory block into a hex representation. Useful for debugging.
+
+
+
+
+
+ a pointer to the memory to dump
+
+
+
+
+
+ the size of the memory block to dump
+
+
+
+
+
+ Adds the fractions @a_n/@a_d and @b_n/@b_d and stores
+the result in @res_n and @res_d.
+
+ %FALSE on overflow, %TRUE otherwise.
+
+
+
+
+ Numerator of first value
+
+
+
+ Denominator of first value
+
+
+
+ Numerator of second value
+
+
+
+ Denominator of second value
+
+
+
+ Pointer to #gint to hold the result numerator
+
+
+
+ Pointer to #gint to hold the result denominator
+
+
+
+
+
+ Compares the fractions @a_n/@a_d and @b_n/@b_d and returns
+-1 if a < b, 0 if a = b and 1 if a > b.
+
+ -1 if a < b; 0 if a = b; 1 if a > b.
+
+
+
+
+ Numerator of first value
+
+
+
+ Denominator of first value
+
+
+
+ Numerator of second value
+
+
+
+ Denominator of second value
+
+
+
+
+
+ Multiplies the fractions @a_n/@a_d and @b_n/@b_d and stores
+the result in @res_n and @res_d.
+
+ %FALSE on overflow, %TRUE otherwise.
+
+
+
+
+ Numerator of first value
+
+
+
+ Denominator of first value
+
+
+
+ Numerator of second value
+
+
+
+ Denominator of second value
+
+
+
+ Pointer to #gint to hold the result numerator
+
+
+
+ Pointer to #gint to hold the result denominator
+
+
+
+
+
+ Transforms a fraction to a #gdouble.
+
+
+
+
+
+ Fraction numerator as #gint
+
+
+
+ Fraction denominator #gint
+
+
+
+ pointer to a #gdouble for the result
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Get a property of type %GST_TYPE_ARRAY and transform it into a
+#GValueArray. This allow language bindings to get GST_TYPE_ARRAY
+properties which are otherwise not an accessible type.
+
+
+
+
+
+ the object to set the array to
+
+
+
+ the name of the property to set
+
+
+
+ a return #GValueArray
+
+
+
+
+
+ Get a timestamp as GstClockTime to be used for interval measurements.
+The timestamp should not be interpreted in any other way.
+
+ the timestamp
+
+
+
+
+ Calculates the greatest common divisor of @a
+and @b.
+
+ Greatest common divisor of @a and @b
+
+
+
+
+ First value as #gint
+
+
+
+ Second value as #gint
+
+
+
+
+
+ Calculates the greatest common divisor of @a
+and @b.
+
+ Greatest common divisor of @a and @b
+
+
+
+
+ First value as #gint64
+
+
+
+ Second value as #gint64
+
+
+
+
+
+ Return a constantly incrementing group id.
+
+This function is used to generate a new group-id for the
+stream-start event.
+
+This function never returns %GST_GROUP_ID_INVALID (which is 0)
+
+ A constantly incrementing unsigned integer, which might
+overflow back to 0 at some point.
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Compare two sequence numbers, handling wraparound.
+
+The current implementation just returns (gint32)(@s1 - @s2).
+
+ A negative number if @s1 is before @s2, 0 if they are equal, or a
+positive number if @s1 is after @s2.
+
+
+
+
+ A sequence number.
+
+
+
+ Another sequence number.
+
+
+
+
+
+ Return a constantly incrementing sequence number.
+
+This function is used internally to GStreamer to be able to determine which
+events and messages are "the same". For example, elements may set the seqnum
+on a segment-done message to be the same as that of the last seek event, to
+indicate that event and the message correspond to the same segment.
+
+This function never returns %GST_SEQNUM_INVALID (which is 0).
+
+ A constantly incrementing 32-bit unsigned integer, which might
+overflow at some point. Use gst_util_seqnum_compare() to make sure
+you handle wraparound correctly.
+
+
+
+
+ Converts the string value to the type of the objects argument and
+sets the argument with it.
+
+Note that this function silently returns if @object has no property named
+@name or when @value cannot be converted to the type of the property.
+
+
+
+
+
+ the object to set the argument of
+
+
+
+ the name of the argument to set
+
+
+
+ the string value to set
+
+
+
+
+
+ Transfer a #GValueArray to %GST_TYPE_ARRAY and set this value on the
+specified property name. This allow language bindings to set GST_TYPE_ARRAY
+properties which are otherwise not an accessible type.
+
+
+
+
+
+ the object to set the array to
+
+
+
+ the name of the property to set
+
+
+
+ a #GValueArray containing the values
+
+
+
+
+
+ Converts the string to the type of the value and
+sets the value with it.
+
+Note that this function is dangerous as it does not return any indication
+if the conversion worked or not.
+
+
+
+
+
+ the value to set
+
+
+
+ the string to get the value from
+
+
+
+
+
+ Scale @val by the rational number @num / @denom, avoiding overflows and
+underflows and without loss of precision.
+
+This function can potentially be very slow if val and num are both
+greater than G_MAXUINT32.
+
+ @val * @num / @denom. In the case of an overflow, this
+function returns G_MAXUINT64. If the result is not exactly
+representable as an integer it is truncated. See also
+gst_util_uint64_scale_round(), gst_util_uint64_scale_ceil(),
+gst_util_uint64_scale_int(), gst_util_uint64_scale_int_round(),
+gst_util_uint64_scale_int_ceil().
+
+
+
+
+ the number to scale
+
+
+
+ the numerator of the scale ratio
+
+
+
+ the denominator of the scale ratio
+
+
+
+
+
+ Scale @val by the rational number @num / @denom, avoiding overflows and
+underflows and without loss of precision.
+
+This function can potentially be very slow if val and num are both
+greater than G_MAXUINT32.
+
+ @val * @num / @denom. In the case of an overflow, this
+function returns G_MAXUINT64. If the result is not exactly
+representable as an integer, it is rounded up. See also
+gst_util_uint64_scale(), gst_util_uint64_scale_round(),
+gst_util_uint64_scale_int(), gst_util_uint64_scale_int_round(),
+gst_util_uint64_scale_int_ceil().
+
+
+
+
+ the number to scale
+
+
+
+ the numerator of the scale ratio
+
+
+
+ the denominator of the scale ratio
+
+
+
+
+
+ Scale @val by the rational number @num / @denom, avoiding overflows and
+underflows and without loss of precision. @num must be non-negative and
+@denom must be positive.
+
+ @val * @num / @denom. In the case of an overflow, this
+function returns G_MAXUINT64. If the result is not exactly
+representable as an integer, it is truncated. See also
+gst_util_uint64_scale_int_round(), gst_util_uint64_scale_int_ceil(),
+gst_util_uint64_scale(), gst_util_uint64_scale_round(),
+gst_util_uint64_scale_ceil().
+
+
+
+
+ guint64 (such as a #GstClockTime) to scale.
+
+
+
+ numerator of the scale factor.
+
+
+
+ denominator of the scale factor.
+
+
+
+
+
+ Scale @val by the rational number @num / @denom, avoiding overflows and
+underflows and without loss of precision. @num must be non-negative and
+@denom must be positive.
+
+ @val * @num / @denom. In the case of an overflow, this
+function returns G_MAXUINT64. If the result is not exactly
+representable as an integer, it is rounded up. See also
+gst_util_uint64_scale_int(), gst_util_uint64_scale_int_round(),
+gst_util_uint64_scale(), gst_util_uint64_scale_round(),
+gst_util_uint64_scale_ceil().
+
+
+
+
+ guint64 (such as a #GstClockTime) to scale.
+
+
+
+ numerator of the scale factor.
+
+
+
+ denominator of the scale factor.
+
+
+
+
+
+ Scale @val by the rational number @num / @denom, avoiding overflows and
+underflows and without loss of precision. @num must be non-negative and
+@denom must be positive.
+
+ @val * @num / @denom. In the case of an overflow, this
+function returns G_MAXUINT64. If the result is not exactly
+representable as an integer, it is rounded to the nearest integer
+(half-way cases are rounded up). See also gst_util_uint64_scale_int(),
+gst_util_uint64_scale_int_ceil(), gst_util_uint64_scale(),
+gst_util_uint64_scale_round(), gst_util_uint64_scale_ceil().
+
+
+
+
+ guint64 (such as a #GstClockTime) to scale.
+
+
+
+ numerator of the scale factor.
+
+
+
+ denominator of the scale factor.
+
+
+
+
+
+ Scale @val by the rational number @num / @denom, avoiding overflows and
+underflows and without loss of precision.
+
+This function can potentially be very slow if val and num are both
+greater than G_MAXUINT32.
+
+ @val * @num / @denom. In the case of an overflow, this
+function returns G_MAXUINT64. If the result is not exactly
+representable as an integer, it is rounded to the nearest integer
+(half-way cases are rounded up). See also gst_util_uint64_scale(),
+gst_util_uint64_scale_ceil(), gst_util_uint64_scale_int(),
+gst_util_uint64_scale_int_round(), gst_util_uint64_scale_int_ceil().
+
+
+
+
+ the number to scale
+
+
+
+ the numerator of the scale ratio
+
+
+
+ the denominator of the scale ratio
+
+
+
+
+
+ Determines if @value1 and @value2 can be compared.
+
+ %TRUE if the values can be compared
+
+
+
+
+ a value to compare
+
+
+
+ another value to compare
+
+
+
+
+
+ Determines if intersecting two values will produce a valid result.
+Two values will produce a valid intersection if they have the same
+type.
+
+ %TRUE if the values can intersect
+
+
+
+
+ a value to intersect
+
+
+
+ another value to intersect
+
+
+
+
+
+ Checks if it's possible to subtract @subtrahend from @minuend.
+
+ %TRUE if a subtraction is possible
+
+
+
+
+ the value to subtract from
+
+
+
+ the value to subtract
+
+
+
+
+
+ Determines if @value1 and @value2 can be non-trivially unioned.
+Any two values can be trivially unioned by adding both of them
+to a GstValueList. However, certain types have the possibility
+to be unioned in a simpler way. For example, an integer range
+and an integer can be unioned if the integer is a subset of the
+integer range. If there is the possibility that two values can
+be unioned, this function returns %TRUE.
+
+ %TRUE if there is a function allowing the two values to
+be unioned.
+
+
+
+
+ a value to union
+
+
+
+ another value to union
+
+
+
+
+
+ Compares @value1 and @value2. If @value1 and @value2 cannot be
+compared, the function returns GST_VALUE_UNORDERED. Otherwise,
+if @value1 is greater than @value2, GST_VALUE_GREATER_THAN is returned.
+If @value1 is less than @value2, GST_VALUE_LESS_THAN is returned.
+If the values are equal, GST_VALUE_EQUAL is returned.
+
+ comparison result
+
+
+
+
+ a value to compare
+
+
+
+ another value to compare
+
+
+
+
+
+ Tries to deserialize a string into the type specified by the given GValue.
+If the operation succeeds, %TRUE is returned, %FALSE otherwise.
+
+ %TRUE on success
+
+
+
+
+ #GValue to fill with contents of
+ deserialization
+
+
+
+ string to deserialize
+
+
+
+
+
+ Fixate @src into a new value @dest.
+For ranges, the first element is taken. For lists and arrays, the
+first item is fixated and returned.
+If @src is already fixed, this function returns %FALSE.
+
+ %TRUE if @dest contains a fixated version of @src.
+
+
+
+
+ the #GValue destination
+
+
+
+ the #GValue to fixate
+
+
+
+
+
+ Multiplies the two #GValue items containing a #GST_TYPE_FRACTION and sets
+@product to the product of the two fractions.
+
+ %FALSE in case of an error (like integer overflow), %TRUE otherwise.
+
+
+
+
+ a GValue initialized to #GST_TYPE_FRACTION
+
+
+
+ a GValue initialized to #GST_TYPE_FRACTION
+
+
+
+ a GValue initialized to #GST_TYPE_FRACTION
+
+
+
+
+
+ Subtracts the @subtrahend from the @minuend and sets @dest to the result.
+
+ %FALSE in case of an error (like integer overflow), %TRUE otherwise.
+
+
+
+
+ a GValue initialized to #GST_TYPE_FRACTION
+
+
+
+ a GValue initialized to #GST_TYPE_FRACTION
+
+
+
+ a GValue initialized to #GST_TYPE_FRACTION
+
+
+
+
+
+ Gets the bitmask specified by @value.
+
+ the bitmask.
+
+
+
+
+ a GValue initialized to #GST_TYPE_BITMASK
+
+
+
+
+
+ Gets the contents of @value. The reference count of the returned
+#GstCaps will not be modified, therefore the caller must take one
+before getting rid of the @value.
+
+ the contents of @value
+
+
+
+
+ a GValue initialized to GST_TYPE_CAPS
+
+
+
+
+
+ Gets the contents of @value.
+
+ the contents of @value
+
+
+
+
+ a GValue initialized to GST_TYPE_CAPS_FEATURES
+
+
+
+
+
+ Gets the maximum of the range specified by @value.
+
+ the maximum of the range
+
+
+
+
+ a GValue initialized to GST_TYPE_DOUBLE_RANGE
+
+
+
+
+
+ Gets the minimum of the range specified by @value.
+
+ the minimum of the range
+
+
+
+
+ a GValue initialized to GST_TYPE_DOUBLE_RANGE
+
+
+
+
+
+ Retrieve the flags field of a GstFlagSet @value.
+
+ the flags field of the flagset instance.
+
+
+
+
+ a GValue initialized to #GST_TYPE_FLAG_SET
+
+
+
+
+
+ Retrieve the mask field of a GstFlagSet @value.
+
+ the mask field of the flagset instance.
+
+
+
+
+ a GValue initialized to #GST_TYPE_FLAG_SET
+
+
+
+
+
+ Gets the denominator of the fraction specified by @value.
+
+ the denominator of the fraction.
+
+
+
+
+ a GValue initialized to #GST_TYPE_FRACTION
+
+
+
+
+
+ Gets the numerator of the fraction specified by @value.
+
+ the numerator of the fraction.
+
+
+
+
+ a GValue initialized to #GST_TYPE_FRACTION
+
+
+
+
+
+ Gets the maximum of the range specified by @value.
+
+ the maximum of the range
+
+
+
+
+ a GValue initialized to GST_TYPE_FRACTION_RANGE
+
+
+
+
+
+ Gets the minimum of the range specified by @value.
+
+ the minimum of the range
+
+
+
+
+ a GValue initialized to GST_TYPE_FRACTION_RANGE
+
+
+
+
+
+ Gets the maximum of the range specified by @value.
+
+ the maximum of the range
+
+
+
+
+ a GValue initialized to GST_TYPE_INT64_RANGE
+
+
+
+
+
+ Gets the minimum of the range specified by @value.
+
+ the minimum of the range
+
+
+
+
+ a GValue initialized to GST_TYPE_INT64_RANGE
+
+
+
+
+
+ Gets the step of the range specified by @value.
+
+ the step of the range
+
+
+
+
+ a GValue initialized to GST_TYPE_INT64_RANGE
+
+
+
+
+
+ Gets the maximum of the range specified by @value.
+
+ the maximum of the range
+
+
+
+
+ a GValue initialized to GST_TYPE_INT_RANGE
+
+
+
+
+
+ Gets the minimum of the range specified by @value.
+
+ the minimum of the range
+
+
+
+
+ a GValue initialized to GST_TYPE_INT_RANGE
+
+
+
+
+
+ Gets the step of the range specified by @value.
+
+ the step of the range
+
+
+
+
+ a GValue initialized to GST_TYPE_INT_RANGE
+
+
+
+
+
+ Gets the contents of @value.
+
+ the contents of @value
+
+
+
+
+ a GValue initialized to GST_TYPE_STRUCTURE
+
+
+
+
+
+ Initialises the target value to be of the same type as source and then copies
+the contents from source to target.
+
+
+
+
+
+ the target value
+
+
+
+ the source value
+
+
+
+
+
+ Calculates the intersection of two values. If the values have
+a non-empty intersection, the value representing the intersection
+is placed in @dest, unless %NULL. If the intersection is non-empty,
+@dest is not modified.
+
+ %TRUE if the intersection is non-empty
+
+
+
+
+
+ a uninitialized #GValue that will hold the calculated
+ intersection value. May be %NULL if the resulting set if not
+ needed.
+
+
+
+ a value to intersect
+
+
+
+ another value to intersect
+
+
+
+
+
+ Tests if the given GValue, if available in a GstStructure (or any other
+container) contains a "fixed" (which means: one value) or an "unfixed"
+(which means: multiple possible values, such as data lists or data
+ranges) value.
+
+ true if the value is "fixed".
+
+
+
+
+ the #GValue to check
+
+
+
+
+
+ Check that @value1 is a subset of @value2.
+
+ %TRUE is @value1 is a subset of @value2
+
+
+
+
+ a #GValue
+
+
+
+ a #GValue
+
+
+
+
+
+ Registers functions to perform calculations on #GValue items of a given
+type. Each type can only be added once.
+
+
+
+
+
+ structure containing functions to register
+
+
+
+
+
+ tries to transform the given @value into a string representation that allows
+getting back this string later on using gst_value_deserialize().
+
+Free-function: g_free
+
+ the serialization for @value
+or %NULL if none exists
+
+
+
+
+ a #GValue to serialize
+
+
+
+
+
+ Sets @value to the bitmask specified by @bitmask.
+
+
+
+
+
+ a GValue initialized to #GST_TYPE_BITMASK
+
+
+
+ the bitmask
+
+
+
+
+
+ Sets the contents of @value to @caps. A reference to the
+provided @caps will be taken by the @value.
+
+
+
+
+
+ a GValue initialized to GST_TYPE_CAPS
+
+
+
+ the caps to set the value to
+
+
+
+
+
+ Sets the contents of @value to @features.
+
+
+
+
+
+ a GValue initialized to GST_TYPE_CAPS_FEATURES
+
+
+
+ the features to set the value to
+
+
+
+
+
+ Sets @value to the range specified by @start and @end.
+
+
+
+
+
+ a GValue initialized to GST_TYPE_DOUBLE_RANGE
+
+
+
+ the start of the range
+
+
+
+ the end of the range
+
+
+
+
+
+ Sets @value to the flags and mask values provided in @flags and @mask.
+The @flags value indicates the values of flags, the @mask represents
+which bits in the flag value have been set, and which are "don't care"
+
+
+
+
+
+ a GValue initialized to %GST_TYPE_FLAG_SET
+
+
+
+ The value of the flags set or unset
+
+
+
+ The mask indicate which flags bits must match for comparisons
+
+
+
+
+
+ Sets @value to the fraction specified by @numerator over @denominator.
+The fraction gets reduced to the smallest numerator and denominator,
+and if necessary the sign is moved to the numerator.
+
+
+
+
+
+ a GValue initialized to #GST_TYPE_FRACTION
+
+
+
+ the numerator of the fraction
+
+
+
+ the denominator of the fraction
+
+
+
+
+
+ Sets @value to the range specified by @start and @end.
+
+
+
+
+
+ a GValue initialized to GST_TYPE_FRACTION_RANGE
+
+
+
+ the start of the range (a GST_TYPE_FRACTION GValue)
+
+
+
+ the end of the range (a GST_TYPE_FRACTION GValue)
+
+
+
+
+
+ Sets @value to the range specified by @numerator_start/@denominator_start
+and @numerator_end/@denominator_end.
+
+
+
+
+
+ a GValue initialized to GST_TYPE_FRACTION_RANGE
+
+
+
+ the numerator start of the range
+
+
+
+ the denominator start of the range
+
+
+
+ the numerator end of the range
+
+
+
+ the denominator end of the range
+
+
+
+
+
+ Sets @value to the range specified by @start and @end.
+
+
+
+
+
+ a GValue initialized to GST_TYPE_INT64_RANGE
+
+
+
+ the start of the range
+
+
+
+ the end of the range
+
+
+
+
+
+ Sets @value to the range specified by @start, @end and @step.
+
+
+
+
+
+ a GValue initialized to GST_TYPE_INT64_RANGE
+
+
+
+ the start of the range
+
+
+
+ the end of the range
+
+
+
+ the step of the range
+
+
+
+
+
+ Sets @value to the range specified by @start and @end.
+
+
+
+
+
+ a GValue initialized to GST_TYPE_INT_RANGE
+
+
+
+ the start of the range
+
+
+
+ the end of the range
+
+
+
+
+
+ Sets @value to the range specified by @start, @end and @step.
+
+
+
+
+
+ a GValue initialized to GST_TYPE_INT_RANGE
+
+
+
+ the start of the range
+
+
+
+ the end of the range
+
+
+
+ the step of the range
+
+
+
+
+
+ Sets the contents of @value to @structure.
+
+
+
+
+
+ a GValue initialized to GST_TYPE_STRUCTURE
+
+
+
+ the structure to set the value to
+
+
+
+
+
+ Subtracts @subtrahend from @minuend and stores the result in @dest.
+Note that this means subtraction as in sets, not as in mathematics.
+
+ %TRUE if the subtraction is not empty
+
+
+
+
+ the destination value
+ for the result if the subtraction is not empty. May be %NULL,
+ in which case the resulting set will not be computed, which can
+ give a fair speedup.
+
+
+
+ the value to subtract from
+
+
+
+ the value to subtract
+
+
+
+
+
+ Creates a GValue corresponding to the union of @value1 and @value2.
+
+ %TRUE if the union succeeded.
+
+
+
+
+ the destination value
+
+
+
+ a value to union
+
+
+
+ another value to union
+
+
+
+
+
+ Gets the version number of the GStreamer library.
+
+
+
+
+
+ pointer to a guint to store the major version number
+
+
+
+ pointer to a guint to store the minor version number
+
+
+
+ pointer to a guint to store the micro version number
+
+
+
+ pointer to a guint to store the nano version number
+
+
+
+
+
+ This function returns a string that is useful for describing this version
+of GStreamer to the outside world: user agent strings, logging, ...
+
+ a newly allocated string describing this version
+ of GStreamer.
+
+
+
+
+
diff --git a/girs/GstApp-1.0.gir b/girs/GstApp-1.0.gir
new file mode 100644
index 0000000000..9913b72d62
--- /dev/null
+++ b/girs/GstApp-1.0.gir
@@ -0,0 +1,1935 @@
+
+
+
+
+
+
+
+
+
+ Appsink is a sink plugin that supports many different methods for making
+the application get a handle on the GStreamer data in a pipeline. Unlike
+most GStreamer elements, Appsink provides external API functions.
+
+appsink can be used by linking to the gstappsink.h header file to access the
+methods or by using the appsink action signals and properties.
+
+The normal way of retrieving samples from appsink is by using the
+gst_app_sink_pull_sample() and gst_app_sink_pull_preroll() methods.
+These methods block until a sample becomes available in the sink or when the
+sink is shut down or reaches EOS. There are also timed variants of these
+methods, gst_app_sink_try_pull_sample() and gst_app_sink_try_pull_preroll(),
+which accept a timeout parameter to limit the amount of time to wait.
+
+Appsink will internally use a queue to collect buffers from the streaming
+thread. If the application is not pulling samples fast enough, this queue
+will consume a lot of memory over time. The "max-buffers" property can be
+used to limit the queue size. The "drop" property controls whether the
+streaming thread blocks or if older buffers are dropped when the maximum
+queue size is reached. Note that blocking the streaming thread can negatively
+affect real-time performance and should be avoided.
+
+If a blocking behaviour is not desirable, setting the "emit-signals" property
+to %TRUE will make appsink emit the "new-sample" and "new-preroll" signals
+when a sample can be pulled without blocking.
+
+The "caps" property on appsink can be used to control the formats that
+appsink can receive. This property can contain non-fixed caps, the format of
+the pulled samples can be obtained by getting the sample caps.
+
+If one of the pull-preroll or pull-sample methods return %NULL, the appsink
+is stopped or in the EOS state. You can check for the EOS state with the
+"eos" property or with the gst_app_sink_is_eos() method.
+
+The eos signal can also be used to be informed when the EOS state is reached
+to avoid polling.
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Get the last preroll sample in @appsink. This was the sample that caused the
+appsink to preroll in the PAUSED state.
+
+This function is typically used when dealing with a pipeline in the PAUSED
+state. Calling this function after doing a seek will give the sample right
+after the seek position.
+
+Calling this function will clear the internal reference to the preroll
+buffer.
+
+Note that the preroll sample will also be returned as the first sample
+when calling gst_app_sink_pull_sample().
+
+If an EOS event was received before any buffers, this function returns
+%NULL. Use gst_app_sink_is_eos () to check for the EOS condition.
+
+This function blocks until a preroll sample or EOS is received or the appsink
+element is set to the READY/NULL state.
+
+ a #GstSample or NULL when the appsink is stopped or EOS.
+ Call gst_sample_unref() after usage.
+
+
+
+
+ a #GstAppSink
+
+
+
+
+
+ This function blocks until a sample or EOS becomes available or the appsink
+element is set to the READY/NULL state.
+
+This function will only return samples when the appsink is in the PLAYING
+state. All rendered buffers will be put in a queue so that the application
+can pull samples at its own rate. Note that when the application does not
+pull samples fast enough, the queued buffers could consume a lot of memory,
+especially when dealing with raw video frames.
+
+If an EOS event was received before any buffers, this function returns
+%NULL. Use gst_app_sink_is_eos () to check for the EOS condition.
+
+ a #GstSample or NULL when the appsink is stopped or EOS.
+ Call gst_sample_unref() after usage.
+
+
+
+
+ a #GstAppSink
+
+
+
+
+
+ Get the last preroll sample in @appsink. This was the sample that caused the
+appsink to preroll in the PAUSED state.
+
+This function is typically used when dealing with a pipeline in the PAUSED
+state. Calling this function after doing a seek will give the sample right
+after the seek position.
+
+Calling this function will clear the internal reference to the preroll
+buffer.
+
+Note that the preroll sample will also be returned as the first sample
+when calling gst_app_sink_pull_sample().
+
+If an EOS event was received before any buffers or the timeout expires,
+this function returns %NULL. Use gst_app_sink_is_eos () to check for the EOS
+condition.
+
+This function blocks until a preroll sample or EOS is received, the appsink
+element is set to the READY/NULL state, or the timeout expires.
+
+ a #GstSample or NULL when the appsink is stopped or EOS or the timeout expires.
+ Call gst_sample_unref() after usage.
+
+
+
+
+ a #GstAppSink
+
+
+
+ the maximum amount of time to wait for the preroll sample
+
+
+
+
+
+ This function blocks until a sample or EOS becomes available or the appsink
+element is set to the READY/NULL state or the timeout expires.
+
+This function will only return samples when the appsink is in the PLAYING
+state. All rendered buffers will be put in a queue so that the application
+can pull samples at its own rate. Note that when the application does not
+pull samples fast enough, the queued buffers could consume a lot of memory,
+especially when dealing with raw video frames.
+
+If an EOS event was received before any buffers or the timeout expires,
+this function returns %NULL. Use gst_app_sink_is_eos () to check for the EOS
+condition.
+
+ a #GstSample or NULL when the appsink is stopped or EOS or the timeout expires.
+Call gst_sample_unref() after usage.
+
+
+
+
+ a #GstAppSink
+
+
+
+ the maximum amount of time to wait for a sample
+
+
+
+
+
+ Check if @appsink supports buffer lists.
+
+ %TRUE if @appsink supports buffer lists.
+
+
+
+
+ a #GstAppSink
+
+
+
+
+
+ Get the configured caps on @appsink.
+
+ the #GstCaps accepted by the sink. gst_caps_unref() after usage.
+
+
+
+
+ a #GstAppSink
+
+
+
+
+
+ Check if @appsink will drop old buffers when the maximum amount of queued
+buffers is reached.
+
+ %TRUE if @appsink is dropping old buffers when the queue is
+filled.
+
+
+
+
+ a #GstAppSink
+
+
+
+
+
+ Check if appsink will emit the "new-preroll" and "new-sample" signals.
+
+ %TRUE if @appsink is emiting the "new-preroll" and "new-sample"
+signals.
+
+
+
+
+ a #GstAppSink
+
+
+
+
+
+ Get the maximum amount of buffers that can be queued in @appsink.
+
+ The maximum amount of buffers that can be queued.
+
+
+
+
+ a #GstAppSink
+
+
+
+
+
+ Check if @appsink will wait for all buffers to be consumed when an EOS is
+received.
+
+ %TRUE if @appsink will wait for all buffers to be consumed when an
+EOS is received.
+
+
+
+
+ a #GstAppSink
+
+
+
+
+
+ Check if @appsink is EOS, which is when no more samples can be pulled because
+an EOS event was received.
+
+This function also returns %TRUE when the appsink is not in the PAUSED or
+PLAYING state.
+
+ %TRUE if no more samples can be pulled and the appsink is EOS.
+
+
+
+
+ a #GstAppSink
+
+
+
+
+
+ Get the last preroll sample in @appsink. This was the sample that caused the
+appsink to preroll in the PAUSED state.
+
+This function is typically used when dealing with a pipeline in the PAUSED
+state. Calling this function after doing a seek will give the sample right
+after the seek position.
+
+Calling this function will clear the internal reference to the preroll
+buffer.
+
+Note that the preroll sample will also be returned as the first sample
+when calling gst_app_sink_pull_sample().
+
+If an EOS event was received before any buffers, this function returns
+%NULL. Use gst_app_sink_is_eos () to check for the EOS condition.
+
+This function blocks until a preroll sample or EOS is received or the appsink
+element is set to the READY/NULL state.
+
+ a #GstSample or NULL when the appsink is stopped or EOS.
+ Call gst_sample_unref() after usage.
+
+
+
+
+ a #GstAppSink
+
+
+
+
+
+ This function blocks until a sample or EOS becomes available or the appsink
+element is set to the READY/NULL state.
+
+This function will only return samples when the appsink is in the PLAYING
+state. All rendered buffers will be put in a queue so that the application
+can pull samples at its own rate. Note that when the application does not
+pull samples fast enough, the queued buffers could consume a lot of memory,
+especially when dealing with raw video frames.
+
+If an EOS event was received before any buffers, this function returns
+%NULL. Use gst_app_sink_is_eos () to check for the EOS condition.
+
+ a #GstSample or NULL when the appsink is stopped or EOS.
+ Call gst_sample_unref() after usage.
+
+
+
+
+ a #GstAppSink
+
+
+
+
+
+ Instruct @appsink to enable or disable buffer list support.
+
+For backwards-compatibility reasons applications need to opt in
+to indicate that they will be able to handle buffer lists.
+
+
+
+
+
+ a #GstAppSink
+
+
+
+ enable or disable buffer list support
+
+
+
+
+
+ Set callbacks which will be executed for each new preroll, new sample and eos.
+This is an alternative to using the signals, it has lower overhead and is thus
+less expensive, but also less flexible.
+
+If callbacks are installed, no signals will be emitted for performance
+reasons.
+
+
+
+
+
+ a #GstAppSink
+
+
+
+ the callbacks
+
+
+
+ a user_data argument for the callbacks
+
+
+
+ a destroy notify function
+
+
+
+
+
+ Set the capabilities on the appsink element. This function takes
+a copy of the caps structure. After calling this method, the sink will only
+accept caps that match @caps. If @caps is non-fixed, or incomplete,
+you must check the caps on the samples to get the actual used caps.
+
+
+
+
+
+ a #GstAppSink
+
+
+
+ caps to set
+
+
+
+
+
+ Instruct @appsink to drop old buffers when the maximum amount of queued
+buffers is reached.
+
+
+
+
+
+ a #GstAppSink
+
+
+
+ the new state
+
+
+
+
+
+ Make appsink emit the "new-preroll" and "new-sample" signals. This option is
+by default disabled because signal emission is expensive and unneeded when
+the application prefers to operate in pull mode.
+
+
+
+
+
+ a #GstAppSink
+
+
+
+ the new state
+
+
+
+
+
+ Set the maximum amount of buffers that can be queued in @appsink. After this
+amount of buffers are queued in appsink, any more buffers will block upstream
+elements until a sample is pulled from @appsink.
+
+
+
+
+
+ a #GstAppSink
+
+
+
+ the maximum number of buffers to queue
+
+
+
+
+
+ Instruct @appsink to wait for all buffers to be consumed when an EOS is received.
+
+
+
+
+
+ a #GstAppSink
+
+
+
+ the new state
+
+
+
+
+
+ Get the last preroll sample in @appsink. This was the sample that caused the
+appsink to preroll in the PAUSED state.
+
+This function is typically used when dealing with a pipeline in the PAUSED
+state. Calling this function after doing a seek will give the sample right
+after the seek position.
+
+Calling this function will clear the internal reference to the preroll
+buffer.
+
+Note that the preroll sample will also be returned as the first sample
+when calling gst_app_sink_pull_sample().
+
+If an EOS event was received before any buffers or the timeout expires,
+this function returns %NULL. Use gst_app_sink_is_eos () to check for the EOS
+condition.
+
+This function blocks until a preroll sample or EOS is received, the appsink
+element is set to the READY/NULL state, or the timeout expires.
+
+ a #GstSample or NULL when the appsink is stopped or EOS or the timeout expires.
+ Call gst_sample_unref() after usage.
+
+
+
+
+ a #GstAppSink
+
+
+
+ the maximum amount of time to wait for the preroll sample
+
+
+
+
+
+ This function blocks until a sample or EOS becomes available or the appsink
+element is set to the READY/NULL state or the timeout expires.
+
+This function will only return samples when the appsink is in the PLAYING
+state. All rendered buffers will be put in a queue so that the application
+can pull samples at its own rate. Note that when the application does not
+pull samples fast enough, the queued buffers could consume a lot of memory,
+especially when dealing with raw video frames.
+
+If an EOS event was received before any buffers or the timeout expires,
+this function returns %NULL. Use gst_app_sink_is_eos () to check for the EOS
+condition.
+
+ a #GstSample or NULL when the appsink is stopped or EOS or the timeout expires.
+Call gst_sample_unref() after usage.
+
+
+
+
+ a #GstAppSink
+
+
+
+ the maximum amount of time to wait for a sample
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Signal that the end-of-stream has been reached. This signal is emitted from
+the streaming thread.
+
+
+
+
+
+ Signal that a new preroll sample is available.
+
+This signal is emitted from the streaming thread and only when the
+"emit-signals" property is %TRUE.
+
+The new preroll sample can be retrieved with the "pull-preroll" action
+signal or gst_app_sink_pull_preroll() either from this signal callback
+or from any other thread.
+
+Note that this signal is only emitted when the "emit-signals" property is
+set to %TRUE, which it is not by default for performance reasons.
+
+
+
+
+
+ Signal that a new sample is available.
+
+This signal is emitted from the streaming thread and only when the
+"emit-signals" property is %TRUE.
+
+The new sample can be retrieved with the "pull-sample" action
+signal or gst_app_sink_pull_sample() either from this signal callback
+or from any other thread.
+
+Note that this signal is only emitted when the "emit-signals" property is
+set to %TRUE, which it is not by default for performance reasons.
+
+
+
+
+
+ Get the last preroll sample in @appsink. This was the sample that caused the
+appsink to preroll in the PAUSED state.
+
+This function is typically used when dealing with a pipeline in the PAUSED
+state. Calling this function after doing a seek will give the sample right
+after the seek position.
+
+Calling this function will clear the internal reference to the preroll
+buffer.
+
+Note that the preroll sample will also be returned as the first sample
+when calling gst_app_sink_pull_sample() or the "pull-sample" action signal.
+
+If an EOS event was received before any buffers, this function returns
+%NULL. Use gst_app_sink_is_eos () to check for the EOS condition.
+
+This function blocks until a preroll sample or EOS is received or the appsink
+element is set to the READY/NULL state.
+
+ a #GstSample or NULL when the appsink is stopped or EOS.
+
+
+
+
+ This function blocks until a sample or EOS becomes available or the appsink
+element is set to the READY/NULL state.
+
+This function will only return samples when the appsink is in the PLAYING
+state. All rendered samples will be put in a queue so that the application
+can pull samples at its own rate.
+
+Note that when the application does not pull samples fast enough, the
+queued samples could consume a lot of memory, especially when dealing with
+raw video frames. It's possible to control the behaviour of the queue with
+the "drop" and "max-buffers" properties.
+
+If an EOS event was received before any buffers, this function returns
+%NULL. Use gst_app_sink_is_eos () to check for the EOS condition.
+
+ a #GstSample or NULL when the appsink is stopped or EOS.
+
+
+
+
+ Get the last preroll sample in @appsink. This was the sample that caused the
+appsink to preroll in the PAUSED state.
+
+This function is typically used when dealing with a pipeline in the PAUSED
+state. Calling this function after doing a seek will give the sample right
+after the seek position.
+
+Calling this function will clear the internal reference to the preroll
+buffer.
+
+Note that the preroll sample will also be returned as the first sample
+when calling gst_app_sink_pull_sample() or the "pull-sample" action signal.
+
+If an EOS event was received before any buffers or the timeout expires,
+this function returns %NULL. Use gst_app_sink_is_eos () to check for the EOS
+condition.
+
+This function blocks until a preroll sample or EOS is received, the appsink
+element is set to the READY/NULL state, or the timeout expires.
+
+ a #GstSample or NULL when the appsink is stopped or EOS or the timeout expires.
+
+
+
+
+ the maximum amount of time to wait for the preroll sample
+
+
+
+
+
+ This function blocks until a sample or EOS becomes available or the appsink
+element is set to the READY/NULL state or the timeout expires.
+
+This function will only return samples when the appsink is in the PLAYING
+state. All rendered samples will be put in a queue so that the application
+can pull samples at its own rate.
+
+Note that when the application does not pull samples fast enough, the
+queued samples could consume a lot of memory, especially when dealing with
+raw video frames. It's possible to control the behaviour of the queue with
+the "drop" and "max-buffers" properties.
+
+If an EOS event was received before any buffers or the timeout expires,
+this function returns %NULL. Use gst_app_sink_is_eos () to check
+for the EOS condition.
+
+ a #GstSample or NULL when the appsink is stopped or EOS or the timeout expires.
+
+
+
+
+ the maximum amount of time to wait for a sample
+
+
+
+
+
+
+ A set of callbacks that can be installed on the appsink with
+gst_app_sink_set_callbacks().
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ a #GstSample or NULL when the appsink is stopped or EOS.
+ Call gst_sample_unref() after usage.
+
+
+
+
+ a #GstAppSink
+
+
+
+
+
+
+
+
+ a #GstSample or NULL when the appsink is stopped or EOS.
+ Call gst_sample_unref() after usage.
+
+
+
+
+ a #GstAppSink
+
+
+
+
+
+
+
+
+ a #GstSample or NULL when the appsink is stopped or EOS or the timeout expires.
+ Call gst_sample_unref() after usage.
+
+
+
+
+ a #GstAppSink
+
+
+
+ the maximum amount of time to wait for the preroll sample
+
+
+
+
+
+
+
+
+ a #GstSample or NULL when the appsink is stopped or EOS or the timeout expires.
+Call gst_sample_unref() after usage.
+
+
+
+
+ a #GstAppSink
+
+
+
+ the maximum amount of time to wait for a sample
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ The appsrc element can be used by applications to insert data into a
+GStreamer pipeline. Unlike most GStreamer elements, appsrc provides
+external API functions.
+
+appsrc can be used by linking with the libgstapp library to access the
+methods directly or by using the appsrc action signals.
+
+Before operating appsrc, the caps property must be set to fixed caps
+describing the format of the data that will be pushed with appsrc. An
+exception to this is when pushing buffers with unknown caps, in which case no
+caps should be set. This is typically true of file-like sources that push raw
+byte buffers. If you don't want to explicitly set the caps, you can use
+gst_app_src_push_sample. This method gets the caps associated with the
+sample and sets them on the appsrc replacing any previously set caps (if
+different from sample's caps).
+
+The main way of handing data to the appsrc element is by calling the
+gst_app_src_push_buffer() method or by emitting the push-buffer action signal.
+This will put the buffer onto a queue from which appsrc will read from in its
+streaming thread. It is important to note that data transport will not happen
+from the thread that performed the push-buffer call.
+
+The "max-bytes" property controls how much data can be queued in appsrc
+before appsrc considers the queue full. A filled internal queue will always
+signal the "enough-data" signal, which signals the application that it should
+stop pushing data into appsrc. The "block" property will cause appsrc to
+block the push-buffer method until free data becomes available again.
+
+When the internal queue is running out of data, the "need-data" signal is
+emitted, which signals the application that it should start pushing more data
+into appsrc.
+
+In addition to the "need-data" and "enough-data" signals, appsrc can emit the
+"seek-data" signal when the "stream-mode" property is set to "seekable" or
+"random-access". The signal argument will contain the new desired position in
+the stream expressed in the unit set with the "format" property. After
+receiving the seek-data signal, the application should push-buffers from the
+new position.
+
+These signals allow the application to operate the appsrc in two different
+ways:
+
+The push mode, in which the application repeatedly calls the push-buffer/push-sample
+method with a new buffer/sample. Optionally, the queue size in the appsrc
+can be controlled with the enough-data and need-data signals by respectively
+stopping/starting the push-buffer/push-sample calls. This is a typical
+mode of operation for the stream-type "stream" and "seekable". Use this
+mode when implementing various network protocols or hardware devices.
+
+The pull mode, in which the need-data signal triggers the next push-buffer call.
+This mode is typically used in the "random-access" stream-type. Use this
+mode for file access or other randomly accessable sources. In this mode, a
+buffer of exactly the amount of bytes given by the need-data signal should be
+pushed into appsrc.
+
+In all modes, the size property on appsrc should contain the total stream
+size in bytes. Setting this property is mandatory in the random-access mode.
+For the stream and seekable modes, setting this property is optional but
+recommended.
+
+When the application has finished pushing data into appsrc, it should call
+gst_app_src_end_of_stream() or emit the end-of-stream action signal. After
+this call, no more buffers can be pushed into appsrc until a flushing seek
+occurs or the state of the appsrc has gone through READY.
+
+
+ Indicates to the appsrc element that the last buffer queued in the
+element is the last buffer of the stream.
+
+ #GST_FLOW_OK when the EOS was successfuly queued.
+#GST_FLOW_FLUSHING when @appsrc is not PAUSED or PLAYING.
+
+
+
+
+ a #GstAppSrc
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Adds a buffer to the queue of buffers that the appsrc element will
+push to its source pad. This function takes ownership of the buffer.
+
+When the block property is TRUE, this function can block until free
+space becomes available in the queue.
+
+ #GST_FLOW_OK when the buffer was successfuly queued.
+#GST_FLOW_FLUSHING when @appsrc is not PAUSED or PLAYING.
+#GST_FLOW_EOS when EOS occured.
+
+
+
+
+ a #GstAppSrc
+
+
+
+ a #GstBuffer to push
+
+
+
+
+
+ Adds a buffer list to the queue of buffers and buffer lists that the
+appsrc element will push to its source pad. This function takes ownership
+of @buffer_list.
+
+When the block property is TRUE, this function can block until free
+space becomes available in the queue.
+
+ #GST_FLOW_OK when the buffer list was successfuly queued.
+#GST_FLOW_FLUSHING when @appsrc is not PAUSED or PLAYING.
+#GST_FLOW_EOS when EOS occured.
+
+
+
+
+ a #GstAppSrc
+
+
+
+ a #GstBufferList to push
+
+
+
+
+
+ Extract a buffer from the provided sample and adds it to the queue of
+buffers that the appsrc element will push to its source pad. Any
+previous caps that were set on appsrc will be replaced by the caps
+associated with the sample if not equal.
+
+When the block property is TRUE, this function can block until free
+space becomes available in the queue.
+
+ #GST_FLOW_OK when the buffer was successfuly queued.
+#GST_FLOW_FLUSHING when @appsrc is not PAUSED or PLAYING.
+#GST_FLOW_EOS when EOS occured.
+
+
+
+
+ a #GstAppSrc
+
+
+
+ a #GstSample from which buffer and caps may be
+extracted
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Indicates to the appsrc element that the last buffer queued in the
+element is the last buffer of the stream.
+
+ #GST_FLOW_OK when the EOS was successfuly queued.
+#GST_FLOW_FLUSHING when @appsrc is not PAUSED or PLAYING.
+
+
+
+
+ a #GstAppSrc
+
+
+
+
+
+ Get the configured caps on @appsrc.
+
+ the #GstCaps produced by the source. gst_caps_unref() after usage.
+
+
+
+
+ a #GstAppSrc
+
+
+
+
+
+ Get the number of currently queued bytes inside @appsrc.
+
+ The number of currently queued bytes.
+
+
+
+
+ a #GstAppSrc
+
+
+
+
+
+ Get the duration of the stream in nanoseconds. A value of GST_CLOCK_TIME_NONE means that the duration is
+not known.
+
+ the duration of the stream previously set with gst_app_src_set_duration();
+
+
+
+
+ a #GstAppSrc
+
+
+
+
+
+ Check if appsrc will emit the "new-preroll" and "new-buffer" signals.
+
+ %TRUE if @appsrc is emitting the "new-preroll" and "new-buffer"
+signals.
+
+
+
+
+ a #GstAppSrc
+
+
+
+
+
+ Retrieve the min and max latencies in @min and @max respectively.
+
+
+
+
+
+ a #GstAppSrc
+
+
+
+ the min latency
+
+
+
+ the max latency
+
+
+
+
+
+ Get the maximum amount of bytes that can be queued in @appsrc.
+
+ The maximum amount of bytes that can be queued.
+
+
+
+
+ a #GstAppSrc
+
+
+
+
+
+ Get the size of the stream in bytes. A value of -1 means that the size is
+not known.
+
+ the size of the stream previously set with gst_app_src_set_size();
+
+
+
+
+ a #GstAppSrc
+
+
+
+
+
+ Get the stream type. Control the stream type of @appsrc
+with gst_app_src_set_stream_type().
+
+ the stream type.
+
+
+
+
+ a #GstAppSrc
+
+
+
+
+
+ Adds a buffer to the queue of buffers that the appsrc element will
+push to its source pad. This function takes ownership of the buffer.
+
+When the block property is TRUE, this function can block until free
+space becomes available in the queue.
+
+ #GST_FLOW_OK when the buffer was successfuly queued.
+#GST_FLOW_FLUSHING when @appsrc is not PAUSED or PLAYING.
+#GST_FLOW_EOS when EOS occured.
+
+
+
+
+ a #GstAppSrc
+
+
+
+ a #GstBuffer to push
+
+
+
+
+
+ Adds a buffer list to the queue of buffers and buffer lists that the
+appsrc element will push to its source pad. This function takes ownership
+of @buffer_list.
+
+When the block property is TRUE, this function can block until free
+space becomes available in the queue.
+
+ #GST_FLOW_OK when the buffer list was successfuly queued.
+#GST_FLOW_FLUSHING when @appsrc is not PAUSED or PLAYING.
+#GST_FLOW_EOS when EOS occured.
+
+
+
+
+ a #GstAppSrc
+
+
+
+ a #GstBufferList to push
+
+
+
+
+
+ Extract a buffer from the provided sample and adds it to the queue of
+buffers that the appsrc element will push to its source pad. Any
+previous caps that were set on appsrc will be replaced by the caps
+associated with the sample if not equal.
+
+When the block property is TRUE, this function can block until free
+space becomes available in the queue.
+
+ #GST_FLOW_OK when the buffer was successfuly queued.
+#GST_FLOW_FLUSHING when @appsrc is not PAUSED or PLAYING.
+#GST_FLOW_EOS when EOS occured.
+
+
+
+
+ a #GstAppSrc
+
+
+
+ a #GstSample from which buffer and caps may be
+extracted
+
+
+
+
+
+ Set callbacks which will be executed when data is needed, enough data has
+been collected or when a seek should be performed.
+This is an alternative to using the signals, it has lower overhead and is thus
+less expensive, but also less flexible.
+
+If callbacks are installed, no signals will be emitted for performance
+reasons.
+
+
+
+
+
+ a #GstAppSrc
+
+
+
+ the callbacks
+
+
+
+ a user_data argument for the callbacks
+
+
+
+ a destroy notify function
+
+
+
+
+
+ Set the capabilities on the appsrc element. This function takes
+a copy of the caps structure. After calling this method, the source will
+only produce caps that match @caps. @caps must be fixed and the caps on the
+buffers must match the caps or left NULL.
+
+
+
+
+
+ a #GstAppSrc
+
+
+
+ caps to set
+
+
+
+
+
+ Set the duration of the stream in nanoseconds. A value of GST_CLOCK_TIME_NONE means that the duration is
+not known.
+
+
+
+
+
+ a #GstAppSrc
+
+
+
+ the duration to set
+
+
+
+
+
+ Make appsrc emit the "new-preroll" and "new-buffer" signals. This option is
+by default disabled because signal emission is expensive and unneeded when
+the application prefers to operate in pull mode.
+
+
+
+
+
+ a #GstAppSrc
+
+
+
+ the new state
+
+
+
+
+
+ Configure the @min and @max latency in @src. If @min is set to -1, the
+default latency calculations for pseudo-live sources will be used.
+
+
+
+
+
+ a #GstAppSrc
+
+
+
+ the min latency
+
+
+
+ the max latency
+
+
+
+
+
+ Set the maximum amount of bytes that can be queued in @appsrc.
+After the maximum amount of bytes are queued, @appsrc will emit the
+"enough-data" signal.
+
+
+
+
+
+ a #GstAppSrc
+
+
+
+ the maximum number of bytes to queue
+
+
+
+
+
+ Set the size of the stream in bytes. A value of -1 means that the size is
+not known.
+
+
+
+
+
+ a #GstAppSrc
+
+
+
+ the size to set
+
+
+
+
+
+ Set the stream type on @appsrc. For seekable streams, the "seek" signal must
+be connected to.
+
+A stream_type stream
+
+
+
+
+
+ a #GstAppSrc
+
+
+
+ the new state
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Notify @appsrc that no more buffer are available.
+
+
+
+
+
+ Signal that the source has enough data. It is recommended that the
+application stops calling push-buffer until the need-data signal is
+emitted again to avoid excessive buffer queueing.
+
+
+
+
+
+ Signal that the source needs more data. In the callback or from another
+thread you should call push-buffer or end-of-stream.
+
+@length is just a hint and when it is set to -1, any number of bytes can be
+pushed into @appsrc.
+
+You can call push-buffer multiple times until the enough-data signal is
+fired.
+
+
+
+
+
+ the amount of bytes needed.
+
+
+
+
+
+ Adds a buffer to the queue of buffers that the appsrc element will
+push to its source pad. This function does not take ownership of the
+buffer so the buffer needs to be unreffed after calling this function.
+
+When the block property is TRUE, this function can block until free space
+becomes available in the queue.
+
+
+
+
+
+ a buffer to push
+
+
+
+
+
+ Adds a buffer list to the queue of buffers and buffer lists that the
+appsrc element will push to its source pad. This function does not take
+ownership of the buffer list so the buffer list needs to be unreffed
+after calling this function.
+
+When the block property is TRUE, this function can block until free space
+becomes available in the queue.
+
+
+
+
+
+ a buffer list to push
+
+
+
+
+
+ Extract a buffer from the provided sample and adds the extracted buffer
+to the queue of buffers that the appsrc element will
+push to its source pad. This function set the appsrc caps based on the caps
+in the sample and reset the caps if they change.
+Only the caps and the buffer of the provided sample are used and not
+for example the segment in the sample.
+This function does not take ownership of the
+sample so the sample needs to be unreffed after calling this function.
+
+When the block property is TRUE, this function can block until free space
+becomes available in the queue.
+
+
+
+
+
+ a sample from which extract buffer to push
+
+
+
+
+
+ Seek to the given offset. The next push-buffer should produce buffers from
+the new @offset.
+This callback is only called for seekable stream types.
+
+ %TRUE if the seek succeeded.
+
+
+
+
+ the offset to seek to
+
+
+
+
+
+
+ A set of callbacks that can be installed on the appsrc with
+gst_app_src_set_callbacks().
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ #GST_FLOW_OK when the buffer was successfuly queued.
+#GST_FLOW_FLUSHING when @appsrc is not PAUSED or PLAYING.
+#GST_FLOW_EOS when EOS occured.
+
+
+
+
+ a #GstAppSrc
+
+
+
+ a #GstBuffer to push
+
+
+
+
+
+
+
+
+ #GST_FLOW_OK when the EOS was successfuly queued.
+#GST_FLOW_FLUSHING when @appsrc is not PAUSED or PLAYING.
+
+
+
+
+ a #GstAppSrc
+
+
+
+
+
+
+
+
+ #GST_FLOW_OK when the buffer was successfuly queued.
+#GST_FLOW_FLUSHING when @appsrc is not PAUSED or PLAYING.
+#GST_FLOW_EOS when EOS occured.
+
+
+
+
+ a #GstAppSrc
+
+
+
+ a #GstSample from which buffer and caps may be
+extracted
+
+
+
+
+
+
+
+
+ #GST_FLOW_OK when the buffer list was successfuly queued.
+#GST_FLOW_FLUSHING when @appsrc is not PAUSED or PLAYING.
+#GST_FLOW_EOS when EOS occured.
+
+
+
+
+ a #GstAppSrc
+
+
+
+ a #GstBufferList to push
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ The stream type.
+
+ No seeking is supported in the stream, such as a
+live stream.
+
+
+ The stream is seekable but seeking might not
+be very fast, such as data from a webserver.
+
+
+ The stream is seekable and seeking is fast,
+such as in a local file.
+
+
+
+
diff --git a/girs/GstAudio-1.0.gir b/girs/GstAudio-1.0.gir
new file mode 100644
index 0000000000..8c8bcd08b8
--- /dev/null
+++ b/girs/GstAudio-1.0.gir
@@ -0,0 +1,9738 @@
+
+
+
+
+
+
+
+
+
+
+
+
+ Maximum range of allowed channels, for use in template caps strings.
+
+
+
+ #GST_TYPE_AUDIO_DITHER_METHOD, The dither method to use when
+changing bit depth.
+Default is #GST_AUDIO_DITHER_NONE.
+
+
+
+ #GST_TYPE_VALUE_LIST, The channel mapping matrix.
+
+The matrix coefficients must be between -1 and 1: the number of rows is equal
+to the number of output channels and the number of columns is equal to the
+number of input channels.
+
+## Example matrix generation code
+To generate the matrix using code:
+
+|[
+GValue v = G_VALUE_INIT;
+GValue v2 = G_VALUE_INIT;
+GValue v3 = G_VALUE_INIT;
+
+g_value_init (&v2, GST_TYPE_ARRAY);
+g_value_init (&v3, G_TYPE_DOUBLE);
+g_value_set_double (&v3, 1);
+gst_value_array_append_value (&v2, &v3);
+g_value_unset (&v3);
+[ Repeat for as many double as your input channels - unset and reinit v3 ]
+g_value_init (&v, GST_TYPE_ARRAY);
+gst_value_array_append_value (&v, &v2);
+g_value_unset (&v2);
+[ Repeat for as many v2's as your output channels - unset and reinit v2]
+g_object_set_property (G_OBJECT (audiomixmatrix), "matrix", &v);
+g_value_unset (&v);
+]|
+
+
+
+ #GST_TYPE_AUDIO_NOISE_SHAPING_METHOD, The noise shaping method to use
+to mask noise from quantization errors.
+Default is #GST_AUDIO_NOISE_SHAPING_NONE.
+
+
+
+ #G_TYPE_UINT, The quantization amount. Components will be
+quantized to multiples of this value.
+Default is 1
+
+
+
+ #GST_TYPE_AUDIO_RESAMPLER_METHOD, The resampler method to use when
+changing sample rates.
+Default is #GST_AUDIO_RESAMPLER_METHOD_BLACKMAN_NUTTALL.
+
+
+
+ Default maximum number of errors tolerated before signaling error.
+
+
+
+ The name of the templates for the sink pad.
+
+
+
+ The name of the templates for the source pad.
+
+
+
+ Standard number of channels used in consumer audio.
+
+
+
+ Standard format used in consumer audio.
+
+
+
+ Standard sampling rate used in consumer audio.
+
+
+
+ the name of the templates for the sink pad
+
+
+
+ the name of the templates for the source pad
+
+
+
+ List of all audio formats, for use in template caps strings.
+
+
+
+ Maximum range of allowed sample rates, for use in template caps strings.
+
+
+
+ G_TYPE_DOUBLE, B parameter of the cubic filter.
+Values between 0.0 and 2.0 are accepted. 1.0 is the default.
+
+Below are some values of popular filters:
+ B C
+Hermite 0.0 0.0
+Spline 1.0 0.0
+Catmull-Rom 0.0 1/2
+
+
+
+ G_TYPE_DOUBLE, C parameter of the cubic filter.
+Values between 0.0 and 2.0 are accepted. 0.0 is the default.
+
+See #GST_AUDIO_RESAMPLER_OPT_CUBIC_B for some more common values
+
+
+
+ G_TYPE_DOUBLE, Cutoff parameter for the filter. 0.940 is the default.
+
+
+
+ GST_TYPE_AUDIO_RESAMPLER_INTERPOLATION: how the filter coeficients should be
+ interpolated.
+GST_AUDIO_RESAMPLER_FILTER_INTERPOLATION_CUBIC is default.
+
+
+
+ GST_TYPE_AUDIO_RESAMPLER_FILTER_MODE: how the filter tables should be
+constructed.
+GST_AUDIO_RESAMPLER_FILTER_MODE_AUTO is the default.
+
+
+
+ G_TYPE_UINT: the amount of memory to use for full filter tables before
+switching to interpolated filter tables.
+1048576 is the default.
+
+
+
+ G_TYPE_UINT, oversampling to use when interpolating filters
+8 is the default.
+
+
+
+ G_TYPE_DOUBLE: The maximum allowed phase error when switching sample
+rates.
+0.1 is the default.
+
+
+
+ G_TYPE_INT: the number of taps to use for the filter.
+0 is the default and selects the taps automatically.
+
+
+
+ G_TYPE_DOUBLE, stopband attenuation in decibels. The attenuation
+after the stopband for the kaiser window. 85 dB is the default.
+
+
+
+ G_TYPE_DOUBLE, transition bandwidth. The width of the
+transition band for the kaiser window. 0.087 is the default.
+
+
+
+
+
+
+
+
+
+
+
+
+ Subclasses must use (a subclass of) #GstAudioAggregatorPad for both
+their source and sink pads,
+gst_element_class_add_static_pad_template_with_gtype() is a convenient
+helper.
+
+#GstAudioAggregator can perform conversion on the data arriving
+on its sink pads, based on the format expected downstream: in order
+to enable that behaviour, the GType of the sink pads must either be
+a (subclass of) #GstAudioAggregatorConvertPad to use the default
+#GstAudioConverter implementation, or a subclass of #GstAudioAggregatorPad
+implementing #GstAudioAggregatorPad.convert_buffer.
+
+To allow for the output caps to change, the mechanism is the same as
+above, with the GType of the source pad.
+
+See #GstAudioMixer for an example.
+
+When conversion is enabled, #GstAudioAggregator will accept
+any type of raw audio caps and perform conversion
+on the data arriving on its sink pads, with whatever downstream
+expects as the target format.
+
+In case downstream caps are not fully fixated, it will use
+the first configured sink pad to finish fixating its source pad
+caps.
+
+A notable exception for now is the sample rate, sink pads must
+have the same sample rate as either the downstream requirement,
+or the first configured pad, or a combination of both (when
+downstream specifies a range or a set of acceptable rates).
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ The parent #GstAggregator
+
+
+
+ The caps set by the subclass
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ An implementation of GstPad that can be used with #GstAudioAggregator.
+
+See #GstAudioAggregator for more details.
+
+
+
+
+ The parent #GstAudioAggregatorPad
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ The default implementation of GstPad used with #GstAudioAggregator
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ The parent #GstAggregatorPad
+
+
+
+ The audio info for this pad set from the incoming caps
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ This is the base class for audio sinks. Subclasses need to implement the
+::create_ringbuffer vmethod. This base class will then take care of
+writing samples to the ringbuffer, synchronisation, clipping and flushing.
+
+ Create and return the #GstAudioRingBuffer for @sink. This function will
+call the ::create_ringbuffer vmethod and will set @sink as the parent of
+the returned buffer (see gst_object_set_parent()).
+
+ The new ringbuffer of @sink.
+
+
+
+
+ a #GstAudioBaseSink.
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Create and return the #GstAudioRingBuffer for @sink. This function will
+call the ::create_ringbuffer vmethod and will set @sink as the parent of
+the returned buffer (see gst_object_set_parent()).
+
+ The new ringbuffer of @sink.
+
+
+
+
+ a #GstAudioBaseSink.
+
+
+
+
+
+ Get the current alignment threshold, in nanoseconds, used by @sink.
+
+ The current alignment threshold used by @sink.
+
+
+
+
+ a #GstAudioBaseSink
+
+
+
+
+
+ Get the current discont wait, in nanoseconds, used by @sink.
+
+ The current discont wait used by @sink.
+
+
+
+
+ a #GstAudioBaseSink
+
+
+
+
+
+ Get the current drift tolerance, in microseconds, used by @sink.
+
+ The current drift tolerance used by @sink.
+
+
+
+
+ a #GstAudioBaseSink
+
+
+
+
+
+ Queries whether @sink will provide a clock or not. See also
+gst_audio_base_sink_set_provide_clock.
+
+ %TRUE if @sink will provide a clock.
+
+
+
+
+ a #GstAudioBaseSink
+
+
+
+
+
+ Get the current slave method used by @sink.
+
+ The current slave method used by @sink.
+
+
+
+
+ a #GstAudioBaseSink
+
+
+
+
+
+ Informs this base class that the audio output device has failed for
+some reason, causing a discontinuity (for example, because the device
+recovered from the error, but lost all contents of its ring buffer).
+This function is typically called by derived classes, and is useful
+for the custom slave method.
+
+
+
+
+
+ a #GstAudioBaseSink
+
+
+
+
+
+ Controls the sink's alignment threshold.
+
+
+
+
+
+ a #GstAudioBaseSink
+
+
+
+ the new alignment threshold in nanoseconds
+
+
+
+
+
+ Sets the custom slaving callback. This callback will
+be invoked if the slave-method property is set to
+GST_AUDIO_BASE_SINK_SLAVE_CUSTOM and the audio sink
+receives and plays samples.
+
+Setting the callback to NULL causes the sink to
+behave as if the GST_AUDIO_BASE_SINK_SLAVE_NONE
+method were used.
+
+
+
+
+
+ a #GstAudioBaseSink
+
+
+
+ a #GstAudioBaseSinkCustomSlavingCallback
+
+
+
+ user data passed to the callback
+
+
+
+ called when user_data becomes unused
+
+
+
+
+
+ Controls how long the sink will wait before creating a discontinuity.
+
+
+
+
+
+ a #GstAudioBaseSink
+
+
+
+ the new discont wait in nanoseconds
+
+
+
+
+
+ Controls the sink's drift tolerance.
+
+
+
+
+
+ a #GstAudioBaseSink
+
+
+
+ the new drift tolerance in microseconds
+
+
+
+
+
+ Controls whether @sink will provide a clock or not. If @provide is %TRUE,
+gst_element_provide_clock() will return a clock that reflects the datarate
+of @sink. If @provide is %FALSE, gst_element_provide_clock() will return
+NULL.
+
+
+
+
+
+ a #GstAudioBaseSink
+
+
+
+ new state
+
+
+
+
+
+ Controls how clock slaving will be performed in @sink.
+
+
+
+
+
+ a #GstAudioBaseSink
+
+
+
+ the new slave method
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ A window of time in nanoseconds to wait before creating a discontinuity as
+a result of breaching the drift-tolerance.
+
+
+
+ Controls the amount of time in microseconds that clocks are allowed
+to drift before resynchronisation happens.
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ #GstAudioBaseSink class. Override the vmethod to implement
+functionality.
+
+ the parent class.
+
+
+
+
+
+ The new ringbuffer of @sink.
+
+
+
+
+ a #GstAudioBaseSink.
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ This function is set with gst_audio_base_sink_set_custom_slaving_callback()
+and is called during playback. It receives the current time of external and
+internal clocks, which the callback can then use to apply any custom
+slaving/synchronization schemes.
+
+The external clock is the sink's element clock, the internal one is the
+internal audio clock. The internal audio clock's calibration is applied to
+the timestamps before they are passed to the callback. The difference between
+etime and itime is the skew; how much internal and external clock lie apart
+from each other. A skew of 0 means both clocks are perfectly in sync.
+itime > etime means the external clock is going slower, while itime < etime
+means it is going faster than the internal clock. etime and itime are always
+valid timestamps, except for when a discontinuity happens.
+
+requested_skew is an output value the callback can write to. It informs the
+sink of whether or not it should move the playout pointer, and if so, by how
+much. This pointer is only NULL if a discontinuity occurs; otherwise, it is
+safe to write to *requested_skew. The default skew is 0.
+
+The sink may experience discontinuities. If one happens, discont is TRUE,
+itime, etime are set to GST_CLOCK_TIME_NONE, and requested_skew is NULL.
+This makes it possible to reset custom clock slaving algorithms when a
+discontinuity happens.
+
+
+
+
+
+ a #GstAudioBaseSink
+
+
+
+ external clock time
+
+
+
+ internal clock time
+
+
+
+ skew amount requested by the callback
+
+
+
+ reason for discontinuity (if any)
+
+
+
+ user data
+
+
+
+
+
+ Different possible reasons for discontinuities. This enum is useful for the custom
+slave method.
+
+ No discontinuity occurred
+
+
+ New caps are set, causing renegotiotion
+
+
+ Samples have been flushed
+
+
+ Sink was synchronized to the estimated latency (occurs during initialization)
+
+
+ Aligning buffers failed because the timestamps are too discontinuous
+
+
+ Audio output device experienced and recovered from an error but introduced latency in the process (see also @gst_audio_base_sink_report_device_failure())
+
+
+
+
+
+ Different possible clock slaving algorithms used when the internal audio
+clock is not selected as the pipeline master clock.
+
+ Resample to match the master clock
+
+
+ Adjust playout pointer when master clock
+drifts too much.
+
+
+ No adjustment is done.
+
+
+ Use custom clock slaving algorithm (Since: 1.6)
+
+
+
+ This is the base class for audio sources. Subclasses need to implement the
+::create_ringbuffer vmethod. This base class will then take care of
+reading samples from the ringbuffer, synchronisation and flushing.
+
+ Create and return the #GstAudioRingBuffer for @src. This function will call
+the ::create_ringbuffer vmethod and will set @src as the parent of the
+returned buffer (see gst_object_set_parent()).
+
+ The new ringbuffer of @src.
+
+
+
+
+ a #GstAudioBaseSrc.
+
+
+
+
+
+ Create and return the #GstAudioRingBuffer for @src. This function will call
+the ::create_ringbuffer vmethod and will set @src as the parent of the
+returned buffer (see gst_object_set_parent()).
+
+ The new ringbuffer of @src.
+
+
+
+
+ a #GstAudioBaseSrc.
+
+
+
+
+
+ Queries whether @src will provide a clock or not. See also
+gst_audio_base_src_set_provide_clock.
+
+ %TRUE if @src will provide a clock.
+
+
+
+
+ a #GstAudioBaseSrc
+
+
+
+
+
+ Get the current slave method used by @src.
+
+ The current slave method used by @src.
+
+
+
+
+ a #GstAudioBaseSrc
+
+
+
+
+
+ Controls whether @src will provide a clock or not. If @provide is %TRUE,
+gst_element_provide_clock() will return a clock that reflects the datarate
+of @src. If @provide is %FALSE, gst_element_provide_clock() will return NULL.
+
+
+
+
+
+ a #GstAudioBaseSrc
+
+
+
+ new state
+
+
+
+
+
+ Controls how clock slaving will be performed in @src.
+
+
+
+
+
+ a #GstAudioBaseSrc
+
+
+
+ the new slave method
+
+
+
+
+
+ Actual configured size of audio buffer in microseconds.
+
+
+
+ Actual configured audio latency in microseconds.
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ #GstAudioBaseSrc class. Override the vmethod to implement
+functionality.
+
+ the parent class.
+
+
+
+
+
+ The new ringbuffer of @src.
+
+
+
+
+ a #GstAudioBaseSrc.
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Different possible clock slaving algorithms when the internal audio clock was
+not selected as the pipeline clock.
+
+ Resample to match the master clock.
+
+
+ Retimestamp output buffers with master
+clock time.
+
+
+ Adjust capture pointer when master clock
+drifts too much.
+
+
+ No adjustment is done.
+
+
+
+ Provides a base class for CD digital audio (CDDA) sources, which handles
+things like seeking, querying, discid calculation, tags, and buffer
+timestamping.
+
+## Using GstAudioCdSrc-based elements in applications
+
+GstAudioCdSrc registers two #GstFormat<!-- -->s of its own, namely
+the "track" format and the "sector" format. Applications will usually
+only find the "track" format interesting. You can retrieve that #GstFormat
+for use in seek events or queries with gst_format_get_by_nick("track").
+
+In order to query the number of tracks, for example, an application would
+set the CDDA source element to READY or PAUSED state and then query the
+the number of tracks via gst_element_query_duration() using the track
+format acquired above. Applications can query the currently playing track
+in the same way.
+
+Alternatively, applications may retrieve the currently playing track and
+the total number of tracks from the taglist that will posted on the bus
+whenever the CD is opened or the currently playing track changes. The
+taglist will contain GST_TAG_TRACK_NUMBER and GST_TAG_TRACK_COUNT tags.
+
+Applications playing back CD audio using playbin and cdda://n URIs should
+issue a seek command in track format to change between tracks, rather than
+setting a new cdda://n+1 URI on playbin (as setting a new URI on playbin
+involves closing and re-opening the CD device, which is much much slower).
+
+## Tags and meta-information
+
+CDDA sources will automatically emit a number of tags, details about which
+can be found in the libgsttag documentation. Those tags are:
+#GST_TAG_CDDA_CDDB_DISCID, #GST_TAG_CDDA_CDDB_DISCID_FULL,
+#GST_TAG_CDDA_MUSICBRAINZ_DISCID, #GST_TAG_CDDA_MUSICBRAINZ_DISCID_FULL,
+among others.
+
+## Tracks and Table of Contents (TOC)
+
+Applications will be informed of the available tracks via a TOC message
+on the pipeline's #GstBus. The #GstToc will contain a #GstTocEntry for
+each track, with information about each track. The duration for each
+track can be retrieved via the #GST_TAG_DURATION tag from each entry's
+tag list, or calculated via gst_toc_entry_get_start_stop_times().
+The track entries in the TOC will be sorted by track number.
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ CDDA sources use this function from their start vfunc to announce the
+available data and audio tracks to the base source class. The caller
+should allocate @track on the stack, the base source will do a shallow
+copy of the structure (and take ownership of the taglist if there is one).
+
+ FALSE on error, otherwise TRUE.
+
+
+
+
+ a #GstAudioCdSrc
+
+
+
+ address of #GstAudioCdSrcTrack to add
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Audio CD source base class.
+
+ the parent class
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Mode in which the CD audio source operates. Influences timestamping,
+EOS handling and seeking.
+
+ each single track is a stream
+
+
+ the entire disc is a single stream
+
+
+
+
+
+ CD track abstraction to communicate TOC entries to the base class.
+
+This structure is only for use by sub-classed in connection with
+gst_audio_cd_src_add_track().
+
+Applications will be informed of the available tracks via a TOC message
+on the pipeline's #GstBus instead.
+
+ Whether this is an audio track
+
+
+
+ Track number in TOC (usually starts from 1, but not always)
+
+
+
+ The first sector of this track (LBA)
+
+
+
+ The last sector of this track (LBA)
+
+
+
+ Track-specific tags (e.g. from cd-text information), or NULL
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Free memory allocated by @mix.
+
+
+
+
+
+ a #GstAudioChannelMixer
+
+
+
+
+
+ Check if @mix is in passthrough.
+
+Only N x N mix identity matrices are considered passthrough,
+this is determined by comparing the contents of the matrix
+with 0.0 and 1.0.
+
+As this is floating point comparisons, if the values have been
+generated, they should be rounded up or down by explicit
+assignment of 0.0 or 1.0 to values within a user-defined
+epsilon, this code doesn't make assumptions as to what may
+constitute an appropriate epsilon.
+
+ %TRUE is @mix is passthrough.
+
+
+
+
+ a #GstAudioChannelMixer
+
+
+
+
+
+ In case the samples are interleaved, @in and @out must point to an
+array with a single element pointing to a block of interleaved samples.
+
+If non-interleaved samples are used, @in and @out must point to an
+array with pointers to memory blocks, one for each channel.
+
+Perform channel mixing on @in_data and write the result to @out_data.
+@in_data and @out_data need to be in @format and @layout.
+
+
+
+
+
+ a #GstAudioChannelMixer
+
+
+
+ input samples
+
+
+
+ output samples
+
+
+
+ number of samples
+
+
+
+
+
+ Create a new channel mixer object for the given parameters.
+
+ a new #GstAudioChannelMixer object, or %NULL if @format isn't supported.
+ Free with gst_audio_channel_mixer_free() after usage.
+
+
+
+
+ #GstAudioChannelMixerFlags
+
+
+
+
+
+
+ number of input channels
+
+
+
+ positions of input channels
+
+
+
+ number of output channels
+
+
+
+ positions of output channels
+
+
+
+
+
+ Create a new channel mixer object for the given parameters.
+
+ a new #GstAudioChannelMixer object, or %NULL if @format isn't supported,
+ @matrix is invalid, or @matrix is %NULL and @in_channels != @out_channels.
+ Free with gst_audio_channel_mixer_free() after usage.
+
+
+
+
+ #GstAudioChannelMixerFlags
+
+
+
+
+
+
+ number of input channels
+
+
+
+ number of output channels
+
+
+
+ channel conversion matrix, m[@in_channels][@out_channels].
+ If identity matrix, passthrough applies. If %NULL, a (potentially truncated)
+ identity matrix is generated.
+
+
+
+
+
+
+ Flags passed to gst_audio_channel_mixer_new()
+
+ no flag
+
+
+ input channels are not interleaved
+
+
+ output channels are not interleaved
+
+
+ input channels are explicitly unpositioned
+
+
+ output channels are explicitly unpositioned
+
+
+
+ Audio channel positions.
+
+These are the channels defined in SMPTE 2036-2-2008
+Table 1 for 22.2 audio systems with the Surround and Wide channels from
+DTS Coherent Acoustics (v.1.3.1) and 10.2 and 7.1 layouts. In the caps the
+actual channel layout is expressed with a channel count and a channel mask,
+which describes the existing channels. The positions in the bit mask correspond
+to the enum values.
+For negotiation it is allowed to have more bits set in the channel mask than
+the number of channels to specify the allowed channel positions but this is
+not allowed in negotiated caps. It is not allowed in any situation other
+than the one mentioned below to have less bits set in the channel mask than
+the number of channels.
+
+@GST_AUDIO_CHANNEL_POSITION_MONO can only be used with a single mono channel that
+has no direction information and would be mixed into all directional channels.
+This is expressed in caps by having a single channel and no channel mask.
+
+@GST_AUDIO_CHANNEL_POSITION_NONE can only be used if all channels have this position.
+This is expressed in caps by having a channel mask with no bits set.
+
+As another special case it is allowed to have two channels without a channel mask.
+This implicitely means that this is a stereo stream with a front left and front right
+channel.
+
+ used for position-less channels, e.g.
+ from a sound card that records 1024 channels; mutually exclusive with
+ any other channel position
+
+
+ Mono without direction;
+ can only be used with 1 channel
+
+
+ invalid position
+
+
+ Front left
+
+
+ Front right
+
+
+ Front center
+
+
+ Low-frequency effects 1 (subwoofer)
+
+
+ Rear left
+
+
+ Rear right
+
+
+ Front left of center
+
+
+ Front right of center
+
+
+ Rear center
+
+
+ Low-frequency effects 2 (subwoofer)
+
+
+ Side left
+
+
+ Side right
+
+
+ Top front left
+
+
+ Top front right
+
+
+ Top front center
+
+
+ Top center
+
+
+ Top rear left
+
+
+ Top rear right
+
+
+ Top side right
+
+
+ Top rear right
+
+
+ Top rear center
+
+
+ Bottom front center
+
+
+ Bottom front left
+
+
+ Bottom front right
+
+
+ Wide left (between front left and side left)
+
+
+ Wide right (between front right and side right)
+
+
+ Surround left (between rear left and side left)
+
+
+ Surround right (between rear right and side right)
+
+
+
+ Extra buffer metadata describing how much audio has to be clipped from
+the start or end of a buffer. This is used for compressed formats, where
+the first frame usually has some additional samples due to encoder and
+decoder delays, and the last frame usually has some additional samples to
+be able to fill the complete last frame.
+
+This is used to ensure that decoded data in the end has the same amount of
+samples, and multiply decoded streams can be gaplessly concatenated.
+
+Note: If clipping of the start is done by adjusting the segment, this meta
+has to be dropped from buffers as otherwise clipping could happen twice.
+
+ parent #GstMeta
+
+
+
+ GstFormat of @start and @stop, GST_FORMAT_DEFAULT is samples
+
+
+
+ Amount of audio to clip from start of buffer
+
+
+
+ Amount of to clip from end of buffer
+
+
+
+
+
+
+
+
+
+ #GstAudioClock makes it easy for elements to implement a #GstClock, they
+simply need to provide a function that returns the current clock time.
+
+This object is internally used to implement the clock in #GstAudioBaseSink.
+
+ Create a new #GstAudioClock instance. Whenever the clock time should be
+calculated it will call @func with @user_data. When @func returns
+#GST_CLOCK_TIME_NONE, the clock will return the last reported time.
+
+ a new #GstAudioClock casted to a #GstClock.
+
+
+
+
+ the name of the clock
+
+
+
+ a function
+
+
+
+ user data
+
+
+
+ #GDestroyNotify for @user_data
+
+
+
+
+
+ Adjust @time with the internal offset of the audio clock.
+
+ @time adjusted with the internal offset.
+
+
+
+
+ a #GstAudioClock
+
+
+
+ a #GstClockTime
+
+
+
+
+
+ Report the time as returned by the #GstAudioClockGetTimeFunc without applying
+any offsets.
+
+ the time as reported by the time function of the audio clock
+
+
+
+
+ a #GstAudioClock
+
+
+
+
+
+ Invalidate the clock function. Call this function when the provided
+#GstAudioClockGetTimeFunc cannot be called anymore, for example, when the
+user_data becomes invalid.
+
+After calling this function, @clock will return the last returned time for
+the rest of its lifetime.
+
+
+
+
+
+ a #GstAudioClock
+
+
+
+
+
+ Inform @clock that future calls to #GstAudioClockGetTimeFunc will return values
+starting from @time. The clock will update an internal offset to make sure that
+future calls to internal_time will return an increasing result as required by
+the #GstClock object.
+
+
+
+
+
+ a #GstAudioClock
+
+
+
+ a #GstClockTime
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ This function will be called whenever the current clock time needs to be
+calculated. If this function returns #GST_CLOCK_TIME_NONE, the last reported
+time will be returned by the clock.
+
+ the current time or #GST_CLOCK_TIME_NONE if the previous time should
+be used.
+
+
+
+
+ the #GstAudioClock
+
+
+
+ user data
+
+
+
+
+
+
+ Create a new #GstAudioConverter that is able to convert between @in and @out
+audio formats.
+
+@config contains extra configuration options, see #GST_VIDEO_CONVERTER_OPT_*
+parameters for details about the options and values.
+
+ a #GstAudioConverter or %NULL if conversion is not possible.
+
+
+
+
+ extra #GstAudioConverterFlags
+
+
+
+ a source #GstAudioInfo
+
+
+
+ a destination #GstAudioInfo
+
+
+
+ a #GstStructure with configuration options
+
+
+
+
+
+ Convenience wrapper around gst_audio_converter_samples(), which will
+perform allocation of the output buffer based on the result from
+gst_audio_converter_get_out_frames().
+
+ %TRUE is the conversion could be performed.
+
+
+
+
+
+
+
+ extra #GstAudioConverterFlags
+
+
+
+ input data
+
+
+
+
+
+ size of @in
+
+
+
+ a pointer where
+ the output data will be written
+
+
+
+
+
+ a pointer where the size of @out will be written
+
+
+
+
+
+ Free a previously allocated @convert instance.
+
+
+
+
+
+ a #GstAudioConverter
+
+
+
+
+
+ Get the current configuration of @convert.
+
+
+ a #GstStructure that remains valid for as long as @convert is valid
+ or until gst_audio_converter_update_config() is called.
+
+
+
+
+ a #GstAudioConverter
+
+
+
+ result input rate
+
+
+
+ result output rate
+
+
+
+
+
+ Calculate how many input frames are currently needed by @convert to produce
+@out_frames of output frames.
+
+ the number of input frames
+
+
+
+
+ a #GstAudioConverter
+
+
+
+ number of output frames
+
+
+
+
+
+ Get the maximum number of input frames that the converter would
+need before producing output.
+
+ the latency of @convert as expressed in the number of
+frames.
+
+
+
+
+ a #GstAudioConverter
+
+
+
+
+
+ Calculate how many output frames can be produced when @in_frames input
+frames are given to @convert.
+
+ the number of output frames
+
+
+
+
+ a #GstAudioConverter
+
+
+
+ number of input frames
+
+
+
+
+
+ Reset @convert to the state it was when it was first created, clearing
+any history it might currently have.
+
+
+
+
+
+ a #GstAudioConverter
+
+
+
+
+
+ Perform the conversion with @in_frames in @in to @out_frames in @out
+using @convert.
+
+In case the samples are interleaved, @in and @out must point to an
+array with a single element pointing to a block of interleaved samples.
+
+If non-interleaved samples are used, @in and @out must point to an
+array with pointers to memory blocks, one for each channel.
+
+@in may be %NULL, in which case @in_frames of silence samples are processed
+by the converter.
+
+This function always produces @out_frames of output and consumes @in_frames of
+input. Use gst_audio_converter_get_out_frames() and
+gst_audio_converter_get_in_frames() to make sure @in_frames and @out_frames
+are matching and @in and @out point to enough memory.
+
+ %TRUE is the conversion could be performed.
+
+
+
+
+ a #GstAudioConverter
+
+
+
+ extra #GstAudioConverterFlags
+
+
+
+ input frames
+
+
+
+ number of input frames
+
+
+
+ output frames
+
+
+
+ number of output frames
+
+
+
+
+
+ Returns whether the audio converter can perform the conversion in-place.
+The return value would be typically input to gst_base_transform_set_in_place()
+
+ %TRUE when the conversion can be done in place.
+
+
+
+
+ a #GstAudioConverter
+
+
+
+
+
+ Set @in_rate, @out_rate and @config as extra configuration for @convert.
+
+@in_rate and @out_rate specify the new sample rates of input and output
+formats. A value of 0 leaves the sample rate unchanged.
+
+@config can be %NULL, in which case, the current configuration is not
+changed.
+
+If the parameters in @config can not be set exactly, this function returns
+%FALSE and will try to update as much state as possible. The new state can
+then be retrieved and refined with gst_audio_converter_get_config().
+
+Look at the #GST_AUDIO_CONVERTER_OPT_* fields to check valid configuration
+option and values.
+
+ %TRUE when the new parameters could be set
+
+
+
+
+ a #GstAudioConverter
+
+
+
+ input rate
+
+
+
+ output rate
+
+
+
+ a #GstStructure or %NULL
+
+
+
+
+
+
+ Extra flags passed to gst_audio_converter_new() and gst_audio_converter_samples().
+
+ no flag
+
+
+ the input sample arrays are writable and can be
+ used as temporary storage during conversion.
+
+
+ allow arbitrary rate updates with
+ gst_audio_converter_update_config().
+
+
+
+ This base class is for audio decoders turning encoded data into
+raw audio samples.
+
+GstAudioDecoder and subclass should cooperate as follows.
+
+## Configuration
+
+ * Initially, GstAudioDecoder calls @start when the decoder element
+ is activated, which allows subclass to perform any global setup.
+ Base class (context) parameters can already be set according to subclass
+ capabilities (or possibly upon receive more information in subsequent
+ @set_format).
+ * GstAudioDecoder calls @set_format to inform subclass of the format
+ of input audio data that it is about to receive.
+ While unlikely, it might be called more than once, if changing input
+ parameters require reconfiguration.
+ * GstAudioDecoder calls @stop at end of all processing.
+
+As of configuration stage, and throughout processing, GstAudioDecoder
+provides various (context) parameters, e.g. describing the format of
+output audio data (valid when output caps have been set) or current parsing state.
+Conversely, subclass can and should configure context to inform
+base class of its expectation w.r.t. buffer handling.
+
+## Data processing
+ * Base class gathers input data, and optionally allows subclass
+ to parse this into subsequently manageable (as defined by subclass)
+ chunks. Such chunks are subsequently referred to as 'frames',
+ though they may or may not correspond to 1 (or more) audio format frame.
+ * Input frame is provided to subclass' @handle_frame.
+ * If codec processing results in decoded data, subclass should call
+ @gst_audio_decoder_finish_frame to have decoded data pushed
+ downstream.
+ * Just prior to actually pushing a buffer downstream,
+ it is passed to @pre_push. Subclass should either use this callback
+ to arrange for additional downstream pushing or otherwise ensure such
+ custom pushing occurs after at least a method call has finished since
+ setting src pad caps.
+ * During the parsing process GstAudioDecoderClass will handle both
+ srcpad and sinkpad events. Sink events will be passed to subclass
+ if @event callback has been provided.
+
+## Shutdown phase
+
+ * GstAudioDecoder class calls @stop to inform the subclass that data
+ parsing will be stopped.
+
+Subclass is responsible for providing pad template caps for
+source and sink pads. The pads need to be named "sink" and "src". It also
+needs to set the fixed caps on srcpad, when the format is ensured. This
+is typically when base class calls subclass' @set_format function, though
+it might be delayed until calling @gst_audio_decoder_finish_frame.
+
+In summary, above process should have subclass concentrating on
+codec data processing while leaving other matters to base class,
+such as most notably timestamp handling. While it may exert more control
+in this area (see e.g. @pre_push), it is very much not recommended.
+
+In particular, base class will try to arrange for perfect output timestamps
+as much as possible while tracking upstream timestamps.
+To this end, if deviation between the next ideal expected perfect timestamp
+and upstream exceeds #GstAudioDecoder:tolerance, then resync to upstream
+occurs (which would happen always if the tolerance mechanism is disabled).
+
+In non-live pipelines, baseclass can also (configurably) arrange for
+output buffer aggregation which may help to redue large(r) numbers of
+small(er) buffers being pushed and processed downstream.
+
+On the other hand, it should be noted that baseclass only provides limited
+seeking support (upon explicit subclass request), as full-fledged support
+should rather be left to upstream demuxer, parser or alike. This simple
+approach caters for seeking and duration reporting using estimated input
+bitrates.
+
+Things that subclass need to take care of:
+
+ * Provide pad templates
+ * Set source pad caps when appropriate
+ * Set user-configurable properties to sane defaults for format and
+ implementing codec at hand, and convey some subclass capabilities and
+ expectations in context.
+
+ * Accept data in @handle_frame and provide encoded results to
+ @gst_audio_decoder_finish_frame. If it is prepared to perform
+ PLC, it should also accept NULL data in @handle_frame and provide for
+ data for indicated duration.
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Negotiate with downstream elements to currently configured #GstAudioInfo.
+Unmark GST_PAD_FLAG_NEED_RECONFIGURE in any case. But mark it again if
+negotiate fails.
+
+ %TRUE if the negotiation succeeded, else %FALSE.
+
+
+
+
+ a #GstAudioDecoder
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Helper function that allocates a buffer to hold an audio frame
+for @dec's current output format.
+
+ allocated buffer
+
+
+
+
+ a #GstAudioDecoder
+
+
+
+ size of the buffer
+
+
+
+
+
+ Collects decoded data and pushes it downstream.
+
+@buf may be NULL in which case the indicated number of frames
+are discarded and considered to have produced no output
+(e.g. lead-in or setup frames).
+Otherwise, source pad caps must be set when it is called with valid
+data in @buf.
+
+Note that a frame received in gst_audio_decoder_handle_frame() may be
+invalidated by a call to this function.
+
+ a #GstFlowReturn that should be escalated to caller (of caller)
+
+
+
+
+ a #GstAudioDecoder
+
+
+
+ decoded data
+
+
+
+ number of decoded frames represented by decoded data
+
+
+
+
+
+ Lets #GstAudioDecoder sub-classes to know the memory @allocator
+used by the base class and its @params.
+
+Unref the @allocator after use it.
+
+
+
+
+
+ a #GstAudioDecoder
+
+
+
+ the #GstAllocator
+used
+
+
+
+ the
+#GstAllocatorParams of @allocator
+
+
+
+
+
+
+ a #GstAudioInfo describing the input audio format
+
+
+
+
+ a #GstAudioDecoder
+
+
+
+
+
+
+ currently configured decoder delay
+
+
+
+
+ a #GstAudioDecoder
+
+
+
+
+
+ Queries decoder drain handling.
+
+ TRUE if drainable handling is enabled.
+
+MT safe.
+
+
+
+
+ a #GstAudioDecoder
+
+
+
+
+
+
+ currently configured byte to time conversion setting
+
+
+
+
+ a #GstAudioDecoder
+
+
+
+
+
+ Sets the variables pointed to by @min and @max to the currently configured
+latency.
+
+
+
+
+
+ a #GstAudioDecoder
+
+
+
+ a pointer to storage to hold minimum latency
+
+
+
+ a pointer to storage to hold maximum latency
+
+
+
+
+
+
+ currently configured decoder tolerated error count.
+
+
+
+
+ a #GstAudioDecoder
+
+
+
+
+
+ Queries decoder's latency aggregation.
+
+ aggregation latency.
+
+MT safe.
+
+
+
+
+ a #GstAudioDecoder
+
+
+
+
+
+ Queries decoder required format handling.
+
+ TRUE if required format handling is enabled.
+
+MT safe.
+
+
+
+
+ a #GstAudioDecoder
+
+
+
+
+
+ Return current parsing (sync and eos) state.
+
+
+
+
+
+ a #GstAudioDecoder
+
+
+
+ a pointer to a variable to hold the current sync state
+
+
+
+ a pointer to a variable to hold the current eos state
+
+
+
+
+
+ Queries decoder packet loss concealment handling.
+
+ TRUE if packet loss concealment is enabled.
+
+MT safe.
+
+
+
+
+ a #GstAudioDecoder
+
+
+
+
+
+
+ currently configured plc handling
+
+
+
+
+ a #GstAudioDecoder
+
+
+
+
+
+ Queries current audio jitter tolerance threshold.
+
+ decoder audio jitter tolerance threshold.
+
+MT safe.
+
+
+
+
+ a #GstAudioDecoder
+
+
+
+
+
+ Sets the audio decoder tags and how they should be merged with any
+upstream stream tags. This will override any tags previously-set
+with gst_audio_decoder_merge_tags().
+
+Note that this is provided for convenience, and the subclass is
+not required to use this and can still do tag handling on its own.
+
+
+
+
+
+ a #GstAudioDecoder
+
+
+
+ a #GstTagList to merge, or NULL
+
+
+
+ the #GstTagMergeMode to use, usually #GST_TAG_MERGE_REPLACE
+
+
+
+
+
+ Negotiate with downstream elements to currently configured #GstAudioInfo.
+Unmark GST_PAD_FLAG_NEED_RECONFIGURE in any case. But mark it again if
+negotiate fails.
+
+ %TRUE if the negotiation succeeded, else %FALSE.
+
+
+
+
+ a #GstAudioDecoder
+
+
+
+
+
+ Returns caps that express @caps (or sink template caps if @caps == NULL)
+restricted to rate/channels/... combinations supported by downstream
+elements.
+
+ a #GstCaps owned by caller
+
+
+
+
+ a #GstAudioDecoder
+
+
+
+ initial caps
+
+
+
+ filter caps
+
+
+
+
+
+ Sets a caps in allocation query which are different from the set
+pad's caps. Use this function before calling
+gst_audio_decoder_negotiate(). Setting to %NULL the allocation
+query will use the caps from the pad.
+
+
+
+
+
+ a #GstAudioDecoder
+
+
+
+ a #GstCaps or %NULL
+
+
+
+
+
+ Configures decoder drain handling. If drainable, subclass might
+be handed a NULL buffer to have it return any leftover decoded data.
+Otherwise, it is not considered so capable and will only ever be passed
+real data.
+
+MT safe.
+
+
+
+
+
+ a #GstAudioDecoder
+
+
+
+ new state
+
+
+
+
+
+ Allows baseclass to perform byte to time estimated conversion.
+
+
+
+
+
+ a #GstAudioDecoder
+
+
+
+ whether to enable byte to time conversion
+
+
+
+
+
+ Sets decoder latency.
+
+
+
+
+
+ a #GstAudioDecoder
+
+
+
+ minimum latency
+
+
+
+ maximum latency
+
+
+
+
+
+ Sets numbers of tolerated decoder errors, where a tolerated one is then only
+warned about, but more than tolerated will lead to fatal error. You can set
+-1 for never returning fatal errors. Default is set to
+GST_AUDIO_DECODER_MAX_ERRORS.
+
+
+
+
+
+ a #GstAudioDecoder
+
+
+
+ max tolerated errors
+
+
+
+
+
+ Sets decoder minimum aggregation latency.
+
+MT safe.
+
+
+
+
+
+ a #GstAudioDecoder
+
+
+
+ new minimum latency
+
+
+
+
+
+ Configures decoder format needs. If enabled, subclass needs to be
+negotiated with format caps before it can process any data. It will then
+never be handed any data before it has been configured.
+Otherwise, it might be handed data without having been configured and
+is then expected being able to do so either by default
+or based on the input data.
+
+MT safe.
+
+
+
+
+
+ a #GstAudioDecoder
+
+
+
+ new state
+
+
+
+
+
+ Configure output info on the srcpad of @dec.
+
+ %TRUE on success.
+
+
+
+
+ a #GstAudioDecoder
+
+
+
+ #GstAudioInfo
+
+
+
+
+
+ Enable or disable decoder packet loss concealment, provided subclass
+and codec are capable and allow handling plc.
+
+MT safe.
+
+
+
+
+
+ a #GstAudioDecoder
+
+
+
+ new state
+
+
+
+
+
+ Indicates whether or not subclass handles packet loss concealment (plc).
+
+
+
+
+
+ a #GstAudioDecoder
+
+
+
+ new plc state
+
+
+
+
+
+ Configures decoder audio jitter tolerance threshold.
+
+MT safe.
+
+
+
+
+
+ a #GstAudioDecoder
+
+
+
+ new tolerance
+
+
+
+
+
+ Lets #GstAudioDecoder sub-classes decide if they want the sink pad
+to use the default pad query handler to reply to accept-caps queries.
+
+By setting this to true it is possible to further customize the default
+handler with %GST_PAD_SET_ACCEPT_INTERSECT and
+%GST_PAD_SET_ACCEPT_TEMPLATE
+
+
+
+
+
+ a #GstAudioDecoder
+
+
+
+ if the default pad accept-caps query handling should be used
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Subclasses can override any of the available virtual methods or not, as
+needed. At minimum @handle_frame (and likely @set_format) needs to be
+overridden.
+
+ The parent class structure
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ %TRUE if the negotiation succeeded, else %FALSE.
+
+
+
+
+ a #GstAudioDecoder
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Set of available dithering methods.
+
+ No dithering
+
+
+ Rectangular dithering
+
+
+ Triangular dithering (default)
+
+
+ High frequency triangular dithering
+
+
+
+ Extra buffer metadata describing audio downmixing matrix. This metadata is
+attached to audio buffers and contains a matrix to downmix the buffer number
+of channels to @channels.
+
+@matrix is an two-dimensional array of @to_channels times @from_channels
+coefficients, i.e. the i-th output channels is constructed by multiplicating
+the input channels with the coefficients in @matrix[i] and taking the sum
+of the results.
+
+ parent #GstMeta
+
+
+
+ the channel positions of the source
+
+
+
+ the channel positions of the destination
+
+
+
+ the number of channels of the source
+
+
+
+ the number of channels of the destination
+
+
+
+ the matrix coefficients.
+
+
+
+
+
+
+
+
+
+ This base class is for audio encoders turning raw audio samples into
+encoded audio data.
+
+GstAudioEncoder and subclass should cooperate as follows.
+
+## Configuration
+
+ * Initially, GstAudioEncoder calls @start when the encoder element
+ is activated, which allows subclass to perform any global setup.
+
+ * GstAudioEncoder calls @set_format to inform subclass of the format
+ of input audio data that it is about to receive. Subclass should
+ setup for encoding and configure various base class parameters
+ appropriately, notably those directing desired input data handling.
+ While unlikely, it might be called more than once, if changing input
+ parameters require reconfiguration.
+
+ * GstAudioEncoder calls @stop at end of all processing.
+
+As of configuration stage, and throughout processing, GstAudioEncoder
+maintains various parameters that provide required context,
+e.g. describing the format of input audio data.
+Conversely, subclass can and should configure these context parameters
+to inform base class of its expectation w.r.t. buffer handling.
+
+## Data processing
+
+ * Base class gathers input sample data (as directed by the context's
+ frame_samples and frame_max) and provides this to subclass' @handle_frame.
+ * If codec processing results in encoded data, subclass should call
+ gst_audio_encoder_finish_frame() to have encoded data pushed
+ downstream. Alternatively, it might also call
+ gst_audio_encoder_finish_frame() (with a NULL buffer and some number of
+ dropped samples) to indicate dropped (non-encoded) samples.
+ * Just prior to actually pushing a buffer downstream,
+ it is passed to @pre_push.
+ * During the parsing process GstAudioEncoderClass will handle both
+ srcpad and sinkpad events. Sink events will be passed to subclass
+ if @event callback has been provided.
+
+## Shutdown phase
+
+ * GstAudioEncoder class calls @stop to inform the subclass that data
+ parsing will be stopped.
+
+Subclass is responsible for providing pad template caps for
+source and sink pads. The pads need to be named "sink" and "src". It also
+needs to set the fixed caps on srcpad, when the format is ensured. This
+is typically when base class calls subclass' @set_format function, though
+it might be delayed until calling @gst_audio_encoder_finish_frame.
+
+In summary, above process should have subclass concentrating on
+codec data processing while leaving other matters to base class,
+such as most notably timestamp handling. While it may exert more control
+in this area (see e.g. @pre_push), it is very much not recommended.
+
+In particular, base class will either favor tracking upstream timestamps
+(at the possible expense of jitter) or aim to arrange for a perfect stream of
+output timestamps, depending on #GstAudioEncoder:perfect-timestamp.
+However, in the latter case, the input may not be so perfect or ideal, which
+is handled as follows. An input timestamp is compared with the expected
+timestamp as dictated by input sample stream and if the deviation is less
+than #GstAudioEncoder:tolerance, the deviation is discarded.
+Otherwise, it is considered a discontuinity and subsequent output timestamp
+is resynced to the new position after performing configured discontinuity
+processing. In the non-perfect-timestamp case, an upstream variation
+exceeding tolerance only leads to marking DISCONT on subsequent outgoing
+(while timestamps are adjusted to upstream regardless of variation).
+While DISCONT is also marked in the perfect-timestamp case, this one
+optionally (see #GstAudioEncoder:hard-resync)
+performs some additional steps, such as clipping of (early) input samples
+or draining all currently remaining input data, depending on the direction
+of the discontuinity.
+
+If perfect timestamps are arranged, it is also possible to request baseclass
+(usually set by subclass) to provide additional buffer metadata (in OFFSET
+and OFFSET_END) fields according to granule defined semantics currently
+needed by oggmux. Specifically, OFFSET is set to granulepos (= sample count
+including buffer) and OFFSET_END to corresponding timestamp (as determined
+by same sample count and sample rate).
+
+Things that subclass need to take care of:
+
+ * Provide pad templates
+ * Set source pad caps when appropriate
+ * Inform base class of buffer processing needs using context's
+ frame_samples and frame_bytes.
+ * Set user-configurable properties to sane defaults for format and
+ implementing codec at hand, e.g. those controlling timestamp behaviour
+ and discontinuity processing.
+ * Accept data in @handle_frame and provide encoded results to
+ gst_audio_encoder_finish_frame().
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Negotiate with downstream elements to currently configured #GstCaps.
+Unmark GST_PAD_FLAG_NEED_RECONFIGURE in any case. But mark it again if
+negotiate fails.
+
+ %TRUE if the negotiation succeeded, else %FALSE.
+
+
+
+
+ a #GstAudioEncoder
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Helper function that allocates a buffer to hold an encoded audio frame
+for @enc's current output format.
+
+ allocated buffer
+
+
+
+
+ a #GstAudioEncoder
+
+
+
+ size of the buffer
+
+
+
+
+
+ Collects encoded data and pushes encoded data downstream.
+Source pad caps must be set when this is called.
+
+If @samples < 0, then best estimate is all samples provided to encoder
+(subclass) so far. @buf may be NULL, in which case next number of @samples
+are considered discarded, e.g. as a result of discontinuous transmission,
+and a discontinuity is marked.
+
+Note that samples received in gst_audio_encoder_handle_frame()
+may be invalidated by a call to this function.
+
+ a #GstFlowReturn that should be escalated to caller (of caller)
+
+
+
+
+ a #GstAudioEncoder
+
+
+
+ encoded data
+
+
+
+ number of samples (per channel) represented by encoded data
+
+
+
+
+
+ Lets #GstAudioEncoder sub-classes to know the memory @allocator
+used by the base class and its @params.
+
+Unref the @allocator after use it.
+
+
+
+
+
+ a #GstAudioEncoder
+
+
+
+ the #GstAllocator
+used
+
+
+
+ the
+#GstAllocatorParams of @allocator
+
+
+
+
+
+
+ a #GstAudioInfo describing the input audio format
+
+
+
+
+ a #GstAudioEncoder
+
+
+
+
+
+ Queries encoder drain handling.
+
+ TRUE if drainable handling is enabled.
+
+MT safe.
+
+
+
+
+ a #GstAudioEncoder
+
+
+
+
+
+
+ currently configured maximum handled frames
+
+
+
+
+ a #GstAudioEncoder
+
+
+
+
+
+
+ currently maximum requested samples per frame
+
+
+
+
+ a #GstAudioEncoder
+
+
+
+
+
+
+ currently minimum requested samples per frame
+
+
+
+
+ a #GstAudioEncoder
+
+
+
+
+
+ Queries encoder hard minimum handling.
+
+ TRUE if hard minimum handling is enabled.
+
+MT safe.
+
+
+
+
+ a #GstAudioEncoder
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Sets the variables pointed to by @min and @max to the currently configured
+latency.
+
+
+
+
+
+ a #GstAudioEncoder
+
+
+
+ a pointer to storage to hold minimum latency
+
+
+
+ a pointer to storage to hold maximum latency
+
+
+
+
+
+
+ currently configured encoder lookahead
+
+
+
+
+ a #GstAudioEncoder
+
+
+
+
+
+ Queries if the encoder will handle granule marking.
+
+ TRUE if granule marking is enabled.
+
+MT safe.
+
+
+
+
+ a #GstAudioEncoder
+
+
+
+
+
+ Queries encoder perfect timestamp behaviour.
+
+ TRUE if perfect timestamp setting enabled.
+
+MT safe.
+
+
+
+
+ a #GstAudioEncoder
+
+
+
+
+
+ Queries current audio jitter tolerance threshold.
+
+ encoder audio jitter tolerance threshold.
+
+MT safe.
+
+
+
+
+ a #GstAudioEncoder
+
+
+
+
+
+ Sets the audio encoder tags and how they should be merged with any
+upstream stream tags. This will override any tags previously-set
+with gst_audio_encoder_merge_tags().
+
+Note that this is provided for convenience, and the subclass is
+not required to use this and can still do tag handling on its own.
+
+MT safe.
+
+
+
+
+
+ a #GstAudioEncoder
+
+
+
+ a #GstTagList to merge, or NULL to unset
+ previously-set tags
+
+
+
+ the #GstTagMergeMode to use, usually #GST_TAG_MERGE_REPLACE
+
+
+
+
+
+ Negotiate with downstream elements to currently configured #GstCaps.
+Unmark GST_PAD_FLAG_NEED_RECONFIGURE in any case. But mark it again if
+negotiate fails.
+
+ %TRUE if the negotiation succeeded, else %FALSE.
+
+
+
+
+ a #GstAudioEncoder
+
+
+
+
+
+ Returns caps that express @caps (or sink template caps if @caps == NULL)
+restricted to channel/rate combinations supported by downstream elements
+(e.g. muxers).
+
+ a #GstCaps owned by caller
+
+
+
+
+ a #GstAudioEncoder
+
+
+
+ initial caps
+
+
+
+ filter caps
+
+
+
+
+
+ Sets a caps in allocation query which are different from the set
+pad's caps. Use this function before calling
+gst_audio_encoder_negotiate(). Setting to %NULL the allocation
+query will use the caps from the pad.
+
+
+
+
+
+ a #GstAudioEncoder
+
+
+
+ a #GstCaps or %NULL
+
+
+
+
+
+ Configures encoder drain handling. If drainable, subclass might
+be handed a NULL buffer to have it return any leftover encoded data.
+Otherwise, it is not considered so capable and will only ever be passed
+real data.
+
+MT safe.
+
+
+
+
+
+ a #GstAudioEncoder
+
+
+
+ new state
+
+
+
+
+
+ Sets max number of frames accepted at once (assumed minimally 1).
+Requires @frame_samples_min and @frame_samples_max to be the equal.
+
+Note: This value will be reset to 0 every time before
+GstAudioEncoder::set_format() is called.
+
+
+
+
+
+ a #GstAudioEncoder
+
+
+
+ number of frames
+
+
+
+
+
+ Sets number of samples (per channel) subclass needs to be handed,
+at most or will be handed all available if 0.
+
+If an exact number of samples is required, gst_audio_encoder_set_frame_samples_min()
+must be called with the same number.
+
+Note: This value will be reset to 0 every time before
+GstAudioEncoder::set_format() is called.
+
+
+
+
+
+ a #GstAudioEncoder
+
+
+
+ number of samples per frame
+
+
+
+
+
+ Sets number of samples (per channel) subclass needs to be handed,
+at least or will be handed all available if 0.
+
+If an exact number of samples is required, gst_audio_encoder_set_frame_samples_max()
+must be called with the same number.
+
+Note: This value will be reset to 0 every time before
+GstAudioEncoder::set_format() is called.
+
+
+
+
+
+ a #GstAudioEncoder
+
+
+
+ number of samples per frame
+
+
+
+
+
+ Configures encoder hard minimum handling. If enabled, subclass
+will never be handed less samples than it configured, which otherwise
+might occur near end-of-data handling. Instead, the leftover samples
+will simply be discarded.
+
+MT safe.
+
+
+
+
+
+ a #GstAudioEncoder
+
+
+
+ new state
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Set the codec headers to be sent downstream whenever requested.
+
+
+
+
+
+ a #GstAudioEncoder
+
+
+
+ a list of
+ #GstBuffer containing the codec header
+
+
+
+
+
+
+
+ Sets encoder latency.
+
+
+
+
+
+ a #GstAudioEncoder
+
+
+
+ minimum latency
+
+
+
+ maximum latency
+
+
+
+
+
+ Sets encoder lookahead (in units of input rate samples)
+
+Note: This value will be reset to 0 every time before
+GstAudioEncoder::set_format() is called.
+
+
+
+
+
+ a #GstAudioEncoder
+
+
+
+ lookahead
+
+
+
+
+
+ Enable or disable encoder granule handling.
+
+MT safe.
+
+
+
+
+
+ a #GstAudioEncoder
+
+
+
+ new state
+
+
+
+
+
+ Configure output caps on the srcpad of @enc.
+
+ %TRUE on success.
+
+
+
+
+ a #GstAudioEncoder
+
+
+
+ #GstCaps
+
+
+
+
+
+ Enable or disable encoder perfect output timestamp preference.
+
+MT safe.
+
+
+
+
+
+ a #GstAudioEncoder
+
+
+
+ new state
+
+
+
+
+
+ Configures encoder audio jitter tolerance threshold.
+
+MT safe.
+
+
+
+
+
+ a #GstAudioEncoder
+
+
+
+ new tolerance
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Subclasses can override any of the available virtual methods or not, as
+needed. At minimum @set_format and @handle_frame needs to be overridden.
+
+ The parent class structure
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ %TRUE if the negotiation succeeded, else %FALSE.
+
+
+
+
+ a #GstAudioEncoder
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ #GstAudioFilter is a #GstBaseTransform<!-- -->-derived base class for simple audio
+filters, ie. those that output the same format that they get as input.
+
+#GstAudioFilter will parse the input format for you (with error checking)
+before calling your setup function. Also, elements deriving from
+#GstAudioFilter may use gst_audio_filter_class_add_pad_templates() from
+their class_init function to easily configure the set of caps/formats that
+the element is able to handle.
+
+Derived classes should override the #GstAudioFilterClass.setup() and
+#GstBaseTransformClass.transform_ip() and/or
+#GstBaseTransformClass.transform()
+virtual functions in their class_init function.
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ In addition to the @setup virtual function, you should also override the
+GstBaseTransform::transform and/or GstBaseTransform::transform_ip virtual
+function.
+
+ parent class
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Convenience function to add pad templates to this element class, with
+@allowed_caps as the caps that can be handled.
+
+This function is usually used from within a GObject class_init function.
+
+
+
+
+
+ an #GstAudioFilterClass
+
+
+
+ what formats the filter can handle, as #GstCaps
+
+
+
+
+
+
+ Extra audio flags
+
+ no valid flag
+
+
+ the position array explicitly
+ contains unpositioned channels.
+
+
+
+ Enum value describing the most common audio formats.
+
+ unknown or unset audio format
+
+
+ encoded audio format
+
+
+ 8 bits in 8 bits, signed
+
+
+ 8 bits in 8 bits, unsigned
+
+
+ 16 bits in 16 bits, signed, little endian
+
+
+ 16 bits in 16 bits, signed, big endian
+
+
+ 16 bits in 16 bits, unsigned, little endian
+
+
+ 16 bits in 16 bits, unsigned, big endian
+
+
+ 24 bits in 32 bits, signed, little endian
+
+
+ 24 bits in 32 bits, signed, big endian
+
+
+ 24 bits in 32 bits, unsigned, little endian
+
+
+ 24 bits in 32 bits, unsigned, big endian
+
+
+ 32 bits in 32 bits, signed, little endian
+
+
+ 32 bits in 32 bits, signed, big endian
+
+
+ 32 bits in 32 bits, unsigned, little endian
+
+
+ 32 bits in 32 bits, unsigned, big endian
+
+
+ 24 bits in 24 bits, signed, little endian
+
+
+ 24 bits in 24 bits, signed, big endian
+
+
+ 24 bits in 24 bits, unsigned, little endian
+
+
+ 24 bits in 24 bits, unsigned, big endian
+
+
+ 20 bits in 24 bits, signed, little endian
+
+
+ 20 bits in 24 bits, signed, big endian
+
+
+ 20 bits in 24 bits, unsigned, little endian
+
+
+ 20 bits in 24 bits, unsigned, big endian
+
+
+ 18 bits in 24 bits, signed, little endian
+
+
+ 18 bits in 24 bits, signed, big endian
+
+
+ 18 bits in 24 bits, unsigned, little endian
+
+
+ 18 bits in 24 bits, unsigned, big endian
+
+
+ 32-bit floating point samples, little endian
+
+
+ 32-bit floating point samples, big endian
+
+
+ 64-bit floating point samples, little endian
+
+
+ 64-bit floating point samples, big endian
+
+
+ 16 bits in 16 bits, signed, native endianness
+
+
+ 16 bits in 16 bits, unsigned, native endianness
+
+
+ 24 bits in 32 bits, signed, native endianness
+
+
+ 24 bits in 32 bits, unsigned, native endianness
+
+
+ 32 bits in 32 bits, signed, native endianness
+
+
+ 32 bits in 32 bits, unsigned, native endianness
+
+
+ 24 bits in 24 bits, signed, native endianness
+
+
+ 24 bits in 24 bits, unsigned, native endianness
+
+
+ 20 bits in 24 bits, signed, native endianness
+
+
+ 20 bits in 24 bits, unsigned, native endianness
+
+
+ 18 bits in 24 bits, signed, native endianness
+
+
+ 18 bits in 24 bits, unsigned, native endianness
+
+
+ 32-bit floating point samples, native endianness
+
+
+ 64-bit floating point samples, native endianness
+
+
+ Construct a #GstAudioFormat with given parameters.
+
+ a #GstAudioFormat or GST_AUDIO_FORMAT_UNKNOWN when no audio format
+exists with the given parameters.
+
+
+
+
+ signed or unsigned format
+
+
+
+ G_LITTLE_ENDIAN or G_BIG_ENDIAN
+
+
+
+ amount of bits used per sample
+
+
+
+ amount of used bits in @width
+
+
+
+
+
+ Fill @length bytes in @dest with silence samples for @info.
+
+
+
+
+
+ a #GstAudioFormatInfo
+
+
+
+ a destination
+ to fill
+
+
+
+
+
+ the length to fill
+
+
+
+
+
+ Convert the @format string to its #GstAudioFormat.
+
+ the #GstAudioFormat for @format or GST_AUDIO_FORMAT_UNKNOWN when the
+string is not a known format.
+
+
+
+
+ a format string
+
+
+
+
+
+ Get the #GstAudioFormatInfo for @format
+
+ The #GstAudioFormatInfo for @format.
+
+
+
+
+ a #GstAudioFormat
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ The different audio flags that a format info can have.
+
+ integer samples
+
+
+ float samples
+
+
+ signed samples
+
+
+ complex layout
+
+
+ the format can be used in
+#GstAudioFormatUnpack and #GstAudioFormatPack functions
+
+
+
+ Information for an audio format.
+
+ #GstAudioFormat
+
+
+
+ string representation of the format
+
+
+
+ user readable description of the format
+
+
+
+ #GstAudioFormatFlags
+
+
+
+ the endianness
+
+
+
+ amount of bits used for one sample
+
+
+
+ amount of valid bits in @width
+
+
+
+ @width/8 bytes with 1 silent sample
+
+
+
+
+
+ the format of the unpacked samples
+
+
+
+ function to unpack samples
+
+
+
+ function to pack samples
+
+
+
+
+
+
+
+
+
+ Packs @length samples from @src to the data array in format @info.
+The samples from source have each channel interleaved
+and will be packed into @data.
+
+
+
+
+
+ a #GstAudioFormatInfo
+
+
+
+
+
+
+ a source array
+
+
+
+
+
+ pointer to the destination
+ data
+
+
+
+
+
+ the amount of samples to pack.
+
+
+
+
+
+ Unpacks @length samples from the given data of format @info.
+The samples will be unpacked into @dest which each channel
+interleaved. @dest should at least be big enough to hold @length *
+channels * size(unpack_format) bytes.
+
+
+
+
+
+ a #GstAudioFormatInfo
+
+
+
+
+
+
+ a destination array
+
+
+
+
+
+ pointer to the audio data
+
+
+
+
+
+ the amount of samples to unpack.
+
+
+
+
+
+ Information describing audio properties. This information can be filled
+in from GstCaps with gst_audio_info_from_caps().
+
+Use the provided macros to access the info in this structure.
+
+ the format info of the audio
+
+
+
+ additional audio flags
+
+
+
+ audio layout
+
+
+
+ the audio sample rate
+
+
+
+ the number of channels
+
+
+
+ the number of bytes for one frame, this is the size of one
+ sample * @channels
+
+
+
+ the positions for each channel
+
+
+
+
+
+
+
+
+
+
+ Allocate a new #GstAudioInfo that is also initialized with
+gst_audio_info_init().
+
+ a new #GstAudioInfo. free with gst_audio_info_free().
+
+
+
+
+ Converts among various #GstFormat types. This function handles
+GST_FORMAT_BYTES, GST_FORMAT_TIME, and GST_FORMAT_DEFAULT. For
+raw audio, GST_FORMAT_DEFAULT corresponds to audio frames. This
+function can be used to handle pad queries of the type GST_QUERY_CONVERT.
+
+ TRUE if the conversion was successful.
+
+
+
+
+ a #GstAudioInfo
+
+
+
+ #GstFormat of the @src_val
+
+
+
+ value to convert
+
+
+
+ #GstFormat of the @dest_val
+
+
+
+ pointer to destination value
+
+
+
+
+
+ Copy a GstAudioInfo structure.
+
+ a new #GstAudioInfo. free with gst_audio_info_free.
+
+
+
+
+ a #GstAudioInfo
+
+
+
+
+
+ Free a GstAudioInfo structure previously allocated with gst_audio_info_new()
+or gst_audio_info_copy().
+
+
+
+
+
+ a #GstAudioInfo
+
+
+
+
+
+ Parse @caps and update @info.
+
+ TRUE if @caps could be parsed
+
+
+
+
+ a #GstAudioInfo
+
+
+
+ a #GstCaps
+
+
+
+
+
+ Initialize @info with default values.
+
+
+
+
+
+ a #GstAudioInfo
+
+
+
+
+
+ Compares two #GstAudioInfo and returns whether they are equal or not
+
+ %TRUE if @info and @other are equal, else %FALSE.
+
+
+
+
+ a #GstAudioInfo
+
+
+
+ a #GstAudioInfo
+
+
+
+
+
+ Set the default info for the audio info of @format and @rate and @channels.
+
+Note: This initializes @info first, no values are preserved.
+
+
+
+
+
+ a #GstAudioInfo
+
+
+
+ the format
+
+
+
+ the samplerate
+
+
+
+ the number of channels
+
+
+
+ the channel positions
+
+
+
+
+
+
+
+ Convert the values of @info into a #GstCaps.
+
+ the new #GstCaps containing the
+ info of @info.
+
+
+
+
+ a #GstAudioInfo
+
+
+
+
+
+
+ Layout of the audio samples for the different channels.
+
+ interleaved audio
+
+
+ non-interleaved audio
+
+
+
+ Set of available noise shaping methods
+
+ No noise shaping (default)
+
+
+ Error feedback
+
+
+ Simple 2-pole noise shaping
+
+
+ Medium 5-pole noise shaping
+
+
+ High 8-pole noise shaping
+
+
+
+ The different flags that can be used when packing and unpacking.
+
+ No flag
+
+
+ When the source has a smaller depth
+ than the target format, set the least significant bits of the target
+ to 0. This is likely sightly faster but less accurate. When this flag
+ is not specified, the most significant bits of the source are duplicated
+ in the least significant bits of the destination.
+
+
+
+
+ Free a #GstAudioQuantize.
+
+
+
+
+
+ a #GstAudioQuantize
+
+
+
+
+
+ Reset @quant to the state is was when created, clearing any
+history it might have.
+
+
+
+
+
+ a #GstAudioQuantize
+
+
+
+
+
+ Perform quantization on @samples in @in and write the result to @out.
+
+In case the samples are interleaved, @in and @out must point to an
+array with a single element pointing to a block of interleaved samples.
+
+If non-interleaved samples are used, @in and @out must point to an
+array with pointers to memory blocks, one for each channel.
+
+@in and @out may point to the same memory location, in which case samples will be
+modified in-place.
+
+
+
+
+
+ a #GstAudioQuantize
+
+
+
+ input samples
+
+
+
+ output samples
+
+
+
+ number of samples
+
+
+
+
+
+ Create a new quantizer object with the given parameters.
+
+Output samples will be quantized to a multiple of @quantizer. Better
+performance is achieved when @quantizer is a power of 2.
+
+Dithering and noise-shaping can be performed during quantization with
+the @dither and @ns parameters.
+
+ a new #GstAudioQuantize. Free with gst_audio_quantize_free().
+
+
+
+
+ a #GstAudioDitherMethod
+
+
+
+ a #GstAudioNoiseShapingMethod
+
+
+
+ #GstAudioQuantizeFlags
+
+
+
+ the #GstAudioFormat of the samples
+
+
+
+ the amount of channels in the samples
+
+
+
+ the quantizer to use
+
+
+
+
+
+
+ Extra flags that can be passed to gst_audio_quantize_new()
+
+ no flags
+
+
+ samples are non-interleaved
+
+
+
+ #GstAudioResampler is a structure which holds the information
+required to perform various kinds of resampling filtering.
+
+ Free a previously allocated #GstAudioResampler @resampler.
+
+
+
+
+
+ a #GstAudioResampler
+
+
+
+
+
+ Get the number of input frames that would currently be needed
+to produce @out_frames from @resampler.
+
+ The number of input frames needed for producing
+@out_frames of data from @resampler.
+
+
+
+
+ a #GstAudioResampler
+
+
+
+ number of input frames
+
+
+
+
+
+ Get the maximum number of input samples that the resampler would
+need before producing output.
+
+ the latency of @resampler as expressed in the number of
+frames.
+
+
+
+
+ a #GstAudioResampler
+
+
+
+
+
+ Get the number of output frames that would be currently available when
+@in_frames are given to @resampler.
+
+ The number of frames that would be availabe after giving
+@in_frames as input to @resampler.
+
+
+
+
+ a #GstAudioResampler
+
+
+
+ number of input frames
+
+
+
+
+
+ Perform resampling on @in_frames frames in @in and write @out_frames to @out.
+
+In case the samples are interleaved, @in and @out must point to an
+array with a single element pointing to a block of interleaved samples.
+
+If non-interleaved samples are used, @in and @out must point to an
+array with pointers to memory blocks, one for each channel.
+
+@in may be %NULL, in which case @in_frames of silence samples are pushed
+into the resampler.
+
+This function always produces @out_frames of output and consumes @in_frames of
+input. Use gst_audio_resampler_get_out_frames() and
+gst_audio_resampler_get_in_frames() to make sure @in_frames and @out_frames
+are matching and @in and @out point to enough memory.
+
+
+
+
+
+ a #GstAudioResampler
+
+
+
+ input samples
+
+
+
+ number of input frames
+
+
+
+ output samples
+
+
+
+ number of output frames
+
+
+
+
+
+ Reset @resampler to the state it was when it was first created, discarding
+all sample history.
+
+
+
+
+
+ a #GstAudioResampler
+
+
+
+
+
+ Update the resampler parameters for @resampler. This function should
+not be called concurrently with any other function on @resampler.
+
+When @in_rate or @out_rate is 0, its value is unchanged.
+
+When @options is %NULL, the previously configured options are reused.
+
+ %TRUE if the new parameters could be set
+
+
+
+
+ a #GstAudioResampler
+
+
+
+ new input rate
+
+
+
+ new output rate
+
+
+
+ new options or %NULL
+
+
+
+
+
+ Make a new resampler.
+
+ %TRUE on success
+
+
+
+
+ a #GstAudioResamplerMethod
+
+
+
+ #GstAudioResamplerFlags
+
+
+
+
+
+
+
+
+
+ input rate
+
+
+
+ output rate
+
+
+
+ extra options
+
+
+
+
+
+ Set the parameters for resampling from @in_rate to @out_rate using @method
+for @quality in @options.
+
+
+
+
+
+ a #GstAudioResamplerMethod
+
+
+
+ the quality
+
+
+
+ the input rate
+
+
+
+ the output rate
+
+
+
+ a #GstStructure
+
+
+
+
+
+
+ The different filter interpolation methods.
+
+ no interpolation
+
+
+ linear interpolation of the
+ filter coeficients.
+
+
+ cubic interpolation of the
+ filter coeficients.
+
+
+
+ Select for the filter tables should be set up.
+
+ Use interpolated filter tables. This
+ uses less memory but more CPU and is slightly less accurate but it allows for more
+ efficient variable rate resampling with gst_audio_resampler_update().
+
+
+ Use full filter table. This uses more memory
+ but less CPU.
+
+
+ Automatically choose between interpolated
+ and full filter tables.
+
+
+
+ Different resampler flags.
+
+ no flags
+
+
+ input samples are non-interleaved.
+ an array of blocks of samples, one for each channel, should be passed to the
+ resample function.
+
+
+ output samples are non-interleaved.
+ an array of blocks of samples, one for each channel, should be passed to the
+ resample function.
+
+
+ optimize for dynamic updates of the sample
+ rates with gst_audio_resampler_update(). This will select an interpolating filter
+ when #GST_AUDIO_RESAMPLER_FILTER_MODE_AUTO is configured.
+
+
+
+ Different subsampling and upsampling methods
+
+ Duplicates the samples when
+ upsampling and drops when downsampling
+
+
+ Uses linear interpolation to reconstruct
+ missing samples and averaging to downsample
+
+
+ Uses cubic interpolation
+
+
+ Uses Blackman-Nuttall windowed sinc interpolation
+
+
+ Uses Kaiser windowed sinc interpolation
+
+
+
+ This object is the base class for audio ringbuffers used by the base
+audio source and sink classes.
+
+The ringbuffer abstracts a circular buffer of data. One reader and
+one writer can operate on the data from different threads in a lockfree
+manner. The base class is sufficiently flexible to be used as an
+abstraction for DMA based ringbuffers as well as a pure software
+implementations.
+
+ Print debug info about the buffer sized in @spec to the debug log.
+
+
+
+
+
+ the spec to debug
+
+
+
+
+
+ Print debug info about the parsed caps in @spec to the debug log.
+
+
+
+
+
+ the spec to debug
+
+
+
+
+
+ Parse @caps into @spec.
+
+ TRUE if the caps could be parsed.
+
+
+
+
+ a spec
+
+
+
+ a #GstCaps
+
+
+
+
+
+ Allocate the resources for the ringbuffer. This function fills
+in the data pointer of the ring buffer with a valid #GstBuffer
+to which samples can be written.
+
+ TRUE if the device could be acquired, FALSE on error.
+
+MT safe.
+
+
+
+
+ the #GstAudioRingBuffer to acquire
+
+
+
+ the specs of the buffer
+
+
+
+
+
+ Activate @buf to start or stop pulling data.
+
+MT safe.
+
+ TRUE if the device could be activated in the requested mode,
+FALSE on error.
+
+
+
+
+ the #GstAudioRingBuffer to activate
+
+
+
+ the new mode
+
+
+
+
+
+ Fill the ringbuffer with silence.
+
+MT safe.
+
+
+
+
+
+ the #GstAudioRingBuffer to clear
+
+
+
+
+
+ Close the audio device associated with the ring buffer. The ring buffer
+should already have been released via gst_audio_ring_buffer_release().
+
+ TRUE if the device could be closed, FALSE on error.
+
+MT safe.
+
+
+
+
+ the #GstAudioRingBuffer
+
+
+
+
+
+ Commit @in_samples samples pointed to by @data to the ringbuffer @buf.
+
+@in_samples and @out_samples define the rate conversion to perform on the
+samples in @data. For negative rates, @out_samples must be negative and
+@in_samples positive.
+
+When @out_samples is positive, the first sample will be written at position @sample
+in the ringbuffer. When @out_samples is negative, the last sample will be written to
+@sample in reverse order.
+
+@out_samples does not need to be a multiple of the segment size of the ringbuffer
+although it is recommended for optimal performance.
+
+@accum will hold a temporary accumulator used in rate conversion and should be
+set to 0 when this function is first called. In case the commit operation is
+interrupted, one can resume the processing by passing the previously returned
+@accum value back to this function.
+
+MT safe.
+
+ The number of samples written to the ringbuffer or -1 on error. The
+number of samples written can be less than @out_samples when @buf was interrupted
+with a flush or stop.
+
+
+
+
+ the #GstAudioRingBuffer to commit
+
+
+
+ the sample position of the data
+
+
+
+ the data to commit
+
+
+
+
+
+ the number of samples in the data to commit
+
+
+
+ the number of samples to write to the ringbuffer
+
+
+
+ accumulator for rate conversion.
+
+
+
+
+
+ Get the number of samples queued in the audio device. This is
+usually less than the segment size but can be bigger when the
+implementation uses another internal buffer between the audio
+device.
+
+For playback ringbuffers this is the amount of samples transfered from the
+ringbuffer to the device but still not played.
+
+For capture ringbuffers this is the amount of samples in the device that are
+not yet transfered to the ringbuffer.
+
+ The number of samples queued in the audio device.
+
+MT safe.
+
+
+
+
+ the #GstAudioRingBuffer to query
+
+
+
+
+
+ Open the audio device associated with the ring buffer. Does not perform any
+setup on the device. You must open the device before acquiring the ring
+buffer.
+
+ TRUE if the device could be opened, FALSE on error.
+
+MT safe.
+
+
+
+
+ the #GstAudioRingBuffer
+
+
+
+
+
+ Pause processing samples from the ringbuffer.
+
+ TRUE if the device could be paused, FALSE on error.
+
+MT safe.
+
+
+
+
+ the #GstAudioRingBuffer to pause
+
+
+
+
+
+ Free the resources of the ringbuffer.
+
+ TRUE if the device could be released, FALSE on error.
+
+MT safe.
+
+
+
+
+ the #GstAudioRingBuffer to release
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Start processing samples from the ringbuffer.
+
+ TRUE if the device could be started, FALSE on error.
+
+MT safe.
+
+
+
+
+ the #GstAudioRingBuffer to start
+
+
+
+
+
+ Stop processing samples from the ringbuffer.
+
+ TRUE if the device could be stopped, FALSE on error.
+
+MT safe.
+
+
+
+
+ the #GstAudioRingBuffer to stop
+
+
+
+
+
+ Allocate the resources for the ringbuffer. This function fills
+in the data pointer of the ring buffer with a valid #GstBuffer
+to which samples can be written.
+
+ TRUE if the device could be acquired, FALSE on error.
+
+MT safe.
+
+
+
+
+ the #GstAudioRingBuffer to acquire
+
+
+
+ the specs of the buffer
+
+
+
+
+
+ Activate @buf to start or stop pulling data.
+
+MT safe.
+
+ TRUE if the device could be activated in the requested mode,
+FALSE on error.
+
+
+
+
+ the #GstAudioRingBuffer to activate
+
+
+
+ the new mode
+
+
+
+
+
+ Subclasses should call this function to notify the fact that
+@advance segments are now processed by the device.
+
+MT safe.
+
+
+
+
+
+ the #GstAudioRingBuffer to advance
+
+
+
+ the number of segments written
+
+
+
+
+
+ Clear the given segment of the buffer with silence samples.
+This function is used by subclasses.
+
+MT safe.
+
+
+
+
+
+ the #GstAudioRingBuffer to clear
+
+
+
+ the segment to clear
+
+
+
+
+
+ Fill the ringbuffer with silence.
+
+MT safe.
+
+
+
+
+
+ the #GstAudioRingBuffer to clear
+
+
+
+
+
+ Close the audio device associated with the ring buffer. The ring buffer
+should already have been released via gst_audio_ring_buffer_release().
+
+ TRUE if the device could be closed, FALSE on error.
+
+MT safe.
+
+
+
+
+ the #GstAudioRingBuffer
+
+
+
+
+
+ Commit @in_samples samples pointed to by @data to the ringbuffer @buf.
+
+@in_samples and @out_samples define the rate conversion to perform on the
+samples in @data. For negative rates, @out_samples must be negative and
+@in_samples positive.
+
+When @out_samples is positive, the first sample will be written at position @sample
+in the ringbuffer. When @out_samples is negative, the last sample will be written to
+@sample in reverse order.
+
+@out_samples does not need to be a multiple of the segment size of the ringbuffer
+although it is recommended for optimal performance.
+
+@accum will hold a temporary accumulator used in rate conversion and should be
+set to 0 when this function is first called. In case the commit operation is
+interrupted, one can resume the processing by passing the previously returned
+@accum value back to this function.
+
+MT safe.
+
+ The number of samples written to the ringbuffer or -1 on error. The
+number of samples written can be less than @out_samples when @buf was interrupted
+with a flush or stop.
+
+
+
+
+ the #GstAudioRingBuffer to commit
+
+
+
+ the sample position of the data
+
+
+
+ the data to commit
+
+
+
+
+
+ the number of samples in the data to commit
+
+
+
+ the number of samples to write to the ringbuffer
+
+
+
+ accumulator for rate conversion.
+
+
+
+
+
+ Convert @src_val in @src_fmt to the equivalent value in @dest_fmt. The result
+will be put in @dest_val.
+
+ TRUE if the conversion succeeded.
+
+
+
+
+ the #GstAudioRingBuffer
+
+
+
+ the source format
+
+
+
+ the source value
+
+
+
+ the destination format
+
+
+
+ a location to store the converted value
+
+
+
+
+
+ Get the number of samples queued in the audio device. This is
+usually less than the segment size but can be bigger when the
+implementation uses another internal buffer between the audio
+device.
+
+For playback ringbuffers this is the amount of samples transfered from the
+ringbuffer to the device but still not played.
+
+For capture ringbuffers this is the amount of samples in the device that are
+not yet transfered to the ringbuffer.
+
+ The number of samples queued in the audio device.
+
+MT safe.
+
+
+
+
+ the #GstAudioRingBuffer to query
+
+
+
+
+
+ Checks the status of the device associated with the ring buffer.
+
+ TRUE if the device was open, FALSE if it was closed.
+
+MT safe.
+
+
+
+
+ the #GstAudioRingBuffer
+
+
+
+
+
+ Check if the ringbuffer is acquired and ready to use.
+
+ TRUE if the ringbuffer is acquired, FALSE on error.
+
+MT safe.
+
+
+
+
+ the #GstAudioRingBuffer to check
+
+
+
+
+
+ Check if @buf is activated.
+
+MT safe.
+
+ TRUE if the device is active.
+
+
+
+
+ the #GstAudioRingBuffer
+
+
+
+
+
+ Check if @buf is flushing.
+
+MT safe.
+
+ TRUE if the device is flushing.
+
+
+
+
+ the #GstAudioRingBuffer
+
+
+
+
+
+ Tell the ringbuffer that it is allowed to start playback when
+the ringbuffer is filled with samples.
+
+MT safe.
+
+
+
+
+
+ the #GstAudioRingBuffer
+
+
+
+ the new value
+
+
+
+
+
+ Open the audio device associated with the ring buffer. Does not perform any
+setup on the device. You must open the device before acquiring the ring
+buffer.
+
+ TRUE if the device could be opened, FALSE on error.
+
+MT safe.
+
+
+
+
+ the #GstAudioRingBuffer
+
+
+
+
+
+ Pause processing samples from the ringbuffer.
+
+ TRUE if the device could be paused, FALSE on error.
+
+MT safe.
+
+
+
+
+ the #GstAudioRingBuffer to pause
+
+
+
+
+
+ Returns a pointer to memory where the data from segment @segment
+can be found. This function is mostly used by subclasses.
+
+ FALSE if the buffer is not started.
+
+MT safe.
+
+
+
+
+ the #GstAudioRingBuffer to read from
+
+
+
+ the segment to read
+
+
+
+
+ the pointer to the memory where samples can be read
+
+
+
+
+
+ the number of bytes to read
+
+
+
+
+
+ Read @len samples from the ringbuffer into the memory pointed
+to by @data.
+The first sample should be read from position @sample in
+the ringbuffer.
+
+@len should not be a multiple of the segment size of the ringbuffer
+although it is recommended.
+
+@timestamp will return the timestamp associated with the data returned.
+
+ The number of samples read from the ringbuffer or -1 on
+error.
+
+MT safe.
+
+
+
+
+ the #GstAudioRingBuffer to read from
+
+
+
+ the sample position of the data
+
+
+
+ where the data should be read
+
+
+
+
+
+ the number of samples in data to read
+
+
+
+ where the timestamp is returned
+
+
+
+
+
+ Free the resources of the ringbuffer.
+
+ TRUE if the device could be released, FALSE on error.
+
+MT safe.
+
+
+
+
+ the #GstAudioRingBuffer to release
+
+
+
+
+
+ Get the number of samples that were processed by the ringbuffer
+since it was last started. This does not include the number of samples not
+yet processed (see gst_audio_ring_buffer_delay()).
+
+ The number of samples processed by the ringbuffer.
+
+MT safe.
+
+
+
+
+ the #GstAudioRingBuffer to query
+
+
+
+
+
+ Sets the given callback function on the buffer. This function
+will be called every time a segment has been written to a device.
+
+MT safe.
+
+
+
+
+
+ the #GstAudioRingBuffer to set the callback on
+
+
+
+ the callback to set
+
+
+
+ user data passed to the callback
+
+
+
+
+
+ Sets the given callback function on the buffer. This function
+will be called every time a segment has been written to a device.
+
+MT safe.
+
+
+
+
+
+ the #GstAudioRingBuffer to set the callback on
+
+
+
+ the callback to set
+
+
+
+ user data passed to the callback
+
+
+
+ function to be called when @user_data is no longer needed
+
+
+
+
+
+ Tell the ringbuffer about the device's channel positions. This must
+be called in when the ringbuffer is acquired.
+
+
+
+
+
+ the #GstAudioRingBuffer
+
+
+
+ the device channel positions
+
+
+
+
+
+
+
+ Set the ringbuffer to flushing mode or normal mode.
+
+MT safe.
+
+
+
+
+
+ the #GstAudioRingBuffer to flush
+
+
+
+ the new mode
+
+
+
+
+
+ Make sure that the next sample written to the device is
+accounted for as being the @sample sample written to the
+device. This value will be used in reporting the current
+sample position of the ringbuffer.
+
+This function will also clear the buffer with silence.
+
+MT safe.
+
+
+
+
+
+ the #GstAudioRingBuffer to use
+
+
+
+ the sample number to set
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Start processing samples from the ringbuffer.
+
+ TRUE if the device could be started, FALSE on error.
+
+MT safe.
+
+
+
+
+ the #GstAudioRingBuffer to start
+
+
+
+
+
+ Stop processing samples from the ringbuffer.
+
+ TRUE if the device could be stopped, FALSE on error.
+
+MT safe.
+
+
+
+
+ the #GstAudioRingBuffer to stop
+
+
+
+
+
+
+
+
+ used to signal start/stop/pause/resume actions
+
+
+
+ boolean indicating that the ringbuffer is open
+
+
+
+ boolean indicating that the ringbuffer is acquired
+
+
+
+ data in the ringbuffer
+
+
+
+ size of data in the ringbuffer
+
+
+
+
+
+
+ format and layout of the ringbuffer data
+
+
+
+ number of samples in one segment
+
+
+
+ pointer to memory holding one segment of silence samples
+
+
+
+ state of the buffer
+
+
+
+ readpointer in the ringbuffer
+
+
+
+ segment corresponding to segment 0 (unused)
+
+
+
+ is a reader or writer waiting for a free segment
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ This function is set with gst_audio_ring_buffer_set_callback() and is
+called to fill the memory at @data with @len bytes of samples.
+
+
+
+
+
+ a #GstAudioRingBuffer
+
+
+
+ target to fill
+
+
+
+
+
+ amount to fill
+
+
+
+ user data
+
+
+
+
+
+ The vmethods that subclasses can override to implement the ringbuffer.
+
+ parent class
+
+
+
+
+
+ TRUE if the device could be opened, FALSE on error.
+
+MT safe.
+
+
+
+
+ the #GstAudioRingBuffer
+
+
+
+
+
+
+
+
+ TRUE if the device could be acquired, FALSE on error.
+
+MT safe.
+
+
+
+
+ the #GstAudioRingBuffer to acquire
+
+
+
+ the specs of the buffer
+
+
+
+
+
+
+
+
+ TRUE if the device could be released, FALSE on error.
+
+MT safe.
+
+
+
+
+ the #GstAudioRingBuffer to release
+
+
+
+
+
+
+
+
+ TRUE if the device could be closed, FALSE on error.
+
+MT safe.
+
+
+
+
+ the #GstAudioRingBuffer
+
+
+
+
+
+
+
+
+ TRUE if the device could be started, FALSE on error.
+
+MT safe.
+
+
+
+
+ the #GstAudioRingBuffer to start
+
+
+
+
+
+
+
+
+ TRUE if the device could be paused, FALSE on error.
+
+MT safe.
+
+
+
+
+ the #GstAudioRingBuffer to pause
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ TRUE if the device could be stopped, FALSE on error.
+
+MT safe.
+
+
+
+
+ the #GstAudioRingBuffer to stop
+
+
+
+
+
+
+
+
+ The number of samples queued in the audio device.
+
+MT safe.
+
+
+
+
+ the #GstAudioRingBuffer to query
+
+
+
+
+
+
+
+
+ TRUE if the device could be activated in the requested mode,
+FALSE on error.
+
+
+
+
+ the #GstAudioRingBuffer to activate
+
+
+
+ the new mode
+
+
+
+
+
+
+
+
+ The number of samples written to the ringbuffer or -1 on error. The
+number of samples written can be less than @out_samples when @buf was interrupted
+with a flush or stop.
+
+
+
+
+ the #GstAudioRingBuffer to commit
+
+
+
+ the sample position of the data
+
+
+
+ the data to commit
+
+
+
+
+
+ the number of samples in the data to commit
+
+
+
+ the number of samples to write to the ringbuffer
+
+
+
+ accumulator for rate conversion.
+
+
+
+
+
+
+
+
+
+
+
+
+ the #GstAudioRingBuffer to clear
+
+
+
+
+
+
+
+
+
+
+
+
+ The format of the samples in the ringbuffer.
+
+ samples in linear or float
+
+
+ samples in mulaw
+
+
+ samples in alaw
+
+
+ samples in ima adpcm
+
+
+ samples in mpeg audio (but not AAC) format
+
+
+ samples in gsm format
+
+
+ samples in IEC958 frames (e.g. AC3)
+
+
+ samples in AC3 format
+
+
+ samples in EAC3 format
+
+
+ samples in DTS format
+
+
+ samples in MPEG-2 AAC ADTS format
+
+
+ samples in MPEG-4 AAC ADTS format
+
+
+ samples in MPEG-2 AAC raw format (Since 1.12)
+
+
+ samples in MPEG-4 AAC raw format (Since 1.12)
+
+
+ samples in FLAC format (Since 1.12)
+
+
+
+ The structure containing the format specification of the ringbuffer.
+
+ The caps that generated the Spec.
+
+
+
+ the sample type
+
+
+
+ the #GstAudioInfo
+
+
+
+ the latency in microseconds
+
+
+
+ the total buffer size in microseconds
+
+
+
+ the size of one segment in bytes
+
+
+
+ the total number of segments
+
+
+
+ number of segments queued in the lower level device,
+ defaults to segtotal
+
+
+
+
+
+
+
+
+
+ The state of the ringbuffer.
+
+ The ringbuffer is stopped
+
+
+ The ringbuffer is paused
+
+
+ The ringbuffer is started
+
+
+ The ringbuffer has encountered an
+ error after it has been started, e.g. because the device was
+ disconnected (Since 1.2)
+
+
+
+ This is the most simple base class for audio sinks that only requires
+subclasses to implement a set of simple functions:
+
+* `open()` :Open the device.
+
+* `prepare()` :Configure the device with the specified format.
+
+* `write()` :Write samples to the device.
+
+* `reset()` :Unblock writes and flush the device.
+
+* `delay()` :Get the number of samples written but not yet played
+by the device.
+
+* `unprepare()` :Undo operations done by prepare.
+
+* `close()` :Close the device.
+
+All scheduling of samples and timestamps is done in this base class
+together with #GstAudioBaseSink using a default implementation of a
+#GstAudioRingBuffer that uses threads.
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ #GstAudioSink class. Override the vmethods to implement functionality.
+
+ the parent class structure.
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ This is the most simple base class for audio sources that only requires
+subclasses to implement a set of simple functions:
+
+* `open()` :Open the device.
+* `prepare()` :Configure the device with the specified format.
+* `read()` :Read samples from the device.
+* `reset()` :Unblock reads and flush the device.
+* `delay()` :Get the number of samples in the device but not yet read.
+* `unprepare()` :Undo operations done by prepare.
+* `close()` :Close the device.
+
+All scheduling of samples and timestamps is done in this base class
+together with #GstAudioBaseSrc using a default implementation of a
+#GstAudioRingBuffer that uses threads.
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ #GstAudioSrc class. Override the vmethod to implement
+functionality.
+
+ the parent class.
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ #GstAudioStreamAlign provides a helper object that helps tracking audio
+stream alignment and discontinuities, and detects discontinuities if
+possible.
+
+See gst_audio_stream_align_new() for a description of its parameters and
+gst_audio_stream_align_process() for the details of the processing.
+
+ Allocate a new #GstAudioStreamAlign with the given configuration. All
+processing happens according to sample rate @rate, until
+gst_audio_discont_wait_set_rate() is called with a new @rate.
+A negative rate can be used for reverse playback.
+
+@alignment_threshold gives the tolerance in nanoseconds after which a
+timestamp difference is considered a discontinuity. Once detected,
+@discont_wait nanoseconds have to pass without going below the threshold
+again until the output buffer is marked as a discontinuity. These can later
+be re-configured with gst_audio_stream_align_set_alignment_threshold() and
+gst_audio_stream_align_set_discont_wait().
+
+ a new #GstAudioStreamAlign. free with gst_audio_stream_align_free().
+
+
+
+
+ a sample rate
+
+
+
+ a alignment threshold in nanoseconds
+
+
+
+ discont wait in nanoseconds
+
+
+
+
+
+ Copy a GstAudioStreamAlign structure.
+
+ a new #GstAudioStreamAlign. free with gst_audio_stream_align_free.
+
+
+
+
+ a #GstAudioStreamAlign
+
+
+
+
+
+ Free a GstAudioStreamAlign structure previously allocated with gst_audio_stream_align_new()
+or gst_audio_stream_align_copy().
+
+
+
+
+
+ a #GstAudioStreamAlign
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Returns the number of samples that were processed since the last
+discontinuity was detected.
+
+ The number of samples processed since the last discontinuity.
+
+
+
+
+ a #GstAudioStreamAlign
+
+
+
+
+
+ Timestamp that was passed when a discontinuity was detected, i.e. the first
+timestamp after the discontinuity.
+
+ The last timestamp at when a discontinuity was detected
+
+
+
+
+ a #GstAudioStreamAlign
+
+
+
+
+
+ Marks the next buffer as discontinuous and resets timestamp tracking.
+
+
+
+
+
+ a #GstAudioStreamAlign
+
+
+
+
+
+ Processes data with @timestamp and @n_samples, and returns the output
+timestamp, duration and sample position together with a boolean to signal
+whether a discontinuity was detected or not. All non-discontinuous data
+will have perfect timestamps and durations.
+
+A discontinuity is detected once the difference between the actual
+timestamp and the timestamp calculated from the sample count since the last
+discontinuity differs by more than the alignment threshold for a duration
+longer than discont wait.
+
+Note: In reverse playback, every buffer is considered discontinuous in the
+context of buffer flags because the last sample of the previous buffer is
+discontinuous with the first sample of the current one. However for this
+function they are only considered discontinuous in reverse playback if the
+first sample of the previous buffer is discontinuous with the last sample
+of the current one.
+
+ %TRUE if a discontinuity was detected, %FALSE otherwise.
+
+
+
+
+ a #GstAudioStreamAlign
+
+
+
+ if this data is considered to be discontinuous
+
+
+
+ a #GstClockTime of the start of the data
+
+
+
+ number of samples to process
+
+
+
+ output timestamp of the data
+
+
+
+ output duration of the data
+
+
+
+ output sample position of the start of the data
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ This metadata stays relevant as long as channels are unchanged.
+
+
+
+ This metadata stays relevant as long as sample rate is unchanged.
+
+
+
+ This metadata is relevant for audio streams.
+
+
+
+ This interface is implemented by elements that provide a stream volume. Examples for
+such elements are #volume and #playbin.
+
+Applications can use this interface to get or set the current stream volume. For this
+the "volume" #GObject property can be used or the helper functions gst_stream_volume_set_volume()
+and gst_stream_volume_get_volume(). This volume is always a linear factor, i.e. 0.0 is muted
+1.0 is 100%. For showing the volume in a GUI it might make sense to convert it to
+a different format by using gst_stream_volume_convert_volume(). Volume sliders should usually
+use a cubic volume.
+
+Separate from the volume the stream can also be muted by the "mute" #GObject property or
+gst_stream_volume_set_mute() and gst_stream_volume_get_mute().
+
+Elements that provide some kind of stream volume should implement the "volume" and
+"mute" #GObject properties and handle setting and getting of them properly.
+The volume property is defined to be a linear volume factor.
+
+
+ the converted volume
+
+
+
+
+ #GstStreamVolumeFormat to convert from
+
+
+
+ #GstStreamVolumeFormat to convert to
+
+
+
+ Volume in @from format that should be converted
+
+
+
+
+
+
+ Returns %TRUE if the stream is muted
+
+
+
+
+ #GstStreamVolume that should be used
+
+
+
+
+
+
+ The current stream volume as linear factor
+
+
+
+
+ #GstStreamVolume that should be used
+
+
+
+ #GstStreamVolumeFormat which should be returned
+
+
+
+
+
+
+
+
+
+
+ #GstStreamVolume that should be used
+
+
+
+ Mute state that should be set
+
+
+
+
+
+
+
+
+
+
+ #GstStreamVolume that should be used
+
+
+
+ #GstStreamVolumeFormat of @val
+
+
+
+ Linear volume factor that should be set
+
+
+
+
+
+
+
+
+
+
+
+
+ Different representations of a stream volume. gst_stream_volume_convert_volume()
+allows to convert between the different representations.
+
+Formulas to convert from a linear to a cubic or dB volume are
+cbrt(val) and 20 * log10 (val).
+
+ Linear scale factor, 1.0 = 100%
+
+
+ Cubic volume scale
+
+
+ Logarithmic volume scale (dB, amplitude not power)
+
+
+
+
+
+
+
+
+ Clip the buffer to the given %GstSegment.
+
+After calling this function the caller does not own a reference to
+@buffer anymore.
+
+ %NULL if the buffer is completely outside the configured segment,
+otherwise the clipped buffer is returned.
+
+If the buffer has no timestamp, it is assumed to be inside the segment and
+is not clipped
+
+
+
+
+ The buffer to clip.
+
+
+
+ Segment in %GST_FORMAT_TIME or %GST_FORMAT_DEFAULT to which
+ the buffer should be clipped.
+
+
+
+ sample rate.
+
+
+
+ size of one audio frame in bytes. This is the size of one sample *
+number of channels.
+
+
+
+
+
+ Reorders @buffer from the channel positions @from to the channel
+positions @to. @from and @to must contain the same number of
+positions and the same positions, only in a different order.
+@buffer must be writable.
+
+ %TRUE if the reordering was possible.
+
+
+
+
+ The buffer to reorder.
+
+
+
+ The %GstAudioFormat of the buffer.
+
+
+
+ The number of channels.
+
+
+
+ The channel positions in the buffer.
+
+
+
+
+
+ The channel positions to convert to.
+
+
+
+
+
+
+
+ Get the fallback channel-mask for the given number of channels.
+
+This function returns a reasonable fallback channel-mask and should be
+called as a last resort when the specific channel map is unknown.
+
+ a fallback channel-mask for @channels or 0 when there is no
+mask and mono.
+
+
+
+
+ the number of channels
+
+
+
+
+
+ Create a new channel mixer object for the given parameters.
+
+ a new #GstAudioChannelMixer object, or %NULL if @format isn't supported.
+ Free with gst_audio_channel_mixer_free() after usage.
+
+
+
+
+ #GstAudioChannelMixerFlags
+
+
+
+
+
+
+ number of input channels
+
+
+
+ positions of input channels
+
+
+
+ number of output channels
+
+
+
+ positions of output channels
+
+
+
+
+
+ Create a new channel mixer object for the given parameters.
+
+ a new #GstAudioChannelMixer object, or %NULL if @format isn't supported,
+ @matrix is invalid, or @matrix is %NULL and @in_channels != @out_channels.
+ Free with gst_audio_channel_mixer_free() after usage.
+
+
+
+
+ #GstAudioChannelMixerFlags
+
+
+
+
+
+
+ number of input channels
+
+
+
+ number of output channels
+
+
+
+ channel conversion matrix, m[@in_channels][@out_channels].
+ If identity matrix, passthrough applies. If %NULL, a (potentially truncated)
+ identity matrix is generated.
+
+
+
+
+
+ Convert the @channels present in @channel_mask to a @position array
+(which should have at least @channels entries ensured by caller).
+If @channel_mask is set to 0, it is considered as 'not present' for purpose
+of conversion.
+A partially valid @channel_mask with less bits set than the number
+of channels is considered valid.
+
+ %TRUE if channel and channel mask are valid and could be converted
+
+
+
+
+ The number of channels
+
+
+
+ The input channel_mask
+
+
+
+ The
+ %GstAudioChannelPosition<!-- -->s
+
+
+
+
+
+
+
+ Convert the @position array of @channels channels to a bitmask.
+
+If @force_order is %TRUE it additionally checks if the channels are
+in the order required by GStreamer.
+
+ %TRUE if the channel positions are valid and could be converted.
+
+
+
+
+ The %GstAudioChannelPositions
+
+
+
+
+
+ The number of channels.
+
+
+
+ Only consider the GStreamer channel order.
+
+
+
+ the output channel mask
+
+
+
+
+
+ Converts @position to a human-readable string representation for
+debugging purposes.
+
+ a newly allocated string representing
+@position
+
+Since 1.10
+
+
+
+
+ The %GstAudioChannelPositions
+ to convert.
+
+
+
+
+
+ The number of channels.
+
+
+
+
+
+ Reorders the channel positions in @position from any order to
+the GStreamer channel order.
+
+ %TRUE if the channel positions are valid and reordering
+was successful.
+
+
+
+
+ The channel positions to
+ reorder to.
+
+
+
+
+
+ The number of channels.
+
+
+
+
+
+ Checks if @position contains valid channel positions for
+@channels channels. If @force_order is %TRUE it additionally
+checks if the channels are in the order required by GStreamer.
+
+ %TRUE if the channel positions are valid.
+
+
+
+
+ The %GstAudioChannelPositions
+ to check.
+
+
+
+
+
+ The number of channels.
+
+
+
+ Only consider the GStreamer channel order.
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Construct a #GstAudioFormat with given parameters.
+
+ a #GstAudioFormat or GST_AUDIO_FORMAT_UNKNOWN when no audio format
+exists with the given parameters.
+
+
+
+
+ signed or unsigned format
+
+
+
+ G_LITTLE_ENDIAN or G_BIG_ENDIAN
+
+
+
+ amount of bits used per sample
+
+
+
+ amount of used bits in @width
+
+
+
+
+
+ Fill @length bytes in @dest with silence samples for @info.
+
+
+
+
+
+ a #GstAudioFormatInfo
+
+
+
+ a destination
+ to fill
+
+
+
+
+
+ the length to fill
+
+
+
+
+
+ Convert the @format string to its #GstAudioFormat.
+
+ the #GstAudioFormat for @format or GST_AUDIO_FORMAT_UNKNOWN when the
+string is not a known format.
+
+
+
+
+ a format string
+
+
+
+
+
+ Get the #GstAudioFormatInfo for @format
+
+ The #GstAudioFormatInfo for @format.
+
+
+
+
+ a #GstAudioFormat
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Returns a reorder map for @from to @to that can be used in
+custom channel reordering code, e.g. to convert from or to the
+GStreamer channel order. @from and @to must contain the same
+number of positions and the same positions, only in a
+different order.
+
+The resulting @reorder_map can be used for reordering by assigning
+channel i of the input to channel reorder_map[i] of the output.
+
+ %TRUE if the channel positions are valid and reordering
+is possible.
+
+
+
+
+ The number of channels.
+
+
+
+ The channel positions to reorder from.
+
+
+
+
+
+ The channel positions to reorder to.
+
+
+
+
+
+ Pointer to the reorder map.
+
+
+
+
+
+
+
+ Calculated the size of the buffer expected by gst_audio_iec61937_payload() for
+payloading type from @spec.
+
+ the size or 0 if the given @type is not supported or cannot be
+payloaded.
+
+
+
+
+ the ringbufer spec
+
+
+
+
+
+ Payloads @src in the form specified by IEC 61937 for the type from @spec and
+stores the result in @dst. @src must contain exactly one frame of data and
+the frame is not checked for errors.
+
+ transfer-full: %TRUE if the payloading was successful, %FALSE
+otherwise.
+
+
+
+
+ a buffer containing the data to payload
+
+
+
+
+
+ size of @src in bytes
+
+
+
+ the destination buffer to store the
+ payloaded contents in. Should not overlap with @src
+
+
+
+
+
+ size of @dst in bytes
+
+
+
+ the ringbufer spec for @src
+
+
+
+ the expected byte order of the payloaded data
+
+
+
+
+
+ Create a new quantizer object with the given parameters.
+
+Output samples will be quantized to a multiple of @quantizer. Better
+performance is achieved when @quantizer is a power of 2.
+
+Dithering and noise-shaping can be performed during quantization with
+the @dither and @ns parameters.
+
+ a new #GstAudioQuantize. Free with gst_audio_quantize_free().
+
+
+
+
+ a #GstAudioDitherMethod
+
+
+
+ a #GstAudioNoiseShapingMethod
+
+
+
+ #GstAudioQuantizeFlags
+
+
+
+ the #GstAudioFormat of the samples
+
+
+
+ the amount of channels in the samples
+
+
+
+ the quantizer to use
+
+
+
+
+
+ Reorders @data from the channel positions @from to the channel
+positions @to. @from and @to must contain the same number of
+positions and the same positions, only in a different order.
+
+ %TRUE if the reordering was possible.
+
+
+
+
+ The pointer to
+ the memory.
+
+
+
+
+
+ The size of the memory.
+
+
+
+ The %GstAudioFormat of the buffer.
+
+
+
+ The number of channels.
+
+
+
+ The channel positions in the buffer.
+
+
+
+
+
+ The channel positions to convert to.
+
+
+
+
+
+
+
+ Make a new resampler.
+
+ %TRUE on success
+
+
+
+
+ a #GstAudioResamplerMethod
+
+
+
+ #GstAudioResamplerFlags
+
+
+
+
+
+
+
+
+
+ input rate
+
+
+
+ output rate
+
+
+
+ extra options
+
+
+
+
+
+ Set the parameters for resampling from @in_rate to @out_rate using @method
+for @quality in @options.
+
+
+
+
+
+ a #GstAudioResamplerMethod
+
+
+
+ the quality
+
+
+
+ the input rate
+
+
+
+ the output rate
+
+
+
+ a #GstStructure
+
+
+
+
+
+ Attaches #GstAudioClippingMeta metadata to @buffer with the given parameters.
+
+ the #GstAudioClippingMeta on @buffer.
+
+
+
+
+ a #GstBuffer
+
+
+
+ GstFormat of @start and @stop, GST_FORMAT_DEFAULT is samples
+
+
+
+ Amount of audio to clip from start of buffer
+
+
+
+ Amount of to clip from end of buffer
+
+
+
+
+
+ Attaches #GstAudioDownmixMeta metadata to @buffer with the given parameters.
+
+@matrix is an two-dimensional array of @to_channels times @from_channels
+coefficients, i.e. the i-th output channels is constructed by multiplicating
+the input channels with the coefficients in @matrix[i] and taking the sum
+of the results.
+
+ the #GstAudioDownmixMeta on @buffer.
+
+
+
+
+ a #GstBuffer
+
+
+
+ the channel positions
+ of the source
+
+
+
+
+
+ The number of channels of the source
+
+
+
+ the channel positions of
+ the destination
+
+
+
+
+
+ The number of channels of the destination
+
+
+
+ The matrix coefficients.
+
+
+
+
+
+ Find the #GstAudioDownmixMeta on @buffer for the given destination
+channel positions.
+
+ the #GstAudioDownmixMeta on @buffer.
+
+
+
+
+ a #GstBuffer
+
+
+
+ the channel positions of
+ the destination
+
+
+
+
+
+ The number of channels of the destination
+
+
+
+
+
+
+ the converted volume
+
+
+
+
+ #GstStreamVolumeFormat to convert from
+
+
+
+ #GstStreamVolumeFormat to convert to
+
+
+
+ Volume in @from format that should be converted
+
+
+
+
+
+
diff --git a/girs/GstBase-1.0.gir b/girs/GstBase-1.0.gir
new file mode 100644
index 0000000000..08ee2c704c
--- /dev/null
+++ b/girs/GstBase-1.0.gir
@@ -0,0 +1,12253 @@
+
+
+
+
+
+
+
+
+
+
+
+ This class is for elements that receive buffers in an undesired size.
+While for example raw video contains one image per buffer, the same is not
+true for a lot of other formats, especially those that come directly from
+a file. So if you have undefined buffer sizes and require a specific size,
+this object is for you.
+
+An adapter is created with gst_adapter_new(). It can be freed again with
+g_object_unref().
+
+The theory of operation is like this: All buffers received are put
+into the adapter using gst_adapter_push() and the data is then read back
+in chunks of the desired size using gst_adapter_map()/gst_adapter_unmap()
+and/or gst_adapter_copy(). After the data has been processed, it is freed
+using gst_adapter_unmap().
+
+Other methods such as gst_adapter_take() and gst_adapter_take_buffer()
+combine gst_adapter_map() and gst_adapter_unmap() in one method and are
+potentially more convenient for some use cases.
+
+For example, a sink pad's chain function that needs to pass data to a library
+in 512-byte chunks could be implemented like this:
+|[<!-- language="C" -->
+static GstFlowReturn
+sink_pad_chain (GstPad *pad, GstObject *parent, GstBuffer *buffer)
+{
+ MyElement *this;
+ GstAdapter *adapter;
+ GstFlowReturn ret = GST_FLOW_OK;
+
+ this = MY_ELEMENT (parent);
+
+ adapter = this->adapter;
+
+ // put buffer into adapter
+ gst_adapter_push (adapter, buffer);
+
+ // while we can read out 512 bytes, process them
+ while (gst_adapter_available (adapter) >= 512 && ret == GST_FLOW_OK) {
+ const guint8 *data = gst_adapter_map (adapter, 512);
+ // use flowreturn as an error value
+ ret = my_library_foo (data);
+ gst_adapter_unmap (adapter);
+ gst_adapter_flush (adapter, 512);
+ }
+ return ret;
+}
+]|
+
+For another example, a simple element inside GStreamer that uses #GstAdapter
+is the libvisual element.
+
+An element using #GstAdapter in its sink pad chain function should ensure that
+when the FLUSH_STOP event is received, that any queued data is cleared using
+gst_adapter_clear(). Data should also be cleared or processed on EOS and
+when changing state from %GST_STATE_PAUSED to %GST_STATE_READY.
+
+Also check the GST_BUFFER_FLAG_DISCONT flag on the buffer. Some elements might
+need to clear the adapter after a discontinuity.
+
+The adapter will keep track of the timestamps of the buffers
+that were pushed. The last seen timestamp before the current position
+can be queried with gst_adapter_prev_pts(). This function can
+optionally return the number of bytes between the start of the buffer that
+carried the timestamp and the current adapter position. The distance is
+useful when dealing with, for example, raw audio samples because it allows
+you to calculate the timestamp of the current adapter position by using the
+last seen timestamp and the amount of bytes since. Additionally, the
+gst_adapter_prev_pts_at_offset() can be used to determine the last
+seen timestamp at a particular offset in the adapter.
+
+The adapter will also keep track of the offset of the buffers
+(#GST_BUFFER_OFFSET) that were pushed. The last seen offset before the
+current position can be queried with gst_adapter_prev_offset(). This function
+can optionally return the number of bytes between the start of the buffer
+that carried the offset and the current adapter position.
+
+Additionally the adapter also keeps track of the PTS, DTS and buffer offset
+at the last discontinuity, which can be retrieved with
+gst_adapter_pts_at_discont(), gst_adapter_dts_at_discont() and
+gst_adapter_offset_at_discont(). The number of bytes that were consumed
+since then can be queried with gst_adapter_distance_from_discont().
+
+A last thing to note is that while #GstAdapter is pretty optimized,
+merging buffers still might be an operation that requires a malloc() and
+memcpy() operation, and these operations are not the fastest. Because of
+this, some functions like gst_adapter_available_fast() are provided to help
+speed up such cases should you want to. To avoid repeated memory allocations,
+gst_adapter_copy() can be used to copy data into a (statically allocated)
+user provided buffer.
+
+#GstAdapter is not MT safe. All operations on an adapter must be serialized by
+the caller. This is not normally a problem, however, as the normal use case
+of #GstAdapter is inside one pad's chain function, in which case access is
+serialized via the pad's STREAM_LOCK.
+
+Note that gst_adapter_push() takes ownership of the buffer passed. Use
+gst_buffer_ref() before pushing it into the adapter if you still want to
+access the buffer later. The adapter will never modify the data in the
+buffer pushed in it.
+
+ Creates a new #GstAdapter. Free with g_object_unref().
+
+ a new #GstAdapter
+
+
+
+
+ Gets the maximum amount of bytes available, that is it returns the maximum
+value that can be supplied to gst_adapter_map() without that function
+returning %NULL.
+
+ number of bytes available in @adapter
+
+
+
+
+ a #GstAdapter
+
+
+
+
+
+ Gets the maximum number of bytes that are immediately available without
+requiring any expensive operations (like copying the data into a
+temporary buffer).
+
+ number of bytes that are available in @adapter without expensive
+operations
+
+
+
+
+ a #GstAdapter
+
+
+
+
+
+ Removes all buffers from @adapter.
+
+
+
+
+
+ a #GstAdapter
+
+
+
+
+
+ Copies @size bytes of data starting at @offset out of the buffers
+contained in #GstAdapter into an array @dest provided by the caller.
+
+The array @dest should be large enough to contain @size bytes.
+The user should check that the adapter has (@offset + @size) bytes
+available before calling this function.
+
+
+
+
+
+ a #GstAdapter
+
+
+
+
+ the memory to copy into
+
+
+
+
+
+ the bytes offset in the adapter to start from
+
+
+
+ the number of bytes to copy
+
+
+
+
+
+ Similar to gst_adapter_copy, but more suitable for language bindings. @size
+bytes of data starting at @offset will be copied out of the buffers contained
+in @adapter and into a new #GBytes structure which is returned. Depending on
+the value of the @size argument an empty #GBytes structure may be returned.
+
+ A new #GBytes structure containing the copied data.
+
+
+
+
+ a #GstAdapter
+
+
+
+ the bytes offset in the adapter to start from
+
+
+
+ the number of bytes to copy
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Get the DTS that was on the last buffer with the GST_BUFFER_FLAG_DISCONT
+flag, or GST_CLOCK_TIME_NONE.
+
+ The DTS at the last discont or GST_CLOCK_TIME_NONE.
+
+
+
+
+ a #GstAdapter
+
+
+
+
+
+ Flushes the first @flush bytes in the @adapter. The caller must ensure that
+at least this many bytes are available.
+
+See also: gst_adapter_map(), gst_adapter_unmap()
+
+
+
+
+
+ a #GstAdapter
+
+
+
+ the number of bytes to flush
+
+
+
+
+
+ Returns a #GstBuffer containing the first @nbytes of the @adapter, but
+does not flush them from the adapter. See gst_adapter_take_buffer()
+for details.
+
+Caller owns a reference to the returned buffer. gst_buffer_unref() after
+usage.
+
+Free-function: gst_buffer_unref
+
+ a #GstBuffer containing the first
+ @nbytes of the adapter, or %NULL if @nbytes bytes are not available.
+ gst_buffer_unref() when no longer needed.
+
+
+
+
+ a #GstAdapter
+
+
+
+ the number of bytes to get
+
+
+
+
+
+ Returns a #GstBuffer containing the first @nbytes of the @adapter, but
+does not flush them from the adapter. See gst_adapter_take_buffer_fast()
+for details.
+
+Caller owns a reference to the returned buffer. gst_buffer_unref() after
+usage.
+
+Free-function: gst_buffer_unref
+
+ a #GstBuffer containing the first
+ @nbytes of the adapter, or %NULL if @nbytes bytes are not available.
+ gst_buffer_unref() when no longer needed.
+
+
+
+
+ a #GstAdapter
+
+
+
+ the number of bytes to get
+
+
+
+
+
+ Returns a #GstBufferList of buffers containing the first @nbytes bytes of
+the @adapter but does not flush them from the adapter. See
+gst_adapter_take_buffer_list() for details.
+
+Caller owns the returned list. Call gst_buffer_list_unref() to free
+the list after usage.
+
+ a #GstBufferList of buffers containing
+ the first @nbytes of the adapter, or %NULL if @nbytes bytes are not
+ available
+
+
+
+
+ a #GstAdapter
+
+
+
+ the number of bytes to get
+
+
+
+
+
+ Returns a #GList of buffers containing the first @nbytes bytes of the
+@adapter, but does not flush them from the adapter. See
+gst_adapter_take_list() for details.
+
+Caller owns returned list and contained buffers. gst_buffer_unref() each
+buffer in the list before freeing the list after usage.
+
+ a #GList of
+ buffers containing the first @nbytes of the adapter, or %NULL if @nbytes
+ bytes are not available
+
+
+
+
+
+
+ a #GstAdapter
+
+
+
+ the number of bytes to get
+
+
+
+
+
+ Gets the first @size bytes stored in the @adapter. The returned pointer is
+valid until the next function is called on the adapter.
+
+Note that setting the returned pointer as the data of a #GstBuffer is
+incorrect for general-purpose plugins. The reason is that if a downstream
+element stores the buffer so that it has access to it outside of the bounds
+of its chain function, the buffer will have an invalid data pointer after
+your element flushes the bytes. In that case you should use
+gst_adapter_take(), which returns a freshly-allocated buffer that you can set
+as #GstBuffer memory or the potentially more performant
+gst_adapter_take_buffer().
+
+Returns %NULL if @size bytes are not available.
+
+
+ a pointer to the first @size bytes of data, or %NULL
+
+
+
+
+
+
+ a #GstAdapter
+
+
+
+ the number of bytes to map/peek
+
+
+
+
+
+ Scan for pattern @pattern with applied mask @mask in the adapter data,
+starting from offset @offset.
+
+The bytes in @pattern and @mask are interpreted left-to-right, regardless
+of endianness. All four bytes of the pattern must be present in the
+adapter for it to match, even if the first or last bytes are masked out.
+
+It is an error to call this function without making sure that there is
+enough data (offset+size bytes) in the adapter.
+
+This function calls gst_adapter_masked_scan_uint32_peek() passing %NULL
+for value.
+
+ offset of the first match, or -1 if no match was found.
+
+Example:
+|[
+// Assume the adapter contains 0x00 0x01 0x02 ... 0xfe 0xff
+
+gst_adapter_masked_scan_uint32 (adapter, 0xffffffff, 0x00010203, 0, 256);
+// -> returns 0
+gst_adapter_masked_scan_uint32 (adapter, 0xffffffff, 0x00010203, 1, 255);
+// -> returns -1
+gst_adapter_masked_scan_uint32 (adapter, 0xffffffff, 0x01020304, 1, 255);
+// -> returns 1
+gst_adapter_masked_scan_uint32 (adapter, 0xffff, 0x0001, 0, 256);
+// -> returns -1
+gst_adapter_masked_scan_uint32 (adapter, 0xffff, 0x0203, 0, 256);
+// -> returns 0
+gst_adapter_masked_scan_uint32 (adapter, 0xffff0000, 0x02030000, 0, 256);
+// -> returns 2
+gst_adapter_masked_scan_uint32 (adapter, 0xffff0000, 0x02030000, 0, 4);
+// -> returns -1
+]|
+
+
+
+
+ a #GstAdapter
+
+
+
+ mask to apply to data before matching against @pattern
+
+
+
+ pattern to match (after mask is applied)
+
+
+
+ offset into the adapter data from which to start scanning, returns
+ the last scanned position.
+
+
+
+ number of bytes to scan from offset
+
+
+
+
+
+ Scan for pattern @pattern with applied mask @mask in the adapter data,
+starting from offset @offset. If a match is found, the value that matched
+is returned through @value, otherwise @value is left untouched.
+
+The bytes in @pattern and @mask are interpreted left-to-right, regardless
+of endianness. All four bytes of the pattern must be present in the
+adapter for it to match, even if the first or last bytes are masked out.
+
+It is an error to call this function without making sure that there is
+enough data (offset+size bytes) in the adapter.
+
+ offset of the first match, or -1 if no match was found.
+
+
+
+
+ a #GstAdapter
+
+
+
+ mask to apply to data before matching against @pattern
+
+
+
+ pattern to match (after mask is applied)
+
+
+
+ offset into the adapter data from which to start scanning, returns
+ the last scanned position.
+
+
+
+ number of bytes to scan from offset
+
+
+
+ pointer to uint32 to return matching data
+
+
+
+
+
+ Get the offset that was on the last buffer with the GST_BUFFER_FLAG_DISCONT
+flag, or GST_BUFFER_OFFSET_NONE.
+
+ The offset at the last discont or GST_BUFFER_OFFSET_NONE.
+
+
+
+
+ a #GstAdapter
+
+
+
+
+
+ Get the dts that was before the current byte in the adapter. When
+@distance is given, the amount of bytes between the dts and the current
+position is returned.
+
+The dts is reset to GST_CLOCK_TIME_NONE and the distance is set to 0 when
+the adapter is first created or when it is cleared. This also means that before
+the first byte with a dts is removed from the adapter, the dts
+and distance returned are GST_CLOCK_TIME_NONE and 0 respectively.
+
+ The previously seen dts.
+
+
+
+
+ a #GstAdapter
+
+
+
+ pointer to location for distance, or %NULL
+
+
+
+
+
+ Get the dts that was before the byte at offset @offset in the adapter. When
+@distance is given, the amount of bytes between the dts and the current
+position is returned.
+
+The dts is reset to GST_CLOCK_TIME_NONE and the distance is set to 0 when
+the adapter is first created or when it is cleared. This also means that before
+the first byte with a dts is removed from the adapter, the dts
+and distance returned are GST_CLOCK_TIME_NONE and 0 respectively.
+
+ The previously seen dts at given offset.
+
+
+
+
+ a #GstAdapter
+
+
+
+ the offset in the adapter at which to get timestamp
+
+
+
+ pointer to location for distance, or %NULL
+
+
+
+
+
+ Get the offset that was before the current byte in the adapter. When
+@distance is given, the amount of bytes between the offset and the current
+position is returned.
+
+The offset is reset to GST_BUFFER_OFFSET_NONE and the distance is set to 0
+when the adapter is first created or when it is cleared. This also means that
+before the first byte with an offset is removed from the adapter, the offset
+and distance returned are GST_BUFFER_OFFSET_NONE and 0 respectively.
+
+ The previous seen offset.
+
+
+
+
+ a #GstAdapter
+
+
+
+ pointer to a location for distance, or %NULL
+
+
+
+
+
+ Get the pts that was before the current byte in the adapter. When
+@distance is given, the amount of bytes between the pts and the current
+position is returned.
+
+The pts is reset to GST_CLOCK_TIME_NONE and the distance is set to 0 when
+the adapter is first created or when it is cleared. This also means that before
+the first byte with a pts is removed from the adapter, the pts
+and distance returned are GST_CLOCK_TIME_NONE and 0 respectively.
+
+ The previously seen pts.
+
+
+
+
+ a #GstAdapter
+
+
+
+ pointer to location for distance, or %NULL
+
+
+
+
+
+ Get the pts that was before the byte at offset @offset in the adapter. When
+@distance is given, the amount of bytes between the pts and the current
+position is returned.
+
+The pts is reset to GST_CLOCK_TIME_NONE and the distance is set to 0 when
+the adapter is first created or when it is cleared. This also means that before
+the first byte with a pts is removed from the adapter, the pts
+and distance returned are GST_CLOCK_TIME_NONE and 0 respectively.
+
+ The previously seen pts at given offset.
+
+
+
+
+ a #GstAdapter
+
+
+
+ the offset in the adapter at which to get timestamp
+
+
+
+ pointer to location for distance, or %NULL
+
+
+
+
+
+ Get the PTS that was on the last buffer with the GST_BUFFER_FLAG_DISCONT
+flag, or GST_CLOCK_TIME_NONE.
+
+ The PTS at the last discont or GST_CLOCK_TIME_NONE.
+
+
+
+
+ a #GstAdapter
+
+
+
+
+
+ Adds the data from @buf to the data stored inside @adapter and takes
+ownership of the buffer.
+
+
+
+
+
+ a #GstAdapter
+
+
+
+ a #GstBuffer to add to queue in the adapter
+
+
+
+
+
+ Returns a freshly allocated buffer containing the first @nbytes bytes of the
+@adapter. The returned bytes will be flushed from the adapter.
+
+Caller owns returned value. g_free after usage.
+
+Free-function: g_free
+
+
+ oven-fresh hot data, or %NULL if @nbytes bytes are not available
+
+
+
+
+
+
+ a #GstAdapter
+
+
+
+ the number of bytes to take
+
+
+
+
+
+ Returns a #GstBuffer containing the first @nbytes bytes of the
+@adapter. The returned bytes will be flushed from the adapter.
+This function is potentially more performant than
+gst_adapter_take() since it can reuse the memory in pushed buffers
+by subbuffering or merging. This function will always return a
+buffer with a single memory region.
+
+Note that no assumptions should be made as to whether certain buffer
+flags such as the DISCONT flag are set on the returned buffer, or not.
+The caller needs to explicitly set or unset flags that should be set or
+unset.
+
+Since 1.6 this will also copy over all GstMeta of the input buffers except
+for meta with the %GST_META_FLAG_POOLED flag or with the "memory" tag.
+
+Caller owns a reference to the returned buffer. gst_buffer_unref() after
+usage.
+
+Free-function: gst_buffer_unref
+
+ a #GstBuffer containing the first
+ @nbytes of the adapter, or %NULL if @nbytes bytes are not available.
+ gst_buffer_unref() when no longer needed.
+
+
+
+
+ a #GstAdapter
+
+
+
+ the number of bytes to take
+
+
+
+
+
+ Returns a #GstBuffer containing the first @nbytes of the @adapter.
+The returned bytes will be flushed from the adapter. This function
+is potentially more performant than gst_adapter_take_buffer() since
+it can reuse the memory in pushed buffers by subbuffering or
+merging. Unlike gst_adapter_take_buffer(), the returned buffer may
+be composed of multiple non-contiguous #GstMemory objects, no
+copies are made.
+
+Note that no assumptions should be made as to whether certain buffer
+flags such as the DISCONT flag are set on the returned buffer, or not.
+The caller needs to explicitly set or unset flags that should be set or
+unset.
+
+This will also copy over all GstMeta of the input buffers except
+for meta with the %GST_META_FLAG_POOLED flag or with the "memory" tag.
+
+This function can return buffer up to the return value of
+gst_adapter_available() without making copies if possible.
+
+Caller owns a reference to the returned buffer. gst_buffer_unref() after
+usage.
+
+Free-function: gst_buffer_unref
+
+ a #GstBuffer containing the first
+ @nbytes of the adapter, or %NULL if @nbytes bytes are not available.
+ gst_buffer_unref() when no longer needed.
+
+
+
+
+ a #GstAdapter
+
+
+
+ the number of bytes to take
+
+
+
+
+
+ Returns a #GstBufferList of buffers containing the first @nbytes bytes of
+the @adapter. The returned bytes will be flushed from the adapter.
+When the caller can deal with individual buffers, this function is more
+performant because no memory should be copied.
+
+Caller owns the returned list. Call gst_buffer_list_unref() to free
+the list after usage.
+
+ a #GstBufferList of buffers containing
+ the first @nbytes of the adapter, or %NULL if @nbytes bytes are not
+ available
+
+
+
+
+ a #GstAdapter
+
+
+
+ the number of bytes to take
+
+
+
+
+
+ Returns a #GList of buffers containing the first @nbytes bytes of the
+@adapter. The returned bytes will be flushed from the adapter.
+When the caller can deal with individual buffers, this function is more
+performant because no memory should be copied.
+
+Caller owns returned list and contained buffers. gst_buffer_unref() each
+buffer in the list before freeing the list after usage.
+
+ a #GList of
+ buffers containing the first @nbytes of the adapter, or %NULL if @nbytes
+ bytes are not available
+
+
+
+
+
+
+ a #GstAdapter
+
+
+
+ the number of bytes to take
+
+
+
+
+
+ Releases the memory obtained with the last gst_adapter_map().
+
+
+
+
+
+ a #GstAdapter
+
+
+
+
+
+
+
+
+ Manages a set of pads with the purpose of aggregating their buffers.
+Control is given to the subclass when all pads have data.
+
+ * Base class for mixers and muxers. Subclasses should at least implement
+ the #GstAggregatorClass.aggregate() virtual method.
+
+ * Installs a #GstPadChainFunction, a #GstPadEventFullFunction and a
+ #GstPadQueryFunction to queue all serialized data packets per sink pad.
+ Subclasses should not overwrite those, but instead implement
+ #GstAggregatorClass.sink_event() and #GstAggregatorClass.sink_query() as
+ needed.
+
+ * When data is queued on all pads, the aggregate vmethod is called.
+
+ * One can peek at the data on any given GstAggregatorPad with the
+ gst_aggregator_pad_peek_buffer () method, and remove it from the pad
+ with the gst_aggregator_pad_pop_buffer () method. When a buffer
+ has been taken with pop_buffer (), a new buffer can be queued
+ on that pad.
+
+ * If the subclass wishes to push a buffer downstream in its aggregate
+ implementation, it should do so through the
+ gst_aggregator_finish_buffer () method. This method will take care
+ of sending and ordering mandatory events such as stream start, caps
+ and segment.
+
+ * Same goes for EOS events, which should not be pushed directly by the
+ subclass, it should instead return GST_FLOW_EOS in its aggregate
+ implementation.
+
+ * Note that the aggregator logic regarding gap event handling is to turn
+ these into gap buffers with matching PTS and duration. It will also
+ flag these buffers with GST_BUFFER_FLAG_GAP and GST_BUFFER_FLAG_DROPPABLE
+ to ease their identification and subsequent processing.
+
+ * Subclasses must use (a subclass of) #GstAggregatorPad for both their
+ sink and source pads.
+ See gst_element_class_add_static_pad_template_with_gtype().
+
+This class used to live in gst-plugins-bad and was moved to core.
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ This method will push the provided output buffer downstream. If needed,
+mandatory events such as stream-start, caps, and segment events will be
+sent before pushing the buffer.
+
+
+
+
+
+ The #GstAggregator
+
+
+
+ the #GstBuffer to push.
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ This method will push the provided output buffer downstream. If needed,
+mandatory events such as stream-start, caps, and segment events will be
+sent before pushing the buffer.
+
+
+
+
+
+ The #GstAggregator
+
+
+
+ the #GstBuffer to push.
+
+
+
+
+
+ Lets #GstAggregator sub-classes get the memory @allocator
+acquired by the base class and its @params.
+
+Unref the @allocator after use it.
+
+
+
+
+
+ a #GstAggregator
+
+
+
+ the #GstAllocator
+used
+
+
+
+ the
+#GstAllocationParams of @allocator
+
+
+
+
+
+
+ the instance of the #GstBufferPool used
+by @trans; free it after use it
+
+
+
+
+ a #GstAggregator
+
+
+
+
+
+ Retrieves the latency values reported by @self in response to the latency
+query, or %GST_CLOCK_TIME_NONE if there is not live source connected and the element
+will not wait for the clock.
+
+Typically only called by subclasses.
+
+ The latency or %GST_CLOCK_TIME_NONE if the element does not sync
+
+
+
+
+ a #GstAggregator
+
+
+
+
+
+ Lets #GstAggregator sub-classes tell the baseclass what their internal
+latency is. Will also post a LATENCY message on the bus so the pipeline
+can reconfigure its global latency.
+
+
+
+
+
+ a #GstAggregator
+
+
+
+ minimum latency
+
+
+
+ maximum latency
+
+
+
+
+
+ Sets the caps to be used on the src pad.
+
+
+
+
+
+ The #GstAggregator
+
+
+
+ The #GstCaps to set on the src pad.
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ the aggregator's source pad
+
+
+
+
+
+
+
+
+
+
+
+
+ The aggregator base class will handle in a thread-safe way all manners of
+concurrent flushes, seeks, pad additions and removals, leaving to the
+subclass the responsibility of clipping buffers, and aggregating buffers in
+the way the implementor sees fit.
+
+It will also take care of event ordering (stream-start, segment, eos).
+
+Basically, a simple implementation will override @aggregate, and call
+_finish_buffer from inside that function.
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ The #GstAggregator
+
+
+
+ the #GstBuffer to push.
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Pads managed by a #GstAggregor subclass.
+
+This class used to live in gst-plugins-bad and was moved to core.
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Drop the buffer currently queued in @pad.
+
+ TRUE if there was a buffer queued in @pad, or FALSE if not.
+
+
+
+
+ the pad where to drop any pending buffer
+
+
+
+
+
+
+ %TRUE if the pad has a buffer available as the next thing.
+
+
+
+
+ the pad to check the buffer on
+
+
+
+
+
+
+ %TRUE if the pad is EOS, otherwise %FALSE.
+
+
+
+
+ an aggregator pad
+
+
+
+
+
+
+ A reference to the buffer in @pad or
+NULL if no buffer was queued. You should unref the buffer after
+usage.
+
+
+
+
+ the pad to get buffer from
+
+
+
+
+
+ Steal the ref to the buffer currently queued in @pad.
+
+ The buffer in @pad or NULL if no buffer was
+ queued. You should unref the buffer after usage.
+
+
+
+
+ the pad to get buffer from
+
+
+
+
+
+
+
+
+ last segment received.
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ The name of the templates for the sink pad.
+
+
+
+ The name of the templates for the source pad.
+
+
+
+ This base class is for parser elements that process data and splits it
+into separate audio/video/whatever frames.
+
+It provides for:
+
+ * provides one sink pad and one source pad
+ * handles state changes
+ * can operate in pull mode or push mode
+ * handles seeking in both modes
+ * handles events (SEGMENT/EOS/FLUSH)
+ * handles queries (POSITION/DURATION/SEEKING/FORMAT/CONVERT)
+ * handles flushing
+
+The purpose of this base class is to provide the basic functionality of
+a parser and share a lot of rather complex code.
+
+# Description of the parsing mechanism:
+
+## Set-up phase
+
+ * #GstBaseParse calls @start to inform subclass that data processing is
+ about to start now.
+
+ * #GstBaseParse class calls @set_sink_caps to inform the subclass about
+ incoming sinkpad caps. Subclass could already set the srcpad caps
+ accordingly, but this might be delayed until calling
+ gst_base_parse_finish_frame() with a non-queued frame.
+
+ * At least at this point subclass needs to tell the #GstBaseParse class
+ how big data chunks it wants to receive (min_frame_size). It can do
+ this with gst_base_parse_set_min_frame_size().
+
+ * #GstBaseParse class sets up appropriate data passing mode (pull/push)
+ and starts to process the data.
+
+## Parsing phase
+
+ * #GstBaseParse gathers at least min_frame_size bytes of data either
+ by pulling it from upstream or collecting buffers in an internal
+ #GstAdapter.
+
+ * A buffer of (at least) min_frame_size bytes is passed to subclass with
+ @handle_frame. Subclass checks the contents and can optionally
+ return GST_FLOW_OK along with an amount of data to be skipped to find
+ a valid frame (which will result in a subsequent DISCONT).
+ If, otherwise, the buffer does not hold a complete frame,
+ @handle_frame can merely return and will be called again when additional
+ data is available. In push mode this amounts to an
+ additional input buffer (thus minimal additional latency), in pull mode
+ this amounts to some arbitrary reasonable buffer size increase.
+ Of course, gst_base_parse_set_min_frame_size() could also be used if a
+ very specific known amount of additional data is required.
+ If, however, the buffer holds a complete valid frame, it can pass
+ the size of this frame to gst_base_parse_finish_frame().
+ If acting as a converter, it can also merely indicate consumed input data
+ while simultaneously providing custom output data.
+ Note that baseclass performs some processing (such as tracking
+ overall consumed data rate versus duration) for each finished frame,
+ but other state is only updated upon each call to @handle_frame
+ (such as tracking upstream input timestamp).
+
+ Subclass is also responsible for setting the buffer metadata
+ (e.g. buffer timestamp and duration, or keyframe if applicable).
+ (although the latter can also be done by #GstBaseParse if it is
+ appropriately configured, see below). Frame is provided with
+ timestamp derived from upstream (as much as generally possible),
+ duration obtained from configuration (see below), and offset
+ if meaningful (in pull mode).
+
+ Note that @check_valid_frame might receive any small
+ amount of input data when leftover data is being drained (e.g. at EOS).
+
+ * As part of finish frame processing,
+ just prior to actually pushing the buffer in question,
+ it is passed to @pre_push_frame which gives subclass yet one
+ last chance to examine buffer metadata, or to send some custom (tag)
+ events, or to perform custom (segment) filtering.
+
+ * During the parsing process #GstBaseParseClass will handle both srcpad
+ and sinkpad events. They will be passed to subclass if @event or
+ @src_event callbacks have been provided.
+
+## Shutdown phase
+
+* #GstBaseParse class calls @stop to inform the subclass that data
+ parsing will be stopped.
+
+Subclass is responsible for providing pad template caps for
+source and sink pads. The pads need to be named "sink" and "src". It also
+needs to set the fixed caps on srcpad, when the format is ensured (e.g.
+when base class calls subclass' @set_sink_caps function).
+
+This base class uses %GST_FORMAT_DEFAULT as a meaning of frames. So,
+subclass conversion routine needs to know that conversion from
+%GST_FORMAT_TIME to %GST_FORMAT_DEFAULT must return the
+frame number that can be found from the given byte position.
+
+#GstBaseParse uses subclasses conversion methods also for seeking (or
+otherwise uses its own default one, see also below).
+
+Subclass @start and @stop functions will be called to inform the beginning
+and end of data processing.
+
+Things that subclass need to take care of:
+
+* Provide pad templates
+* Fixate the source pad caps when appropriate
+* Inform base class how big data chunks should be retrieved. This is
+ done with gst_base_parse_set_min_frame_size() function.
+* Examine data chunks passed to subclass with @handle_frame and pass
+ proper frame(s) to gst_base_parse_finish_frame(), and setting src pad
+ caps and timestamps on frame.
+* Provide conversion functions
+* Update the duration information with gst_base_parse_set_duration()
+* Optionally passthrough using gst_base_parse_set_passthrough()
+* Configure various baseparse parameters using
+ gst_base_parse_set_average_bitrate(), gst_base_parse_set_syncable()
+ and gst_base_parse_set_frame_rate().
+
+* In particular, if subclass is unable to determine a duration, but
+ parsing (or specs) yields a frames per seconds rate, then this can be
+ provided to #GstBaseParse to enable it to cater for
+ buffer time metadata (which will be taken from upstream as much as
+ possible). Internally keeping track of frame durations and respective
+ sizes that have been pushed provides #GstBaseParse with an estimated
+ bitrate. A default @convert (used if not overridden) will then use these
+ rates to perform obvious conversions. These rates are also used to
+ update (estimated) duration at regular frame intervals.
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Adds an entry to the index associating @offset to @ts. It is recommended
+to only add keyframe entries. @force allows to bypass checks, such as
+whether the stream is (upstream) seekable, another entry is already "close"
+to the new entry, etc.
+
+ #gboolean indicating whether entry was added
+
+
+
+
+ #GstBaseParse.
+
+
+
+ offset of entry
+
+
+
+ timestamp associated with offset
+
+
+
+ whether entry refers to keyframe
+
+
+
+ add entry disregarding sanity checks
+
+
+
+
+
+ Default implementation of "convert" vmethod in #GstBaseParse class.
+
+ %TRUE if conversion was successful.
+
+
+
+
+ #GstBaseParse.
+
+
+
+ #GstFormat describing the source format.
+
+
+
+ Source value to be converted.
+
+
+
+ #GstFormat defining the converted format.
+
+
+
+ Pointer where the conversion result will be put.
+
+
+
+
+
+ Drains the adapter until it is empty. It decreases the min_frame_size to
+match the current adapter size and calls chain method until the adapter
+is emptied or chain returns with error.
+
+
+
+
+
+ a #GstBaseParse
+
+
+
+
+
+ Collects parsed data and pushes this downstream.
+Source pad caps must be set when this is called.
+
+If @frame's out_buffer is set, that will be used as subsequent frame data.
+Otherwise, @size samples will be taken from the input and used for output,
+and the output's metadata (timestamps etc) will be taken as (optionally)
+set by the subclass on @frame's (input) buffer (which is otherwise
+ignored for any but the above purpose/information).
+
+Note that the latter buffer is invalidated by this call, whereas the
+caller retains ownership of @frame.
+
+ a #GstFlowReturn that should be escalated to caller (of caller)
+
+
+
+
+ a #GstBaseParse
+
+
+
+ a #GstBaseParseFrame
+
+
+
+ consumed input data represented by frame
+
+
+
+
+
+ Sets the parser subclass's tags and how they should be merged with any
+upstream stream tags. This will override any tags previously-set
+with gst_base_parse_merge_tags().
+
+Note that this is provided for convenience, and the subclass is
+not required to use this and can still do tag handling on its own.
+
+
+
+
+
+ a #GstBaseParse
+
+
+
+ a #GstTagList to merge, or NULL to unset
+ previously-set tags
+
+
+
+ the #GstTagMergeMode to use, usually #GST_TAG_MERGE_REPLACE
+
+
+
+
+
+ Pushes the frame's buffer downstream, sends any pending events and
+does some timestamp and segment handling. Takes ownership of
+frame's buffer, though caller retains ownership of @frame.
+
+This must be called with sinkpad STREAM_LOCK held.
+
+ #GstFlowReturn
+
+
+
+
+ #GstBaseParse.
+
+
+
+ a #GstBaseParseFrame
+
+
+
+
+
+ Optionally sets the average bitrate detected in media (if non-zero),
+e.g. based on metadata, as it will be posted to the application.
+
+By default, announced average bitrate is estimated. The average bitrate
+is used to estimate the total duration of the stream and to estimate
+a seek position, if there's no index and the format is syncable
+(see gst_base_parse_set_syncable()).
+
+
+
+
+
+ #GstBaseParse.
+
+
+
+ average bitrate in bits/second
+
+
+
+
+
+ Sets the duration of the currently playing media. Subclass can use this
+when it is able to determine duration and/or notices a change in the media
+duration. Alternatively, if @interval is non-zero (default), then stream
+duration is determined based on estimated bitrate, and updated every @interval
+frames.
+
+
+
+
+
+ #GstBaseParse.
+
+
+
+ #GstFormat.
+
+
+
+ duration value.
+
+
+
+ how often to update the duration estimate based on bitrate, or 0.
+
+
+
+
+
+ If frames per second is configured, parser can take care of buffer duration
+and timestamping. When performing segment clipping, or seeking to a specific
+location, a corresponding decoder might need an initial @lead_in and a
+following @lead_out number of frames to ensure the desired segment is
+entirely filled upon decoding.
+
+
+
+
+
+ the #GstBaseParse to set
+
+
+
+ frames per second (numerator).
+
+
+
+ frames per second (denominator).
+
+
+
+ frames needed before a segment for subsequent decode
+
+
+
+ frames needed after a segment
+
+
+
+
+
+ Set if frames carry timing information which the subclass can (generally)
+parse and provide. In particular, intrinsic (rather than estimated) time
+can be obtained following a seek.
+
+
+
+
+
+ a #GstBaseParse
+
+
+
+ whether frames carry timing information
+
+
+
+
+
+ By default, the base class might try to infer PTS from DTS and vice
+versa. While this is generally correct for audio data, it may not
+be otherwise. Sub-classes implementing such formats should disable
+timestamp inferring.
+
+
+
+
+
+ a #GstBaseParse
+
+
+
+ %TRUE if parser should infer DTS/PTS from each other
+
+
+
+
+
+ Sets the minimum and maximum (which may likely be equal) latency introduced
+by the parsing process. If there is such a latency, which depends on the
+particular parsing of the format, it typically corresponds to 1 frame duration.
+
+
+
+
+
+ a #GstBaseParse
+
+
+
+ minimum parse latency
+
+
+
+ maximum parse latency
+
+
+
+
+
+ Subclass can use this function to tell the base class that it needs to
+give at least #min_size buffers.
+
+
+
+
+
+ #GstBaseParse.
+
+
+
+ Minimum size of the data that this base class should give to
+ subclass.
+
+
+
+
+
+ Set if the nature of the format or configuration does not allow (much)
+parsing, and the parser should operate in passthrough mode (which only
+applies when operating in push mode). That is, incoming buffers are
+pushed through unmodified, i.e. no @check_valid_frame or @parse_frame
+callbacks will be invoked, but @pre_push_frame will still be invoked,
+so subclass can perform as much or as little is appropriate for
+passthrough semantics in @pre_push_frame.
+
+
+
+
+
+ a #GstBaseParse
+
+
+
+ %TRUE if parser should run in passthrough mode
+
+
+
+
+
+ By default, the base class will guess PTS timestamps using a simple
+interpolation (previous timestamp + duration), which is incorrect for
+data streams with reordering, where PTS can go backward. Sub-classes
+implementing such formats should disable PTS interpolation.
+
+
+
+
+
+ a #GstBaseParse
+
+
+
+ %TRUE if parser should interpolate PTS timestamps
+
+
+
+
+
+ Set if frame starts can be identified. This is set by default and
+determines whether seeking based on bitrate averages
+is possible for a format/stream.
+
+
+
+
+
+ a #GstBaseParse
+
+
+
+ set if frame starts can be identified
+
+
+
+
+
+ This function should only be called from a @handle_frame implementation.
+
+#GstBaseParse creates initial timestamps for frames by using the last
+timestamp seen in the stream before the frame starts. In certain
+cases, the correct timestamps will occur in the stream after the
+start of the frame, but before the start of the actual picture data.
+This function can be used to set the timestamps based on the offset
+into the frame data that the picture starts.
+
+
+
+
+
+ a #GstBaseParse
+
+
+
+ offset into current buffer
+
+
+
+
+
+ If set to %TRUE, baseparse will unconditionally force parsing of the
+incoming data. This can be required in the rare cases where the incoming
+side-data (caps, pts, dts, ...) is not trusted by the user and wants to
+force validation and parsing of the incoming data.
+If set to %FALSE, decision of whether to parse the data or not is up to
+the implementation (standard behaviour).
+
+
+
+ the parent element.
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Subclasses can override any of the available virtual methods or not, as
+needed. At minimum @handle_frame needs to be overridden.
+
+ the parent class
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Frame (context) data passed to each frame parsing virtual methods. In
+addition to providing the data to be checked for a valid frame or an already
+identified frame, it conveys additional metadata or control information
+from and to the subclass w.r.t. the particular frame in question (rather
+than global parameters). Some of these may apply to each parsing stage, others
+only to some a particular one. These parameters are effectively zeroed at start
+of each frame's processing, i.e. parsing virtual method invocation sequence.
+
+ input data to be parsed for frames.
+
+
+
+ output data.
+
+
+
+ a combination of input and output #GstBaseParseFrameFlags that
+ convey additional context to subclass or allow subclass to tune
+ subsequent #GstBaseParse actions.
+
+
+
+ media specific offset of input frame
+ Note that a converter may have a different one on the frame's buffer.
+
+
+
+ subclass can set this to indicates the metadata overhead
+ for the given frame, which is then used to enable more accurate bitrate
+ computations. If this is -1, it is assumed that this frame should be
+ skipped in bitrate calculation.
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Allocates a new #GstBaseParseFrame. This function is mainly for bindings,
+elements written in C should usually allocate the frame on the stack and
+then use gst_base_parse_frame_init() to initialise it.
+
+ a newly-allocated #GstBaseParseFrame. Free with
+ gst_base_parse_frame_free() when no longer needed.
+
+
+
+
+ a #GstBuffer
+
+
+
+ the flags
+
+
+
+ number of bytes in this frame which should be counted as
+ metadata overhead, ie. not used to calculate the average bitrate.
+ Set to -1 to mark the entire frame as metadata. If in doubt, set to 0.
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Sets a #GstBaseParseFrame to initial state. Currently this means
+all public fields are zero-ed and a private flag is set to make
+sure gst_base_parse_frame_free() only frees the contents but not
+the actual frame. Use this function to initialise a #GstBaseParseFrame
+allocated on the stack.
+
+
+
+
+
+ #GstBaseParseFrame.
+
+
+
+
+
+
+ Flags to be used in a #GstBaseParseFrame.
+
+ no flag
+
+
+ set by baseclass if current frame
+ is passed for processing to the subclass for the first time
+ (and not set on subsequent calls with same data).
+
+
+ set to indicate this buffer should not be
+ counted as frame, e.g. if this frame is dependent on a previous one.
+ As it is not counted as a frame, bitrate increases but frame to time
+ conversions are maintained.
+
+
+ @pre_push_frame can set this to indicate
+ that regular segment clipping can still be performed (as opposed to
+ any custom one having been done).
+
+
+ indicates to @finish_frame that the
+ the frame should be dropped (and might be handled internally by subclass)
+
+
+ indicates to @finish_frame that the
+ the frame should be queued for now and processed fully later
+ when the first non-queued frame is finished
+
+
+
+
+
+ #GstBaseSink is the base class for sink elements in GStreamer, such as
+xvimagesink or filesink. It is a layer on top of #GstElement that provides a
+simplified interface to plugin writers. #GstBaseSink handles many details
+for you, for example: preroll, clock synchronization, state changes,
+activation in push or pull mode, and queries.
+
+In most cases, when writing sink elements, there is no need to implement
+class methods from #GstElement or to set functions on pads, because the
+#GstBaseSink infrastructure should be sufficient.
+
+#GstBaseSink provides support for exactly one sink pad, which should be
+named "sink". A sink implementation (subclass of #GstBaseSink) should
+install a pad template in its class_init function, like so:
+|[<!-- language="C" -->
+static void
+my_element_class_init (GstMyElementClass *klass)
+{
+ GstElementClass *gstelement_class = GST_ELEMENT_CLASS (klass);
+
+ // sinktemplate should be a #GstStaticPadTemplate with direction
+ // %GST_PAD_SINK and name "sink"
+ gst_element_class_add_static_pad_template (gstelement_class, &sinktemplate);
+
+ gst_element_class_set_static_metadata (gstelement_class,
+ "Sink name",
+ "Sink",
+ "My Sink element",
+ "The author <my.sink@my.email>");
+}
+]|
+
+#GstBaseSink will handle the prerolling correctly. This means that it will
+return %GST_STATE_CHANGE_ASYNC from a state change to PAUSED until the first
+buffer arrives in this element. The base class will call the
+#GstBaseSinkClass.preroll() vmethod with this preroll buffer and will then
+commit the state change to the next asynchronously pending state.
+
+When the element is set to PLAYING, #GstBaseSink will synchronise on the
+clock using the times returned from #GstBaseSinkClass.get_times(). If this
+function returns %GST_CLOCK_TIME_NONE for the start time, no synchronisation
+will be done. Synchronisation can be disabled entirely by setting the object
+#GstBaseSink:sync property to %FALSE.
+
+After synchronisation the virtual method #GstBaseSinkClass.render() will be
+called. Subclasses should minimally implement this method.
+
+Subclasses that synchronise on the clock in the #GstBaseSinkClass.render()
+method are supported as well. These classes typically receive a buffer in
+the render method and can then potentially block on the clock while
+rendering. A typical example is an audiosink.
+These subclasses can use gst_base_sink_wait_preroll() to perform the
+blocking wait.
+
+Upon receiving the EOS event in the PLAYING state, #GstBaseSink will wait
+for the clock to reach the time indicated by the stop time of the last
+#GstBaseSinkClass.get_times() call before posting an EOS message. When the
+element receives EOS in PAUSED, preroll completes, the event is queued and an
+EOS message is posted when going to PLAYING.
+
+#GstBaseSink will internally use the %GST_EVENT_SEGMENT events to schedule
+synchronisation and clipping of buffers. Buffers that fall completely outside
+of the current segment are dropped. Buffers that fall partially in the
+segment are rendered (and prerolled). Subclasses should do any subbuffer
+clipping themselves when needed.
+
+#GstBaseSink will by default report the current playback position in
+%GST_FORMAT_TIME based on the current clock time and segment information.
+If no clock has been set on the element, the query will be forwarded
+upstream.
+
+The #GstBaseSinkClass.set_caps() function will be called when the subclass
+should configure itself to process a specific media type.
+
+The #GstBaseSinkClass.start() and #GstBaseSinkClass.stop() virtual methods
+will be called when resources should be allocated. Any
+#GstBaseSinkClass.preroll(), #GstBaseSinkClass.render() and
+#GstBaseSinkClass.set_caps() function will be called between the
+#GstBaseSinkClass.start() and #GstBaseSinkClass.stop() calls.
+
+The #GstBaseSinkClass.event() virtual method will be called when an event is
+received by #GstBaseSink. Normally this method should only be overridden by
+very specific elements (such as file sinks) which need to handle the
+newsegment event specially.
+
+The #GstBaseSinkClass.unlock() method is called when the elements should
+unblock any blocking operations they perform in the
+#GstBaseSinkClass.render() method. This is mostly useful when the
+#GstBaseSinkClass.render() method performs a blocking write on a file
+descriptor, for example.
+
+The #GstBaseSink:max-lateness property affects how the sink deals with
+buffers that arrive too late in the sink. A buffer arrives too late in the
+sink when the presentation time (as a combination of the last segment, buffer
+timestamp and element base_time) plus the duration is before the current
+time of the clock.
+If the frame is later than max-lateness, the sink will drop the buffer
+without calling the render method.
+This feature is disabled if sync is disabled, the
+#GstBaseSinkClass.get_times() method does not return a valid start time or
+max-lateness is set to -1 (the default).
+Subclasses can use gst_base_sink_set_max_lateness() to configure the
+max-lateness value.
+
+The #GstBaseSink:qos property will enable the quality-of-service features of
+the basesink which gather statistics about the real-time performance of the
+clock synchronisation. For each buffer received in the sink, statistics are
+gathered and a QOS event is sent upstream with these numbers. This
+information can then be used by upstream elements to reduce their processing
+rate, for example.
+
+The #GstBaseSink:async property can be used to instruct the sink to never
+perform an ASYNC state change. This feature is mostly usable when dealing
+with non-synchronized streams or sparse streams.
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ If the @sink spawns its own thread for pulling buffers from upstream it
+should call this method after it has pulled a buffer. If the element needed
+to preroll, this function will perform the preroll and will then block
+until the element state is changed.
+
+This function should be called with the PREROLL_LOCK held.
+
+ %GST_FLOW_OK if the preroll completed and processing can
+continue. Any other return value should be returned from the render vmethod.
+
+
+
+
+ the sink
+
+
+
+ the mini object that caused the preroll
+
+
+
+
+
+ Get the number of bytes that the sink will pull when it is operating in pull
+mode.
+
+ the number of bytes @sink will pull in pull mode.
+
+
+
+
+ a #GstBaseSink
+
+
+
+
+
+ Checks if @sink is currently configured to drop buffers which are outside
+the current segment
+
+ %TRUE if the sink is configured to drop buffers outside the
+current segment.
+
+
+
+
+ the sink
+
+
+
+
+
+ Get the last sample that arrived in the sink and was used for preroll or for
+rendering. This property can be used to generate thumbnails.
+
+The #GstCaps on the sample can be used to determine the type of the buffer.
+
+Free-function: gst_sample_unref
+
+ a #GstSample. gst_sample_unref() after
+ usage. This function returns %NULL when no buffer has arrived in the
+ sink yet or when the sink is not in PAUSED or PLAYING.
+
+
+
+
+ the sink
+
+
+
+
+
+ Get the currently configured latency.
+
+ The configured latency.
+
+
+
+
+ the sink
+
+
+
+
+
+ Get the maximum amount of bits per second that the sink will render.
+
+ the maximum number of bits per second @sink will render.
+
+
+
+
+ a #GstBaseSink
+
+
+
+
+
+ Gets the max lateness value. See gst_base_sink_set_max_lateness() for
+more details.
+
+ The maximum time in nanoseconds that a buffer can be late
+before it is dropped and not rendered. A value of -1 means an
+unlimited time.
+
+
+
+
+ the sink
+
+
+
+
+
+ Get the render delay of @sink. see gst_base_sink_set_render_delay() for more
+information about the render delay.
+
+ the render delay of @sink.
+
+
+
+
+ a #GstBaseSink
+
+
+
+
+
+ Checks if @sink is currently configured to synchronize against the
+clock.
+
+ %TRUE if the sink is configured to synchronize against the clock.
+
+
+
+
+ the sink
+
+
+
+
+
+ Get the time that will be inserted between frames to control the
+maximum buffers per second.
+
+ the number of nanoseconds @sink will put between frames.
+
+
+
+
+ a #GstBaseSink
+
+
+
+
+
+ Get the synchronisation offset of @sink.
+
+ The synchronisation offset.
+
+
+
+
+ the sink
+
+
+
+
+
+ Checks if @sink is currently configured to perform asynchronous state
+changes to PAUSED.
+
+ %TRUE if the sink is configured to perform asynchronous state
+changes.
+
+
+
+
+ the sink
+
+
+
+
+
+ Checks if @sink is currently configured to store the last received sample in
+the last-sample property.
+
+ %TRUE if the sink is configured to store the last received sample.
+
+
+
+
+ the sink
+
+
+
+
+
+ Checks if @sink is currently configured to send Quality-of-Service events
+upstream.
+
+ %TRUE if the sink is configured to perform Quality-of-Service.
+
+
+
+
+ the sink
+
+
+
+
+
+ Query the sink for the latency parameters. The latency will be queried from
+the upstream elements. @live will be %TRUE if @sink is configured to
+synchronize against the clock. @upstream_live will be %TRUE if an upstream
+element is live.
+
+If both @live and @upstream_live are %TRUE, the sink will want to compensate
+for the latency introduced by the upstream elements by setting the
+@min_latency to a strictly positive value.
+
+This function is mostly used by subclasses.
+
+ %TRUE if the query succeeded.
+
+
+
+
+ the sink
+
+
+
+ if the sink is live
+
+
+
+ if an upstream element is live
+
+
+
+ the min latency of the upstream elements
+
+
+
+ the max latency of the upstream elements
+
+
+
+
+
+ Configures @sink to perform all state changes asynchronously. When async is
+disabled, the sink will immediately go to PAUSED instead of waiting for a
+preroll buffer. This feature is useful if the sink does not synchronize
+against the clock or when it is dealing with sparse streams.
+
+
+
+
+
+ the sink
+
+
+
+ the new async value.
+
+
+
+
+
+ Set the number of bytes that the sink will pull when it is operating in pull
+mode.
+
+
+
+
+
+ a #GstBaseSink
+
+
+
+ the blocksize in bytes
+
+
+
+
+
+ Configure @sink to drop buffers which are outside the current segment
+
+
+
+
+
+ the sink
+
+
+
+ drop buffers outside the segment
+
+
+
+
+
+ Configures @sink to store the last received sample in the last-sample
+property.
+
+
+
+
+
+ the sink
+
+
+
+ the new enable-last-sample value.
+
+
+
+
+
+ Set the maximum amount of bits per second that the sink will render.
+
+
+
+
+
+ a #GstBaseSink
+
+
+
+ the max_bitrate in bits per second
+
+
+
+
+
+ Sets the new max lateness value to @max_lateness. This value is
+used to decide if a buffer should be dropped or not based on the
+buffer timestamp and the current clock time. A value of -1 means
+an unlimited time.
+
+
+
+
+
+ the sink
+
+
+
+ the new max lateness value.
+
+
+
+
+
+ Configures @sink to send Quality-of-Service events upstream.
+
+
+
+
+
+ the sink
+
+
+
+ the new qos value.
+
+
+
+
+
+ Set the render delay in @sink to @delay. The render delay is the time
+between actual rendering of a buffer and its synchronisation time. Some
+devices might delay media rendering which can be compensated for with this
+function.
+
+After calling this function, this sink will report additional latency and
+other sinks will adjust their latency to delay the rendering of their media.
+
+This function is usually called by subclasses.
+
+
+
+
+
+ a #GstBaseSink
+
+
+
+ the new delay
+
+
+
+
+
+ Configures @sink to synchronize on the clock or not. When
+@sync is %FALSE, incoming samples will be played as fast as
+possible. If @sync is %TRUE, the timestamps of the incoming
+buffers will be used to schedule the exact render time of its
+contents.
+
+
+
+
+
+ the sink
+
+
+
+ the new sync value.
+
+
+
+
+
+ Set the time that will be inserted between rendered buffers. This
+can be used to control the maximum buffers per second that the sink
+will render.
+
+
+
+
+
+ a #GstBaseSink
+
+
+
+ the throttle time in nanoseconds
+
+
+
+
+
+ Adjust the synchronisation of @sink with @offset. A negative value will
+render buffers earlier than their timestamp. A positive value will delay
+rendering. This function can be used to fix playback of badly timestamped
+buffers.
+
+
+
+
+
+ the sink
+
+
+
+ the new offset
+
+
+
+
+
+ This function will wait for preroll to complete and will then block until @time
+is reached. It is usually called by subclasses that use their own internal
+synchronisation but want to let some synchronization (like EOS) be handled
+by the base class.
+
+This function should only be called with the PREROLL_LOCK held (like when
+receiving an EOS event in the ::event vmethod or when handling buffers in
+::render).
+
+The @time argument should be the running_time of when the timeout should happen
+and will be adjusted with any latency and offset configured in the sink.
+
+ #GstFlowReturn
+
+
+
+
+ the sink
+
+
+
+ the running_time to be reached
+
+
+
+ the jitter to be filled with time diff, or %NULL
+
+
+
+
+
+ This function will block until @time is reached. It is usually called by
+subclasses that use their own internal synchronisation.
+
+If @time is not valid, no synchronisation is done and %GST_CLOCK_BADTIME is
+returned. Likewise, if synchronisation is disabled in the element or there
+is no clock, no synchronisation is done and %GST_CLOCK_BADTIME is returned.
+
+This function should only be called with the PREROLL_LOCK held, like when
+receiving an EOS event in the #GstBaseSinkClass.event() vmethod or when
+receiving a buffer in
+the #GstBaseSinkClass.render() vmethod.
+
+The @time argument should be the running_time of when this method should
+return and is not adjusted with any latency or offset configured in the
+sink.
+
+ #GstClockReturn
+
+
+
+
+ the sink
+
+
+
+ the running_time to be reached
+
+
+
+ the jitter to be filled with time diff, or %NULL
+
+
+
+
+
+ If the #GstBaseSinkClass.render() method performs its own synchronisation
+against the clock it must unblock when going from PLAYING to the PAUSED state
+and call this method before continuing to render the remaining data.
+
+If the #GstBaseSinkClass.render() method can block on something else than
+the clock, it must also be ready to unblock immediately on
+the #GstBaseSinkClass.unlock() method and cause the
+#GstBaseSinkClass.render() method to immediately call this function.
+In this case, the subclass must be prepared to continue rendering where it
+left off if this function returns %GST_FLOW_OK.
+
+This function will block until a state change to PLAYING happens (in which
+case this function returns %GST_FLOW_OK) or the processing must be stopped due
+to a state change to READY or a FLUSH event (in which case this function
+returns %GST_FLOW_FLUSHING).
+
+This function should only be called with the PREROLL_LOCK held, like in the
+render function.
+
+ %GST_FLOW_OK if the preroll completed and processing can
+continue. Any other return value should be returned from the render vmethod.
+
+
+
+
+ the sink
+
+
+
+
+
+ If set to %TRUE, the basesink will perform asynchronous state changes.
+When set to %FALSE, the sink will not signal the parent when it prerolls.
+Use this option when dealing with sparse streams or when synchronisation is
+not required.
+
+
+
+ The amount of bytes to pull when operating in pull mode.
+
+
+
+ Enable the last-sample property. If %FALSE, basesink doesn't keep a
+reference to the last buffer arrived and the last-sample property is always
+set to %NULL. This can be useful if you need buffers to be released as soon
+as possible, eg. if you're using a buffer pool.
+
+
+
+ The last buffer that arrived in the sink and was used for preroll or for
+rendering. This property can be used to generate thumbnails. This property
+can be %NULL when the sink has not yet received a buffer.
+
+
+
+ Control the maximum amount of bits that will be rendered per second.
+Setting this property to a value bigger than 0 will make the sink delay
+rendering of the buffers when it would exceed to max-bitrate.
+
+
+
+
+
+
+
+
+
+ The additional delay between synchronisation and actual rendering of the
+media. This property will add additional latency to the device in order to
+make other sinks compensate for the delay.
+
+
+
+
+
+
+ The time to insert between buffers. This property can be used to control
+the maximum amount of buffers per second to render. Setting this property
+to a value bigger than 0 will make the sink create THROTTLE QoS events.
+
+
+
+ Controls the final synchronisation, a negative value will render the buffer
+earlier while a positive value delays playback. This property can be
+used to fix synchronisation in bad files.
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Subclasses can override any of the available virtual methods or not, as
+needed. At the minimum, the @render method should be overridden to
+output/present buffers.
+
+ Element parent class
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ This is a generic base class for source elements. The following
+types of sources are supported:
+
+ * random access sources like files
+ * seekable sources
+ * live sources
+
+The source can be configured to operate in any #GstFormat with the
+gst_base_src_set_format() method. The currently set format determines
+the format of the internal #GstSegment and any %GST_EVENT_SEGMENT
+events. The default format for #GstBaseSrc is %GST_FORMAT_BYTES.
+
+#GstBaseSrc always supports push mode scheduling. If the following
+conditions are met, it also supports pull mode scheduling:
+
+ * The format is set to %GST_FORMAT_BYTES (default).
+ * #GstBaseSrcClass.is_seekable() returns %TRUE.
+
+If all the conditions are met for operating in pull mode, #GstBaseSrc is
+automatically seekable in push mode as well. The following conditions must
+be met to make the element seekable in push mode when the format is not
+%GST_FORMAT_BYTES:
+
+* #GstBaseSrcClass.is_seekable() returns %TRUE.
+* #GstBaseSrcClass.query() can convert all supported seek formats to the
+ internal format as set with gst_base_src_set_format().
+* #GstBaseSrcClass.do_seek() is implemented, performs the seek and returns
+ %TRUE.
+
+When the element does not meet the requirements to operate in pull mode, the
+offset and length in the #GstBaseSrcClass.create() method should be ignored.
+It is recommended to subclass #GstPushSrc instead, in this situation. If the
+element can operate in pull mode but only with specific offsets and
+lengths, it is allowed to generate an error when the wrong values are passed
+to the #GstBaseSrcClass.create() function.
+
+#GstBaseSrc has support for live sources. Live sources are sources that when
+paused discard data, such as audio or video capture devices. A typical live
+source also produces data at a fixed rate and thus provides a clock to publish
+this rate.
+Use gst_base_src_set_live() to activate the live source mode.
+
+A live source does not produce data in the PAUSED state. This means that the
+#GstBaseSrcClass.create() method will not be called in PAUSED but only in
+PLAYING. To signal the pipeline that the element will not produce data, the
+return value from the READY to PAUSED state will be
+%GST_STATE_CHANGE_NO_PREROLL.
+
+A typical live source will timestamp the buffers it creates with the
+current running time of the pipeline. This is one reason why a live source
+can only produce data in the PLAYING state, when the clock is actually
+distributed and running.
+
+Live sources that synchronize and block on the clock (an audio source, for
+example) can use gst_base_src_wait_playing() when the
+#GstBaseSrcClass.create() function was interrupted by a state change to
+PAUSED.
+
+The #GstBaseSrcClass.get_times() method can be used to implement pseudo-live
+sources. It only makes sense to implement the #GstBaseSrcClass.get_times()
+function if the source is a live source. The #GstBaseSrcClass.get_times()
+function should return timestamps starting from 0, as if it were a non-live
+source. The base class will make sure that the timestamps are transformed
+into the current running_time. The base source will then wait for the
+calculated running_time before pushing out the buffer.
+
+For live sources, the base class will by default report a latency of 0.
+For pseudo live sources, the base class will by default measure the difference
+between the first buffer timestamp and the start time of get_times and will
+report this value as the latency.
+Subclasses should override the query function when this behaviour is not
+acceptable.
+
+There is only support in #GstBaseSrc for exactly one source pad, which
+should be named "src". A source implementation (subclass of #GstBaseSrc)
+should install a pad template in its class_init function, like so:
+|[<!-- language="C" -->
+static void
+my_element_class_init (GstMyElementClass *klass)
+{
+ GstElementClass *gstelement_class = GST_ELEMENT_CLASS (klass);
+ // srctemplate should be a #GstStaticPadTemplate with direction
+ // %GST_PAD_SRC and name "src"
+ gst_element_class_add_static_pad_template (gstelement_class, &srctemplate);
+
+ gst_element_class_set_static_metadata (gstelement_class,
+ "Source name",
+ "Source",
+ "My Source element",
+ "The author <my.sink@my.email>");
+}
+]|
+
+## Controlled shutdown of live sources in applications
+
+Applications that record from a live source may want to stop recording
+in a controlled way, so that the recording is stopped, but the data
+already in the pipeline is processed to the end (remember that many live
+sources would go on recording forever otherwise). For that to happen the
+application needs to make the source stop recording and send an EOS
+event down the pipeline. The application would then wait for an
+EOS message posted on the pipeline's bus to know when all data has
+been processed and the pipeline can safely be stopped.
+
+An application may send an EOS event to a source element to make it
+perform the EOS logic (send EOS event downstream or post a
+%GST_MESSAGE_SEGMENT_DONE on the bus). This can typically be done
+with the gst_element_send_event() function on the element or its parent bin.
+
+After the EOS has been sent to the element, the application should wait for
+an EOS message to be posted on the pipeline's bus. Once this EOS message is
+received, it may safely shut down the entire pipeline.
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Ask the subclass to create a buffer with @offset and @size, the default
+implementation will call alloc and fill.
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Given @buffer, return @start and @end time when it should be pushed
+out. The base class will sync on the clock using these times.
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Set new caps on the basesrc source pad.
+
+ %TRUE if the caps could be set
+
+
+
+
+ a #GstBaseSrc
+
+
+
+ a #GstCaps
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Lets #GstBaseSrc sub-classes to know the memory @allocator
+used by the base class and its @params.
+
+Unref the @allocator after usage.
+
+
+
+
+
+ a #GstBaseSrc
+
+
+
+ the #GstAllocator
+used
+
+
+
+ the
+#GstAllocationParams of @allocator
+
+
+
+
+
+ Get the number of bytes that @src will push out with each buffer.
+
+ the number of bytes pushed with each buffer.
+
+
+
+
+ the source
+
+
+
+
+
+
+ the instance of the #GstBufferPool used
+by the src; unref it after usage.
+
+
+
+
+ a #GstBaseSrc
+
+
+
+
+
+ Query if @src timestamps outgoing buffers based on the current running_time.
+
+ %TRUE if the base class will automatically timestamp outgoing buffers.
+
+
+
+
+ the source
+
+
+
+
+
+ Get the current async behaviour of @src. See also gst_base_src_set_async().
+
+ %TRUE if @src is operating in async mode.
+
+
+
+
+ base source instance
+
+
+
+
+
+ Check if an element is in live mode.
+
+ %TRUE if element is in live mode.
+
+
+
+
+ base source instance
+
+
+
+
+
+ Prepare a new seamless segment for emission downstream. This function must
+only be called by derived sub-classes, and only from the create() function,
+as the stream-lock needs to be held.
+
+The format for the new segment will be the current format of the source, as
+configured with gst_base_src_set_format()
+
+ %TRUE if preparation of the seamless segment succeeded.
+
+
+
+
+ The source
+
+
+
+ The new start value for the segment
+
+
+
+ Stop value for the new segment
+
+
+
+ The new time value for the start of the new segment
+
+
+
+
+
+ Query the source for the latency parameters. @live will be %TRUE when @src is
+configured as a live source. @min_latency and @max_latency will be set
+to the difference between the running time and the timestamp of the first
+buffer.
+
+This function is mostly used by subclasses.
+
+ %TRUE if the query succeeded.
+
+
+
+
+ the source
+
+
+
+ if the source is live
+
+
+
+ the min latency of the source
+
+
+
+ the max latency of the source
+
+
+
+
+
+ Configure async behaviour in @src, no state change will block. The open,
+close, start, stop, play and pause virtual methods will be executed in a
+different thread and are thus allowed to perform blocking operations. Any
+blocking operation should be unblocked with the unlock vmethod.
+
+
+
+
+
+ base source instance
+
+
+
+ new async mode
+
+
+
+
+
+ If @automatic_eos is %TRUE, @src will automatically go EOS if a buffer
+after the total size is returned. By default this is %TRUE but sources
+that can't return an authoritative size and only know that they're EOS
+when trying to read more should set this to %FALSE.
+
+
+
+
+
+ base source instance
+
+
+
+ automatic eos
+
+
+
+
+
+ Set the number of bytes that @src will push out with each buffer. When
+@blocksize is set to -1, a default length will be used.
+
+
+
+
+
+ the source
+
+
+
+ the new blocksize in bytes
+
+
+
+
+
+ Set new caps on the basesrc source pad.
+
+ %TRUE if the caps could be set
+
+
+
+
+ a #GstBaseSrc
+
+
+
+ a #GstCaps
+
+
+
+
+
+ Configure @src to automatically timestamp outgoing buffers based on the
+current running_time of the pipeline. This property is mostly useful for live
+sources.
+
+
+
+
+
+ the source
+
+
+
+ enable or disable timestamping
+
+
+
+
+
+ If not @dynamic, size is only updated when needed, such as when trying to
+read past current tracked size. Otherwise, size is checked for upon each
+read.
+
+
+
+
+
+ base source instance
+
+
+
+ new dynamic size mode
+
+
+
+
+
+ Sets the default format of the source. This will be the format used
+for sending SEGMENT events and for performing seeks.
+
+If a format of GST_FORMAT_BYTES is set, the element will be able to
+operate in pull mode if the #GstBaseSrcClass.is_seekable() returns %TRUE.
+
+This function must only be called in states < %GST_STATE_PAUSED.
+
+
+
+
+
+ base source instance
+
+
+
+ the format to use
+
+
+
+
+
+ If the element listens to a live source, @live should
+be set to %TRUE.
+
+A live source will not produce data in the PAUSED state and
+will therefore not be able to participate in the PREROLL phase
+of a pipeline. To signal this fact to the application and the
+pipeline, the state change return value of the live source will
+be GST_STATE_CHANGE_NO_PREROLL.
+
+
+
+
+
+ base source instance
+
+
+
+ new live-mode
+
+
+
+
+
+ Complete an asynchronous start operation. When the subclass overrides the
+start method, it should call gst_base_src_start_complete() when the start
+operation completes either from the same thread or from an asynchronous
+helper thread.
+
+
+
+
+
+ base source instance
+
+
+
+ a #GstFlowReturn
+
+
+
+
+
+ Wait until the start operation completes.
+
+ a #GstFlowReturn.
+
+
+
+
+ base source instance
+
+
+
+
+
+ Subclasses can call this from their create virtual method implementation
+to submit a buffer list to be pushed out later. This is useful in
+cases where the create function wants to produce multiple buffers to be
+pushed out in one go in form of a #GstBufferList, which can reduce overhead
+drastically, especially for packetised inputs (for data streams where
+the packetisation/chunking is not important it is usually more efficient
+to return larger buffers instead).
+
+Subclasses that use this function from their create function must return
+%GST_FLOW_OK and no buffer from their create virtual method implementation.
+If a buffer is returned after a buffer list has also been submitted via this
+function the behaviour is undefined.
+
+Subclasses must only call this function once per create function call and
+subclasses must only call this function when the source operates in push
+mode.
+
+
+
+
+
+ a #GstBaseSrc
+
+
+
+ a #GstBufferList
+
+
+
+
+
+ If the #GstBaseSrcClass.create() method performs its own synchronisation
+against the clock it must unblock when going from PLAYING to the PAUSED state
+and call this method before continuing to produce the remaining data.
+
+This function will block until a state change to PLAYING happens (in which
+case this function returns %GST_FLOW_OK) or the processing must be stopped due
+to a state change to READY or a FLUSH event (in which case this function
+returns %GST_FLOW_FLUSHING).
+
+ %GST_FLOW_OK if @src is PLAYING and processing can
+continue. Any other return value should be returned from the create vmethod.
+
+
+
+
+ the src
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Subclasses can override any of the available virtual methods or not, as
+needed. At the minimum, the @create method should be overridden to produce
+buffers.
+
+ Element parent class
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ %TRUE if the caps could be set
+
+
+
+
+ a #GstBaseSrc
+
+
+
+ a #GstCaps
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ The #GstElement flags that a basesrc element may have.
+
+ has source is starting
+
+
+ has source been started
+
+
+ offset to define more flags
+
+
+
+
+
+ This base class is for filter elements that process data. Elements
+that are suitable for implementation using #GstBaseTransform are ones
+where the size and caps of the output is known entirely from the input
+caps and buffer sizes. These include elements that directly transform
+one buffer into another, modify the contents of a buffer in-place, as
+well as elements that collate multiple input buffers into one output buffer,
+or that expand one input buffer into multiple output buffers. See below
+for more concrete use cases.
+
+It provides for:
+
+* one sinkpad and one srcpad
+* Possible formats on sink and source pad implemented
+ with custom transform_caps function. By default uses
+ same format on sink and source.
+
+* Handles state changes
+* Does flushing
+* Push mode
+* Pull mode if the sub-class transform can operate on arbitrary data
+
+# Use Cases
+
+## Passthrough mode
+
+ * Element has no interest in modifying the buffer. It may want to inspect it,
+ in which case the element should have a transform_ip function. If there
+ is no transform_ip function in passthrough mode, the buffer is pushed
+ intact.
+
+ * The #GstBaseTransformClass.passthrough_on_same_caps variable
+ will automatically set/unset passthrough based on whether the
+ element negotiates the same caps on both pads.
+
+ * #GstBaseTransformClass.passthrough_on_same_caps on an element that
+ doesn't implement a transform_caps function is useful for elements that
+ only inspect data (such as level)
+
+ * Example elements
+
+ * Level
+ * Videoscale, audioconvert, videoconvert, audioresample in certain modes.
+
+## Modifications in-place - input buffer and output buffer are the same thing.
+
+* The element must implement a transform_ip function.
+* Output buffer size must <= input buffer size
+* If the always_in_place flag is set, non-writable buffers will be copied
+ and passed to the transform_ip function, otherwise a new buffer will be
+ created and the transform function called.
+
+* Incoming writable buffers will be passed to the transform_ip function
+ immediately.
+* only implementing transform_ip and not transform implies always_in_place = %TRUE
+
+ * Example elements:
+ * Volume
+ * Audioconvert in certain modes (signed/unsigned conversion)
+ * videoconvert in certain modes (endianness swapping)
+
+## Modifications only to the caps/metadata of a buffer
+
+* The element does not require writable data, but non-writable buffers
+ should be subbuffered so that the meta-information can be replaced.
+
+* Elements wishing to operate in this mode should replace the
+ prepare_output_buffer method to create subbuffers of the input buffer
+ and set always_in_place to %TRUE
+
+* Example elements
+ * Capsfilter when setting caps on outgoing buffers that have
+ none.
+ * identity when it is going to re-timestamp buffers by
+ datarate.
+
+## Normal mode
+ * always_in_place flag is not set, or there is no transform_ip function
+ * Element will receive an input buffer and output buffer to operate on.
+ * Output buffer is allocated by calling the prepare_output_buffer function.
+ * Example elements:
+ * Videoscale, videoconvert, audioconvert when doing
+ scaling/conversions
+
+## Special output buffer allocations
+ * Elements which need to do special allocation of their output buffers
+ beyond allocating output buffers via the negotiated allocator or
+ buffer pool should implement the prepare_output_buffer method.
+
+ * Example elements:
+ * efence
+
+# Sub-class settable flags on GstBaseTransform
+
+* passthrough
+
+ * Implies that in the current configuration, the sub-class is not interested in modifying the buffers.
+ * Elements which are always in passthrough mode whenever the same caps has been negotiated on both pads can set the class variable passthrough_on_same_caps to have this behaviour automatically.
+
+* always_in_place
+ * Determines whether a non-writable buffer will be copied before passing
+ to the transform_ip function.
+
+ * Implied %TRUE if no transform function is implemented.
+ * Implied %FALSE if ONLY transform function is implemented.
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Lets #GstBaseTransform sub-classes to know the memory @allocator
+used by the base class and its @params.
+
+Unref the @allocator after use it.
+
+
+
+
+
+ a #GstBaseTransform
+
+
+
+ the #GstAllocator
+used
+
+
+
+ the
+#GstAllocationParams of @allocator
+
+
+
+
+
+
+ the instance of the #GstBufferPool used
+by @trans; free it after use it
+
+
+
+
+ a #GstBaseTransform
+
+
+
+
+
+ See if @trans is configured as a in_place transform.
+
+ %TRUE is the transform is configured in in_place mode.
+
+MT safe.
+
+
+
+
+ the #GstBaseTransform to query
+
+
+
+
+
+ See if @trans is configured as a passthrough transform.
+
+ %TRUE is the transform is configured in passthrough mode.
+
+MT safe.
+
+
+
+
+ the #GstBaseTransform to query
+
+
+
+
+
+ Queries if the transform will handle QoS.
+
+ %TRUE if QoS is enabled.
+
+MT safe.
+
+
+
+
+ a #GstBaseTransform
+
+
+
+
+
+ Instructs @trans to request renegotiation upstream. This function is
+typically called after properties on the transform were set that
+influence the input format.
+
+
+
+
+
+ a #GstBaseTransform
+
+
+
+
+
+ Instructs @trans to renegotiate a new downstream transform on the next
+buffer. This function is typically called after properties on the transform
+were set that influence the output format.
+
+
+
+
+
+ a #GstBaseTransform
+
+
+
+
+
+ If @gap_aware is %FALSE (the default), output buffers will have the
+%GST_BUFFER_FLAG_GAP flag unset.
+
+If set to %TRUE, the element must handle output buffers with this flag set
+correctly, i.e. it can assume that the buffer contains neutral data but must
+unset the flag if the output is no neutral data.
+
+MT safe.
+
+
+
+
+
+ a #GstBaseTransform
+
+
+
+ New state
+
+
+
+
+
+ Determines whether a non-writable buffer will be copied before passing
+to the transform_ip function.
+
+ * Always %TRUE if no transform function is implemented.
+ * Always %FALSE if ONLY transform function is implemented.
+
+MT safe.
+
+
+
+
+
+ the #GstBaseTransform to modify
+
+
+
+ Boolean value indicating that we would like to operate
+on in_place buffers.
+
+
+
+
+
+ Set passthrough mode for this filter by default. This is mostly
+useful for filters that do not care about negotiation.
+
+Always %TRUE for filters which don't implement either a transform
+or transform_ip method.
+
+MT safe.
+
+
+
+
+
+ the #GstBaseTransform to set
+
+
+
+ boolean indicating passthrough mode.
+
+
+
+
+
+ If @prefer_passthrough is %TRUE (the default), @trans will check and
+prefer passthrough caps from the list of caps returned by the
+transform_caps vmethod.
+
+If set to %FALSE, the element must order the caps returned from the
+transform_caps function in such a way that the preferred format is
+first in the list. This can be interesting for transforms that can do
+passthrough transforms but prefer to do something else, like a
+capsfilter.
+
+MT safe.
+
+
+
+
+
+ a #GstBaseTransform
+
+
+
+ New state
+
+
+
+
+
+ Enable or disable QoS handling in the transform.
+
+MT safe.
+
+
+
+
+
+ a #GstBaseTransform
+
+
+
+ new state
+
+
+
+
+
+ Set the QoS parameters in the transform. This function is called internally
+when a QOS event is received but subclasses can provide custom information
+when needed.
+
+MT safe.
+
+
+
+
+
+ a #GstBaseTransform
+
+
+
+ the proportion
+
+
+
+ the diff against the clock
+
+
+
+ the timestamp of the buffer generating the QoS expressed in
+running_time.
+
+
+
+
+
+ Updates the srcpad caps and send the caps downstream. This function
+can be used by subclasses when they have already negotiated their caps
+but found a change in them (or computed new informations). This way,
+they can notify downstream about that change without loosing any
+buffer.
+
+ %TRUE if the caps could be send downstream %FALSE otherwise
+
+
+
+
+ a #GstBaseTransform
+
+
+
+ An updated version of the srcpad caps to be pushed
+downstream
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Subclasses can override any of the available virtual methods or not, as
+needed. At minimum either @transform or @transform_ip need to be overridden.
+If the element can overwrite the input data with the results (data is of the
+same type and quantity) it should provide @transform_ip.
+
+ Element parent class
+
+
+
+ If set to %TRUE, passthrough mode will be
+ automatically enabled if the caps are the same.
+ Set to %FALSE by default.
+
+
+
+ If set to %TRUE, @transform_ip will be called in
+ passthrough mode. The passed buffer might not be
+ writable. When %FALSE, neither @transform nor
+ @transform_ip will be called in passthrough mode.
+ Set to %TRUE by default.
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ #GstBitReader provides a bit reader that can read any number of bits
+from a memory buffer. It provides functions for reading any number of bits
+into 8, 16, 32 and 64 bit variables.
+
+ Data from which the bit reader will
+ read
+
+
+
+
+
+ Size of @data in bytes
+
+
+
+ Current byte position
+
+
+
+ Bit position in the current byte
+
+
+
+
+
+
+
+
+ Frees a #GstBitReader instance, which was previously allocated by
+gst_bit_reader_new().
+
+
+
+
+
+ a #GstBitReader instance
+
+
+
+
+
+ Read @nbits bits into @val and update the current position.
+
+ %TRUE if successful, %FALSE otherwise.
+
+
+
+
+ a #GstBitReader instance
+
+
+
+ Pointer to a #guint16 to store the result
+
+
+
+ number of bits to read
+
+
+
+
+
+ Read @nbits bits into @val and update the current position.
+
+ %TRUE if successful, %FALSE otherwise.
+
+
+
+
+ a #GstBitReader instance
+
+
+
+ Pointer to a #guint32 to store the result
+
+
+
+ number of bits to read
+
+
+
+
+
+ Read @nbits bits into @val and update the current position.
+
+ %TRUE if successful, %FALSE otherwise.
+
+
+
+
+ a #GstBitReader instance
+
+
+
+ Pointer to a #guint64 to store the result
+
+
+
+ number of bits to read
+
+
+
+
+
+ Read @nbits bits into @val and update the current position.
+
+ %TRUE if successful, %FALSE otherwise.
+
+
+
+
+ a #GstBitReader instance
+
+
+
+ Pointer to a #guint8 to store the result
+
+
+
+ number of bits to read
+
+
+
+
+
+ Returns the current position of a #GstBitReader instance in bits.
+
+ The current position of @reader in bits.
+
+
+
+
+ a #GstBitReader instance
+
+
+
+
+
+ Returns the remaining number of bits of a #GstBitReader instance.
+
+ The remaining number of bits of @reader instance.
+
+
+
+
+ a #GstBitReader instance
+
+
+
+
+
+ Returns the total number of bits of a #GstBitReader instance.
+
+ The total number of bits of @reader instance.
+
+
+
+
+ a #GstBitReader instance
+
+
+
+
+
+ Initializes a #GstBitReader instance to read from @data. This function
+can be called on already initialized instances.
+
+
+
+
+
+ a #GstBitReader instance
+
+
+
+ data from which the bit reader should read
+
+
+
+
+
+ Size of @data in bytes
+
+
+
+
+
+ Read @nbits bits into @val but keep the current position.
+
+ %TRUE if successful, %FALSE otherwise.
+
+
+
+
+ a #GstBitReader instance
+
+
+
+ Pointer to a #guint16 to store the result
+
+
+
+ number of bits to read
+
+
+
+
+
+ Read @nbits bits into @val but keep the current position.
+
+ %TRUE if successful, %FALSE otherwise.
+
+
+
+
+ a #GstBitReader instance
+
+
+
+ Pointer to a #guint32 to store the result
+
+
+
+ number of bits to read
+
+
+
+
+
+ Read @nbits bits into @val but keep the current position.
+
+ %TRUE if successful, %FALSE otherwise.
+
+
+
+
+ a #GstBitReader instance
+
+
+
+ Pointer to a #guint64 to store the result
+
+
+
+ number of bits to read
+
+
+
+
+
+ Read @nbits bits into @val but keep the current position.
+
+ %TRUE if successful, %FALSE otherwise.
+
+
+
+
+ a #GstBitReader instance
+
+
+
+ Pointer to a #guint8 to store the result
+
+
+
+ number of bits to read
+
+
+
+
+
+ Sets the new position of a #GstBitReader instance to @pos in bits.
+
+ %TRUE if the position could be set successfully, %FALSE
+otherwise.
+
+
+
+
+ a #GstBitReader instance
+
+
+
+ The new position in bits
+
+
+
+
+
+ Skips @nbits bits of the #GstBitReader instance.
+
+ %TRUE if @nbits bits could be skipped, %FALSE otherwise.
+
+
+
+
+ a #GstBitReader instance
+
+
+
+ the number of bits to skip
+
+
+
+
+
+ Skips until the next byte.
+
+ %TRUE if successful, %FALSE otherwise.
+
+
+
+
+ a #GstBitReader instance
+
+
+
+
+
+ Create a new #GstBitReader instance, which will read from @data.
+
+Free-function: gst_bit_reader_free
+
+ a new #GstBitReader instance
+
+
+
+
+ Data from which the #GstBitReader
+ should read
+
+
+
+
+
+ Size of @data in bytes
+
+
+
+
+
+
+ #GstByteReader provides a byte reader that can read different integer and
+floating point types from a memory buffer. It provides functions for reading
+signed/unsigned, little/big endian integers of 8, 16, 24, 32 and 64 bits
+and functions for reading little/big endian floating points numbers of
+32 and 64 bits. It also provides functions to read NUL-terminated strings
+in various character encodings.
+
+ Data from which the bit reader will
+ read
+
+
+
+
+
+ Size of @data in bytes
+
+
+
+ Current byte position
+
+
+
+
+
+
+
+
+ Free-function: g_free
+
+Returns a newly-allocated copy of the current data
+position if at least @size bytes are left and
+updates the current position. Free with g_free() when no longer needed.
+
+ %TRUE if successful, %FALSE otherwise.
+
+
+
+
+ a #GstByteReader instance
+
+
+
+ Size in bytes
+
+
+
+ address of a
+ #guint8 pointer variable in which to store the result
+
+
+
+
+
+
+
+ Free-function: g_free
+
+Returns a newly-allocated copy of the current data position if there is
+a NUL-terminated UTF-16 string in the data (this could be an empty string
+as well), and advances the current position.
+
+No input checking for valid UTF-16 is done. This function is endianness
+agnostic - you should not assume the UTF-16 characters are in host
+endianness.
+
+This function will fail if no NUL-terminator was found in in the data.
+
+Note: there is no peek or get variant of this function to ensure correct
+byte alignment of the UTF-16 string.
+
+ %TRUE if a string could be read, %FALSE otherwise. The
+ string put into @str must be freed with g_free() when no longer needed.
+
+
+
+
+ a #GstByteReader instance
+
+
+
+ address of a
+ #guint16 pointer variable in which to store the result
+
+
+
+
+
+
+
+ Free-function: g_free
+
+Returns a newly-allocated copy of the current data position if there is
+a NUL-terminated UTF-32 string in the data (this could be an empty string
+as well), and advances the current position.
+
+No input checking for valid UTF-32 is done. This function is endianness
+agnostic - you should not assume the UTF-32 characters are in host
+endianness.
+
+This function will fail if no NUL-terminator was found in in the data.
+
+Note: there is no peek or get variant of this function to ensure correct
+byte alignment of the UTF-32 string.
+
+ %TRUE if a string could be read, %FALSE otherwise. The
+ string put into @str must be freed with g_free() when no longer needed.
+
+
+
+
+ a #GstByteReader instance
+
+
+
+ address of a
+ #guint32 pointer variable in which to store the result
+
+
+
+
+
+
+
+ Free-function: g_free
+
+FIXME:Reads (copies) a NUL-terminated string in the #GstByteReader instance,
+advancing the current position to the byte after the string. This will work
+for any NUL-terminated string with a character width of 8 bits, so ASCII,
+UTF-8, ISO-8859-N etc. No input checking for valid UTF-8 is done.
+
+This function will fail if no NUL-terminator was found in in the data.
+
+ %TRUE if a string could be read into @str, %FALSE otherwise. The
+ string put into @str must be freed with g_free() when no longer needed.
+
+
+
+
+ a #GstByteReader instance
+
+
+
+ address of a
+ #gchar pointer variable in which to store the result
+
+
+
+
+
+
+
+ Frees a #GstByteReader instance, which was previously allocated by
+gst_byte_reader_new().
+
+
+
+
+
+ a #GstByteReader instance
+
+
+
+
+
+ Returns a constant pointer to the current data
+position if at least @size bytes are left and
+updates the current position.
+
+ %TRUE if successful, %FALSE otherwise.
+
+
+
+
+ a #GstByteReader instance
+
+
+
+ Size in bytes
+
+
+
+ address of a
+ #guint8 pointer variable in which to store the result
+
+
+
+
+
+
+
+ Read a 32 bit big endian floating point value into @val
+and update the current position.
+
+ %TRUE if successful, %FALSE otherwise.
+
+
+
+
+ a #GstByteReader instance
+
+
+
+ Pointer to a #gfloat to store the result
+
+
+
+
+
+ Read a 32 bit little endian floating point value into @val
+and update the current position.
+
+ %TRUE if successful, %FALSE otherwise.
+
+
+
+
+ a #GstByteReader instance
+
+
+
+ Pointer to a #gfloat to store the result
+
+
+
+
+
+ Read a 64 bit big endian floating point value into @val
+and update the current position.
+
+ %TRUE if successful, %FALSE otherwise.
+
+
+
+
+ a #GstByteReader instance
+
+
+
+ Pointer to a #gdouble to store the result
+
+
+
+
+
+ Read a 64 bit little endian floating point value into @val
+and update the current position.
+
+ %TRUE if successful, %FALSE otherwise.
+
+
+
+
+ a #GstByteReader instance
+
+
+
+ Pointer to a #gdouble to store the result
+
+
+
+
+
+ Read a signed 16 bit big endian integer into @val
+and update the current position.
+
+ %TRUE if successful, %FALSE otherwise.
+
+
+
+
+ a #GstByteReader instance
+
+
+
+ Pointer to a #gint16 to store the result
+
+
+
+
+
+ Read a signed 16 bit little endian integer into @val
+and update the current position.
+
+ %TRUE if successful, %FALSE otherwise.
+
+
+
+
+ a #GstByteReader instance
+
+
+
+ Pointer to a #gint16 to store the result
+
+
+
+
+
+ Read a signed 24 bit big endian integer into @val
+and update the current position.
+
+ %TRUE if successful, %FALSE otherwise.
+
+
+
+
+ a #GstByteReader instance
+
+
+
+ Pointer to a #gint32 to store the result
+
+
+
+
+
+ Read a signed 24 bit little endian integer into @val
+and update the current position.
+
+ %TRUE if successful, %FALSE otherwise.
+
+
+
+
+ a #GstByteReader instance
+
+
+
+ Pointer to a #gint32 to store the result
+
+
+
+
+
+ Read a signed 32 bit big endian integer into @val
+and update the current position.
+
+ %TRUE if successful, %FALSE otherwise.
+
+
+
+
+ a #GstByteReader instance
+
+
+
+ Pointer to a #gint32 to store the result
+
+
+
+
+
+ Read a signed 32 bit little endian integer into @val
+and update the current position.
+
+ %TRUE if successful, %FALSE otherwise.
+
+
+
+
+ a #GstByteReader instance
+
+
+
+ Pointer to a #gint32 to store the result
+
+
+
+
+
+ Read a signed 64 bit big endian integer into @val
+and update the current position.
+
+ %TRUE if successful, %FALSE otherwise.
+
+
+
+
+ a #GstByteReader instance
+
+
+
+ Pointer to a #gint64 to store the result
+
+
+
+
+
+ Read a signed 64 bit little endian integer into @val
+and update the current position.
+
+ %TRUE if successful, %FALSE otherwise.
+
+
+
+
+ a #GstByteReader instance
+
+
+
+ Pointer to a #gint64 to store the result
+
+
+
+
+
+ Read a signed 8 bit integer into @val and update the current position.
+
+ %TRUE if successful, %FALSE otherwise.
+
+
+
+
+ a #GstByteReader instance
+
+
+
+ Pointer to a #gint8 to store the result
+
+
+
+
+
+ Returns the current position of a #GstByteReader instance in bytes.
+
+ The current position of @reader in bytes.
+
+
+
+
+ a #GstByteReader instance
+
+
+
+
+
+ Returns the remaining number of bytes of a #GstByteReader instance.
+
+ The remaining number of bytes of @reader instance.
+
+
+
+
+ a #GstByteReader instance
+
+
+
+
+
+ Returns the total number of bytes of a #GstByteReader instance.
+
+ The total number of bytes of @reader instance.
+
+
+
+
+ a #GstByteReader instance
+
+
+
+
+
+ Returns a constant pointer to the current data position if there is
+a NUL-terminated string in the data (this could be just a NUL terminator),
+advancing the current position to the byte after the string. This will work
+for any NUL-terminated string with a character width of 8 bits, so ASCII,
+UTF-8, ISO-8859-N etc.
+
+No input checking for valid UTF-8 is done.
+
+This function will fail if no NUL-terminator was found in in the data.
+
+ %TRUE if a string could be found, %FALSE otherwise.
+
+
+
+
+ a #GstByteReader instance
+
+
+
+ address of a
+ #gchar pointer variable in which to store the result
+
+
+
+
+
+
+
+ Initializes a #GstByteReader sub-reader instance to contain @size bytes of
+data from the current position of @reader. This is useful to read chunked
+formats and make sure that one doesn't read beyond the size of the sub-chunk.
+
+Unlike gst_byte_reader_peek_sub_reader(), this function also modifies the
+position of @reader and moves it forward by @size bytes.
+
+ FALSE on error or if @reader does not contain @size more bytes from
+ the current position, and otherwise TRUE
+
+
+
+
+ an existing and initialized #GstByteReader instance
+
+
+
+ a #GstByteReader instance to initialize as sub-reader
+
+
+
+ size of @sub_reader in bytes
+
+
+
+
+
+ Read an unsigned 16 bit big endian integer into @val
+and update the current position.
+
+ %TRUE if successful, %FALSE otherwise.
+
+
+
+
+ a #GstByteReader instance
+
+
+
+ Pointer to a #guint16 to store the result
+
+
+
+
+
+ Read an unsigned 16 bit little endian integer into @val
+and update the current position.
+
+ %TRUE if successful, %FALSE otherwise.
+
+
+
+
+ a #GstByteReader instance
+
+
+
+ Pointer to a #guint16 to store the result
+
+
+
+
+
+ Read an unsigned 24 bit big endian integer into @val
+and update the current position.
+
+ %TRUE if successful, %FALSE otherwise.
+
+
+
+
+ a #GstByteReader instance
+
+
+
+ Pointer to a #guint32 to store the result
+
+
+
+
+
+ Read an unsigned 24 bit little endian integer into @val
+and update the current position.
+
+ %TRUE if successful, %FALSE otherwise.
+
+
+
+
+ a #GstByteReader instance
+
+
+
+ Pointer to a #guint32 to store the result
+
+
+
+
+
+ Read an unsigned 32 bit big endian integer into @val
+and update the current position.
+
+ %TRUE if successful, %FALSE otherwise.
+
+
+
+
+ a #GstByteReader instance
+
+
+
+ Pointer to a #guint32 to store the result
+
+
+
+
+
+ Read an unsigned 32 bit little endian integer into @val
+and update the current position.
+
+ %TRUE if successful, %FALSE otherwise.
+
+
+
+
+ a #GstByteReader instance
+
+
+
+ Pointer to a #guint32 to store the result
+
+
+
+
+
+ Read an unsigned 64 bit big endian integer into @val
+and update the current position.
+
+ %TRUE if successful, %FALSE otherwise.
+
+
+
+
+ a #GstByteReader instance
+
+
+
+ Pointer to a #guint64 to store the result
+
+
+
+
+
+ Read an unsigned 64 bit little endian integer into @val
+and update the current position.
+
+ %TRUE if successful, %FALSE otherwise.
+
+
+
+
+ a #GstByteReader instance
+
+
+
+ Pointer to a #guint64 to store the result
+
+
+
+
+
+ Read an unsigned 8 bit integer into @val and update the current position.
+
+ %TRUE if successful, %FALSE otherwise.
+
+
+
+
+ a #GstByteReader instance
+
+
+
+ Pointer to a #guint8 to store the result
+
+
+
+
+
+ Initializes a #GstByteReader instance to read from @data. This function
+can be called on already initialized instances.
+
+
+
+
+
+ a #GstByteReader instance
+
+
+
+ data from which
+ the #GstByteReader should read
+
+
+
+
+
+ Size of @data in bytes
+
+
+
+
+
+ Scan for pattern @pattern with applied mask @mask in the byte reader data,
+starting from offset @offset relative to the current position.
+
+The bytes in @pattern and @mask are interpreted left-to-right, regardless
+of endianness. All four bytes of the pattern must be present in the
+byte reader data for it to match, even if the first or last bytes are masked
+out.
+
+It is an error to call this function without making sure that there is
+enough data (offset+size bytes) in the byte reader.
+
+ offset of the first match, or -1 if no match was found.
+
+Example:
+|[
+// Assume the reader contains 0x00 0x01 0x02 ... 0xfe 0xff
+
+gst_byte_reader_masked_scan_uint32 (reader, 0xffffffff, 0x00010203, 0, 256);
+// -> returns 0
+gst_byte_reader_masked_scan_uint32 (reader, 0xffffffff, 0x00010203, 1, 255);
+// -> returns -1
+gst_byte_reader_masked_scan_uint32 (reader, 0xffffffff, 0x01020304, 1, 255);
+// -> returns 1
+gst_byte_reader_masked_scan_uint32 (reader, 0xffff, 0x0001, 0, 256);
+// -> returns -1
+gst_byte_reader_masked_scan_uint32 (reader, 0xffff, 0x0203, 0, 256);
+// -> returns 0
+gst_byte_reader_masked_scan_uint32 (reader, 0xffff0000, 0x02030000, 0, 256);
+// -> returns 2
+gst_byte_reader_masked_scan_uint32 (reader, 0xffff0000, 0x02030000, 0, 4);
+// -> returns -1
+]|
+
+
+
+
+ a #GstByteReader
+
+
+
+ mask to apply to data before matching against @pattern
+
+
+
+ pattern to match (after mask is applied)
+
+
+
+ offset from which to start scanning, relative to the current
+ position
+
+
+
+ number of bytes to scan from offset
+
+
+
+
+
+ Scan for pattern @pattern with applied mask @mask in the byte reader data,
+starting from offset @offset relative to the current position.
+
+The bytes in @pattern and @mask are interpreted left-to-right, regardless
+of endianness. All four bytes of the pattern must be present in the
+byte reader data for it to match, even if the first or last bytes are masked
+out.
+
+It is an error to call this function without making sure that there is
+enough data (offset+size bytes) in the byte reader.
+
+ offset of the first match, or -1 if no match was found.
+
+
+
+
+ a #GstByteReader
+
+
+
+ mask to apply to data before matching against @pattern
+
+
+
+ pattern to match (after mask is applied)
+
+
+
+ offset from which to start scanning, relative to the current
+ position
+
+
+
+ number of bytes to scan from offset
+
+
+
+ pointer to uint32 to return matching data
+
+
+
+
+
+ Returns a constant pointer to the current data
+position if at least @size bytes are left and
+keeps the current position.
+
+ %TRUE if successful, %FALSE otherwise.
+
+
+
+
+ a #GstByteReader instance
+
+
+
+ Size in bytes
+
+
+
+ address of a
+ #guint8 pointer variable in which to store the result
+
+
+
+
+
+
+
+ Read a 32 bit big endian floating point value into @val
+but keep the current position.
+
+ %TRUE if successful, %FALSE otherwise.
+
+
+
+
+ a #GstByteReader instance
+
+
+
+ Pointer to a #gfloat to store the result
+
+
+
+
+
+ Read a 32 bit little endian floating point value into @val
+but keep the current position.
+
+ %TRUE if successful, %FALSE otherwise.
+
+
+
+
+ a #GstByteReader instance
+
+
+
+ Pointer to a #gfloat to store the result
+
+
+
+
+
+ Read a 64 bit big endian floating point value into @val
+but keep the current position.
+
+ %TRUE if successful, %FALSE otherwise.
+
+
+
+
+ a #GstByteReader instance
+
+
+
+ Pointer to a #gdouble to store the result
+
+
+
+
+
+ Read a 64 bit little endian floating point value into @val
+but keep the current position.
+
+ %TRUE if successful, %FALSE otherwise.
+
+
+
+
+ a #GstByteReader instance
+
+
+
+ Pointer to a #gdouble to store the result
+
+
+
+
+
+ Read a signed 16 bit big endian integer into @val
+but keep the current position.
+
+ %TRUE if successful, %FALSE otherwise.
+
+
+
+
+ a #GstByteReader instance
+
+
+
+ Pointer to a #gint16 to store the result
+
+
+
+
+
+ Read a signed 16 bit little endian integer into @val
+but keep the current position.
+
+ %TRUE if successful, %FALSE otherwise.
+
+
+
+
+ a #GstByteReader instance
+
+
+
+ Pointer to a #gint16 to store the result
+
+
+
+
+
+ Read a signed 24 bit big endian integer into @val
+but keep the current position.
+
+ %TRUE if successful, %FALSE otherwise.
+
+
+
+
+ a #GstByteReader instance
+
+
+
+ Pointer to a #gint32 to store the result
+
+
+
+
+
+ Read a signed 24 bit little endian integer into @val
+but keep the current position.
+
+ %TRUE if successful, %FALSE otherwise.
+
+
+
+
+ a #GstByteReader instance
+
+
+
+ Pointer to a #gint32 to store the result
+
+
+
+
+
+ Read a signed 32 bit big endian integer into @val
+but keep the current position.
+
+ %TRUE if successful, %FALSE otherwise.
+
+
+
+
+ a #GstByteReader instance
+
+
+
+ Pointer to a #gint32 to store the result
+
+
+
+
+
+ Read a signed 32 bit little endian integer into @val
+but keep the current position.
+
+ %TRUE if successful, %FALSE otherwise.
+
+
+
+
+ a #GstByteReader instance
+
+
+
+ Pointer to a #gint32 to store the result
+
+
+
+
+
+ Read a signed 64 bit big endian integer into @val
+but keep the current position.
+
+ %TRUE if successful, %FALSE otherwise.
+
+
+
+
+ a #GstByteReader instance
+
+
+
+ Pointer to a #gint64 to store the result
+
+
+
+
+
+ Read a signed 64 bit little endian integer into @val
+but keep the current position.
+
+ %TRUE if successful, %FALSE otherwise.
+
+
+
+
+ a #GstByteReader instance
+
+
+
+ Pointer to a #gint64 to store the result
+
+
+
+
+
+ Read a signed 8 bit integer into @val but keep the current position.
+
+ %TRUE if successful, %FALSE otherwise.
+
+
+
+
+ a #GstByteReader instance
+
+
+
+ Pointer to a #gint8 to store the result
+
+
+
+
+
+ Returns a constant pointer to the current data position if there is
+a NUL-terminated string in the data (this could be just a NUL terminator).
+The current position will be maintained. This will work for any
+NUL-terminated string with a character width of 8 bits, so ASCII,
+UTF-8, ISO-8859-N etc.
+
+No input checking for valid UTF-8 is done.
+
+This function will fail if no NUL-terminator was found in in the data.
+
+ %TRUE if a string could be skipped, %FALSE otherwise.
+
+
+
+
+ a #GstByteReader instance
+
+
+
+ address of a
+ #gchar pointer variable in which to store the result
+
+
+
+
+
+
+
+ Initializes a #GstByteReader sub-reader instance to contain @size bytes of
+data from the current position of @reader. This is useful to read chunked
+formats and make sure that one doesn't read beyond the size of the sub-chunk.
+
+Unlike gst_byte_reader_get_sub_reader(), this function does not modify the
+current position of @reader.
+
+ FALSE on error or if @reader does not contain @size more bytes from
+ the current position, and otherwise TRUE
+
+
+
+
+ an existing and initialized #GstByteReader instance
+
+
+
+ a #GstByteReader instance to initialize as sub-reader
+
+
+
+ size of @sub_reader in bytes
+
+
+
+
+
+ Read an unsigned 16 bit big endian integer into @val
+but keep the current position.
+
+ %TRUE if successful, %FALSE otherwise.
+
+
+
+
+ a #GstByteReader instance
+
+
+
+ Pointer to a #guint16 to store the result
+
+
+
+
+
+ Read an unsigned 16 bit little endian integer into @val
+but keep the current position.
+
+ %TRUE if successful, %FALSE otherwise.
+
+
+
+
+ a #GstByteReader instance
+
+
+
+ Pointer to a #guint16 to store the result
+
+
+
+
+
+ Read an unsigned 24 bit big endian integer into @val
+but keep the current position.
+
+ %TRUE if successful, %FALSE otherwise.
+
+
+
+
+ a #GstByteReader instance
+
+
+
+ Pointer to a #guint32 to store the result
+
+
+
+
+
+ Read an unsigned 24 bit little endian integer into @val
+but keep the current position.
+
+ %TRUE if successful, %FALSE otherwise.
+
+
+
+
+ a #GstByteReader instance
+
+
+
+ Pointer to a #guint32 to store the result
+
+
+
+
+
+ Read an unsigned 32 bit big endian integer into @val
+but keep the current position.
+
+ %TRUE if successful, %FALSE otherwise.
+
+
+
+
+ a #GstByteReader instance
+
+
+
+ Pointer to a #guint32 to store the result
+
+
+
+
+
+ Read an unsigned 32 bit little endian integer into @val
+but keep the current position.
+
+ %TRUE if successful, %FALSE otherwise.
+
+
+
+
+ a #GstByteReader instance
+
+
+
+ Pointer to a #guint32 to store the result
+
+
+
+
+
+ Read an unsigned 64 bit big endian integer into @val
+but keep the current position.
+
+ %TRUE if successful, %FALSE otherwise.
+
+
+
+
+ a #GstByteReader instance
+
+
+
+ Pointer to a #guint64 to store the result
+
+
+
+
+
+ Read an unsigned 64 bit little endian integer into @val
+but keep the current position.
+
+ %TRUE if successful, %FALSE otherwise.
+
+
+
+
+ a #GstByteReader instance
+
+
+
+ Pointer to a #guint64 to store the result
+
+
+
+
+
+ Read an unsigned 8 bit integer into @val but keep the current position.
+
+ %TRUE if successful, %FALSE otherwise.
+
+
+
+
+ a #GstByteReader instance
+
+
+
+ Pointer to a #guint8 to store the result
+
+
+
+
+
+ Sets the new position of a #GstByteReader instance to @pos in bytes.
+
+ %TRUE if the position could be set successfully, %FALSE
+otherwise.
+
+
+
+
+ a #GstByteReader instance
+
+
+
+ The new position in bytes
+
+
+
+
+
+ Skips @nbytes bytes of the #GstByteReader instance.
+
+ %TRUE if @nbytes bytes could be skipped, %FALSE otherwise.
+
+
+
+
+ a #GstByteReader instance
+
+
+
+ the number of bytes to skip
+
+
+
+
+
+ Skips a NUL-terminated UTF-16 string in the #GstByteReader instance,
+advancing the current position to the byte after the string.
+
+No input checking for valid UTF-16 is done.
+
+This function will fail if no NUL-terminator was found in in the data.
+
+ %TRUE if a string could be skipped, %FALSE otherwise.
+
+
+
+
+ a #GstByteReader instance
+
+
+
+
+
+ Skips a NUL-terminated UTF-32 string in the #GstByteReader instance,
+advancing the current position to the byte after the string.
+
+No input checking for valid UTF-32 is done.
+
+This function will fail if no NUL-terminator was found in in the data.
+
+ %TRUE if a string could be skipped, %FALSE otherwise.
+
+
+
+
+ a #GstByteReader instance
+
+
+
+
+
+ Skips a NUL-terminated string in the #GstByteReader instance, advancing
+the current position to the byte after the string. This will work for
+any NUL-terminated string with a character width of 8 bits, so ASCII,
+UTF-8, ISO-8859-N etc. No input checking for valid UTF-8 is done.
+
+This function will fail if no NUL-terminator was found in in the data.
+
+ %TRUE if a string could be skipped, %FALSE otherwise.
+
+
+
+
+ a #GstByteReader instance
+
+
+
+
+
+ Create a new #GstByteReader instance, which will read from @data.
+
+Free-function: gst_byte_reader_free
+
+ a new #GstByteReader instance
+
+
+
+
+ data from which the
+ #GstByteReader should read
+
+
+
+
+
+ Size of @data in bytes
+
+
+
+
+
+
+ #GstByteWriter provides a byte writer and reader that can write/read different
+integer and floating point types to/from a memory buffer. It provides functions
+for writing/reading signed/unsigned, little/big endian integers of 8, 16, 24,
+32 and 64 bits and functions for reading little/big endian floating points numbers of
+32 and 64 bits. It also provides functions to write/read NUL-terminated strings
+in various character encodings.
+
+ #GstByteReader parent
+
+
+
+ Allocation size of the data
+
+
+
+ If %TRUE no reallocations are allowed
+
+
+
+ If %FALSE no reallocations are allowed and copies of data are returned
+
+
+
+
+
+
+
+
+ Checks if enough free space from the current write cursor is
+available and reallocates if necessary.
+
+ %TRUE if at least @size bytes are still available
+
+
+
+
+ #GstByteWriter instance
+
+
+
+ Number of bytes that should be available
+
+
+
+
+
+ Writes @size bytes containing @value to @writer.
+
+ %TRUE if the value could be written
+
+
+
+
+ #GstByteWriter instance
+
+
+
+ Value to be written
+
+
+
+ Number of bytes to be written
+
+
+
+
+
+ Frees @writer and all memory allocated by it.
+
+
+
+
+
+ #GstByteWriter instance
+
+
+
+
+
+ Frees @writer and all memory allocated by it except
+the current data, which is returned as #GstBuffer.
+
+Free-function: gst_buffer_unref
+
+ the current data as buffer. gst_buffer_unref()
+ after usage.
+
+
+
+
+ #GstByteWriter instance
+
+
+
+
+
+ Frees @writer and all memory allocated by it except
+the current data, which is returned.
+
+Free-function: g_free
+
+ the current data. g_free() after usage.
+
+
+
+
+ #GstByteWriter instance
+
+
+
+
+
+ Returns the remaining size of data that can still be written. If
+-1 is returned the remaining size is only limited by system resources.
+
+ the remaining size of data that can still be written
+
+
+
+
+ #GstByteWriter instance
+
+
+
+
+
+ Initializes @writer to an empty instance
+
+
+
+
+
+ #GstByteWriter instance
+
+
+
+
+
+ Initializes @writer with the given
+memory area. If @initialized is %TRUE it is possible to
+read @size bytes from the #GstByteWriter from the beginning.
+
+
+
+
+
+ #GstByteWriter instance
+
+
+
+ Memory area for writing
+
+
+
+
+
+ Size of @data in bytes
+
+
+
+ If %TRUE the complete data can be read from the beginning
+
+
+
+
+
+ Initializes @writer with the given initial data size.
+
+
+
+
+
+ #GstByteWriter instance
+
+
+
+ Initial size of data
+
+
+
+ If %TRUE the data can't be reallocated
+
+
+
+
+
+ Writes @size bytes of @data to @writer.
+
+ %TRUE if the data could be written
+
+
+
+
+ #GstByteWriter instance
+
+
+
+ source #GstBuffer
+
+
+
+ offset to copy from
+
+
+
+ total size to copy. If -1, all data is copied
+
+
+
+
+
+ Writes @size bytes of @data to @writer.
+
+ %TRUE if the value could be written
+
+
+
+
+ #GstByteWriter instance
+
+
+
+ Data to write
+
+
+
+
+
+ Size of @data in bytes
+
+
+
+
+
+ Writes a big endian 32 bit float to @writer.
+
+ %TRUE if the value could be written
+
+
+
+
+ #GstByteWriter instance
+
+
+
+ Value to write
+
+
+
+
+
+ Writes a little endian 32 bit float to @writer.
+
+ %TRUE if the value could be written
+
+
+
+
+ #GstByteWriter instance
+
+
+
+ Value to write
+
+
+
+
+
+ Writes a big endian 64 bit float to @writer.
+
+ %TRUE if the value could be written
+
+
+
+
+ #GstByteWriter instance
+
+
+
+ Value to write
+
+
+
+
+
+ Writes a little endian 64 bit float to @writer.
+
+ %TRUE if the value could be written
+
+
+
+
+ #GstByteWriter instance
+
+
+
+ Value to write
+
+
+
+
+
+ Writes a signed big endian 16 bit integer to @writer.
+
+ %TRUE if the value could be written
+
+
+
+
+ #GstByteWriter instance
+
+
+
+ Value to write
+
+
+
+
+
+ Writes a signed little endian 16 bit integer to @writer.
+
+ %TRUE if the value could be written
+
+
+
+
+ #GstByteWriter instance
+
+
+
+ Value to write
+
+
+
+
+
+ Writes a signed big endian 24 bit integer to @writer.
+
+ %TRUE if the value could be written
+
+
+
+
+ #GstByteWriter instance
+
+
+
+ Value to write
+
+
+
+
+
+ Writes a signed little endian 24 bit integer to @writer.
+
+ %TRUE if the value could be written
+
+
+
+
+ #GstByteWriter instance
+
+
+
+ Value to write
+
+
+
+
+
+ Writes a signed big endian 32 bit integer to @writer.
+
+ %TRUE if the value could be written
+
+
+
+
+ #GstByteWriter instance
+
+
+
+ Value to write
+
+
+
+
+
+ Writes a signed little endian 32 bit integer to @writer.
+
+ %TRUE if the value could be written
+
+
+
+
+ #GstByteWriter instance
+
+
+
+ Value to write
+
+
+
+
+
+ Writes a signed big endian 64 bit integer to @writer.
+
+ %TRUE if the value could be written
+
+
+
+
+ #GstByteWriter instance
+
+
+
+ Value to write
+
+
+
+
+
+ Writes a signed little endian 64 bit integer to @writer.
+
+ %TRUE if the value could be written
+
+
+
+
+ #GstByteWriter instance
+
+
+
+ Value to write
+
+
+
+
+
+ Writes a signed 8 bit integer to @writer.
+
+ %TRUE if the value could be written
+
+
+
+
+ #GstByteWriter instance
+
+
+
+ Value to write
+
+
+
+
+
+ Writes a NUL-terminated UTF16 string to @writer (including the terminator).
+
+ %TRUE if the value could be written
+
+
+
+
+ #GstByteWriter instance
+
+
+
+ UTF16 string to write
+
+
+
+
+
+
+
+ Writes a NUL-terminated UTF32 string to @writer (including the terminator).
+
+ %TRUE if the value could be written
+
+
+
+
+ #GstByteWriter instance
+
+
+
+ UTF32 string to write
+
+
+
+
+
+
+
+ Writes a NUL-terminated UTF8 string to @writer (including the terminator).
+
+ %TRUE if the value could be written
+
+
+
+
+ #GstByteWriter instance
+
+
+
+ UTF8 string to write
+
+
+
+
+
+ Writes a unsigned big endian 16 bit integer to @writer.
+
+ %TRUE if the value could be written
+
+
+
+
+ #GstByteWriter instance
+
+
+
+ Value to write
+
+
+
+
+
+ Writes a unsigned little endian 16 bit integer to @writer.
+
+ %TRUE if the value could be written
+
+
+
+
+ #GstByteWriter instance
+
+
+
+ Value to write
+
+
+
+
+
+ Writes a unsigned big endian 24 bit integer to @writer.
+
+ %TRUE if the value could be written
+
+
+
+
+ #GstByteWriter instance
+
+
+
+ Value to write
+
+
+
+
+
+ Writes a unsigned little endian 24 bit integer to @writer.
+
+ %TRUE if the value could be written
+
+
+
+
+ #GstByteWriter instance
+
+
+
+ Value to write
+
+
+
+
+
+ Writes a unsigned big endian 32 bit integer to @writer.
+
+ %TRUE if the value could be written
+
+
+
+
+ #GstByteWriter instance
+
+
+
+ Value to write
+
+
+
+
+
+ Writes a unsigned little endian 32 bit integer to @writer.
+
+ %TRUE if the value could be written
+
+
+
+
+ #GstByteWriter instance
+
+
+
+ Value to write
+
+
+
+
+
+ Writes a unsigned big endian 64 bit integer to @writer.
+
+ %TRUE if the value could be written
+
+
+
+
+ #GstByteWriter instance
+
+
+
+ Value to write
+
+
+
+
+
+ Writes a unsigned little endian 64 bit integer to @writer.
+
+ %TRUE if the value could be written
+
+
+
+
+ #GstByteWriter instance
+
+
+
+ Value to write
+
+
+
+
+
+ Writes a unsigned 8 bit integer to @writer.
+
+ %TRUE if the value could be written
+
+
+
+
+ #GstByteWriter instance
+
+
+
+ Value to write
+
+
+
+
+
+ Resets @writer and frees the data if it's
+owned by @writer.
+
+
+
+
+
+ #GstByteWriter instance
+
+
+
+
+
+ Resets @writer and returns the current data as buffer.
+
+Free-function: gst_buffer_unref
+
+ the current data as buffer. gst_buffer_unref()
+ after usage.
+
+
+
+
+ #GstByteWriter instance
+
+
+
+
+
+ Resets @writer and returns the current data.
+
+Free-function: g_free
+
+ the current data. g_free() after
+usage.
+
+
+
+
+
+
+ #GstByteWriter instance
+
+
+
+
+
+ Creates a new, empty #GstByteWriter instance
+
+Free-function: gst_byte_writer_free
+
+ a new, empty #GstByteWriter instance
+
+
+
+
+ Creates a new #GstByteWriter instance with the given
+memory area. If @initialized is %TRUE it is possible to
+read @size bytes from the #GstByteWriter from the beginning.
+
+Free-function: gst_byte_writer_free
+
+ a new #GstByteWriter instance
+
+
+
+
+ Memory area for writing
+
+
+
+ Size of @data in bytes
+
+
+
+ If %TRUE the complete data can be read from the beginning
+
+
+
+
+
+ Creates a new #GstByteWriter instance with the given
+initial data size.
+
+Free-function: gst_byte_writer_free
+
+ a new #GstByteWriter instance
+
+
+
+
+ Initial size of data
+
+
+
+ If %TRUE the data can't be reallocated
+
+
+
+
+
+
+ Structure used by the collect_pads.
+
+ owner #GstCollectPads
+
+
+
+ #GstPad managed by this data
+
+
+
+ currently queued buffer.
+
+
+
+ position in the buffer
+
+
+
+ last segment received.
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ A function that will be called when the #GstCollectData will be freed.
+It is passed the pointer to the structure and should free any custom
+memory and resources allocated for it.
+
+
+
+
+
+ the #GstCollectData that will be freed
+
+
+
+
+
+
+
+ Manages a set of pads that operate in collect mode. This means that control
+is given to the manager of this object when all pads have data.
+
+ * Collectpads are created with gst_collect_pads_new(). A callback should then
+ be installed with gst_collect_pads_set_function ().
+
+ * Pads are added to the collection with gst_collect_pads_add_pad()/
+ gst_collect_pads_remove_pad(). The pad has to be a sinkpad. When added,
+ the chain, event and query functions of the pad are overridden. The
+ element_private of the pad is used to store private information for the
+ collectpads.
+
+ * For each pad, data is queued in the _chain function or by
+ performing a pull_range.
+
+ * When data is queued on all pads in waiting mode, the callback function is called.
+
+ * Data can be dequeued from the pad with the gst_collect_pads_pop() method.
+ One can peek at the data with the gst_collect_pads_peek() function.
+ These functions will return %NULL if the pad received an EOS event. When all
+ pads return %NULL from a gst_collect_pads_peek(), the element can emit an EOS
+ event itself.
+
+ * Data can also be dequeued in byte units using the gst_collect_pads_available(),
+ gst_collect_pads_read_buffer() and gst_collect_pads_flush() calls.
+
+ * Elements should call gst_collect_pads_start() and gst_collect_pads_stop() in
+ their state change functions to start and stop the processing of the collectpads.
+ The gst_collect_pads_stop() call should be called before calling the parent
+ element state change function in the PAUSED_TO_READY state change to ensure
+ no pad is blocked and the element can finish streaming.
+
+ * gst_collect_pads_set_waiting() sets a pad to waiting or non-waiting mode.
+ CollectPads element is not waiting for data to be collected on non-waiting pads.
+ Thus these pads may but need not have data when the callback is called.
+ All pads are in waiting mode by default.
+
+ Create a new instance of #GstCollectPads.
+
+MT safe.
+
+ a new #GstCollectPads, or %NULL in case of an error.
+
+
+
+
+ Add a pad to the collection of collect pads. The pad has to be
+a sinkpad. The refcount of the pad is incremented. Use
+gst_collect_pads_remove_pad() to remove the pad from the collection
+again.
+
+You specify a size for the returned #GstCollectData structure
+so that you can use it to store additional information.
+
+You can also specify a #GstCollectDataDestroyNotify that will be called
+just before the #GstCollectData structure is freed. It is passed the
+pointer to the structure and should free any custom memory and resources
+allocated for it.
+
+Keeping a pad locked in waiting state is only relevant when using
+the default collection algorithm (providing the oldest buffer).
+It ensures a buffer must be available on this pad for a collection
+to take place. This is of typical use to a muxer element where
+non-subtitle streams should always be in waiting state,
+e.g. to assure that caps information is available on all these streams
+when initial headers have to be written.
+
+The pad will be automatically activated in push mode when @pads is
+started.
+
+MT safe.
+
+ a new #GstCollectData to identify the
+ new pad. Or %NULL if wrong parameters are supplied.
+
+
+
+
+ the collectpads to use
+
+
+
+ the pad to add
+
+
+
+ the size of the returned #GstCollectData structure
+
+
+
+ function to be called before the returned
+ #GstCollectData structure is freed
+
+
+
+ whether to lock this pad in usual waiting state
+
+
+
+
+
+ Query how much bytes can be read from each queued buffer. This means
+that the result of this call is the maximum number of bytes that can
+be read from each of the pads.
+
+This function should be called with @pads STREAM_LOCK held, such as
+in the callback.
+
+MT safe.
+
+ The maximum number of bytes queued on all pads. This function
+returns 0 if a pad has no queued buffer.
+
+
+
+
+ the collectpads to query
+
+
+
+
+
+ Convenience clipping function that converts incoming buffer's timestamp
+to running time, or clips the buffer if outside configured segment.
+
+Since 1.6, this clipping function also sets the DTS parameter of the
+GstCollectData structure. This version of the running time DTS can be
+negative. G_MININT64 is used to indicate invalid value.
+
+
+
+
+
+ the collectpads to use
+
+
+
+ collect data of corresponding pad
+
+
+
+ buffer being clipped
+
+
+
+ output buffer with running time, or NULL if clipped
+
+
+
+ user data (unused)
+
+
+
+
+
+ Default #GstCollectPads event handling that elements should always
+chain up to to ensure proper operation. Element might however indicate
+event should not be forwarded downstream.
+
+
+
+
+
+ the collectpads to use
+
+
+
+ collect data of corresponding pad
+
+
+
+ event being processed
+
+
+
+ process but do not send event downstream
+
+
+
+
+
+ Flush @size bytes from the pad @data.
+
+This function should be called with @pads STREAM_LOCK held, such as
+in the callback.
+
+MT safe.
+
+ The number of bytes flushed This can be less than @size and
+is 0 if the pad was end-of-stream.
+
+
+
+
+ the collectpads to query
+
+
+
+ the data to use
+
+
+
+ the number of bytes to flush
+
+
+
+
+
+ Peek at the buffer currently queued in @data. This function
+should be called with the @pads STREAM_LOCK held, such as in the callback
+handler.
+
+MT safe.
+
+ The buffer in @data or %NULL if no
+buffer is queued. should unref the buffer after usage.
+
+
+
+
+ the collectpads to peek
+
+
+
+ the data to use
+
+
+
+
+
+ Pop the buffer currently queued in @data. This function
+should be called with the @pads STREAM_LOCK held, such as in the callback
+handler.
+
+MT safe.
+
+ The buffer in @data or %NULL if no
+buffer was queued. You should unref the buffer after usage.
+
+
+
+
+ the collectpads to pop
+
+
+
+ the data to use
+
+
+
+
+
+ Default #GstCollectPads query handling that elements should always
+chain up to to ensure proper operation. Element might however indicate
+query should not be forwarded downstream.
+
+
+
+
+
+ the collectpads to use
+
+
+
+ collect data of corresponding pad
+
+
+
+ query being processed
+
+
+
+ process but do not send event downstream
+
+
+
+
+
+ Get a subbuffer of @size bytes from the given pad @data.
+
+This function should be called with @pads STREAM_LOCK held, such as in the
+callback.
+
+MT safe.
+
+ A sub buffer. The size of the buffer can
+be less that requested. A return of %NULL signals that the pad is
+end-of-stream. Unref the buffer after use.
+
+
+
+
+ the collectpads to query
+
+
+
+ the data to use
+
+
+
+ the number of bytes to read
+
+
+
+
+
+ Remove a pad from the collection of collect pads. This function will also
+free the #GstCollectData and all the resources that were allocated with
+gst_collect_pads_add_pad().
+
+The pad will be deactivated automatically when @pads is stopped.
+
+MT safe.
+
+ %TRUE if the pad could be removed.
+
+
+
+
+ the collectpads to use
+
+
+
+ the pad to remove
+
+
+
+
+
+ Set the callback function and user data that will be called with
+the oldest buffer when all pads have been collected, or %NULL on EOS.
+If a buffer is passed, the callback owns a reference and must unref
+it.
+
+MT safe.
+
+
+
+
+
+ the collectpads to use
+
+
+
+ the function to set
+
+
+
+ user data passed to the function
+
+
+
+
+
+ Install a clipping function that is called right after a buffer is received
+on a pad managed by @pads. See #GstCollectPadsClipFunction for more info.
+
+
+
+
+
+ the collectpads to use
+
+
+
+ clip function to install
+
+
+
+ user data to pass to @clip_func
+
+
+
+
+
+ Set the timestamp comparison function.
+
+MT safe.
+
+
+
+
+
+ the pads to use
+
+
+
+ the function to set
+
+
+
+ user data passed to the function
+
+
+
+
+
+ Set the event callback function and user data that will be called when
+collectpads has received an event originating from one of the collected
+pads. If the event being processed is a serialized one, this callback is
+called with @pads STREAM_LOCK held, otherwise not. As this lock should be
+held when calling a number of CollectPads functions, it should be acquired
+if so (unusually) needed.
+
+MT safe.
+
+
+
+
+
+ the collectpads to use
+
+
+
+ the function to set
+
+
+
+ user data passed to the function
+
+
+
+
+
+ Install a flush function that is called when the internal
+state of all pads should be flushed as part of flushing seek
+handling. See #GstCollectPadsFlushFunction for more info.
+
+
+
+
+
+ the collectpads to use
+
+
+
+ flush function to install
+
+
+
+ user data to pass to @func
+
+
+
+
+
+ Change the flushing state of all the pads in the collection. No pad
+is able to accept anymore data when @flushing is %TRUE. Calling this
+function with @flushing %FALSE makes @pads accept data again.
+Caller must ensure that downstream streaming (thread) is not blocked,
+e.g. by sending a FLUSH_START downstream.
+
+MT safe.
+
+
+
+
+
+ the collectpads to use
+
+
+
+ desired state of the pads
+
+
+
+
+
+ CollectPads provides a default collection algorithm that will determine
+the oldest buffer available on all of its pads, and then delegate
+to a configured callback.
+However, if circumstances are more complicated and/or more control
+is desired, this sets a callback that will be invoked instead when
+all the pads added to the collection have buffers queued.
+Evidently, this callback is not compatible with
+gst_collect_pads_set_buffer_function() callback.
+If this callback is set, the former will be unset.
+
+MT safe.
+
+
+
+
+
+ the collectpads to use
+
+
+
+ the function to set
+
+
+
+ user data passed to the function
+
+
+
+
+
+ Set the query callback function and user data that will be called after
+collectpads has received a query originating from one of the collected
+pads. If the query being processed is a serialized one, this callback is
+called with @pads STREAM_LOCK held, otherwise not. As this lock should be
+held when calling a number of CollectPads functions, it should be acquired
+if so (unusually) needed.
+
+MT safe.
+
+
+
+
+
+ the collectpads to use
+
+
+
+ the function to set
+
+
+
+ user data passed to the function
+
+
+
+
+
+ Sets a pad to waiting or non-waiting mode, if at least this pad
+has not been created with locked waiting state,
+in which case nothing happens.
+
+This function should be called with @pads STREAM_LOCK held, such as
+in the callback.
+
+MT safe.
+
+
+
+
+
+ the collectpads
+
+
+
+ the data to use
+
+
+
+ boolean indicating whether this pad should operate
+ in waiting or non-waiting mode
+
+
+
+
+
+ Default #GstCollectPads event handling for the src pad of elements.
+Elements can chain up to this to let flushing seek event handling
+be done by #GstCollectPads.
+
+
+
+
+
+ the #GstCollectPads to use
+
+
+
+ src #GstPad that received the event
+
+
+
+ event being processed
+
+
+
+
+
+ Starts the processing of data in the collect_pads.
+
+MT safe.
+
+
+
+
+
+ the collectpads to use
+
+
+
+
+
+ Stops the processing of data in the collect_pads. this function
+will also unblock any blocking operations.
+
+MT safe.
+
+
+
+
+
+ the collectpads to use
+
+
+
+
+
+ Get a subbuffer of @size bytes from the given pad @data. Flushes the amount
+of read bytes.
+
+This function should be called with @pads STREAM_LOCK held, such as in the
+callback.
+
+MT safe.
+
+ A sub buffer. The size of the buffer can
+be less that requested. A return of %NULL signals that the pad is
+end-of-stream. Unref the buffer after use.
+
+
+
+
+ the collectpads to query
+
+
+
+ the data to use
+
+
+
+ the number of bytes to read
+
+
+
+
+
+
+
+
+ #GList of #GstCollectData managed
+ by this #GstCollectPads.
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ A function that will be called when a (considered oldest) buffer can be muxed.
+If all pads have reached EOS, this function is called with %NULL @buffer
+and %NULL @data.
+
+ %GST_FLOW_OK for success
+
+
+
+
+ the #GstCollectPads that triggered the callback
+
+
+
+ the #GstCollectData of pad that has received the buffer
+
+
+
+ the #GstBuffer
+
+
+
+ user data passed to gst_collect_pads_set_buffer_function()
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ A function that will be called when @inbuffer is received on the pad managed
+by @data in the collectpad object @pads.
+
+The function should use the segment of @data and the negotiated media type on
+the pad to perform clipping of @inbuffer.
+
+This function takes ownership of @inbuffer and should output a buffer in
+@outbuffer or return %NULL in @outbuffer if the buffer should be dropped.
+
+ a #GstFlowReturn that corresponds to the result of clipping.
+
+
+
+
+ a #GstCollectPads
+
+
+
+ a #GstCollectData
+
+
+
+ the input #GstBuffer
+
+
+
+ the output #GstBuffer
+
+
+
+ user data
+
+
+
+
+
+ A function for comparing two timestamps of buffers or newsegments collected on one pad.
+
+ Integer less than zero when first timestamp is deemed older than the second one.
+ Zero if the timestamps are deemed equally old.
+ Integer greater than zero when second timestamp is deemed older than the first one.
+
+
+
+
+ the #GstCollectPads that is comparing the timestamps
+
+
+
+ the first #GstCollectData
+
+
+
+ the first timestamp
+
+
+
+ the second #GstCollectData
+
+
+
+ the second timestamp
+
+
+
+ user data passed to gst_collect_pads_set_compare_function()
+
+
+
+
+
+ A function that will be called while processing an event. It takes
+ownership of the event and is responsible for chaining up (to
+gst_collect_pads_event_default()) or dropping events (such typical cases
+being handled by the default handler).
+
+ %TRUE if the pad could handle the event
+
+
+
+
+ the #GstCollectPads that triggered the callback
+
+
+
+ the #GstPad that received an event
+
+
+
+ the #GstEvent received
+
+
+
+ user data passed to gst_collect_pads_set_event_function()
+
+
+
+
+
+ A function that will be called while processing a flushing seek event.
+
+The function should flush any internal state of the element and the state of
+all the pads. It should clear only the state not directly managed by the
+@pads object. It is therefore not necessary to call
+gst_collect_pads_set_flushing nor gst_collect_pads_clear from this function.
+
+
+
+
+
+ a #GstCollectPads
+
+
+
+ user data
+
+
+
+
+
+ A function that will be called when all pads have received data.
+
+ %GST_FLOW_OK for success
+
+
+
+
+ the #GstCollectPads that triggered the callback
+
+
+
+ user data passed to gst_collect_pads_set_function()
+
+
+
+
+
+
+
+ A function that will be called while processing a query. It takes
+ownership of the query and is responsible for chaining up (to
+events downstream (with gst_pad_event_default()).
+
+ %TRUE if the pad could handle the event
+
+
+
+
+ the #GstCollectPads that triggered the callback
+
+
+
+ the #GstPad that received an event
+
+
+
+ the #GstEvent received
+
+
+
+ user data passed to gst_collect_pads_set_query_function()
+
+
+
+
+
+
+ Set if collectdata's pad is EOS.
+
+
+ Set if collectdata's pad is flushing.
+
+
+ Set if collectdata's pad received a
+ new_segment event.
+
+
+ Set if collectdata's pad must be waited
+ for when collecting.
+
+
+ Set collectdata's pad WAITING state must
+ not be changed.
+#GstCollectPadsStateFlags indicate private state of a collectdata('s pad).
+
+
+
+ #GstDataQueue is an object that handles threadsafe queueing of objects. It
+also provides size-related functionality. This object should be used for
+any #GstElement that wishes to provide some sort of queueing functionality.
+
+ Creates a new #GstDataQueue. If @fullcallback or @emptycallback are supplied, then
+the #GstDataQueue will call the respective callback to signal full or empty condition.
+If the callbacks are NULL the #GstDataQueue will instead emit 'full' and 'empty'
+signals.
+
+ a new #GstDataQueue.
+
+
+
+
+ the callback used to tell if the element considers the queue full
+or not.
+
+
+
+ the callback which will be called when the queue is considered full.
+
+
+
+ the callback which will be called when the queue is considered empty.
+
+
+
+ a #gpointer that will be passed to the @checkfull, @fullcallback,
+ and @emptycallback callbacks.
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Pop and unref the head-most #GstMiniObject with the given #GType.
+
+ %TRUE if an element was removed.
+
+
+
+
+ The #GstDataQueue to drop an item from.
+
+
+
+ The #GType of the item to drop.
+
+
+
+
+
+ Flushes all the contents of the @queue. Any call to #gst_data_queue_push and
+#gst_data_queue_pop will be released.
+MT safe.
+
+
+
+
+
+ a #GstDataQueue.
+
+
+
+
+
+ Get the current level of the queue.
+
+
+
+
+
+ The #GstDataQueue
+
+
+
+ the location to store the result
+
+
+
+
+
+ Queries if there are any items in the @queue.
+MT safe.
+
+ %TRUE if @queue is empty.
+
+
+
+
+ a #GstDataQueue.
+
+
+
+
+
+ Queries if @queue is full. This check will be done using the
+#GstDataQueueCheckFullFunction registered with @queue.
+MT safe.
+
+ %TRUE if @queue is full.
+
+
+
+
+ a #GstDataQueue.
+
+
+
+
+
+ Inform the queue that the limits for the fullness check have changed and that
+any blocking gst_data_queue_push() should be unblocked to recheck the limits.
+
+
+
+
+
+ The #GstDataQueue
+
+
+
+
+
+ Retrieves the first @item available on the @queue without removing it.
+If the queue is currently empty, the call will block until at least
+one item is available, OR the @queue is set to the flushing state.
+MT safe.
+
+ %TRUE if an @item was successfully retrieved from the @queue.
+
+
+
+
+ a #GstDataQueue.
+
+
+
+ pointer to store the returned #GstDataQueueItem.
+
+
+
+
+
+ Retrieves the first @item available on the @queue. If the queue is currently
+empty, the call will block until at least one item is available, OR the
+@queue is set to the flushing state.
+MT safe.
+
+ %TRUE if an @item was successfully retrieved from the @queue.
+
+
+
+
+ a #GstDataQueue.
+
+
+
+ pointer to store the returned #GstDataQueueItem.
+
+
+
+
+
+ Pushes a #GstDataQueueItem (or a structure that begins with the same fields)
+on the @queue. If the @queue is full, the call will block until space is
+available, OR the @queue is set to flushing state.
+MT safe.
+
+Note that this function has slightly different semantics than gst_pad_push()
+and gst_pad_push_event(): this function only takes ownership of @item and
+the #GstMiniObject contained in @item if the push was successful. If %FALSE
+is returned, the caller is responsible for freeing @item and its contents.
+
+ %TRUE if the @item was successfully pushed on the @queue.
+
+
+
+
+ a #GstDataQueue.
+
+
+
+ a #GstDataQueueItem.
+
+
+
+
+
+ Pushes a #GstDataQueueItem (or a structure that begins with the same fields)
+on the @queue. It ignores if the @queue is full or not and forces the @item
+to be pushed anyway.
+MT safe.
+
+Note that this function has slightly different semantics than gst_pad_push()
+and gst_pad_push_event(): this function only takes ownership of @item and
+the #GstMiniObject contained in @item if the push was successful. If %FALSE
+is returned, the caller is responsible for freeing @item and its contents.
+
+ %TRUE if the @item was successfully pushed on the @queue.
+
+
+
+
+ a #GstDataQueue.
+
+
+
+ a #GstDataQueueItem.
+
+
+
+
+
+ Sets the queue to flushing state if @flushing is %TRUE. If set to flushing
+state, any incoming data on the @queue will be discarded. Any call currently
+blocking on #gst_data_queue_push or #gst_data_queue_pop will return straight
+away with a return value of %FALSE. While the @queue is in flushing state,
+all calls to those two functions will return %FALSE.
+
+MT Safe.
+
+
+
+
+
+ a #GstDataQueue.
+
+
+
+ a #gboolean stating if the queue will be flushing or not.
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ the parent structure
+
+
+
+
+
+
+
+
+
+
+
+ Reports that the queue became empty (empty).
+A queue is empty if the total amount of visible items inside it (num-visible, time,
+size) is lower than the boundary values which can be set through the GObject
+properties.
+
+
+
+
+
+ Reports that the queue became full (full).
+A queue is full if the total amount of data inside it (num-visible, time,
+size) is higher than the boundary values which can be set through the GObject
+properties.
+
+
+
+
+
+
+ The prototype of the function used to inform the queue that it should be
+considered as full.
+
+ %TRUE if the queue should be considered full.
+
+
+
+
+ a #GstDataQueue.
+
+
+
+ The number of visible items currently in the queue.
+
+
+
+ The amount of bytes currently in the queue.
+
+
+
+ The accumulated duration of the items currently in the queue.
+
+
+
+ The #gpointer registered when the #GstDataQueue was created.
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Structure used by #GstDataQueue. You can supply a different structure, as
+long as the top of the structure is identical to this structure.
+
+ the #GstMiniObject to queue.
+
+
+
+ the size in bytes of the miniobject.
+
+
+
+ the duration in #GstClockTime of the miniobject. Can not be
+%GST_CLOCK_TIME_NONE.
+
+
+
+ %TRUE if @object should be considered as a visible object.
+
+
+
+ The #GDestroyNotify function to use to free the #GstDataQueueItem.
+This function should also drop the reference to @object the owner of the
+#GstDataQueueItem is assumed to hold.
+
+
+
+
+
+
+
+
+
+
+
+ Structure describing the size of a queue.
+
+ number of buffers
+
+
+
+ number of bytes
+
+
+
+ amount of time
+
+
+
+
+ Utility struct to help handling #GstFlowReturn combination. Useful for
+#GstElement<!-- -->s that have multiple source pads and need to combine
+the different #GstFlowReturn for those pads.
+
+#GstFlowCombiner works by using the last #GstFlowReturn for all #GstPad
+it has in its list and computes the combined return value and provides
+it to the caller.
+
+To add a new pad to the #GstFlowCombiner use gst_flow_combiner_add_pad().
+The new #GstPad is stored with a default value of %GST_FLOW_OK.
+
+In case you want a #GstPad to be removed, use gst_flow_combiner_remove_pad().
+
+Please be aware that this struct isn't thread safe as its designed to be
+ used by demuxers, those usually will have a single thread operating it.
+
+These functions will take refs on the passed #GstPad<!-- -->s.
+
+Aside from reducing the user's code size, the main advantage of using this
+helper struct is to follow the standard rules for #GstFlowReturn combination.
+These rules are:
+
+* %GST_FLOW_EOS: only if all returns are EOS too
+* %GST_FLOW_NOT_LINKED: only if all returns are NOT_LINKED too
+* %GST_FLOW_ERROR or below: if at least one returns an error return
+* %GST_FLOW_NOT_NEGOTIATED: if at least one returns a not-negotiated return
+* %GST_FLOW_FLUSHING: if at least one returns flushing
+* %GST_FLOW_OK: otherwise
+
+%GST_FLOW_ERROR or below, GST_FLOW_NOT_NEGOTIATED and GST_FLOW_FLUSHING are
+returned immediatelly from the gst_flow_combiner_update_flow() function.
+
+ Creates a new #GstFlowCombiner, use gst_flow_combiner_free() to free it.
+
+ A new #GstFlowCombiner
+
+
+
+
+ Adds a new #GstPad to the #GstFlowCombiner.
+
+
+
+
+
+ the #GstFlowCombiner
+
+
+
+ the #GstPad that is being added
+
+
+
+
+
+ Removes all pads from a #GstFlowCombiner and resets it to its initial state.
+
+
+
+
+
+ the #GstFlowCombiner to clear
+
+
+
+
+
+ Frees a #GstFlowCombiner struct and all its internal data.
+
+
+
+
+
+ the #GstFlowCombiner to free
+
+
+
+
+
+ Increments the reference count on the #GstFlowCombiner.
+
+ the #GstFlowCombiner.
+
+
+
+
+ the #GstFlowCombiner to add a reference to.
+
+
+
+
+
+ Removes a #GstPad from the #GstFlowCombiner.
+
+
+
+
+
+ the #GstFlowCombiner
+
+
+
+ the #GstPad to remove
+
+
+
+
+
+ Reset flow combiner and all pads to their initial state without removing pads.
+
+
+
+
+
+ the #GstFlowCombiner to clear
+
+
+
+
+
+ Decrements the reference count on the #GstFlowCombiner.
+
+
+
+
+
+ the #GstFlowCombiner to unreference.
+
+
+
+
+
+ Computes the combined flow return for the pads in it.
+
+The #GstFlowReturn parameter should be the last flow return update for a pad
+in this #GstFlowCombiner. It will use this value to be able to shortcut some
+combinations and avoid looking over all pads again. e.g. The last combined
+return is the same as the latest obtained #GstFlowReturn.
+
+ The combined #GstFlowReturn
+
+
+
+
+ the #GstFlowCombiner
+
+
+
+ the latest #GstFlowReturn received for a pad in this #GstFlowCombiner
+
+
+
+
+
+ Sets the provided pad's last flow return to provided value and computes
+the combined flow return for the pads in it.
+
+The #GstFlowReturn parameter should be the last flow return update for a pad
+in this #GstFlowCombiner. It will use this value to be able to shortcut some
+combinations and avoid looking over all pads again. e.g. The last combined
+return is the same as the latest obtained #GstFlowReturn.
+
+ The combined #GstFlowReturn
+
+
+
+
+ the #GstFlowCombiner
+
+
+
+ the #GstPad whose #GstFlowReturn to update
+
+
+
+ the latest #GstFlowReturn received for a pad in this #GstFlowCombiner
+
+
+
+
+
+
+ This class is mostly useful for elements that cannot do
+random access, or at least very slowly. The source usually
+prefers to push out a fixed size buffer.
+
+Subclasses usually operate in a format that is different from the
+default GST_FORMAT_BYTES format of #GstBaseSrc.
+
+Classes extending this base class will usually be scheduled
+in a push based mode. If the peer accepts to operate without
+offsets and within the limits of the allowed block size, this
+class can operate in getrange based mode automatically. To make
+this possible, the subclass should implement and override the
+SCHEDULING query.
+
+The subclass should extend the methods from the baseclass in
+addition to the ::create method.
+
+Seeking, flushing, scheduling and sync is all handled by this
+base class.
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Subclasses can override any of the available virtual methods or not, as
+needed. At the minimum, the @fill method should be overridden to produce
+buffers.
+
+ Element parent class
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ #GstQueueArray is an object that provides standard queue functionality
+based on an array instead of linked lists. This reduces the overhead
+caused by memory management by a large factor.
+
+ Drops the queue element at position @idx from queue @array.
+
+ the dropped element
+
+
+
+
+ a #GstQueueArray object
+
+
+
+ index to drop
+
+
+
+
+
+ Drops the queue element at position @idx from queue @array and copies the
+data of the element or structure that was removed into @p_struct if
+@p_struct is set (not NULL).
+
+ TRUE on success, or FALSE on error
+
+
+
+
+ a #GstQueueArray object
+
+
+
+ index to drop
+
+
+
+ address into which to store the data of the dropped structure, or NULL
+
+
+
+
+
+ Finds an element in the queue @array, either by comparing every element
+with @func or by looking up @data if no compare function @func is provided,
+and returning the index of the found element.
+
+Note that the index is not 0-based, but an internal index number with a
+random offset. The index can be used in connection with
+gst_queue_array_drop_element(). FIXME: return index 0-based and make
+gst_queue_array_drop_element() take a 0-based index.
+
+ Index of the found element or -1 if nothing was found.
+
+
+
+
+ a #GstQueueArray object
+
+
+
+ comparison function, or %NULL to find @data by value
+
+
+
+ data for comparison function
+
+
+
+
+
+ Frees queue @array and all memory associated to it.
+
+
+
+
+
+ a #GstQueueArray object
+
+
+
+
+
+ Returns the length of the queue @array
+
+ the length of the queue @array.
+
+
+
+
+ a #GstQueueArray object
+
+
+
+
+
+ Checks if the queue @array is empty.
+
+ %TRUE if the queue @array is empty
+
+
+
+
+ a #GstQueueArray object
+
+
+
+
+
+ Returns the head of the queue @array and does not
+remove it from the queue.
+
+ The head of the queue
+
+
+
+
+ a #GstQueueArray object
+
+
+
+
+
+ Returns the head of the queue @array without removing it from the queue.
+
+ pointer to element or struct, or NULL if @array was empty. The
+ data pointed to by the returned pointer stays valid only as long as
+ the queue array is not modified further!
+
+
+
+
+ a #GstQueueArray object
+
+
+
+
+
+ Returns the tail of the queue @array, but does not remove it from the queue.
+
+ The tail of the queue
+
+
+
+
+ a #GstQueueArray object
+
+
+
+
+
+ Returns the tail of the queue @array, but does not remove it from the queue.
+
+ The tail of the queue
+
+
+
+
+ a #GstQueueArray object
+
+
+
+
+
+ Returns and head of the queue @array and removes
+it from the queue.
+
+ The head of the queue
+
+
+
+
+ a #GstQueueArray object
+
+
+
+
+
+ Returns the head of the queue @array and removes it from the queue.
+
+ pointer to element or struct, or NULL if @array was empty. The
+ data pointed to by the returned pointer stays valid only as long as
+ the queue array is not modified further!
+
+
+
+
+ a #GstQueueArray object
+
+
+
+
+
+ Returns the tail of the queue @array and removes
+it from the queue.
+
+ The tail of the queue
+
+
+
+
+ a #GstQueueArray object
+
+
+
+
+
+ Returns the tail of the queue @array and removes
+it from the queue.
+
+ The tail of the queue
+
+
+
+
+ a #GstQueueArray object
+
+
+
+
+
+ Pushes @data to the tail of the queue @array.
+
+
+
+
+
+ a #GstQueueArray object
+
+
+
+ object to push
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Allocates a new #GstQueueArray object with an initial
+queue size of @initial_size.
+
+ a new #GstQueueArray object
+
+
+
+
+ Initial size of the new queue
+
+
+
+
+
+ Allocates a new #GstQueueArray object for elements (e.g. structures)
+of size @struct_size, with an initial queue size of @initial_size.
+
+ a new #GstQueueArray object
+
+
+
+
+ Size of each element (e.g. structure) in the array
+
+
+
+ Initial size of the new queue
+
+
+
+
+
+
+ This function will be called by gst_type_find_helper_get_range() when
+typefinding functions request to peek at the data of a stream at certain
+offsets. If this function returns GST_FLOW_OK, the result buffer will be
+stored in @buffer. The contents of @buffer is invalid for any other
+return value.
+
+This function is supposed to behave exactly like a #GstPadGetRangeFunction.
+
+ GST_FLOW_OK for success
+
+
+
+
+ a #GstObject that will handle the getrange request
+
+
+
+ the parent of @obj or %NULL
+
+
+
+ the offset of the range
+
+
+
+ the length of the range
+
+
+
+ a memory location to hold the result buffer
+
+
+
+
+
+ Create a new #GstBitReader instance, which will read from @data.
+
+Free-function: gst_bit_reader_free
+
+ a new #GstBitReader instance
+
+
+
+
+ Data from which the #GstBitReader
+ should read
+
+
+
+
+
+ Size of @data in bytes
+
+
+
+
+
+ Create a new #GstByteReader instance, which will read from @data.
+
+Free-function: gst_byte_reader_free
+
+ a new #GstByteReader instance
+
+
+
+
+ data from which the
+ #GstByteReader should read
+
+
+
+
+
+ Size of @data in bytes
+
+
+
+
+
+ Creates a new, empty #GstByteWriter instance
+
+Free-function: gst_byte_writer_free
+
+ a new, empty #GstByteWriter instance
+
+
+
+
+ Creates a new #GstByteWriter instance with the given
+memory area. If @initialized is %TRUE it is possible to
+read @size bytes from the #GstByteWriter from the beginning.
+
+Free-function: gst_byte_writer_free
+
+ a new #GstByteWriter instance
+
+
+
+
+ Memory area for writing
+
+
+
+ Size of @data in bytes
+
+
+
+ If %TRUE the complete data can be read from the beginning
+
+
+
+
+
+ Creates a new #GstByteWriter instance with the given
+initial data size.
+
+Free-function: gst_byte_writer_free
+
+ a new #GstByteWriter instance
+
+
+
+
+ Initial size of data
+
+
+
+ If %TRUE the data can't be reallocated
+
+
+
+
+
+ Allocates a new #GstQueueArray object with an initial
+queue size of @initial_size.
+
+ a new #GstQueueArray object
+
+
+
+
+ Initial size of the new queue
+
+
+
+
+
+ Allocates a new #GstQueueArray object for elements (e.g. structures)
+of size @struct_size, with an initial queue size of @initial_size.
+
+ a new #GstQueueArray object
+
+
+
+
+ Size of each element (e.g. structure) in the array
+
+
+
+ Initial size of the new queue
+
+
+
+
+
+ Tries to find what type of data is flowing from the given source #GstPad.
+
+Free-function: gst_caps_unref
+
+ the #GstCaps corresponding to the data
+ stream. Returns %NULL if no #GstCaps matches the data stream.
+
+
+
+
+ A source #GstPad
+
+
+
+ The length in bytes
+
+
+
+
+
+ Tries to find what type of data is contained in the given #GstBuffer, the
+assumption being that the buffer represents the beginning of the stream or
+file.
+
+All available typefinders will be called on the data in order of rank. If
+a typefinding function returns a probability of %GST_TYPE_FIND_MAXIMUM,
+typefinding is stopped immediately and the found caps will be returned
+right away. Otherwise, all available typefind functions will the tried,
+and the caps with the highest probability will be returned, or %NULL if
+the content of the buffer could not be identified.
+
+Free-function: gst_caps_unref
+
+ the #GstCaps corresponding to the data,
+ or %NULL if no type could be found. The caller should free the caps
+ returned with gst_caps_unref().
+
+
+
+
+ object doing the typefinding, or %NULL (used for logging)
+
+
+
+ a #GstBuffer with data to typefind
+
+
+
+ location to store the probability of the found
+ caps, or %NULL
+
+
+
+
+
+ Tries to find what type of data is contained in the given @data, the
+assumption being that the data represents the beginning of the stream or
+file.
+
+All available typefinders will be called on the data in order of rank. If
+a typefinding function returns a probability of %GST_TYPE_FIND_MAXIMUM,
+typefinding is stopped immediately and the found caps will be returned
+right away. Otherwise, all available typefind functions will the tried,
+and the caps with the highest probability will be returned, or %NULL if
+the content of @data could not be identified.
+
+Free-function: gst_caps_unref
+
+ the #GstCaps corresponding to the data,
+ or %NULL if no type could be found. The caller should free the caps
+ returned with gst_caps_unref().
+
+
+
+
+ object doing the typefinding, or %NULL (used for logging)
+
+
+
+ * a pointer with data to typefind
+
+
+
+
+
+ the size of @data
+
+
+
+ location to store the probability of the found
+ caps, or %NULL
+
+
+
+
+
+ Tries to find the best #GstCaps associated with @extension.
+
+All available typefinders will be checked against the extension in order
+of rank. The caps of the first typefinder that can handle @extension will be
+returned.
+
+Free-function: gst_caps_unref
+
+ the #GstCaps corresponding to
+ @extension, or %NULL if no type could be found. The caller should free
+ the caps returned with gst_caps_unref().
+
+
+
+
+ object doing the typefinding, or %NULL (used for logging)
+
+
+
+ an extension
+
+
+
+
+
+ Utility function to do pull-based typefinding. Unlike gst_type_find_helper()
+however, this function will use the specified function @func to obtain the
+data needed by the typefind functions, rather than operating on a given
+source pad. This is useful mostly for elements like tag demuxers which
+strip off data at the beginning and/or end of a file and want to typefind
+the stripped data stream before adding their own source pad (the specified
+callback can then call the upstream peer pad with offsets adjusted for the
+tag size, for example).
+
+When @extension is not %NULL, this function will first try the typefind
+functions for the given extension, which might speed up the typefinding
+in many cases.
+
+Free-function: gst_caps_unref
+
+ the #GstCaps corresponding to the data
+ stream. Returns %NULL if no #GstCaps matches the data stream.
+
+
+
+
+ A #GstObject that will be passed as first argument to @func
+
+
+
+ the parent of @obj or %NULL
+
+
+
+ A generic #GstTypeFindHelperGetRangeFunction that will
+ be used to access data at random offsets when doing the typefinding
+
+
+
+ The length in bytes
+
+
+
+ extension of the media
+
+
+
+ location to store the probability of the found
+ caps, or %NULL
+
+
+
+
+
+
diff --git a/girs/GstController-1.0.gir b/girs/GstController-1.0.gir
new file mode 100644
index 0000000000..af78936baf
--- /dev/null
+++ b/girs/GstController-1.0.gir
@@ -0,0 +1,931 @@
+
+
+
+
+
+
+
+
+
+
+
+ A value mapping object that attaches multiple control sources to a guint
+gobject properties representing a color. A control value of 0.0 will turn the
+color component off and a value of 1.0 will be the color level.
+
+ Create a new control-binding that attaches the given #GstControlSource to the
+#GObject property.
+
+ the new #GstARGBControlBinding
+
+
+
+
+ the object of the property
+
+
+
+ the property-name to attach the control source
+
+
+
+ the control source for the alpha channel
+
+
+
+ the control source for the red channel
+
+
+
+ the control source for the green channel
+
+
+
+ the control source for the blue channel
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ The class structure of #GstARGBControlBinding.
+
+ Parent class
+
+
+
+
+
+
+
+
+
+ An internal structure for value+time and various temporary
+values used for interpolation. This "inherits" from
+GstTimedValue.
+
+ timestamp of the value change
+
+
+
+ the new value
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Frees all data allocated by a #GstControlPoint instance.
+
+
+
+
+
+ the object to free
+
+
+
+
+
+
+ A value mapping object that attaches control sources to gobject properties. It
+will map the control values directly to the target property range. If a
+non-absolute direct control binding is used, the value range [0.0 ... 1.0]
+is mapped to full target property range, and all values outside the range
+will be clipped. An absolute control binding will not do any value
+transformations.
+
+ Create a new control-binding that attaches the #GstControlSource to the
+#GObject property. It will map the control source range [0.0 ... 1.0] to
+the full target property range, and clip all values outside this range.
+
+ the new #GstDirectControlBinding
+
+
+
+
+ the object of the property
+
+
+
+ the property-name to attach the control source
+
+
+
+ the control source
+
+
+
+
+
+ Create a new control-binding that attaches the #GstControlSource to the
+#GObject property. It will directly map the control source values to the
+target property range without any transformations.
+
+ the new #GstDirectControlBinding
+
+
+
+
+ the object of the property
+
+
+
+ the property-name to attach the control source
+
+
+
+ the control source
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ The class structure of #GstDirectControlBinding.
+
+ Parent class
+
+
+
+
+
+
+
+
+
+ Function to map a control-value to the target GValue.
+
+
+
+
+
+ the #GstDirectControlBinding instance
+
+
+
+ the value returned by the cotnrol source
+
+
+
+ the target GValue
+
+
+
+
+
+ Function to map a control-value to the target plain data type.
+
+
+
+
+
+ the #GstDirectControlBinding instance
+
+
+
+ the value returned by the cotnrol source
+
+
+
+ the target location
+
+
+
+
+
+ #GstInterpolationControlSource is a #GstControlSource, that interpolates values between user-given
+control points. It supports several interpolation modes and property types.
+
+To use #GstInterpolationControlSource get a new instance by calling
+gst_interpolation_control_source_new(), bind it to a #GParamSpec and set some
+control points by calling gst_timed_value_control_source_set().
+
+All functions are MT-safe.
+
+ This returns a new, unbound #GstInterpolationControlSource.
+
+ a new, unbound #GstInterpolationControlSource.
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ The various interpolation modes available.
+
+ steps-like interpolation, default
+
+
+ linear interpolation
+
+
+ cubic interpolation (natural), may overshoot
+ the min or max values set by the control point, but is more 'curvy'
+
+
+ monotonic cubic interpolation, will not
+ produce any values outside of the min-max range set by the control points
+ (Since 1.8)
+
+
+
+ #GstLFOControlSource is a #GstControlSource, that provides several periodic
+waveforms as control values.
+
+To use #GstLFOControlSource get a new instance by calling
+gst_lfo_control_source_new(), bind it to a #GParamSpec and set the relevant
+properties.
+
+All functions are MT-safe.
+
+ This returns a new, unbound #GstLFOControlSource.
+
+ a new, unbound #GstLFOControlSource.
+
+
+
+
+ Specifies the amplitude for the waveform of this #GstLFOControlSource.
+
+
+
+ Specifies the frequency that should be used for the waveform
+of this #GstLFOControlSource. It should be large enough
+so that the period is longer than one nanosecond.
+
+
+
+ Specifies the value offset for the waveform of this #GstLFOControlSource.
+
+
+
+ Specifies the timeshift to the right that should be used for the waveform
+of this #GstLFOControlSource in nanoseconds.
+
+To get a n nanosecond shift to the left use
+"(GST_SECOND / frequency) - n".
+
+
+
+ Specifies the waveform that should be used for this #GstLFOControlSource.
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ The various waveform modes available.
+
+ sine waveform
+
+
+ square waveform
+
+
+ saw waveform
+
+
+ reverse saw waveform
+
+
+ triangle waveform
+
+
+
+ A #GstControlBinding that forwards requests to another #GstControlBinding
+
+ #GstProxyControlBinding forwards all access to data or sync_values()
+requests from @property_name on @object to the control binding at
+@ref_property_name on @ref_object.
+
+ a new #GstControlBinding that proxies the control interface between
+properties on different #GstObject's
+
+
+
+
+ a #GstObject
+
+
+
+ the property name in @object to control
+
+
+
+ a #GstObject to forward all
+ #GstControlBinding requests to
+
+
+
+ the property_name in @ref_object to control
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Opaque #GstProxyControlBindingClass struct
+
+
+
+
+
+
+
+
+
+
+ Base class for #GstControlSource that use time-stamped values.
+
+When overriding bind, chain up first to give this bind implementation a
+chance to setup things.
+
+All functions are MT-safe.
+
+ Find last value before given timestamp in control point list.
+If all values in the control point list come after the given
+timestamp or no values exist, %NULL is returned.
+
+For use in control source implementations.
+
+ the found #GSequenceIter or %NULL
+
+
+
+
+ the control source to search in
+
+
+
+ the search key
+
+
+
+
+
+ Returns a read-only copy of the list of #GstTimedValue for the given property.
+Free the list after done with it.
+
+ a copy
+of the list, or %NULL if the property isn't handled by the controller
+
+
+
+
+
+
+ the #GstTimedValueControlSource to get the list from
+
+
+
+
+
+ Get the number of control points that are set.
+
+ the number of control points that are set.
+
+
+
+
+ the #GstTimedValueControlSource to get the number of values from
+
+
+
+
+
+ Set the value of given controller-handled property at a certain time.
+
+ FALSE if the values couldn't be set, TRUE otherwise.
+
+
+
+
+ the #GstTimedValueControlSource object
+
+
+
+ the time the control-change is scheduled for
+
+
+
+ the control-value
+
+
+
+
+
+ Sets multiple timed values at once.
+
+ FALSE if the values couldn't be set, TRUE otherwise.
+
+
+
+
+ the #GstTimedValueControlSource object
+
+
+
+ a list
+with #GstTimedValue items
+
+
+
+
+
+
+
+ Used to remove the value of given controller-handled property at a certain
+time.
+
+ FALSE if the value couldn't be unset (i.e. not found, TRUE otherwise.
+
+
+
+
+ the #GstTimedValueControlSource object
+
+
+
+ the time the control-change should be removed from
+
+
+
+
+
+ Used to remove all time-stamped values of given controller-handled property
+
+
+
+
+
+ the #GstTimedValueControlSource object
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Emited right after the new value has been added to @self
+
+
+
+
+
+ The newly added #GstTimedValue
+
+
+
+
+
+ Emited right after the new value has been set on @timed_signals
+
+
+
+
+
+ The #GstTimedValue where the value changed
+
+
+
+
+
+ Emited when @timed_value is removed from @self
+
+
+
+
+
+ The removed #GstTimedValue
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ #GstTriggerControlSource is a #GstControlSource, that returns values from user-given
+control points. It allows for a tolerance on the time-stamps.
+
+To use #GstTriggerControlSource get a new instance by calling
+gst_trigger_control_source_new(), bind it to a #GParamSpec and set some
+control points by calling gst_timed_value_control_source_set().
+
+All functions are MT-safe.
+
+ This returns a new, unbound #GstTriggerControlSource.
+
+ a new, unbound #GstTriggerControlSource.
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Reset the controlled value cache.
+
+
+
+
+
+ the #GstTimedValueControlSource
+
+
+
+
+
+
diff --git a/girs/GstNet-1.0.gir b/girs/GstNet-1.0.gir
new file mode 100644
index 0000000000..076f880690
--- /dev/null
+++ b/girs/GstNet-1.0.gir
@@ -0,0 +1,861 @@
+
+
+
+
+
+
+
+
+
+
+
+
+ The size of the packets sent between network clocks.
+
+
+
+ #GstNetAddressMeta can be used to store a network address (a #GSocketAddress)
+in a #GstBuffer so that it network elements can track the to and from address
+of the buffer.
+
+ the parent type
+
+
+
+ a #GSocketAddress stored as metadata
+
+
+
+
+
+
+
+
+
+ #GstNetClientClock implements a custom #GstClock that synchronizes its time
+to a remote time provider such as #GstNetTimeProvider. #GstNtpClock
+implements a #GstClock that synchronizes its time to a remote NTPv4 server.
+
+A new clock is created with gst_net_client_clock_new() or
+gst_ntp_clock_new(), which takes the address and port of the remote time
+provider along with a name and an initial time.
+
+This clock will poll the time provider and will update its calibration
+parameters based on the local and remote observations.
+
+The "round-trip" property limits the maximum round trip packets can take.
+
+Various parameters of the clock can be configured with the parent #GstClock
+"timeout", "window-size" and "window-threshold" object properties.
+
+A #GstNetClientClock and #GstNtpClock is typically set on a #GstPipeline with
+gst_pipeline_use_clock().
+
+If you set a #GstBus on the clock via the "bus" object property, it will
+send @GST_MESSAGE_ELEMENT messages with an attached #GstStructure containing
+statistics about clock accuracy and network traffic.
+
+ Create a new #GstNetClientInternalClock that will report the time
+provided by the #GstNetTimeProvider on @remote_address and
+@remote_port.
+
+ a new #GstClock that receives a time from the remote
+clock.
+
+
+
+
+ a name for the clock
+
+
+
+ the address or hostname of the remote clock provider
+
+
+
+ the port of the remote clock provider
+
+
+
+ initial time of the clock
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ #GstNetControlMessageMeta can be used to store control messages (ancillary
+data) which was received with or is to be sent alongside the buffer data.
+When used with socket sinks and sources which understand this meta it allows
+sending and receiving ancillary data such as unix credentials (See
+#GUnixCredentialsMessage) and Unix file descriptions (See #GUnixFDMessage).
+
+ the parent type
+
+
+
+ a #GSocketControlMessage stored as metadata
+
+
+
+
+
+
+
+
+
+ Various functions for receiving, sending an serializing #GstNetTimePacket
+structures.
+
+ the local time when this packet was sent
+
+
+
+ the remote time observation
+
+
+
+ Creates a new #GstNetTimePacket from a buffer received over the network. The
+caller is responsible for ensuring that @buffer is at least
+#GST_NET_TIME_PACKET_SIZE bytes long.
+
+If @buffer is %NULL, the local and remote times will be set to
+#GST_CLOCK_TIME_NONE.
+
+MT safe. Caller owns return value (gst_net_time_packet_free to free).
+
+ The new #GstNetTimePacket.
+
+
+
+
+ a buffer from which to construct the packet, or NULL
+
+
+
+
+
+
+
+ Make a copy of @packet.
+
+ a copy of @packet, free with gst_net_time_packet_free().
+
+
+
+
+ the #GstNetTimePacket
+
+
+
+
+
+ Free @packet.
+
+
+
+
+
+ the #GstNetTimePacket
+
+
+
+
+
+ Sends a #GstNetTimePacket over a socket.
+
+MT safe.
+
+ TRUE if successful, FALSE in case an error occurred.
+
+
+
+
+ the #GstNetTimePacket to send
+
+
+
+ socket to send the time packet on
+
+
+
+ address to send the time packet to
+
+
+
+
+
+ Serialized a #GstNetTimePacket into a newly-allocated sequence of
+#GST_NET_TIME_PACKET_SIZE bytes, in network byte order. The value returned is
+suitable for passing to write(2) or sendto(2) for communication over the
+network.
+
+MT safe. Caller owns return value (g_free to free).
+
+ A newly allocated sequence of #GST_NET_TIME_PACKET_SIZE bytes.
+
+
+
+
+ the #GstNetTimePacket
+
+
+
+
+
+ Receives a #GstNetTimePacket over a socket. Handles interrupted system
+calls, but otherwise returns NULL on error.
+
+ a new #GstNetTimePacket, or NULL on error. Free
+ with gst_net_time_packet_free() when done.
+
+
+
+
+ socket to receive the time packet on
+
+
+
+ address of variable to return sender address
+
+
+
+
+
+
+ This object exposes the time of a #GstClock on the network.
+
+A #GstNetTimeProvider is created with gst_net_time_provider_new() which
+takes a #GstClock, an address and a port number as arguments.
+
+After creating the object, a client clock such as #GstNetClientClock can
+query the exposed clock over the network for its values.
+
+The #GstNetTimeProvider typically wraps the clock used by a #GstPipeline.
+
+
+ Allows network clients to get the current time of @clock.
+
+ the new #GstNetTimeProvider, or NULL on error
+
+
+
+
+ a #GstClock to export over the network
+
+
+
+ an address to bind on as a dotted quad
+ (xxx.xxx.xxx.xxx), IPv6 address, or NULL to bind to all addresses
+
+
+
+ a port to bind on, or 0 to let the kernel choose
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Create a new #GstNtpClock that will report the time provided by
+the NTPv4 server on @remote_address and @remote_port.
+
+ a new #GstClock that receives a time from the remote
+clock.
+
+
+
+
+ a name for the clock
+
+
+
+ the address or hostname of the remote clock provider
+
+
+
+ the port of the remote clock provider
+
+
+
+ initial time of the clock
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ PTP clock identification that can be passed to gst_ptp_init() to
+automatically select one based on the MAC address of interfaces
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ GstPtpClock implements a PTP (IEEE1588:2008) ordinary clock in slave-only
+mode, that allows a GStreamer pipeline to synchronize to a PTP network
+clock in some specific domain.
+
+The PTP subsystem can be initialized with gst_ptp_init(), which then starts
+a helper process to do the actual communication via the PTP ports. This is
+required as PTP listens on ports < 1024 and thus requires special
+privileges. Once this helper process is started, the main process will
+synchronize to all PTP domains that are detected on the selected
+interfaces.
+
+gst_ptp_clock_new() then allows to create a GstClock that provides the PTP
+time from a master clock inside a specific PTP domain. This clock will only
+return valid timestamps once the timestamps in the PTP domain are known. To
+check this, you can use gst_clock_wait_for_sync(), the GstClock::synced
+signal and gst_clock_is_synced().
+
+To gather statistics about the PTP clock synchronization,
+gst_ptp_statistics_callback_add() can be used. This gives the application
+the possibility to collect all kinds of statistics from the clock
+synchronization.
+
+ Creates a new PTP clock instance that exports the PTP time of the master
+clock in @domain. This clock can be slaved to other clocks as needed.
+
+If gst_ptp_init() was not called before, this will call gst_ptp_init() with
+default parameters.
+
+This clock only returns valid timestamps after it received the first
+times from the PTP master clock on the network. Once this happens the
+GstPtpClock::internal-clock property will become non-NULL. You can
+check this with gst_clock_wait_for_sync(), the GstClock::synced signal and
+gst_clock_is_synced().
+
+ A new #GstClock
+
+
+
+
+ Name of the clock
+
+
+
+ PTP domain
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Opaque #GstPtpClockClass structure.
+
+ parented to #GstSystemClockClass
+
+
+
+
+
+
+
+
+
+
+
+ The statistics can be the following structures:
+
+GST_PTP_STATISTICS_NEW_DOMAIN_FOUND:
+"domain" G_TYPE_UINT The domain identifier of the domain
+"clock" GST_TYPE_CLOCK The internal clock that is slaved to the
+ PTP domain
+
+GST_PTP_STATISTICS_BEST_MASTER_CLOCK_SELECTED:
+"domain" G_TYPE_UINT The domain identifier of the domain
+"master-clock-id" G_TYPE_UINT64 PTP clock identifier of the selected master
+ clock
+"master-clock-port" G_TYPE_UINT PTP port number of the selected master clock
+"grandmaster-clock-id" G_TYPE_UINT64 PTP clock identifier of the grandmaster clock
+
+GST_PTP_STATISTICS_PATH_DELAY_MEASURED:
+"domain" G_TYPE_UINT The domain identifier of the domain
+"mean-path-delay-avg" GST_TYPE_CLOCK_TIME Average mean path delay
+"mean-path-delay" GST_TYPE_CLOCK_TIME Latest mean path delay
+"delay-request-delay" GST_TYPE_CLOCK_TIME Delay of DELAY_REQ / DELAY_RESP messages
+
+GST_PTP_STATISTICS_TIME_UPDATED:
+"domain" G_TYPE_UINT The domain identifier of the domain
+"mean-path-delay-avg" GST_TYPE_CLOCK_TIME Average mean path delay
+"local-time" GST_TYPE_CLOCK_TIME Local time that corresponds to ptp-time
+"ptp-time" GST_TYPE_CLOCK_TIME Newly measured PTP time at local-time
+"estimated-ptp-time" GST_TYPE_CLOCK_TIME Estimated PTP time based on previous measurements
+"discontinuity" G_TYPE_INT64 Difference between estimated and measured PTP time
+"synced" G_TYPE_BOOLEAN Currently synced to the remote clock
+"r-squared" G_TYPE_DOUBLE R² of clock estimation regression
+"internal-time" GST_TYPE_CLOCK_TIME Internal time clock parameter
+"external-time" GST_TYPE_CLOCK_TIME External time clock parameter
+"rate-num" G_TYPE_UINT64 Internal/external rate numerator
+"rate-den" G_TYPE_UINT64 Internal/external rate denominator
+"rate" G_TYPE_DOUBLE Internal/external rate
+
+If %FALSE is returned, the callback is removed and never called again.
+
+
+
+
+
+ PTP domain identifier
+
+
+
+ New statistics
+
+
+
+ Data passed to gst_ptp_statistics_callback_add()
+
+
+
+
+
+ Attaches @addr as metadata in a #GstNetAddressMeta to @buffer.
+
+ a #GstNetAddressMeta connected to @buffer
+
+
+
+
+ a #GstBuffer
+
+
+
+ a @GSocketAddress to connect to @buffer
+
+
+
+
+
+ Attaches @message as metadata in a #GstNetControlMessageMeta to @buffer.
+
+ a #GstNetControlMessageMeta connected to @buffer
+
+
+
+
+ a #GstBuffer
+
+
+
+ a @GSocketControlMessage to attach to @buffer
+
+
+
+
+
+ Find the #GstNetAddressMeta on @buffer.
+
+ the #GstNetAddressMeta or %NULL when there
+is no such metadata on @buffer.
+
+
+
+
+ a #GstBuffer
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Receives a #GstNetTimePacket over a socket. Handles interrupted system
+calls, but otherwise returns NULL on error.
+
+ a new #GstNetTimePacket, or NULL on error. Free
+ with gst_net_time_packet_free() when done.
+
+
+
+
+ socket to receive the time packet on
+
+
+
+ address of variable to return sender address
+
+
+
+
+
+ Deinitialize the GStreamer PTP subsystem and stop the PTP clock. If there
+are any remaining GstPtpClock instances, they won't be further synchronized
+to the PTP network clock.
+
+
+
+
+
+ Initialize the GStreamer PTP subsystem and create a PTP ordinary clock in
+slave-only mode for all domains on the given @interfaces with the
+given @clock_id.
+
+If @clock_id is %GST_PTP_CLOCK_ID_NONE, a clock id is automatically
+generated from the MAC address of the first network interface.
+
+This function is automatically called by gst_ptp_clock_new() with default
+parameters if it wasn't called before.
+
+ %TRUE if the GStreamer PTP clock subsystem could be initialized.
+
+
+
+
+ PTP clock id of this process' clock or %GST_PTP_CLOCK_ID_NONE
+
+
+
+ network interfaces to run the clock on
+
+
+
+
+
+
+
+ Check if the GStreamer PTP clock subsystem is initialized.
+
+ %TRUE if the GStreamer PTP clock subsystem is intialized.
+
+
+
+
+ Check if PTP clocks are generally supported on this system, and if previous
+initializations did not fail.
+
+ %TRUE if PTP clocks are generally supported on this system, and
+previous initializations did not fail.
+
+
+
+
+ Installs a new statistics callback for gathering PTP statistics. See
+GstPtpStatisticsCallback for a list of statistics that are provided.
+
+ Id for the callback that can be passed to
+gst_ptp_statistics_callback_remove()
+
+
+
+
+ GstPtpStatisticsCallback to call
+
+
+
+ Data to pass to the callback
+
+
+
+ GDestroyNotify to destroy the data
+
+
+
+
+
+ Removes a PTP statistics callback that was previously added with
+gst_ptp_statistics_callback_add().
+
+
+
+
+
+ Callback id to remove
+
+
+
+
+
+
diff --git a/girs/GstPbutils-1.0.gir b/girs/GstPbutils-1.0.gir
new file mode 100644
index 0000000000..5cb76d9144
--- /dev/null
+++ b/girs/GstPbutils-1.0.gir
@@ -0,0 +1,4126 @@
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ A baseclass for scopes (visualizers). It takes care of re-fitting the
+audio-rate to video-rate and handles renegotiation (downstream video size
+changes).
+
+It also provides several background shading effects. These effects are
+applied to a previous picture before the render() implementation can draw a
+new frame.
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Different types of supported background shading functions.
+
+ no shading
+
+
+ plain fading
+
+
+ fade and move up
+
+
+ fade and move down
+
+
+ fade and move left
+
+
+ fade and move right
+
+
+ fade and move horizontally out
+
+
+ fade and move horizontally in
+
+
+ fade and move vertically out
+
+
+ fade and move vertically in
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ The #GstDiscoverer is a utility object which allows to get as much
+information as possible from one or many URIs.
+
+It provides two APIs, allowing usage in blocking or non-blocking mode.
+
+The blocking mode just requires calling gst_discoverer_discover_uri()
+with the URI one wishes to discover.
+
+The non-blocking mode requires a running #GMainLoop iterating a
+#GMainContext, where one connects to the various signals, appends the
+URIs to be processed (through gst_discoverer_discover_uri_async()) and then
+asks for the discovery to begin (through gst_discoverer_start()).
+By default this will use the GLib default main context unless you have
+set a custom context using g_main_context_push_thread_default().
+
+All the information is returned in a #GstDiscovererInfo structure.
+
+ Creates a new #GstDiscoverer with the provided timeout.
+
+ The new #GstDiscoverer.
+If an error occurred when creating the discoverer, @err will be set
+accordingly and %NULL will be returned. If @err is set, the caller must
+free it when no longer needed using g_error_free().
+
+
+
+
+ timeout per file, in nanoseconds. Allowed are values between
+ one second (#GST_SECOND) and one hour (3600 * #GST_SECOND)
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Synchronously discovers the given @uri.
+
+A copy of @uri will be made internally, so the caller can safely g_free()
+afterwards.
+
+ the result of the scanning. Can be %NULL if an
+error occurred.
+
+
+
+
+ A #GstDiscoverer
+
+
+
+ The URI to run on.
+
+
+
+
+
+ Appends the given @uri to the list of URIs to discoverer. The actual
+discovery of the @uri will only take place if gst_discoverer_start() has
+been called.
+
+A copy of @uri will be made internally, so the caller can safely g_free()
+afterwards.
+
+ %TRUE if the @uri was successfully appended to the list of pending
+uris, else %FALSE
+
+
+
+
+ A #GstDiscoverer
+
+
+
+ the URI to add.
+
+
+
+
+
+ Allow asynchronous discovering of URIs to take place.
+A #GMainLoop must be available for #GstDiscoverer to properly work in
+asynchronous mode.
+
+
+
+
+
+ A #GstDiscoverer
+
+
+
+
+
+ Stop the discovery of any pending URIs and clears the list of
+pending URIS (if any).
+
+
+
+
+
+ A #GstDiscoverer
+
+
+
+
+
+ The duration (in nanoseconds) after which the discovery of an individual
+URI will timeout.
+
+If the discovery of a URI times out, the %GST_DISCOVERER_TIMEOUT will be
+set on the result flags.
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Will be emitted in async mode when all information on a URI could be
+discovered, or an error occurred.
+
+When an error occurs, @info might still contain some partial information,
+depending on the circumstances of the error.
+
+
+
+
+
+ the results #GstDiscovererInfo
+
+
+
+ #GError, which will be non-NULL
+ if an error occurred during
+ discovery. You must not free
+ this #GError, it will be freed by
+ the discoverer.
+
+
+
+
+
+ Will be emitted in async mode when all pending URIs have been processed.
+
+
+
+
+
+ This signal is emitted after the source element has been created for, so
+the URI being discovered, so it can be configured by setting additional
+properties (e.g. set a proxy server for an http source, or set the device
+and read speed for an audio cd source).
+
+This signal is usually emitted from the context of a GStreamer streaming
+thread.
+
+
+
+
+
+ source element
+
+
+
+
+
+ Will be emitted when the discover starts analyzing the pending URIs
+
+
+
+
+
+
+ #GstDiscovererStreamInfo specific to audio streams.
+
+
+ the average or nominal bitrate of the stream in bits/second.
+
+
+
+
+ a #GstDiscovererAudioInfo
+
+
+
+
+
+
+ the channel-mask of the stream, refer to
+gst_audio_channel_positions_from_mask() for more
+information.
+
+
+
+
+ a #GstDiscovererAudioInfo
+
+
+
+
+
+
+ the number of channels in the stream.
+
+
+
+
+ a #GstDiscovererAudioInfo
+
+
+
+
+
+
+ the number of bits used per sample in each channel.
+
+
+
+
+ a #GstDiscovererAudioInfo
+
+
+
+
+
+
+ the language of the stream, or NULL if unknown.
+
+
+
+
+ a #GstDiscovererAudioInfo
+
+
+
+
+
+
+ the maximum bitrate of the stream in bits/second.
+
+
+
+
+ a #GstDiscovererAudioInfo
+
+
+
+
+
+
+ the sample rate of the stream in Hertz.
+
+
+
+
+ a #GstDiscovererAudioInfo
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ #GstDiscovererStreamInfo specific to container streams.
+
+
+ the list of
+#GstDiscovererStreamInfo this container stream offers.
+Free with gst_discoverer_stream_info_list_free() after usage.
+
+
+
+
+
+
+ a #GstDiscovererStreamInfo
+
+
+
+
+
+
+ Structure containing the information of a URI analyzed by #GstDiscoverer.
+
+ Parses a #GVariant as produced by gst_discoverer_info_to_variant()
+back to a #GstDiscovererInfo.
+
+ A newly-allocated #GstDiscovererInfo.
+
+
+
+
+ A #GVariant to deserialize into a #GstDiscovererInfo.
+
+
+
+
+
+
+ A copy of the #GstDiscovererInfo
+
+
+
+
+ a #GstDiscovererInfo
+
+
+
+
+
+ Finds all the #GstDiscovererAudioInfo contained in @info
+
+ A #GList of
+matching #GstDiscovererStreamInfo. The caller should free it with
+gst_discoverer_stream_info_list_free().
+
+
+
+
+
+
+ a #GstDiscovererInfo
+
+
+
+
+
+ Finds all the #GstDiscovererContainerInfo contained in @info
+
+ A #GList of
+matching #GstDiscovererStreamInfo. The caller should free it with
+gst_discoverer_stream_info_list_free().
+
+
+
+
+
+
+ a #GstDiscovererInfo
+
+
+
+
+
+
+ the duration of the URI in #GstClockTime (nanoseconds).
+
+
+
+
+ a #GstDiscovererInfo
+
+
+
+
+
+
+ whether the URI is live.
+
+
+
+
+ a #GstDiscovererInfo
+
+
+
+
+
+ This functions is deprecated since version 1.4, use
+#gst_discoverer_info_get_missing_elements_installer_details
+
+ Miscellaneous information stored as a #GstStructure
+(for example: information about missing plugins). If you wish to use the
+#GstStructure after the life-time of @info, you will need to copy it.
+
+
+
+
+ a #GstDiscovererInfo
+
+
+
+
+
+ Get the installer details for missing elements
+
+ An array of strings
+containing informations about how to install the various missing elements
+for @info to be usable. If you wish to use the strings after the life-time
+of @info, you will need to copy them.
+
+
+
+
+
+
+ a #GstDiscovererStreamInfo to retrieve installer detail
+for the missing element
+
+
+
+
+
+
+ the result of the discovery as a #GstDiscovererResult.
+
+
+
+
+ a #GstDiscovererInfo
+
+
+
+
+
+
+ the whether the URI is seekable.
+
+
+
+
+ a #GstDiscovererInfo
+
+
+
+
+
+
+ the structure (or topology) of the URI as a
+#GstDiscovererStreamInfo.
+This structure can be traversed to see the original hierarchy. Unref with
+gst_discoverer_stream_info_unref() after usage.
+
+
+
+
+ a #GstDiscovererInfo
+
+
+
+
+
+
+ the list of
+all streams contained in the #info. Free after usage
+with gst_discoverer_stream_info_list_free().
+
+
+
+
+
+
+ a #GstDiscovererInfo
+
+
+
+
+
+ Finds the #GstDiscovererStreamInfo contained in @info that match the
+given @streamtype.
+
+ A #GList of
+matching #GstDiscovererStreamInfo. The caller should free it with
+gst_discoverer_stream_info_list_free().
+
+
+
+
+
+
+ a #GstDiscovererInfo
+
+
+
+ a #GType derived from #GstDiscovererStreamInfo
+
+
+
+
+
+ Finds all the #GstDiscovererSubtitleInfo contained in @info
+
+ A #GList of
+matching #GstDiscovererStreamInfo. The caller should free it with
+gst_discoverer_stream_info_list_free().
+
+
+
+
+
+
+ a #GstDiscovererInfo
+
+
+
+
+
+
+ all tags contained in the URI. If you wish to use
+the tags after the life-time of @info, you will need to copy them.
+
+
+
+
+ a #GstDiscovererInfo
+
+
+
+
+
+
+ TOC contained in the URI. If you wish to use
+the TOC after the life-time of @info, you will need to copy it.
+
+
+
+
+ a #GstDiscovererInfo
+
+
+
+
+
+
+ the URI to which this information corresponds to.
+Copy it if you wish to use it after the life-time of @info.
+
+
+
+
+ a #GstDiscovererInfo
+
+
+
+
+
+ Finds all the #GstDiscovererVideoInfo contained in @info
+
+ A #GList of
+matching #GstDiscovererStreamInfo. The caller should free it with
+gst_discoverer_stream_info_list_free().
+
+
+
+
+
+
+ a #GstDiscovererInfo
+
+
+
+
+
+ Serializes @info to a #GVariant that can be parsed again
+through gst_discoverer_info_from_variant().
+
+Note that any #GstToc (s) that might have been discovered will not be serialized
+for now.
+
+ A newly-allocated #GVariant representing @info.
+
+
+
+
+ A #GstDiscovererInfo
+
+
+
+ A combination of #GstDiscovererSerializeFlags to specify
+what needs to be serialized.
+
+
+
+
+
+
+
+
+ Result values for the discovery process.
+
+ The discovery was successful
+
+
+ the URI is invalid
+
+
+ an error happened and the GError is set
+
+
+ the discovery timed-out
+
+
+ the discoverer was already discovering a file
+
+
+ Some plugins are missing for full discovery
+
+
+
+ You can use these flags to control what is serialized by
+gst_discoverer_info_to_variant()
+
+ Serialize only basic information, excluding
+caps, tags and miscellaneous information
+
+
+ Serialize the caps for each stream
+
+
+ Serialize the tags for each stream
+
+
+ Serialize miscellaneous information for each stream
+
+
+ Serialize all the available info, including
+caps, tags and miscellaneous information
+
+
+
+ Base structure for information concerning a media stream. Depending on the
+stream type, one can find more media-specific information in
+#GstDiscovererAudioInfo, #GstDiscovererVideoInfo, and
+#GstDiscovererContainerInfo.
+
+The #GstDiscovererStreamInfo represents the topology of the stream. Siblings
+can be iterated over with gst_discoverer_stream_info_get_next() and
+gst_discoverer_stream_info_get_previous(). Children (sub-streams) of a
+stream can be accessed using the #GstDiscovererContainerInfo API.
+
+As a simple example, if you run #GstDiscoverer on an AVI file with one audio
+and one video stream, you will get a #GstDiscovererContainerInfo
+corresponding to the AVI container, which in turn will have a
+#GstDiscovererAudioInfo sub-stream and a #GstDiscovererVideoInfo sub-stream
+for the audio and video streams respectively.
+
+ Decrements the reference count of all contained #GstDiscovererStreamInfo
+and fress the #GList.
+
+
+
+
+
+ a #GList of #GstDiscovererStreamInfo
+
+
+
+
+
+
+
+
+ the #GstCaps of the stream. Unref with
+#gst_caps_unref after usage.
+
+
+
+
+ a #GstDiscovererStreamInfo
+
+
+
+
+
+ This functions is deprecated since version 1.4, use
+#gst_discoverer_info_get_missing_elements_installer_details
+
+ additional information regarding the stream (for
+example codec version, profile, etc..). If you wish to use the #GstStructure
+after the life-time of @info you will need to copy it.
+
+
+
+
+ a #GstDiscovererStreamInfo
+
+
+
+
+
+
+ the next #GstDiscovererStreamInfo in a chain. %NULL
+for final streams.
+Unref with #gst_discoverer_stream_info_unref after usage.
+
+
+
+
+ a #GstDiscovererStreamInfo
+
+
+
+
+
+
+ the previous #GstDiscovererStreamInfo in a chain.
+%NULL for starting points. Unref with #gst_discoverer_stream_info_unref
+after usage.
+
+
+
+
+ a #GstDiscovererStreamInfo
+
+
+
+
+
+
+ the stream ID of this stream. If you wish to
+use the stream ID after the life-time of @info you will need to copy it.
+
+
+
+
+ a #GstDiscovererStreamInfo
+
+
+
+
+
+
+ a human readable name for the stream type of the given @info (ex : "audio",
+"container",...).
+
+
+
+
+ a #GstDiscovererStreamInfo
+
+
+
+
+
+
+ the tags contained in this stream. If you wish to
+use the tags after the life-time of @info you will need to copy them.
+
+
+
+
+ a #GstDiscovererStreamInfo
+
+
+
+
+
+
+ the TOC contained in this stream. If you wish to
+use the TOC after the life-time of @info you will need to copy it.
+
+
+
+
+ a #GstDiscovererStreamInfo
+
+
+
+
+
+
+ #GstDiscovererStreamInfo specific to subtitle streams (this includes text and
+image based ones).
+
+
+ the language of the stream, or NULL if unknown.
+
+
+
+
+ a #GstDiscovererSubtitleInfo
+
+
+
+
+
+
+ #GstDiscovererStreamInfo specific to video streams (this includes images).
+
+
+ the average or nominal bitrate of the video stream in bits/second.
+
+
+
+
+ a #GstDiscovererVideoInfo
+
+
+
+
+
+
+ the depth in bits of the video stream.
+
+
+
+
+ a #GstDiscovererVideoInfo
+
+
+
+
+
+
+ the framerate of the video stream (denominator).
+
+
+
+
+ a #GstDiscovererVideoInfo
+
+
+
+
+
+
+ the framerate of the video stream (numerator).
+
+
+
+
+ a #GstDiscovererVideoInfo
+
+
+
+
+
+
+ the height of the video stream in pixels.
+
+
+
+
+ a #GstDiscovererVideoInfo
+
+
+
+
+
+
+ the maximum bitrate of the video stream in bits/second.
+
+
+
+
+ a #GstDiscovererVideoInfo
+
+
+
+
+
+
+ the Pixel Aspect Ratio (PAR) of the video stream (denominator).
+
+
+
+
+ a #GstDiscovererVideoInfo
+
+
+
+
+
+
+ the Pixel Aspect Ratio (PAR) of the video stream (numerator).
+
+
+
+
+ a #GstDiscovererVideoInfo
+
+
+
+
+
+
+ the width of the video stream in pixels.
+
+
+
+
+ a #GstDiscovererVideoInfo
+
+
+
+
+
+
+ %TRUE if the video stream corresponds to an image (i.e. only contains
+one frame).
+
+
+
+
+ a #GstDiscovererVideoInfo
+
+
+
+
+
+
+ %TRUE if the stream is interlaced, else %FALSE.
+
+
+
+
+ a #GstDiscovererVideoInfo
+
+
+
+
+
+
+ #GstEncodingTarget category for recording and capture.
+Targets within this category are optimized for low latency encoding.
+
+
+
+ #GstEncodingTarget category for device-specific targets.
+The name of the target will usually be the constructor and model of the device,
+and that target will contain #GstEncodingProfiles suitable for that device.
+
+
+
+ #GstEncodingTarget category for file extensions.
+The name of the target will be the name of the file extensions possible
+for a particular target. Those targets are defining like 'default' formats
+usually used for a particular file extension.
+
+
+
+ #GstEncodingTarget category for online-services.
+The name of the target will usually be the name of the online service
+and that target will contain #GstEncodingProfiles suitable for that online
+service.
+
+
+
+ #GstEncodingTarget category for storage, archiving and editing targets.
+Those targets can be lossless and/or provide very fast random access content.
+The name of the target will usually be the container type or editing target,
+and that target will contain #GstEncodingProfiles suitable for editing or
+storage.
+
+
+
+ Variant of #GstEncodingProfile for audio streams.
+
+ Creates a new #GstEncodingAudioProfile
+
+All provided allocatable arguments will be internally copied, so can be
+safely freed/unreferenced after calling this method.
+
+ the newly created #GstEncodingAudioProfile.
+
+
+
+
+ the #GstCaps
+
+
+
+ the preset(s) to use on the encoder, can be %NULL
+
+
+
+ the #GstCaps used to restrict the input to the encoder, can be
+NULL. See gst_encoding_profile_get_restriction() for more details.
+
+
+
+ the number of time this stream must be used. 0 means any number of
+ times (including never)
+
+
+
+
+
+
+
+
+ Encoding profiles for containers. Keeps track of a list of #GstEncodingProfile
+
+ Creates a new #GstEncodingContainerProfile.
+
+ The newly created #GstEncodingContainerProfile.
+
+
+
+
+ The name of the container profile, can be %NULL
+
+
+
+ The description of the container profile,
+ can be %NULL
+
+
+
+ The format to use for this profile
+
+
+
+ The preset to use for this profile.
+
+
+
+
+
+ Add a #GstEncodingProfile to the list of profiles handled by @container.
+
+No copy of @profile will be made, if you wish to use it elsewhere after this
+method you should increment its reference count.
+
+ %TRUE if the @stream was properly added, else %FALSE.
+
+
+
+
+ the #GstEncodingContainerProfile to use
+
+
+
+ the #GstEncodingProfile to add.
+
+
+
+
+
+ Checks if @container contains a #GstEncodingProfile identical to
+@profile.
+
+ %TRUE if @container contains a #GstEncodingProfile identical
+to @profile, else %FALSE.
+
+
+
+
+ a #GstEncodingContainerProfile
+
+
+
+ a #GstEncodingProfile
+
+
+
+
+
+
+
+the list of contained #GstEncodingProfile.
+
+
+
+
+
+
+ a #GstEncodingContainerProfile
+
+
+
+
+
+
+
+
+ The opaque base class object for all encoding profiles. This contains generic
+information like name, description, format and preset.
+
+ Find the #GstEncodingProfile with the specified name and category.
+
+ The matching #GstEncodingProfile or %NULL.
+
+
+
+
+ The name of the target
+
+
+
+ (allow-none): The name of the profile, if %NULL
+provided, it will default to the encoding profile called `default`.
+
+
+
+ The target category. Can be %NULL
+
+
+
+
+
+ Creates a #GstEncodingProfile matching the formats from the given
+#GstDiscovererInfo. Streams other than audio or video (eg,
+subtitles), are currently ignored.
+
+ The new #GstEncodingProfile or %NULL.
+
+
+
+
+ The #GstDiscovererInfo to read from
+
+
+
+
+
+ Makes a deep copy of @self
+
+ The copy of @self
+
+Since 1.12
+
+
+
+
+ The #GstEncodingProfile to copy
+
+
+
+
+
+ Get whether the format that has been negotiated in at some point can be renegotiated
+later during the encoding.
+
+
+
+
+
+ a #GstEncodingProfile
+
+
+
+
+
+
+ the description of the profile, can be %NULL.
+
+
+
+
+ a #GstEncodingProfile
+
+
+
+
+
+
+ a suitable file extension for @profile, or NULL.
+
+
+
+
+ a #GstEncodingProfile
+
+
+
+
+
+
+ the #GstCaps corresponding to the media format used
+in the profile. Unref after usage.
+
+
+
+
+ a #GstEncodingProfile
+
+
+
+
+
+ Computes the full output caps that this @profile will be able to consume.
+
+ The full caps the given @profile can consume. Call
+gst_caps_unref() when you are done with the caps.
+
+
+
+
+ a #GstEncodingProfile
+
+
+
+
+
+
+ the name of the profile, can be %NULL.
+
+
+
+
+ a #GstEncodingProfile
+
+
+
+
+
+
+ The number of times the profile is used in its parent
+container profile. If 0, it is not a mandatory stream.
+
+
+
+
+ a #GstEncodingProfile
+
+
+
+
+
+
+ the name of the #GstPreset to be used in the profile.
+This is the name that has been set when saving the preset.
+
+
+
+
+ a #GstEncodingProfile
+
+
+
+
+
+
+ the name of the #GstPreset factory to be used in the profile.
+
+
+
+
+ a #GstEncodingProfile
+
+
+
+
+
+
+ The restriction #GstCaps to apply before the encoder
+that will be used in the profile. The fields present in restriction caps are
+properties of the raw stream (that is before encoding), such as height and
+width for video and depth and sampling rate for audio. Does not apply to
+#GstEncodingContainerProfile (since there is no corresponding raw stream).
+Can be %NULL. Unref after usage.
+
+
+
+
+ a #GstEncodingProfile
+
+
+
+
+
+
+ the human-readable name of the type of @profile.
+
+
+
+
+ a #GstEncodingProfile
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Checks whether the two #GstEncodingProfile are equal
+
+ %TRUE if @a and @b are equal, else %FALSE.
+
+
+
+
+ a #GstEncodingProfile
+
+
+
+ a #GstEncodingProfile
+
+
+
+
+
+ Sets whether the format that has been negotiated in at some point can be renegotiated
+later during the encoding.
+
+
+
+
+
+ a #GstEncodingProfile
+
+
+
+ Whether the format that has been negotiated first can be renegotiated
+during the encoding
+
+
+
+
+
+ Set @description as the given description for the @profile. A copy of
+@description will be made internally.
+
+
+
+
+
+ a #GstEncodingProfile
+
+
+
+ the description to set on the profile
+
+
+
+
+
+ Set whether the profile should be used or not.
+
+Since 1.6
+
+
+
+
+
+ a #GstEncodingProfile
+
+
+
+ %FALSE to disable #profile, %TRUE to enable it
+
+
+
+
+
+ Sets the media format used in the profile.
+
+
+
+
+
+ a #GstEncodingProfile
+
+
+
+ the media format to use in the profile.
+
+
+
+
+
+ Set @name as the given name for the @profile. A copy of @name will be made
+internally.
+
+
+
+
+
+ a #GstEncodingProfile
+
+
+
+ the name to set on the profile
+
+
+
+
+
+ Set the number of time the profile is used in its parent
+container profile. If 0, it is not a mandatory stream
+
+
+
+
+
+ a #GstEncodingProfile
+
+
+
+ the number of time the profile can be used
+
+
+
+
+
+ Sets the name of the #GstElement that implements the #GstPreset interface
+to use for the profile.
+This is the name that has been set when saving the preset.
+
+
+
+
+
+ a #GstEncodingProfile
+
+
+
+ the element preset to use
+
+
+
+
+
+ Sets the name of the #GstPreset's factory to be used in the profile.
+
+
+
+
+
+ a #GstEncodingProfile
+
+
+
+ The name of the preset to use in this @profile.
+
+
+
+
+
+ Set the restriction #GstCaps to apply before the encoder
+that will be used in the profile. See gst_encoding_profile_get_restriction()
+for more about restrictions. Does not apply to #GstEncodingContainerProfile.
+
+
+
+
+
+ a #GstEncodingProfile
+
+
+
+ the restriction to apply
+
+
+
+
+
+
+
+
+
+
+
+ Collection of #GstEncodingProfile for a specific target or use-case.
+
+When being stored/loaded, targets come from a specific category, like
+#GST_ENCODING_CATEGORY_DEVICE.
+
+ Creates a new #GstEncodingTarget.
+
+The name and category can only consist of lowercase ASCII letters for the
+first character, followed by either lowercase ASCII letters, digits or
+hyphens ('-').
+
+The @category <emphasis>should</emphasis> be one of the existing
+well-defined categories, like #GST_ENCODING_CATEGORY_DEVICE, but it
+<emphasis>can</emphasis> be a application or user specific category if
+needed.
+
+ The newly created #GstEncodingTarget or %NULL if
+there was an error.
+
+
+
+
+ The name of the target.
+
+
+
+ The name of the category to which this @target
+belongs. For example: #GST_ENCODING_CATEGORY_DEVICE.
+
+
+
+ A description of #GstEncodingTarget in the
+current locale.
+
+
+
+ A #GList of
+#GstEncodingProfile.
+
+
+
+
+
+
+
+ Searches for the #GstEncodingTarget with the given name, loads it
+and returns it.
+
+If the category name is specified only targets from that category will be
+searched for.
+
+ The #GstEncodingTarget if available, else %NULL.
+
+
+
+
+ the name of the #GstEncodingTarget to load (automatically
+converted to lower case internally as capital letters are not
+valid for target names).
+
+
+
+ the name of the target category, like
+#GST_ENCODING_CATEGORY_DEVICE. Can be %NULL
+
+
+
+
+
+ Opens the provided file and returns the contained #GstEncodingTarget.
+
+ The #GstEncodingTarget contained in the file, else
+%NULL
+
+
+
+
+ The file location to load the #GstEncodingTarget from
+
+
+
+
+
+ Adds the given @profile to the @target. Each added profile must have
+a unique name within the profile.
+
+The @target will steal a reference to the @profile. If you wish to use
+the profile after calling this method, you should increase its reference
+count.
+
+ %TRUE if the profile was added, else %FALSE.
+
+
+
+
+ the #GstEncodingTarget to add a profile to
+
+
+
+ the #GstEncodingProfile to add
+
+
+
+
+
+
+ The category of the @target. For example:
+#GST_ENCODING_CATEGORY_DEVICE.
+
+
+
+
+ a #GstEncodingTarget
+
+
+
+
+
+
+ The description of the @target.
+
+
+
+
+ a #GstEncodingTarget
+
+
+
+
+
+
+ The name of the @target.
+
+
+
+
+ a #GstEncodingTarget
+
+
+
+
+
+
+ The matching #GstEncodingProfile, or %NULL.
+
+
+
+
+ a #GstEncodingTarget
+
+
+
+ the name of the profile to retrieve
+
+
+
+
+
+
+ A list of
+#GstEncodingProfile(s) this @target handles.
+
+
+
+
+
+
+ a #GstEncodingTarget
+
+
+
+
+
+ Saves the @target to a default user-local directory.
+
+ %TRUE if the target was correctly saved, else %FALSE.
+
+
+
+
+ a #GstEncodingTarget
+
+
+
+
+
+ Saves the @target to the provided file location.
+
+ %TRUE if the target was correctly saved, else %FALSE.
+
+
+
+
+ a #GstEncodingTarget
+
+
+
+ the location to store the @target at.
+
+
+
+
+
+
+ Variant of #GstEncodingProfile for video streams, allows specifying the @pass.
+
+ Creates a new #GstEncodingVideoProfile
+
+All provided allocatable arguments will be internally copied, so can be
+safely freed/unreferenced after calling this method.
+
+If you wish to control the pass number (in case of multi-pass scenarios),
+please refer to the gst_encoding_video_profile_set_pass() documentation.
+
+If you wish to use/force a constant framerate please refer to the
+gst_encoding_video_profile_set_variableframerate() documentation.
+
+ the newly created #GstEncodingVideoProfile.
+
+
+
+
+ the #GstCaps
+
+
+
+ the preset(s) to use on the encoder, can be %NULL
+
+
+
+ the #GstCaps used to restrict the input to the encoder, can be
+NULL. See gst_encoding_profile_get_restriction() for more details.
+
+
+
+ the number of time this stream must be used. 0 means any number of
+ times (including never)
+
+
+
+
+
+ Get the pass number if this is part of a multi-pass profile.
+
+ The pass number. Starts at 1 for multi-pass. 0 if this is
+not a multi-pass profile
+
+
+
+
+ a #GstEncodingVideoProfile
+
+
+
+
+
+
+ Whether non-constant video framerate is allowed for encoding.
+
+
+
+
+ a #GstEncodingVideoProfile
+
+
+
+
+
+ Sets the pass number of this video profile. The first pass profile should have
+this value set to 1. If this video profile isn't part of a multi-pass profile,
+you may set it to 0 (the default value).
+
+
+
+
+
+ a #GstEncodingVideoProfile
+
+
+
+ the pass number for this profile
+
+
+
+
+
+ If set to %TRUE, then the incoming stream will be allowed to have non-constant
+framerate. If set to %FALSE (default value), then the incoming stream will
+be normalized by dropping/duplicating frames in order to produce a
+constance framerate.
+
+
+
+
+
+ a #GstEncodingVideoProfile
+
+
+
+ a boolean
+
+
+
+
+
+
+
+
+ Opaque context structure for the plugin installation. Use the provided
+API to set details on it.
+
+ Creates a new #GstInstallPluginsContext.
+
+ a new #GstInstallPluginsContext. Free with
+gst_install_plugins_context_free() when no longer needed
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Frees a #GstInstallPluginsContext.
+
+
+
+
+
+ a #GstInstallPluginsContext
+
+
+
+
+
+ This function is used to tell the external installer process whether it
+should ask for confirmation or not before searching for missing plugins.
+
+If set, this option will be passed to the installer via a
+--interaction=[show-confirm-search|hide-confirm-search] command line option.
+
+
+
+
+
+ a #GstInstallPluginsContext
+
+
+
+ whether to ask for confirmation before searching for plugins
+
+
+
+
+
+ This function is used to pass the calling application's desktop file ID to
+the external installer process.
+
+A desktop file ID is the basename of the desktop file, including the
+.desktop extension.
+
+If set, the desktop file ID will be passed to the installer via a
+--desktop-id= command line option.
+
+
+
+
+
+ a #GstInstallPluginsContext
+
+
+
+ the desktop file ID of the calling application
+
+
+
+
+
+ Sets the startup notification ID for the launched process.
+
+This is typically used to to pass the current X11 event timestamp to the
+external installer process.
+
+Startup notification IDs are defined in the
+[FreeDesktop.Org Startup Notifications standard](http://standards.freedesktop.org/startup-notification-spec/startup-notification-latest.txt).
+
+If set, the ID will be passed to the installer via a
+--startup-notification-id= command line option.
+
+GTK+/GNOME applications should be able to create a startup notification ID
+like this:
+|[
+ timestamp = gtk_get_current_event_time ();
+ startup_id = g_strdup_printf ("_TIME%u", timestamp);
+...
+]|
+
+
+
+
+
+ a #GstInstallPluginsContext
+
+
+
+ the startup notification ID
+
+
+
+
+
+ This function is for X11-based applications (such as most Gtk/Qt
+applications on linux/unix) only. You can use it to tell the external
+installer the XID of your main application window. That way the installer
+can make its own window transient to your application window during the
+installation.
+
+If set, the XID will be passed to the installer via a --transient-for=XID
+command line option.
+
+Gtk+/Gnome application should be able to obtain the XID of the top-level
+window like this:
+|[
+##include <gtk/gtk.h>
+##ifdef GDK_WINDOWING_X11
+##include <gdk/gdkx.h>
+##endif
+...
+##ifdef GDK_WINDOWING_X11
+ xid = GDK_WINDOW_XWINDOW (GTK_WIDGET (application_window)->window);
+##endif
+...
+]|
+
+
+
+
+
+ a #GstInstallPluginsContext
+
+
+
+ the XWindow ID (XID) of the top-level application
+
+
+
+
+
+
+ The prototype of the callback function that will be called once the
+external plugin installer program has returned. You only need to provide
+a callback function if you are using the asynchronous interface.
+
+
+
+
+
+ whether the installation of the requested plugins succeeded or not
+
+
+
+ the user data passed to gst_install_plugins_async()
+
+
+
+
+
+ Result codes returned by gst_install_plugins_async() and
+gst_install_plugins_sync(), and also the result code passed to the
+#GstInstallPluginsResultFunc specified with gst_install_plugins_async().
+
+These codes indicate success or failure of starting an external installer
+program and to what extent the requested plugins could be installed.
+
+ all of the requested plugins could be
+ installed
+
+
+ no appropriate installation candidate for
+ any of the requested plugins could be found. Only return this if nothing
+ has been installed. Return #GST_INSTALL_PLUGINS_PARTIAL_SUCCESS if
+ some (but not all) of the requested plugins could be installed.
+
+
+ an error occured during the installation. If
+ this happens, the user has already seen an error message and another
+ one should not be displayed
+
+
+ some of the requested plugins could
+ be installed, but not all
+
+
+ the user has aborted the installation
+
+
+ the installer had an unclean exit code
+ (ie. death by signal)
+
+
+ the helper returned an invalid status code
+
+
+ returned by gst_install_plugins_async() to
+ indicate that everything went fine so far and the provided callback
+ will be called with the result of the installation later
+
+
+ some internal failure has
+ occured when trying to start the installer
+
+
+ the helper script to call the
+ actual installer is not installed
+
+
+ a previously-started plugin
+ installation is still in progress, try again later
+
+
+ Convenience function to return the descriptive string associated
+with a status code. This function returns English strings and
+should not be used for user messages. It is here only to assist
+in debugging.
+
+ a descriptive string for the status code in @ret
+
+
+
+
+ the return status code
+
+
+
+
+
+
+ The major version of GStreamer's gst-plugins-base libraries at compile time.
+
+
+
+ The micro version of GStreamer's gst-plugins-base libraries at compile time.
+
+
+
+ The minor version of GStreamer's gst-plugins-base libraries at compile time.
+
+
+
+ The nano version of GStreamer's gst-plugins-base libraries at compile time.
+Actual releases have 0, GIT versions have 1, prerelease versions have 2-...
+
+
+
+ Sets the level and profile on @caps if it can be determined from
+@audio_config. See gst_codec_utils_aac_get_level() and
+gst_codec_utils_aac_get_profile() for more details on the parameters.
+@caps must be audio/mpeg caps with an "mpegversion" field of either 2 or 4.
+If mpegversion is 4, the "base-profile" field is also set in @caps.
+
+ %TRUE if the level and profile could be set, %FALSE otherwise.
+
+
+
+
+ the #GstCaps to which level and profile fields are to be added
+
+
+
+ a pointer to the AudioSpecificConfig
+ as specified in the Elementary Stream Descriptor (esds)
+ in ISO/IEC 14496-1. (See below for more details)
+
+
+
+
+
+ Length of @audio_config in bytes
+
+
+
+
+
+ Returns the channels of the given AAC stream.
+
+ The channels or 0 if the channel could not be determined.
+
+Since 1.10
+
+
+
+
+ a pointer to the AudioSpecificConfig
+ as specified in the Elementary Stream Descriptor (esds)
+ in ISO/IEC 14496-1.
+
+
+
+
+
+ Length of @audio_config in bytes
+
+
+
+
+
+ Translates the sample rate to the index corresponding to it in AAC spec.
+
+ The AAC index for this sample rate, -1 if the rate is not a
+valid AAC sample rate.
+
+
+
+
+ Sample rate
+
+
+
+
+
+ Determines the level of a stream as defined in ISO/IEC 14496-3. For AAC LC
+streams, the constraints from the AAC audio profile are applied. For AAC
+Main, LTP, SSR and others, the Main profile is used.
+
+The @audio_config parameter follows the following format, starting from the
+most significant bit of the first byte:
+
+ * Bit 0:4 contains the AudioObjectType
+ * Bit 5:8 contains the sample frequency index (if this is 0xf, then the
+ next 24 bits define the actual sample frequency, and subsequent
+ fields are appropriately shifted).
+ * Bit 9:12 contains the channel configuration
+
+> HE-AAC support has not yet been implemented.
+
+ The level as a const string and %NULL if the level could not be
+determined.
+
+
+
+
+ a pointer to the AudioSpecificConfig
+ as specified in the Elementary Stream Descriptor (esds)
+ in ISO/IEC 14496-1.
+
+
+
+
+
+ Length of @audio_config in bytes
+
+
+
+
+
+ Returns the profile of the given AAC stream as a string. The profile is
+determined using the AudioObjectType field which is in the first 5 bits of
+@audio_config.
+
+> HE-AAC support has not yet been implemented.
+
+ The profile as a const string and %NULL if the profile could not be
+determined.
+
+
+
+
+ a pointer to the AudioSpecificConfig
+ as specified in the Elementary Stream Descriptor (esds)
+ in ISO/IEC 14496-1.
+
+
+
+
+
+ Length of @audio_config in bytes
+
+
+
+
+
+ Translates the sample rate index found in AAC headers to the actual sample
+rate.
+
+ The sample rate if sr_idx is valid, 0 otherwise.
+
+Since 1.10
+
+
+
+
+ a pointer to the AudioSpecificConfig
+ as specified in the Elementary Stream Descriptor (esds)
+ in ISO/IEC 14496-1.
+
+
+
+
+
+ Length of @audio_config
+
+
+
+
+
+ Translates the sample rate index found in AAC headers to the actual sample
+rate.
+
+ The sample rate if @sr_idx is valid, 0 otherwise.
+
+
+
+
+ Sample rate index as from the AudioSpecificConfig (MPEG-4
+ container) or ADTS frame header
+
+
+
+
+
+ Sets the level and profile in @caps if it can be determined from @sps. See
+gst_codec_utils_h264_get_level() and gst_codec_utils_h264_get_profile()
+for more details on the parameters.
+
+ %TRUE if the level and profile could be set, %FALSE otherwise.
+
+
+
+
+ the #GstCaps to which the level and profile are to be added
+
+
+
+ Pointer to the sequence parameter set for the stream.
+
+
+
+
+
+ Length of the data available in @sps.
+
+
+
+
+
+ Converts the level indication (level_idc) in the stream's
+sequence parameter set into a string. The SPS is expected to have the
+same format as for gst_codec_utils_h264_get_profile().
+
+ The level as a const string, or %NULL if there is an error.
+
+
+
+
+ Pointer to the sequence parameter set for the stream.
+
+
+
+
+
+ Length of the data available in @sps.
+
+
+
+
+
+ Transform a level string from the caps into the level_idc
+
+ the level_idc or 0 if the level is unknown
+
+
+
+
+ A level string from caps
+
+
+
+
+
+ Converts the profile indication (profile_idc) in the stream's
+sequence parameter set into a string. The SPS is expected to have the
+following format, as defined in the H.264 specification. The SPS is viewed
+as a bitstream here, with bit 0 being the most significant bit of the first
+byte.
+
+* Bit 0:7 - Profile indication
+* Bit 8 - constraint_set0_flag
+* Bit 9 - constraint_set1_flag
+* Bit 10 - constraint_set2_flag
+* Bit 11 - constraint_set3_flag
+* Bit 12 - constraint_set3_flag
+* Bit 13:15 - Reserved
+* Bit 16:24 - Level indication
+
+ The profile as a const string, or %NULL if there is an error.
+
+
+
+
+ Pointer to the sequence parameter set for the stream.
+
+
+
+
+
+ Length of the data available in @sps.
+
+
+
+
+
+ Sets the level, tier and profile in @caps if it can be determined from
+@profile_tier_level. See gst_codec_utils_h265_get_level(),
+gst_codec_utils_h265_get_tier() and gst_codec_utils_h265_get_profile()
+for more details on the parameters.
+
+ %TRUE if the level, tier, profile could be set, %FALSE otherwise.
+
+Since 1.4
+
+
+
+
+ the #GstCaps to which the level, tier and profile are to be added
+
+
+
+ Pointer to the profile_tier_level
+ struct
+
+
+
+
+
+ Length of the data available in @profile_tier_level.
+
+
+
+
+
+ Converts the level indication (general_level_idc) in the stream's
+profile_tier_level structure into a string. The profiel_tier_level is
+expected to have the same format as for gst_codec_utils_h264_get_profile().
+
+ The level as a const string, or %NULL if there is an error.
+
+Since 1.4
+
+
+
+
+ Pointer to the profile_tier_level
+ for the stream
+
+
+
+
+
+ Length of the data available in @profile_tier_level.
+
+
+
+
+
+ Transform a level string from the caps into the level_idc
+
+ the level_idc or 0 if the level is unknown
+
+Since 1.4
+
+
+
+
+ A level string from caps
+
+
+
+
+
+ Converts the profile indication (general_profile_idc) in the stream's
+profile_level_tier structure into a string. The profile_tier_level is
+expected to have the following format, as defined in the H.265
+specification. The profile_tier_level is viewed as a bitstream here,
+with bit 0 being the most significant bit of the first byte.
+
+* Bit 0:1 - general_profile_space
+* Bit 2 - general_tier_flag
+* Bit 3:7 - general_profile_idc
+* Bit 8:39 - gernal_profile_compatibility_flags
+* Bit 40 - general_progressive_source_flag
+* Bit 41 - general_interlaced_source_flag
+* Bit 42 - general_non_packed_constraint_flag
+* Bit 43 - general_frame_only_constraint_flag
+* Bit 44:87 - general_reserved_zero_44bits
+* Bit 88:95 - general_level_idc
+
+ The profile as a const string, or %NULL if there is an error.
+
+Since 1.4
+
+
+
+
+ Pointer to the profile_tier_level
+ structure for the stream.
+
+
+
+
+
+ Length of the data available in @profile_tier_level
+
+
+
+
+
+ Converts the tier indication (general_tier_flag) in the stream's
+profile_tier_level structure into a string. The profile_tier_level
+is expected to have the same format as for gst_codec_utils_h264_get_profile().
+
+ The tier as a const string, or %NULL if there is an error.
+
+Since 1.4
+
+
+
+
+ Pointer to the profile_tier_level
+ for the stream.
+
+
+
+
+
+ Length of the data available in @profile_tier_level.
+
+
+
+
+
+ Sets the level and profile in @caps if it can be determined from
+@vis_obj_seq. See gst_codec_utils_mpeg4video_get_level() and
+gst_codec_utils_mpeg4video_get_profile() for more details on the
+parameters.
+
+ %TRUE if the level and profile could be set, %FALSE otherwise.
+
+
+
+
+ the #GstCaps to which the level and profile are to be added
+
+
+
+ Pointer to the visual object
+ sequence for the stream.
+
+
+
+
+
+ Length of the data available in @sps.
+
+
+
+
+
+ Converts the level indication in the stream's visual object sequence into
+a string. @vis_obj_seq is expected to be the data following the visual
+object sequence start code. Only the first byte
+(profile_and_level_indication) is used.
+
+ The level as a const string, or NULL if there is an error.
+
+
+
+
+ Pointer to the visual object
+ sequence for the stream.
+
+
+
+
+
+ Length of the data available in @sps.
+
+
+
+
+
+ Converts the profile indication in the stream's visual object sequence into
+a string. @vis_obj_seq is expected to be the data following the visual
+object sequence start code. Only the first byte
+(profile_and_level_indication) is used.
+
+ The profile as a const string, or NULL if there is an error.
+
+
+
+
+ Pointer to the visual object
+ sequence for the stream.
+
+
+
+
+
+ Length of the data available in @sps.
+
+
+
+
+
+ Creates Opus caps from the given parameters.
+
+ The #GstCaps, or %NULL if the parameters would lead to
+invalid Opus caps.
+
+
+
+
+ the sample rate
+
+
+
+ the number of channels
+
+
+
+ the channel mapping family
+
+
+
+ the number of independent streams
+
+
+
+ the number of stereo streams
+
+
+
+ the mapping between the streams
+
+
+
+
+
+
+
+ Creates Opus caps from the given OpusHead @header and comment header
+@comments.
+
+ The #GstCaps.
+
+
+
+
+ OpusHead header
+
+
+
+ Comment header or NULL
+
+
+
+
+
+ Creates OpusHead header from the given parameters.
+
+ The #GstBuffer containing the OpusHead.
+
+
+
+
+ the sample rate
+
+
+
+ the number of channels
+
+
+
+ the channel mapping family
+
+
+
+ the number of independent streams
+
+
+
+ the number of stereo streams
+
+
+
+ the mapping between the streams
+
+
+
+
+
+ Pre-skip in 48kHz samples or 0
+
+
+
+ Output gain or 0
+
+
+
+
+
+ Parses Opus caps and fills the different fields with defaults if possible.
+
+ %TRUE if parsing was successful, %FALSE otherwise.
+
+
+
+
+ the #GstCaps to parse the data from
+
+
+
+ the sample rate
+
+
+
+ the number of channels
+
+
+
+ the channel mapping family
+
+
+
+ the number of independent streams
+
+
+
+ the number of stereo streams
+
+
+
+ the mapping between the streams
+
+
+
+
+
+
+
+ Parses the OpusHead header.
+
+ %TRUE if parsing was successful, %FALSE otherwise.
+
+
+
+
+ the OpusHead #GstBuffer
+
+
+
+ the sample rate
+
+
+
+ the number of channels
+
+
+
+ the channel mapping family
+
+
+
+ the number of independent streams
+
+
+
+ the number of stereo streams
+
+
+
+ the mapping between the streams
+
+
+
+
+
+ Pre-skip in 48kHz samples or 0
+
+
+
+ Output gain or 0
+
+
+
+
+
+ List all available #GstEncodingTarget for the specified category, or all categories
+if @categoryname is %NULL.
+
+ The list of #GstEncodingTarget
+
+
+
+
+
+
+ The category, for ex: #GST_ENCODING_CATEGORY_DEVICE.
+Can be %NULL.
+
+
+
+
+
+ Lists all #GstEncodingTarget categories present on disk.
+
+ A list
+of #GstEncodingTarget categories.
+
+
+
+
+
+
+ Requests plugin installation without blocking. Once the plugins have been
+installed or installation has failed, @func will be called with the result
+of the installation and your provided @user_data pointer.
+
+This function requires a running GLib/Gtk main loop. If you are not
+running a GLib/Gtk main loop, make sure to regularly call
+g_main_context_iteration(NULL,FALSE).
+
+The installer strings that make up @detail are typically obtained by
+calling gst_missing_plugin_message_get_installer_detail() on missing-plugin
+messages that have been caught on a pipeline's bus or created by the
+application via the provided API, such as gst_missing_element_message_new().
+
+It is possible to request the installation of multiple missing plugins in
+one go (as might be required if there is a demuxer for a certain format
+installed but no suitable video decoder and no suitable audio decoder).
+
+ result code whether an external installer could be started
+
+
+
+
+ NULL-terminated array
+ of installer string details (see below)
+
+
+
+
+
+ a #GstInstallPluginsContext, or NULL
+
+
+
+ the function to call when the installer program returns
+
+
+
+ the user data to pass to @func when called, or NULL
+
+
+
+
+
+ Checks whether plugin installation (initiated by this application only)
+is currently in progress.
+
+ TRUE if plugin installation is in progress, otherwise FALSE
+
+
+
+
+ Convenience function to return the descriptive string associated
+with a status code. This function returns English strings and
+should not be used for user messages. It is here only to assist
+in debugging.
+
+ a descriptive string for the status code in @ret
+
+
+
+
+ the return status code
+
+
+
+
+
+ Checks whether plugin installation is likely to be supported by the
+current environment. This currently only checks whether the helper script
+that is to be provided by the distribution or operating system vendor
+exists.
+
+ TRUE if plugin installation is likely to be supported.
+
+
+
+
+ Requests plugin installation and block until the plugins have been
+installed or installation has failed.
+
+This function should almost never be used, it only exists for cases where
+a non-GLib main loop is running and the user wants to run it in a separate
+thread and marshal the result back asynchronously into the main thread
+using the other non-GLib main loop. You should almost always use
+gst_install_plugins_async() instead of this function.
+
+ the result of the installation.
+
+
+
+
+ NULL-terminated array
+ of installer string details
+
+
+
+
+
+ a #GstInstallPluginsContext, or NULL
+
+
+
+
+
+ Checks whether @msg is a missing plugins message.
+
+ %TRUE if @msg is a missing-plugins message, otherwise %FALSE.
+
+
+
+
+ a #GstMessage
+
+
+
+
+
+ Returns an opaque string containing all the details about the missing
+element to be passed to an external installer called via
+gst_install_plugins_async() or gst_install_plugins_sync().
+
+This function is mainly for applications that call external plugin
+installation mechanisms using one of the two above-mentioned functions in
+the case where the application knows exactly what kind of plugin it is
+missing.
+
+ a newly-allocated detail string, or NULL on error. Free string
+ with g_free() when not needed any longer.
+
+
+
+
+ the (fixed) caps for which a decoder element is needed
+
+
+
+
+
+ Creates a missing-plugin message for @element to notify the application
+that a decoder element for a particular set of (fixed) caps is missing.
+This function is mainly for use in plugins.
+
+ a new #GstMessage, or NULL on error
+
+
+
+
+ the #GstElement posting the message
+
+
+
+ the (fixed) caps for which a decoder element is needed
+
+
+
+
+
+ Returns an opaque string containing all the details about the missing
+element to be passed to an external installer called via
+gst_install_plugins_async() or gst_install_plugins_sync().
+
+This function is mainly for applications that call external plugin
+installation mechanisms using one of the two above-mentioned functions in
+the case where the application knows exactly what kind of plugin it is
+missing.
+
+ a newly-allocated detail string, or NULL on error. Free string
+ with g_free() when not needed any longer.
+
+
+
+
+ the name of the missing element (element factory),
+ e.g. "videoscale" or "cdparanoiasrc"
+
+
+
+
+
+ Creates a missing-plugin message for @element to notify the application
+that a certain required element is missing. This function is mainly for
+use in plugins.
+
+ a new #GstMessage, or NULL on error
+
+
+
+
+ the #GstElement posting the message
+
+
+
+ the name of the missing element (element factory),
+ e.g. "videoscale" or "cdparanoiasrc"
+
+
+
+
+
+ Returns an opaque string containing all the details about the missing
+element to be passed to an external installer called via
+gst_install_plugins_async() or gst_install_plugins_sync().
+
+This function is mainly for applications that call external plugin
+installation mechanisms using one of the two above-mentioned functions in
+the case where the application knows exactly what kind of plugin it is
+missing.
+
+ a newly-allocated detail string, or NULL on error. Free string
+ with g_free() when not needed any longer.
+
+
+
+
+ the (fixed) caps for which an encoder element is needed
+
+
+
+
+
+ Creates a missing-plugin message for @element to notify the application
+that an encoder element for a particular set of (fixed) caps is missing.
+This function is mainly for use in plugins.
+
+ a new #GstMessage, or NULL on error
+
+
+
+
+ the #GstElement posting the message
+
+
+
+ the (fixed) caps for which an encoder element is needed
+
+
+
+
+
+ Returns a localised string describing the missing feature, for use in
+error dialogs and the like. Should never return NULL unless @msg is not
+a valid missing-plugin message.
+
+This function is mainly for applications that need a human-readable string
+describing a missing plugin, given a previously collected missing-plugin
+message
+
+ a newly-allocated description string, or NULL on error. Free
+ string with g_free() when not needed any longer.
+
+
+
+
+ a missing-plugin #GstMessage of type #GST_MESSAGE_ELEMENT
+
+
+
+
+
+ Returns an opaque string containing all the details about the missing
+element to be passed to an external installer called via
+gst_install_plugins_async() or gst_install_plugins_sync().
+
+This function is mainly for applications that call external plugin
+installation mechanisms using one of the two above-mentioned functions.
+
+ a newly-allocated detail string, or NULL on error. Free string
+ with g_free() when not needed any longer.
+
+
+
+
+ a missing-plugin #GstMessage of type #GST_MESSAGE_ELEMENT
+
+
+
+
+
+ Returns an opaque string containing all the details about the missing
+element to be passed to an external installer called via
+gst_install_plugins_async() or gst_install_plugins_sync().
+
+This function is mainly for applications that call external plugin
+installation mechanisms using one of the two above-mentioned functions in
+the case where the application knows exactly what kind of plugin it is
+missing.
+
+ a newly-allocated detail string, or NULL on error. Free string
+ with g_free() when not needed any longer.
+
+
+
+
+ the URI protocol the missing source needs to implement,
+ e.g. "http" or "mms"
+
+
+
+
+
+ Creates a missing-plugin message for @element to notify the application
+that a sink element for a particular URI protocol is missing. This
+function is mainly for use in plugins.
+
+ a new #GstMessage, or NULL on error
+
+
+
+
+ the #GstElement posting the message
+
+
+
+ the URI protocol the missing sink needs to implement,
+ e.g. "http" or "smb"
+
+
+
+
+
+ Returns an opaque string containing all the details about the missing
+element to be passed to an external installer called via
+gst_install_plugins_async() or gst_install_plugins_sync().
+
+This function is mainly for applications that call external plugin
+installation mechanisms using one of the two above-mentioned functions in
+the case where the application knows exactly what kind of plugin it is
+missing.
+
+ a newly-allocated detail string, or NULL on error. Free string
+ with g_free() when not needed any longer.
+
+
+
+
+ the URI protocol the missing source needs to implement,
+ e.g. "http" or "mms"
+
+
+
+
+
+ Creates a missing-plugin message for @element to notify the application
+that a source element for a particular URI protocol is missing. This
+function is mainly for use in plugins.
+
+ a new #GstMessage, or NULL on error
+
+
+
+
+ the #GstElement posting the message
+
+
+
+ the URI protocol the missing source needs to implement,
+ e.g. "http" or "mms"
+
+
+
+
+
+ Adds a codec tag describing the format specified by @caps to @taglist.
+
+ TRUE if a codec tag was added, FALSE otherwise.
+
+
+
+
+ a #GstTagList
+
+
+
+ a GStreamer codec tag such as #GST_TAG_AUDIO_CODEC,
+ #GST_TAG_VIDEO_CODEC or #GST_TAG_CODEC. If none is specified,
+ the function will attempt to detect the appropriate category.
+
+
+
+ the (fixed) #GstCaps for which a codec tag should be added.
+
+
+
+
+
+ Returns a localised (as far as this is possible) string describing the
+media format specified in @caps, for use in error dialogs or other messages
+to be seen by the user. Should never return NULL unless @caps is invalid.
+
+Also see the convenience function
+gst_pb_utils_add_codec_description_to_tag_list().
+
+ a newly-allocated description string, or NULL on error. Free
+ string with g_free() when not needed any longer.
+
+
+
+
+ the (fixed) #GstCaps for which an format description is needed
+
+
+
+
+
+ Returns a localised string describing an decoder for the format specified
+in @caps, for use in error dialogs or other messages to be seen by the user.
+Should never return NULL unless @factory_name or @caps are invalid.
+
+This function is mainly for internal use, applications would typically
+use gst_missing_plugin_message_get_description() to get a description of
+a missing feature from a missing-plugin message.
+
+ a newly-allocated description string, or NULL on error. Free
+ string with g_free() when not needed any longer.
+
+
+
+
+ the (fixed) #GstCaps for which an decoder description is needed
+
+
+
+
+
+ Returns a localised string describing the given element, for use in
+error dialogs or other messages to be seen by the user. Should never
+return NULL unless @factory_name is invalid.
+
+This function is mainly for internal use, applications would typically
+use gst_missing_plugin_message_get_description() to get a description of
+a missing feature from a missing-plugin message.
+
+ a newly-allocated description string, or NULL on error. Free
+ string with g_free() when not needed any longer.
+
+
+
+
+ the name of the element, e.g. "giosrc"
+
+
+
+
+
+ Returns a localised string describing an encoder for the format specified
+in @caps, for use in error dialogs or other messages to be seen by the user.
+Should never return NULL unless @factory_name or @caps are invalid.
+
+This function is mainly for internal use, applications would typically
+use gst_missing_plugin_message_get_description() to get a description of
+a missing feature from a missing-plugin message.
+
+ a newly-allocated description string, or NULL on error. Free
+ string with g_free() when not needed any longer.
+
+
+
+
+ the (fixed) #GstCaps for which an encoder description is needed
+
+
+
+
+
+ Returns a localised string describing a sink element handling the protocol
+specified in @protocol, for use in error dialogs or other messages to be
+seen by the user. Should never return NULL unless @protocol is invalid.
+
+This function is mainly for internal use, applications would typically
+use gst_missing_plugin_message_get_description() to get a description of
+a missing feature from a missing-plugin message.
+
+ a newly-allocated description string, or NULL on error. Free
+ string with g_free() when not needed any longer.
+
+
+
+
+ the protocol the sink element needs to handle, e.g. "http"
+
+
+
+
+
+ Returns a localised string describing a source element handling the protocol
+specified in @protocol, for use in error dialogs or other messages to be
+seen by the user. Should never return NULL unless @protocol is invalid.
+
+This function is mainly for internal use, applications would typically
+use gst_missing_plugin_message_get_description() to get a description of
+a missing feature from a missing-plugin message.
+
+ a newly-allocated description string, or NULL on error. Free
+ string with g_free() when not needed any longer.
+
+
+
+
+ the protocol the source element needs to handle, e.g. "http"
+
+
+
+
+
+ Initialises the base utils support library. This function is not
+thread-safe. Applications should call it after calling gst_init(),
+plugins should call it from their plugin_init function.
+
+This function may be called multiple times. It will do nothing if the
+library has already been initialised.
+
+
+
+
+
+ Gets the version number of the GStreamer Plugins Base libraries.
+
+
+
+
+
+ pointer to a guint to store the major version number, or %NULL
+
+
+
+ pointer to a guint to store the minor version number, or %NULL
+
+
+
+ pointer to a guint to store the micro version number, or %NULL
+
+
+
+ pointer to a guint to store the nano version number, or %NULL
+
+
+
+
+
+ This function returns a string that is useful for describing this version
+of GStreamer's gst-plugins-base libraries to the outside world: user agent
+strings, logging, about dialogs ...
+
+ a newly allocated string describing this version of gst-plugins-base
+
+
+
+
+
diff --git a/girs/GstRtp-1.0.gir b/girs/GstRtp-1.0.gir
new file mode 100644
index 0000000000..d80e51259f
--- /dev/null
+++ b/girs/GstRtp-1.0.gir
@@ -0,0 +1,4824 @@
+
+
+
+
+
+
+
+
+
+ Note: The API in this module is not yet declared stable.
+
+The GstRTPCBuffer helper functions makes it easy to parse and create regular
+#GstBuffer objects that contain compound RTCP packets. These buffers are typically
+of 'application/x-rtcp' #GstCaps.
+
+An RTCP buffer consists of 1 or more #GstRTCPPacket structures that you can
+retrieve with gst_rtcp_buffer_get_first_packet(). #GstRTCPPacket acts as a pointer
+into the RTCP buffer; you can move to the next packet with
+gst_rtcp_packet_move_to_next().
+
+
+
+
+
+
+
+ Add a new packet of @type to @rtcp. @packet will point to the newly created
+packet.
+
+ %TRUE if the packet could be created. This function returns %FALSE
+if the max mtu is exceeded for the buffer.
+
+
+
+
+ a valid RTCP buffer
+
+
+
+ the #GstRTCPType of the new packet
+
+
+
+ pointer to new packet
+
+
+
+
+
+ Initialize a new #GstRTCPPacket pointer that points to the first packet in
+@rtcp.
+
+ TRUE if the packet existed in @rtcp.
+
+
+
+
+ a valid RTCP buffer
+
+
+
+ a #GstRTCPPacket
+
+
+
+
+
+ Get the number of RTCP packets in @rtcp.
+
+ the number of RTCP packets in @rtcp.
+
+
+
+
+ a valid RTCP buffer
+
+
+
+
+
+ Finish @rtcp after being constructed. This function is usually called
+after gst_rtcp_buffer_map() and after adding the RTCP items to the new buffer.
+
+The function adjusts the size of @rtcp with the total length of all the
+added packets.
+
+
+
+
+
+ a buffer with an RTCP packet
+
+
+
+
+
+ Open @buffer for reading or writing, depending on @flags. The resulting RTCP
+buffer state is stored in @rtcp.
+
+
+
+
+
+ a buffer with an RTCP packet
+
+
+
+ flags for the mapping
+
+
+
+ resulting #GstRTCPBuffer
+
+
+
+
+
+ Create a new buffer for constructing RTCP packets. The packet will have a
+maximum size of @mtu.
+
+ A newly allocated buffer.
+
+
+
+
+ the maximum mtu size.
+
+
+
+
+
+ Create a new buffer and set the data to a copy of @len
+bytes of @data and the size to @len. The data will be freed when the buffer
+is freed.
+
+ A newly allocated buffer with a copy of @data and of size @len.
+
+
+
+
+ data for the new buffer
+
+
+
+
+
+ the length of data
+
+
+
+
+
+ Create a new buffer and set the data and size of the buffer to @data and @len
+respectively. @data will be freed when the buffer is unreffed, so this
+function transfers ownership of @data to the new buffer.
+
+ A newly allocated buffer with @data and of size @len.
+
+
+
+
+ data for the new buffer
+
+
+
+
+
+ the length of data
+
+
+
+
+
+ Check if the data pointed to by @buffer is a valid RTCP packet using
+gst_rtcp_buffer_validate_data().
+
+ TRUE if @buffer is a valid RTCP packet.
+
+
+
+
+ the buffer to validate
+
+
+
+
+
+ Check if the @data and @size point to the data of a valid compound,
+non-reduced size RTCP packet.
+Use this function to validate a packet before using the other functions in
+this module.
+
+ TRUE if the data points to a valid RTCP packet.
+
+
+
+
+ the data to validate
+
+
+
+
+
+ the length of @data to validate
+
+
+
+
+
+ Check if the @data and @size point to the data of a valid RTCP packet.
+Use this function to validate a packet before using the other functions in
+this module.
+
+This function is updated to support reduced size rtcp packets according to
+RFC 5506 and will validate full compound RTCP packets as well as reduced
+size RTCP packets.
+
+ TRUE if the data points to a valid RTCP packet.
+
+
+
+
+ the data to validate
+
+
+
+
+
+ the length of @data to validate
+
+
+
+
+
+ Check if the data pointed to by @buffer is a valid RTCP packet using
+gst_rtcp_buffer_validate_reduced().
+
+ TRUE if @buffer is a valid RTCP packet.
+
+
+
+
+ the buffer to validate
+
+
+
+
+
+
+ Different types of feedback messages.
+
+ Invalid type
+
+
+ Generic NACK
+
+
+ Temporary Maximum Media Stream Bit Rate Request
+
+
+ Temporary Maximum Media Stream Bit Rate
+ Notification
+
+
+ Request an SR packet for early
+ synchronization
+
+
+ Picture Loss Indication
+
+
+ Slice Loss Indication
+
+
+ Reference Picture Selection Indication
+
+
+ Application layer Feedback
+
+
+ Full Intra Request Command
+
+
+ Temporal-Spatial Trade-off Request
+
+
+ Temporal-Spatial Trade-off Notification
+
+
+ Video Back Channel Message
+
+
+
+ Data structure that points to a packet at @offset in @buffer.
+The size of the structure is made public to allow stack allocations.
+
+ pointer to RTCP buffer
+
+
+
+ offset of packet in buffer data
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Add profile-specific extension @data to @packet. If @packet already
+contains profile-specific extension @data will be appended to the existing
+extension.
+
+ %TRUE if the profile specific extension data was added.
+
+
+
+
+ a valid SR or RR #GstRTCPPacket
+
+
+
+ profile-specific data
+
+
+
+
+
+ length of the profile-specific data in bytes
+
+
+
+
+
+ Add a new report block to @packet with the given values.
+
+ %TRUE if the packet was created. This function can return %FALSE if
+the max MTU is exceeded or the number of report blocks is greater than
+#GST_RTCP_MAX_RB_COUNT.
+
+
+
+
+ a valid SR or RR #GstRTCPPacket
+
+
+
+ data source being reported
+
+
+
+ fraction lost since last SR/RR
+
+
+
+ the cumululative number of packets lost
+
+
+
+ the extended last sequence number received
+
+
+
+ the interarrival jitter
+
+
+
+ the last SR packet from this source
+
+
+
+ the delay since last SR packet
+
+
+
+
+
+ Get the application-dependent data attached to a RTPFB or PSFB @packet.
+
+ A pointer to the data
+
+
+
+
+ a valid APP #GstRTCPPacket
+
+
+
+
+
+ Get the length of the application-dependent data attached to an APP
+@packet.
+
+ The length of data in 32-bit words.
+
+
+
+
+ a valid APP #GstRTCPPacket
+
+
+
+
+
+ Get the name field of the APP @packet.
+
+ The 4-byte name field, not zero-terminated.
+
+
+
+
+ a valid APP #GstRTCPPacket
+
+
+
+
+
+ Get the SSRC/CSRC field of the APP @packet.
+
+ The SSRC/CSRC.
+
+
+
+
+ a valid APP #GstRTCPPacket
+
+
+
+
+
+ Get the subtype field of the APP @packet.
+
+ The subtype.
+
+
+
+
+ a valid APP #GstRTCPPacket
+
+
+
+
+
+ Set the length of the application-dependent data attached to an APP
+@packet.
+
+ %TRUE if there was enough space in the packet to add this much
+data.
+
+
+
+
+ a valid APP #GstRTCPPacket
+
+
+
+ Length of the data in 32-bit words
+
+
+
+
+
+ Set the name field of the APP @packet.
+
+
+
+
+
+ a valid APP #GstRTCPPacket
+
+
+
+ 4-byte ASCII name
+
+
+
+
+
+ Set the SSRC/CSRC field of the APP @packet.
+
+
+
+
+
+ a valid APP #GstRTCPPacket
+
+
+
+ SSRC/CSRC of the packet
+
+
+
+
+
+ Set the subtype field of the APP @packet.
+
+
+
+
+
+ a valid APP #GstRTCPPacket
+
+
+
+ subtype of the packet
+
+
+
+
+
+ Add @ssrc to the BYE @packet.
+
+ %TRUE if the ssrc was added. This function can return %FALSE if
+the max MTU is exceeded or the number of sources blocks is greater than
+#GST_RTCP_MAX_BYE_SSRC_COUNT.
+
+
+
+
+ a valid BYE #GstRTCPPacket
+
+
+
+ an SSRC to add
+
+
+
+
+
+ Adds @len SSRCs in @ssrc to BYE @packet.
+
+ %TRUE if the all the SSRCs were added. This function can return %FALSE if
+the max MTU is exceeded or the number of sources blocks is greater than
+#GST_RTCP_MAX_BYE_SSRC_COUNT.
+
+
+
+
+ a valid BYE #GstRTCPPacket
+
+
+
+ an array of SSRCs to add
+
+
+
+
+
+ number of elements in @ssrc
+
+
+
+
+
+ Get the @nth SSRC of the BYE @packet.
+
+ The @nth SSRC of @packet.
+
+
+
+
+ a valid BYE #GstRTCPPacket
+
+
+
+ the nth SSRC to get
+
+
+
+
+
+ Get the reason in @packet.
+
+ The reason for the BYE @packet or NULL if the packet did not contain
+a reason string. The string must be freed with g_free() after usage.
+
+
+
+
+ a valid BYE #GstRTCPPacket
+
+
+
+
+
+ Get the length of the reason string.
+
+ The length of the reason string or 0 when there is no reason string
+present.
+
+
+
+
+ a valid BYE #GstRTCPPacket
+
+
+
+
+
+ Get the number of SSRC fields in @packet.
+
+ The number of SSRC fields in @packet.
+
+
+
+
+ a valid BYE #GstRTCPPacket
+
+
+
+
+
+ Set the reason string to @reason in @packet.
+
+ TRUE if the string could be set.
+
+
+
+
+ a valid BYE #GstRTCPPacket
+
+
+
+ a reason string
+
+
+
+
+
+ The profile-specific extension data is copied into a new allocated
+memory area @data. This must be freed with g_free() after usage.
+
+ %TRUE if there was valid data.
+
+
+
+
+ a valid SR or RR #GstRTCPPacket
+
+
+
+ result profile-specific data
+
+
+
+
+
+ length of the profile-specific extension data
+
+
+
+
+
+ Get the Feedback Control Information attached to a RTPFB or PSFB @packet.
+
+ a pointer to the FCI
+
+
+
+
+ a valid RTPFB or PSFB #GstRTCPPacket
+
+
+
+
+
+ Get the length of the Feedback Control Information attached to a
+RTPFB or PSFB @packet.
+
+ The length of the FCI in 32-bit words.
+
+
+
+
+ a valid RTPFB or PSFB #GstRTCPPacket
+
+
+
+
+
+ Get the media SSRC field of the RTPFB or PSFB @packet.
+
+ the media SSRC.
+
+
+
+
+ a valid RTPFB or PSFB #GstRTCPPacket
+
+
+
+
+
+ Get the sender SSRC field of the RTPFB or PSFB @packet.
+
+ the sender SSRC.
+
+
+
+
+ a valid RTPFB or PSFB #GstRTCPPacket
+
+
+
+
+
+ Get the feedback message type of the FB @packet.
+
+ The feedback message type.
+
+
+
+
+ a valid RTPFB or PSFB #GstRTCPPacket
+
+
+
+
+
+ Set the length of the Feedback Control Information attached to a
+RTPFB or PSFB @packet.
+
+ %TRUE if there was enough space in the packet to add this much FCI
+
+
+
+
+ a valid RTPFB or PSFB #GstRTCPPacket
+
+
+
+ Length of the FCI in 32-bit words
+
+
+
+
+
+ Set the media SSRC field of the RTPFB or PSFB @packet.
+
+
+
+
+
+ a valid RTPFB or PSFB #GstRTCPPacket
+
+
+
+ a media SSRC
+
+
+
+
+
+ Set the sender SSRC field of the RTPFB or PSFB @packet.
+
+
+
+
+
+ a valid RTPFB or PSFB #GstRTCPPacket
+
+
+
+ a sender SSRC
+
+
+
+
+
+ Set the feedback message type of the FB @packet.
+
+
+
+
+
+ a valid RTPFB or PSFB #GstRTCPPacket
+
+
+
+ the #GstRTCPFBType to set
+
+
+
+
+
+ Get the count field in @packet.
+
+ The count field in @packet or -1 if @packet does not point to a
+valid packet.
+
+
+
+
+ a valid #GstRTCPPacket
+
+
+
+
+
+ Get the length field of @packet. This is the length of the packet in
+32-bit words minus one.
+
+ The length field of @packet.
+
+
+
+
+ a valid #GstRTCPPacket
+
+
+
+
+
+ Get the packet padding of the packet pointed to by @packet.
+
+ If the packet has the padding bit set.
+
+
+
+
+ a valid #GstRTCPPacket
+
+
+
+
+
+
+ %TRUE if there was valid data.
+
+
+
+
+ a valid SR or RR #GstRTCPPacket
+
+
+
+ result profile-specific data
+
+
+
+
+
+ result length of the profile-specific data
+
+
+
+
+
+
+ The number of 32-bit words containing profile-specific extension
+ data from @packet.
+
+
+
+
+ a valid SR or RR #GstRTCPPacket
+
+
+
+
+
+ Parse the values of the @nth report block in @packet and store the result in
+the values.
+
+
+
+
+
+ a valid SR or RR #GstRTCPPacket
+
+
+
+ the nth report block in @packet
+
+
+
+ result for data source being reported
+
+
+
+ result for fraction lost since last SR/RR
+
+
+
+ result for the cumululative number of packets lost
+
+
+
+ result for the extended last sequence number received
+
+
+
+ result for the interarrival jitter
+
+
+
+ result for the last SR packet from this source
+
+
+
+ result for the delay since last SR packet
+
+
+
+
+
+ Get the number of report blocks in @packet.
+
+ The number of report blocks in @packet.
+
+
+
+
+ a valid SR or RR #GstRTCPPacket
+
+
+
+
+
+ Get the packet type of the packet pointed to by @packet.
+
+ The packet type or GST_RTCP_TYPE_INVALID when @packet is not
+pointing to a valid packet.
+
+
+
+
+ a valid #GstRTCPPacket
+
+
+
+
+
+ Move the packet pointer @packet to the next packet in the payload.
+Use gst_rtcp_buffer_get_first_packet() to initialize @packet.
+
+ TRUE if @packet is pointing to a valid packet after calling this
+function.
+
+
+
+
+ a #GstRTCPPacket
+
+
+
+
+
+ Removes the packet pointed to by @packet and moves pointer to the next one
+
+ TRUE if @packet is pointing to a valid packet after calling this
+function.
+
+
+
+
+ a #GstRTCPPacket
+
+
+
+
+
+ Get the ssrc field of the RR @packet.
+
+ the ssrc.
+
+
+
+
+ a valid RR #GstRTCPPacket
+
+
+
+
+
+ Set the ssrc field of the RR @packet.
+
+
+
+
+
+ a valid RR #GstRTCPPacket
+
+
+
+ the SSRC to set
+
+
+
+
+
+ Add a new SDES entry to the current item in @packet.
+
+ %TRUE if the item could be added, %FALSE if the MTU has been
+reached.
+
+
+
+
+ a valid SDES #GstRTCPPacket
+
+
+
+ the #GstRTCPSDESType of the SDES entry
+
+
+
+ the data length
+
+
+
+ the data
+
+
+
+
+
+
+
+ Add a new SDES item for @ssrc to @packet.
+
+ %TRUE if the item could be added, %FALSE if the maximum amount of
+items has been exceeded for the SDES packet or the MTU has been reached.
+
+
+
+
+ a valid SDES #GstRTCPPacket
+
+
+
+ the SSRC of the new item to add
+
+
+
+
+
+ This function is like gst_rtcp_packet_sdes_get_entry() but it returns a
+null-terminated copy of the data instead. use g_free() after usage.
+
+ %TRUE if there was valid data.
+
+
+
+
+ a valid SDES #GstRTCPPacket
+
+
+
+ result of the entry type
+
+
+
+ result length of the entry data
+
+
+
+ result entry data
+
+
+
+
+
+
+
+ Move to the first SDES entry in the current item.
+
+ %TRUE if there was a first entry.
+
+
+
+
+ a valid SDES #GstRTCPPacket
+
+
+
+
+
+ Move to the first SDES item in @packet.
+
+ TRUE if there was a first item.
+
+
+
+
+ a valid SDES #GstRTCPPacket
+
+
+
+
+
+ Get the data of the current SDES item entry. @type (when not NULL) will
+contain the type of the entry. @data (when not NULL) will point to @len
+bytes.
+
+When @type refers to a text item, @data will point to a UTF8 string. Note
+that this UTF8 string is NOT null-terminated. Use
+gst_rtcp_packet_sdes_copy_entry() to get a null-terminated copy of the entry.
+
+ %TRUE if there was valid data.
+
+
+
+
+ a valid SDES #GstRTCPPacket
+
+
+
+ result of the entry type
+
+
+
+ result length of the entry data
+
+
+
+ result entry data
+
+
+
+
+
+
+
+ Get the number of items in the SDES packet @packet.
+
+ The number of items in @packet.
+
+
+
+
+ a valid SDES #GstRTCPPacket
+
+
+
+
+
+ Get the SSRC of the current SDES item.
+
+ the SSRC of the current item.
+
+
+
+
+ a valid SDES #GstRTCPPacket
+
+
+
+
+
+ Move to the next SDES entry in the current item.
+
+ %TRUE if there was a next entry.
+
+
+
+
+ a valid SDES #GstRTCPPacket
+
+
+
+
+
+ Move to the next SDES item in @packet.
+
+ TRUE if there was a next item.
+
+
+
+
+ a valid SDES #GstRTCPPacket
+
+
+
+
+
+ Set the @nth new report block in @packet with the given values.
+
+Note: Not implemented.
+
+
+
+
+
+ a valid SR or RR #GstRTCPPacket
+
+
+
+ the nth report block to set
+
+
+
+ data source being reported
+
+
+
+ fraction lost since last SR/RR
+
+
+
+ the cumululative number of packets lost
+
+
+
+ the extended last sequence number received
+
+
+
+ the interarrival jitter
+
+
+
+ the last SR packet from this source
+
+
+
+ the delay since last SR packet
+
+
+
+
+
+ Parse the SR sender info and store the values.
+
+
+
+
+
+ a valid SR #GstRTCPPacket
+
+
+
+ result SSRC
+
+
+
+ result NTP time
+
+
+
+ result RTP time
+
+
+
+ result packet count
+
+
+
+ result octet count
+
+
+
+
+
+ Set the given values in the SR packet @packet.
+
+
+
+
+
+ a valid SR #GstRTCPPacket
+
+
+
+ the SSRC
+
+
+
+ the NTP time
+
+
+
+ the RTP time
+
+
+
+ the packet count
+
+
+
+ the octet count
+
+
+
+
+
+
+ Different types of SDES content.
+
+ Invalid SDES entry
+
+
+ End of SDES list
+
+
+ Canonical name
+
+
+ User name
+
+
+ User's electronic mail address
+
+
+ User's phone number
+
+
+ Geographic user location
+
+
+ Name of application or tool
+
+
+ Notice about the source
+
+
+ Private extensions
+
+
+
+ Different RTCP packet types.
+
+ Invalid type
+
+
+ Sender report
+
+
+ Receiver report
+
+
+ Source description
+
+
+ Goodbye
+
+
+ Application defined
+
+
+ Transport layer feedback.
+
+
+ Payload-specific feedback.
+
+
+ Extended report.
+
+
+
+ The maximum amount of SSRCs in a BYE packet.
+
+
+
+ The maximum amount of Receiver report blocks in RR and SR messages.
+
+
+
+ The maximum text length for an SDES item.
+
+
+
+ The maximum amount of SDES items.
+
+
+
+ Mask for version, padding bit and packet type pair allowing reduced size
+packets, basically it accepts other types than RR and SR
+
+
+
+ Mask for version, padding bit and packet type pair
+
+
+
+ Valid value for the first two bytes of an RTCP packet after applying
+#GST_RTCP_VALID_MASK to them.
+
+
+
+ The supported RTCP version 2.
+
+
+
+ Provides a base class for audio RTP payloaders for frame or sample based
+audio codecs (constant bitrate)
+
+This class derives from GstRTPBasePayload. It can be used for payloading
+audio codecs. It will only work with constant bitrate codecs. It supports
+both frame based and sample based codecs. It takes care of packing up the
+audio data into RTP packets and filling up the headers accordingly. The
+payloading is done based on the maximum MTU (mtu) and the maximum time per
+packet (max-ptime). The general idea is to divide large data buffers into
+smaller RTP packets. The RTP packet size is the minimum of either the MTU,
+max-ptime (if set) or available data. The RTP packet size is always larger or
+equal to min-ptime (if set). If min-ptime is not set, any residual data is
+sent in a last RTP packet. In the case of frame based codecs, the resulting
+RTP packets always contain full frames.
+
+## Usage
+
+To use this base class, your child element needs to call either
+gst_rtp_base_audio_payload_set_frame_based() or
+gst_rtp_base_audio_payload_set_sample_based(). This is usually done in the
+element's _init() function. Then, the child element must call either
+gst_rtp_base_audio_payload_set_frame_options(),
+gst_rtp_base_audio_payload_set_sample_options() or
+gst_rtp_base_audio_payload_set_samplebits_options. Since
+GstRTPBaseAudioPayload derives from GstRTPBasePayload, the child element
+must set any variables or call/override any functions required by that base
+class. The child element does not need to override any other functions
+specific to GstRTPBaseAudioPayload.
+
+ Create an RTP buffer and store @payload_len bytes of the adapter as the
+payload. Set the timestamp on the new buffer to @timestamp before pushing
+the buffer downstream.
+
+If @payload_len is -1, all pending bytes will be flushed. If @timestamp is
+-1, the timestamp will be calculated automatically.
+
+ a #GstFlowReturn
+
+
+
+
+ a #GstRTPBasePayload
+
+
+
+ length of payload
+
+
+
+ a #GstClockTime
+
+
+
+
+
+ Gets the internal adapter used by the depayloader.
+
+ a #GstAdapter.
+
+
+
+
+ a #GstRTPBaseAudioPayload
+
+
+
+
+
+ Create an RTP buffer and store @payload_len bytes of @data as the
+payload. Set the timestamp on the new buffer to @timestamp before pushing
+the buffer downstream.
+
+ a #GstFlowReturn
+
+
+
+
+ a #GstRTPBasePayload
+
+
+
+ data to set as payload
+
+
+
+
+
+ length of payload
+
+
+
+ a #GstClockTime
+
+
+
+
+
+ Tells #GstRTPBaseAudioPayload that the child element is for a frame based
+audio codec
+
+
+
+
+
+ a pointer to the element.
+
+
+
+
+
+ Sets the options for frame based audio codecs.
+
+
+
+
+
+ a pointer to the element.
+
+
+
+ The duraction of an audio frame in milliseconds.
+
+
+
+ The size of an audio frame in bytes.
+
+
+
+
+
+ Tells #GstRTPBaseAudioPayload that the child element is for a sample based
+audio codec
+
+
+
+
+
+ a pointer to the element.
+
+
+
+
+
+ Sets the options for sample based audio codecs.
+
+
+
+
+
+ a pointer to the element.
+
+
+
+ Size per sample in bytes.
+
+
+
+
+
+ Sets the options for sample based audio codecs.
+
+
+
+
+
+ a pointer to the element.
+
+
+
+ Size per sample in bits.
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Base class for audio RTP payloader.
+
+ the parent class
+
+
+
+
+
+
+
+
+
+
+
+ Provides a base class for RTP depayloaders
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Push @out_buf to the peer of @filter. This function takes ownership of
+@out_buf.
+
+This function will by default apply the last incomming timestamp on
+the outgoing buffer when it didn't have a timestamp already.
+
+ a #GstFlowReturn.
+
+
+
+
+ a #GstRTPBaseDepayload
+
+
+
+ a #GstBuffer
+
+
+
+
+
+ Push @out_list to the peer of @filter. This function takes ownership of
+@out_list.
+
+ a #GstFlowReturn.
+
+
+
+
+ a #GstRTPBaseDepayload
+
+
+
+ a #GstBufferList
+
+
+
+
+
+ Various depayloader statistics retrieved atomically (and are therefore
+synchroized with each other). This property return a GstStructure named
+application/x-rtp-depayload-stats containing the following fields relating to
+the last processed buffer and current state of the stream being depayloaded:
+
+ * `clock-rate`: #G_TYPE_UINT, clock-rate of the stream
+ * `npt-start`: #G_TYPE_UINT64, time of playback start
+ * `npt-stop`: #G_TYPE_UINT64, time of playback stop
+ * `play-speed`: #G_TYPE_DOUBLE, the playback speed
+ * `play-scale`: #G_TYPE_DOUBLE, the playback scale
+ * `running-time-dts`: #G_TYPE_UINT64, the last running-time of the
+ last DTS
+ * `running-time-pts`: #G_TYPE_UINT64, the last running-time of the
+ last PTS
+ * `seqnum`: #G_TYPE_UINT, the last seen seqnum
+ * `timestamp`: #G_TYPE_UINT, the last seen RTP timestamp
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Base class for RTP depayloaders.
+
+ the parent class
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Provides a base class for RTP payloaders
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Check if the packet with @size and @duration would exceed the configured
+maximum size.
+
+ %TRUE if the packet of @size and @duration would exceed the
+configured MTU or max_ptime.
+
+
+
+
+ a #GstRTPBasePayload
+
+
+
+ the size of the packet
+
+
+
+ the duration of the packet
+
+
+
+
+
+ Push @buffer to the peer element of the payloader. The SSRC, payload type,
+seqnum and timestamp of the RTP buffer will be updated first.
+
+This function takes ownership of @buffer.
+
+ a #GstFlowReturn.
+
+
+
+
+ a #GstRTPBasePayload
+
+
+
+ a #GstBuffer
+
+
+
+
+
+ Push @list to the peer element of the payloader. The SSRC, payload type,
+seqnum and timestamp of the RTP buffer will be updated first.
+
+This function takes ownership of @list.
+
+ a #GstFlowReturn.
+
+
+
+
+ a #GstRTPBasePayload
+
+
+
+ a #GstBufferList
+
+
+
+
+
+ Set the rtp options of the payloader. These options will be set in the caps
+of the payloader. Subclasses must call this method before calling
+gst_rtp_base_payload_push() or gst_rtp_base_payload_set_outcaps().
+
+
+
+
+
+ a #GstRTPBasePayload
+
+
+
+ the media type (typically "audio" or "video")
+
+
+
+ if the payload type is dynamic
+
+
+
+ the encoding name
+
+
+
+ the clock rate of the media
+
+
+
+
+
+ Configure the output caps with the optional parameters.
+
+Variable arguments should be in the form field name, field type
+(as a GType), value(s). The last variable argument should be NULL.
+
+ %TRUE if the caps could be set.
+
+
+
+
+ a #GstRTPBasePayload
+
+
+
+ the first field name or %NULL
+
+
+
+ field values
+
+
+
+
+
+
+
+
+ Minimum duration of the packet data in ns (can't go above MTU)
+
+
+
+
+
+
+ Try to use the offset fields to generate perfect RTP timestamps. When this
+option is disabled, RTP timestamps are generated from GST_BUFFER_PTS of
+each payloaded buffer. The PTSes of buffers may not necessarily increment
+with the amount of data in each input buffer, consider e.g. the case where
+the buffer arrives from a network which means that the PTS is unrelated to
+the amount of data. Because the RTP timestamps are generated from
+GST_BUFFER_PTS this can result in RTP timestamps that also don't increment
+with the amount of data in the payloaded packet. To circumvent this it is
+possible to set the perfect rtptime option enabled. When this option is
+enabled the payloader will increment the RTP timestamps based on
+GST_BUFFER_OFFSET which relates to the amount of data in each packet
+rather than the GST_BUFFER_PTS of each buffer and therefore the RTP
+timestamps will more closely correlate with the amount of data in each
+buffer. Currently GstRTPBasePayload is limited to handling perfect RTP
+timestamps for audio streams.
+
+
+
+
+
+
+ Force buffers to be multiples of this duration in ns (0 disables)
+
+
+
+
+
+
+
+
+
+
+
+
+ Various payloader statistics retrieved atomically (and are therefore
+synchroized with each other), these can be used e.g. to generate an
+RTP-Info header. This property return a GstStructure named
+application/x-rtp-payload-stats containing the following fields relating to
+the last processed buffer and current state of the stream being payloaded:
+
+ * `clock-rate` :#G_TYPE_UINT, clock-rate of the stream
+ * `running-time` :#G_TYPE_UINT64, running time
+ * `seqnum` :#G_TYPE_UINT, sequence number, same as #GstRTPBasePayload:seqnum
+ * `timestamp` :#G_TYPE_UINT, RTP timestamp, same as #GstRTPBasePayload:timestamp
+ * `ssrc` :#G_TYPE_UINT, The SSRC in use
+ * `pt` :#G_TYPE_UINT, The Payload type in use, same as #GstRTPBasePayload:pt
+ * `seqnum-offset` :#G_TYPE_UINT, The current offset added to the seqnum
+ * `timestamp-offset` :#G_TYPE_UINT, The current offset added to the timestamp
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Base class for audio RTP payloader.
+
+ the parent class
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ The GstRTPBuffer helper functions makes it easy to parse and create regular
+#GstBuffer objects that contain RTP payloads. These buffers are typically of
+'application/x-rtp' #GstCaps.
+
+ pointer to RTP buffer
+
+
+
+ internal state
+
+
+
+ array of data
+
+
+
+
+
+ array of size
+
+
+
+
+
+ array of #GstMapInfo
+
+
+
+
+
+ Adds a RFC 5285 header extension with a one byte header to the end of the
+RTP header. If there is already a RFC 5285 header extension with a one byte
+header, the new extension will be appended.
+It will not work if there is already a header extension that does not follow
+the mecanism described in RFC 5285 or if there is a header extension with
+a two bytes header as described in RFC 5285. In that case, use
+gst_rtp_buffer_add_extension_twobytes_header()
+
+ %TRUE if header extension could be added
+
+
+
+
+ the RTP packet
+
+
+
+ The ID of the header extension (between 1 and 14).
+
+
+
+ location for data
+
+
+
+
+
+ the size of the data in bytes
+
+
+
+
+
+ Adds a RFC 5285 header extension with a two bytes header to the end of the
+RTP header. If there is already a RFC 5285 header extension with a two bytes
+header, the new extension will be appended.
+It will not work if there is already a header extension that does not follow
+the mecanism described in RFC 5285 or if there is a header extension with
+a one byte header as described in RFC 5285. In that case, use
+gst_rtp_buffer_add_extension_onebyte_header()
+
+ %TRUE if header extension could be added
+
+
+
+
+ the RTP packet
+
+
+
+ Application specific bits
+
+
+
+ The ID of the header extension
+
+
+
+ location for data
+
+
+
+
+
+ the size of the data in bytes
+
+
+
+
+
+ Get the CSRC at index @idx in @buffer.
+
+ the CSRC at index @idx in host order.
+
+
+
+
+ the RTP packet
+
+
+
+ the index of the CSRC to get
+
+
+
+
+
+ Get the CSRC count of the RTP packet in @buffer.
+
+ the CSRC count of @buffer.
+
+
+
+
+ the RTP packet
+
+
+
+
+
+ Check if the extension bit is set on the RTP packet in @buffer.
+
+ TRUE if @buffer has the extension bit set.
+
+
+
+
+ the RTP packet
+
+
+
+
+
+ Similar to gst_rtp_buffer_get_extension_data, but more suitable for language
+bindings usage. @bits will contain the extension 16 bits of custom data and
+the extension data (not including the extension header) is placed in a new
+#GBytes structure.
+
+If @rtp did not contain an extension, this function will return %NULL, with
+@bits unchanged. If there is an extension header but no extension data then
+an empty #GBytes will be returned.
+
+ A new #GBytes if an extension header was present
+and %NULL otherwise.
+
+
+
+
+ the RTP packet
+
+
+
+ location for header bits
+
+
+
+
+
+ Get the extension data. @bits will contain the extension 16 bits of custom
+data. @data will point to the data in the extension and @wordlen will contain
+the length of @data in 32 bits words.
+
+If @buffer did not contain an extension, this function will return %FALSE
+with @bits, @data and @wordlen unchanged.
+
+ TRUE if @buffer had the extension bit set.
+
+
+
+
+ the RTP packet
+
+
+
+ location for result bits
+
+
+
+ location for data
+
+
+
+
+
+ location for length of @data in 32 bits words
+
+
+
+
+
+ Parses RFC 5285 style header extensions with a one byte header. It will
+return the nth extension with the requested id.
+
+ TRUE if @buffer had the requested header extension
+
+
+
+
+ the RTP packet
+
+
+
+ The ID of the header extension to be read (between 1 and 14).
+
+
+
+ Read the nth extension packet with the requested ID
+
+
+
+
+ location for data
+
+
+
+
+
+ the size of the data in bytes
+
+
+
+
+
+ Parses RFC 5285 style header extensions with a two bytes header. It will
+return the nth extension with the requested id.
+
+ TRUE if @buffer had the requested header extension
+
+
+
+
+ the RTP packet
+
+
+
+ Application specific bits
+
+
+
+ The ID of the header extension to be read (between 1 and 14).
+
+
+
+ Read the nth extension packet with the requested ID
+
+
+
+
+ location for data
+
+
+
+
+
+ the size of the data in bytes
+
+
+
+
+
+ Return the total length of the header in @buffer. This include the length of
+the fixed header, the CSRC list and the extension header.
+
+ The total length of the header in @buffer.
+
+
+
+
+ the RTP packet
+
+
+
+
+
+ Check if the marker bit is set on the RTP packet in @buffer.
+
+ TRUE if @buffer has the marker bit set.
+
+
+
+
+ the RTP packet
+
+
+
+
+
+ Return the total length of the packet in @buffer.
+
+ The total length of the packet in @buffer.
+
+
+
+
+ the RTP packet
+
+
+
+
+
+ Check if the padding bit is set on the RTP packet in @buffer.
+
+ TRUE if @buffer has the padding bit set.
+
+
+
+
+ the RTP packet
+
+
+
+
+
+ Get a pointer to the payload data in @buffer. This pointer is valid as long
+as a reference to @buffer is held.
+
+ A pointer
+to the payload data in @buffer.
+
+
+
+
+
+
+ the RTP packet
+
+
+
+
+
+ Create a buffer of the payload of the RTP packet in @buffer. This function
+will internally create a subbuffer of @buffer so that a memcpy can be
+avoided.
+
+ A new buffer with the data of the payload.
+
+
+
+
+ the RTP packet
+
+
+
+
+
+ Similar to gst_rtp_buffer_get_payload, but more suitable for language
+bindings usage. The return value is a pointer to a #GBytes structure
+containing the payload data in @rtp.
+
+ A new #GBytes containing the payload data in @rtp.
+
+
+
+
+ the RTP packet
+
+
+
+
+
+ Get the length of the payload of the RTP packet in @buffer.
+
+ The length of the payload in @buffer.
+
+
+
+
+ the RTP packet
+
+
+
+
+
+ Create a subbuffer of the payload of the RTP packet in @buffer. @offset bytes
+are skipped in the payload and the subbuffer will be of size @len.
+If @len is -1 the total payload starting from @offset is subbuffered.
+
+ A new buffer with the specified data of the payload.
+
+
+
+
+ the RTP packet
+
+
+
+ the offset in the payload
+
+
+
+ the length in the payload
+
+
+
+
+
+ Get the payload type of the RTP packet in @buffer.
+
+ The payload type.
+
+
+
+
+ the RTP packet
+
+
+
+
+
+ Get the sequence number of the RTP packet in @buffer.
+
+ The sequence number in host order.
+
+
+
+
+ the RTP packet
+
+
+
+
+
+ Get the SSRC of the RTP packet in @buffer.
+
+ the SSRC of @buffer in host order.
+
+
+
+
+ the RTP packet
+
+
+
+
+
+ Get the timestamp of the RTP packet in @buffer.
+
+ The timestamp in host order.
+
+
+
+
+ the RTP packet
+
+
+
+
+
+ Get the version number of the RTP packet in @buffer.
+
+ The version of @buffer.
+
+
+
+
+ the RTP packet
+
+
+
+
+
+ Set the amount of padding in the RTP packet in @buffer to
+@len. If @len is 0, the padding is removed.
+
+NOTE: This function does not work correctly.
+
+
+
+
+
+ the RTP packet
+
+
+
+ the new amount of padding
+
+
+
+
+
+ Modify the CSRC at index @idx in @buffer to @csrc.
+
+
+
+
+
+ the RTP packet
+
+
+
+ the CSRC index to set
+
+
+
+ the CSRC in host order to set at @idx
+
+
+
+
+
+ Set the extension bit on the RTP packet in @buffer to @extension.
+
+
+
+
+
+ the RTP packet
+
+
+
+ the new extension
+
+
+
+
+
+ Set the extension bit of the rtp buffer and fill in the @bits and @length of the
+extension header. If the existing extension data is not large enough, it will
+be made larger.
+
+ True if done.
+
+
+
+
+ the RTP packet
+
+
+
+ the bits specific for the extension
+
+
+
+ the length that counts the number of 32-bit words in
+the extension, excluding the extension header ( therefore zero is a valid length)
+
+
+
+
+
+ Set the marker bit on the RTP packet in @buffer to @marker.
+
+
+
+
+
+ the RTP packet
+
+
+
+ the new marker
+
+
+
+
+
+ Set the total @rtp size to @len. The data in the buffer will be made
+larger if needed. Any padding will be removed from the packet.
+
+
+
+
+
+ the RTP packet
+
+
+
+ the new packet length
+
+
+
+
+
+ Set the padding bit on the RTP packet in @buffer to @padding.
+
+
+
+
+
+ the buffer
+
+
+
+ the new padding
+
+
+
+
+
+ Set the payload type of the RTP packet in @buffer to @payload_type.
+
+
+
+
+
+ the RTP packet
+
+
+
+ the new type
+
+
+
+
+
+ Set the sequence number of the RTP packet in @buffer to @seq.
+
+
+
+
+
+ the RTP packet
+
+
+
+ the new sequence number
+
+
+
+
+
+ Set the SSRC on the RTP packet in @buffer to @ssrc.
+
+
+
+
+
+ the RTP packet
+
+
+
+ the new SSRC
+
+
+
+
+
+ Set the timestamp of the RTP packet in @buffer to @timestamp.
+
+
+
+
+
+ the RTP packet
+
+
+
+ the new timestamp
+
+
+
+
+
+ Set the version of the RTP packet in @buffer to @version.
+
+
+
+
+
+ the RTP packet
+
+
+
+ the new version
+
+
+
+
+
+ Unmap @rtp previously mapped with gst_rtp_buffer_map().
+
+
+
+
+
+ a #GstRTPBuffer
+
+
+
+
+
+ Allocate enough data in @buffer to hold an RTP packet with @csrc_count CSRCs,
+a payload length of @payload_len and padding of @pad_len.
+@buffer must be writable and all previous memory in @buffer will be freed.
+If @pad_len is >0, the padding bit will be set. All other RTP header fields
+will be set to 0/FALSE.
+
+
+
+
+
+ a #GstBuffer
+
+
+
+ the length of the payload
+
+
+
+ the amount of padding
+
+
+
+ the number of CSRC entries
+
+
+
+
+
+ Calculate the header length of an RTP packet with @csrc_count CSRC entries.
+An RTP packet can have at most 15 CSRC entries.
+
+ The length of an RTP header with @csrc_count CSRC entries.
+
+
+
+
+ the number of CSRC entries
+
+
+
+
+
+ Calculate the total length of an RTP packet with a payload size of @payload_len,
+a padding of @pad_len and a @csrc_count CSRC entries.
+
+ The total length of an RTP header with given parameters.
+
+
+
+
+ the length of the payload
+
+
+
+ the amount of padding
+
+
+
+ the number of CSRC entries
+
+
+
+
+
+ Calculate the length of the payload of an RTP packet with size @packet_len,
+a padding of @pad_len and a @csrc_count CSRC entries.
+
+ The length of the payload of an RTP packet with given parameters.
+
+
+
+
+ the length of the total RTP packet
+
+
+
+ the amount of padding
+
+
+
+ the number of CSRC entries
+
+
+
+
+
+ Compare two sequence numbers, taking care of wraparounds. This function
+returns the difference between @seqnum1 and @seqnum2.
+
+ a negative value if @seqnum1 is bigger than @seqnum2, 0 if they
+are equal or a positive value if @seqnum1 is smaller than @segnum2.
+
+
+
+
+ a sequence number
+
+
+
+ a sequence number
+
+
+
+
+
+ Get the default clock-rate for the static payload type @payload_type.
+
+ the default clock rate or -1 if the payload type is not static or
+the clock-rate is undefined.
+
+
+
+
+ the static payload type
+
+
+
+
+
+ Update the @exttimestamp field with the extended timestamp of @timestamp
+For the first call of the method, @exttimestamp should point to a location
+with a value of -1.
+
+This function is able to handle both forward and backward timestamps taking
+into account:
+ - timestamp wraparound making sure that the returned value is properly increased.
+ - timestamp unwraparound making sure that the returned value is properly decreased.
+
+ The extended timestamp of @timestamp or 0 if the result can't go anywhere backwards.
+
+
+
+
+ a previous extended timestamp
+
+
+
+ a new timestamp
+
+
+
+
+
+ Map the contents of @buffer into @rtp.
+
+ %TRUE if @buffer could be mapped.
+
+
+
+
+ a #GstBuffer
+
+
+
+ #GstMapFlags
+
+
+
+ a #GstRTPBuffer
+
+
+
+
+
+ Allocate a new #GstBuffer with enough data to hold an RTP packet with
+@csrc_count CSRCs, a payload length of @payload_len and padding of @pad_len.
+All other RTP header fields will be set to 0/FALSE.
+
+ A newly allocated buffer that can hold an RTP packet with given
+parameters.
+
+
+
+
+ the length of the payload
+
+
+
+ the amount of padding
+
+
+
+ the number of CSRC entries
+
+
+
+
+
+ Create a new #GstBuffer that can hold an RTP packet that is exactly
+@packet_len long. The length of the payload depends on @pad_len and
+@csrc_count and can be calculated with gst_rtp_buffer_calc_payload_len().
+All RTP header fields will be set to 0/FALSE.
+
+ A newly allocated buffer that can hold an RTP packet of @packet_len.
+
+
+
+
+ the total length of the packet
+
+
+
+ the amount of padding
+
+
+
+ the number of CSRC entries
+
+
+
+
+
+ Create a new buffer and set the data to a copy of @len
+bytes of @data and the size to @len. The data will be freed when the buffer
+is freed.
+
+ A newly allocated buffer with a copy of @data and of size @len.
+
+
+
+
+ data for the new
+ buffer
+
+
+
+
+
+ the length of data
+
+
+
+
+
+ Create a new buffer and set the data and size of the buffer to @data and @len
+respectively. @data will be freed when the buffer is unreffed, so this
+function transfers ownership of @data to the new buffer.
+
+ A newly allocated buffer with @data and of size @len.
+
+
+
+
+
+ data for the new buffer
+
+
+
+
+
+ the length of data
+
+
+
+
+
+
+ Additional RTP buffer flags. These flags can potentially be used on any
+buffers carrying RTP packets.
+
+Note that these are only valid for #GstCaps of type: application/x-rtp (x-rtcp).
+They can conflict with other extended buffer flags.
+
+ The #GstBuffer was once wrapped
+ in a retransmitted packet as specified by RFC 4588.
+
+
+ The packet represents redundant RTP packet.
+ The flag is used in gstrtpstorage to be able to hold the packetback
+ and use it only for recovery from packet loss.
+ Since: 1.14
+
+
+ Offset to define more flags.
+
+
+
+ Additional mapping flags for gst_rtp_buffer_map().
+
+ Skip mapping and validation of RTP
+ padding and RTP pad count when present. Useful for buffers where
+ the padding may be encrypted.
+
+
+ Offset to define more flags
+
+
+
+ Standard predefined fixed payload types.
+
+The official list is at:
+http://www.iana.org/assignments/rtp-parameters
+
+Audio:
+reserved: 19
+unassigned: 20-23,
+
+Video:
+unassigned: 24, 27, 29, 30, 35-71, 77-95
+Reserved for RTCP conflict avoidance: 72-76
+
+ ITU-T G.711. mu-law audio (RFC 3551)
+
+
+ RFC 3551 says reserved
+
+
+ RFC 3551 says reserved
+
+
+ GSM audio
+
+
+ ITU G.723.1 audio
+
+
+ IMA ADPCM wave type (RFC 3551)
+
+
+ IMA ADPCM wave type (RFC 3551)
+
+
+ experimental linear predictive encoding
+
+
+ ITU-T G.711 A-law audio (RFC 3551)
+
+
+ ITU-T G.722 (RFC 3551)
+
+
+ stereo PCM
+
+
+ mono PCM
+
+
+ EIA & TIA standard IS-733
+
+
+ Comfort Noise (RFC 3389)
+
+
+ Audio MPEG 1-3.
+
+
+ ITU-T G.728 Speech coder (RFC 3551)
+
+
+ IMA ADPCM wave type (RFC 3551)
+
+
+ IMA ADPCM wave type (RFC 3551)
+
+
+ ITU-T G.729 Speech coder (RFC 3551)
+
+
+ See RFC 2029
+
+
+ ISO Standards 10918-1 and 10918-2 (RFC 2435)
+
+
+ nv encoding by Ron Frederick
+
+
+ ITU-T Recommendation H.261 (RFC 2032)
+
+
+ Video MPEG 1 & 2 (RFC 2250)
+
+
+ MPEG-2 transport stream (RFC 2250)
+
+
+ Video H263 (RFC 2190)
+
+
+
+ Structure holding default payload type information.
+
+ payload type, -1 means dynamic
+
+
+
+ the media type(s), usually "audio", "video", "application", "text",
+"message".
+
+
+
+ the encoding name of @pt
+
+
+
+ default clock rate, 0 = unknown/variable
+
+
+
+ encoding parameters. For audio this is the number of
+channels. NULL = not applicable.
+
+
+
+ the bitrate of the media. 0 = unknown/variable.
+
+
+
+
+
+
+
+
+ Get the #GstRTPPayloadInfo for @media and @encoding_name. This function is
+mostly used to get the default clock-rate and bandwidth for dynamic payload
+types specified with @media and @encoding name.
+
+The search for @encoding_name will be performed in a case insensitve way.
+
+ a #GstRTPPayloadInfo or NULL when no info could be found.
+
+
+
+
+ the media to find
+
+
+
+ the encoding name to find
+
+
+
+
+
+ Get the #GstRTPPayloadInfo for @payload_type. This function is
+mostly used to get the default clock-rate and bandwidth for static payload
+types specified with @payload_type.
+
+ a #GstRTPPayloadInfo or NULL when no info could be found.
+
+
+
+
+ the payload_type to find
+
+
+
+
+
+
+ The transfer profile to use.
+
+ invalid profile
+
+
+ the Audio/Visual profile (RFC 3551)
+
+
+ the secure Audio/Visual profile (RFC 3711)
+
+
+ the Audio/Visual profile with feedback (RFC 4585)
+
+
+ the secure Audio/Visual profile with feedback (RFC 5124)
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ The supported RTP version 2.
+
+
+
+ Open @buffer for reading or writing, depending on @flags. The resulting RTCP
+buffer state is stored in @rtcp.
+
+
+
+
+
+ a buffer with an RTCP packet
+
+
+
+ flags for the mapping
+
+
+
+ resulting #GstRTCPBuffer
+
+
+
+
+
+ Create a new buffer for constructing RTCP packets. The packet will have a
+maximum size of @mtu.
+
+ A newly allocated buffer.
+
+
+
+
+ the maximum mtu size.
+
+
+
+
+
+ Create a new buffer and set the data to a copy of @len
+bytes of @data and the size to @len. The data will be freed when the buffer
+is freed.
+
+ A newly allocated buffer with a copy of @data and of size @len.
+
+
+
+
+ data for the new buffer
+
+
+
+
+
+ the length of data
+
+
+
+
+
+ Create a new buffer and set the data and size of the buffer to @data and @len
+respectively. @data will be freed when the buffer is unreffed, so this
+function transfers ownership of @data to the new buffer.
+
+ A newly allocated buffer with @data and of size @len.
+
+
+
+
+ data for the new buffer
+
+
+
+
+
+ the length of data
+
+
+
+
+
+ Check if the data pointed to by @buffer is a valid RTCP packet using
+gst_rtcp_buffer_validate_data().
+
+ TRUE if @buffer is a valid RTCP packet.
+
+
+
+
+ the buffer to validate
+
+
+
+
+
+ Check if the @data and @size point to the data of a valid compound,
+non-reduced size RTCP packet.
+Use this function to validate a packet before using the other functions in
+this module.
+
+ TRUE if the data points to a valid RTCP packet.
+
+
+
+
+ the data to validate
+
+
+
+
+
+ the length of @data to validate
+
+
+
+
+
+ Check if the @data and @size point to the data of a valid RTCP packet.
+Use this function to validate a packet before using the other functions in
+this module.
+
+This function is updated to support reduced size rtcp packets according to
+RFC 5506 and will validate full compound RTCP packets as well as reduced
+size RTCP packets.
+
+ TRUE if the data points to a valid RTCP packet.
+
+
+
+
+ the data to validate
+
+
+
+
+
+ the length of @data to validate
+
+
+
+
+
+ Check if the data pointed to by @buffer is a valid RTCP packet using
+gst_rtcp_buffer_validate_reduced().
+
+ TRUE if @buffer is a valid RTCP packet.
+
+
+
+
+ the buffer to validate
+
+
+
+
+
+ Converts an NTP time to UNIX nanoseconds. @ntptime can typically be
+the NTP time of an SR RTCP message and contains, in the upper 32 bits, the
+number of seconds since 1900 and, in the lower 32 bits, the fractional
+seconds. The resulting value will be the number of nanoseconds since 1970.
+
+ the UNIX time for @ntptime in nanoseconds.
+
+
+
+
+ an NTP timestamp
+
+
+
+
+
+ Convert @name into a @GstRTCPSDESType. @name is typically a key in a
+#GstStructure containing SDES items.
+
+ the #GstRTCPSDESType for @name or #GST_RTCP_SDES_PRIV when @name
+is a private sdes item.
+
+
+
+
+ a SDES name
+
+
+
+
+
+ Converts @type to the string equivalent. The string is typically used as a
+key in a #GstStructure containing SDES items.
+
+ the string equivalent of @type
+
+
+
+
+ a #GstRTCPSDESType
+
+
+
+
+
+ Converts a UNIX timestamp in nanoseconds to an NTP time. The caller should
+pass a value with nanoseconds since 1970. The NTP time will, in the upper
+32 bits, contain the number of seconds since 1900 and, in the lower 32
+bits, the fractional seconds. The resulting value can be used as an ntptime
+for constructing SR RTCP packets.
+
+ the NTP time for @unixtime.
+
+
+
+
+ an UNIX timestamp in nanoseconds
+
+
+
+
+
+ Allocate enough data in @buffer to hold an RTP packet with @csrc_count CSRCs,
+a payload length of @payload_len and padding of @pad_len.
+@buffer must be writable and all previous memory in @buffer will be freed.
+If @pad_len is >0, the padding bit will be set. All other RTP header fields
+will be set to 0/FALSE.
+
+
+
+
+
+ a #GstBuffer
+
+
+
+ the length of the payload
+
+
+
+ the amount of padding
+
+
+
+ the number of CSRC entries
+
+
+
+
+
+ Calculate the header length of an RTP packet with @csrc_count CSRC entries.
+An RTP packet can have at most 15 CSRC entries.
+
+ The length of an RTP header with @csrc_count CSRC entries.
+
+
+
+
+ the number of CSRC entries
+
+
+
+
+
+ Calculate the total length of an RTP packet with a payload size of @payload_len,
+a padding of @pad_len and a @csrc_count CSRC entries.
+
+ The total length of an RTP header with given parameters.
+
+
+
+
+ the length of the payload
+
+
+
+ the amount of padding
+
+
+
+ the number of CSRC entries
+
+
+
+
+
+ Calculate the length of the payload of an RTP packet with size @packet_len,
+a padding of @pad_len and a @csrc_count CSRC entries.
+
+ The length of the payload of an RTP packet with given parameters.
+
+
+
+
+ the length of the total RTP packet
+
+
+
+ the amount of padding
+
+
+
+ the number of CSRC entries
+
+
+
+
+
+ Compare two sequence numbers, taking care of wraparounds. This function
+returns the difference between @seqnum1 and @seqnum2.
+
+ a negative value if @seqnum1 is bigger than @seqnum2, 0 if they
+are equal or a positive value if @seqnum1 is smaller than @segnum2.
+
+
+
+
+ a sequence number
+
+
+
+ a sequence number
+
+
+
+
+
+ Get the default clock-rate for the static payload type @payload_type.
+
+ the default clock rate or -1 if the payload type is not static or
+the clock-rate is undefined.
+
+
+
+
+ the static payload type
+
+
+
+
+
+ Update the @exttimestamp field with the extended timestamp of @timestamp
+For the first call of the method, @exttimestamp should point to a location
+with a value of -1.
+
+This function is able to handle both forward and backward timestamps taking
+into account:
+ - timestamp wraparound making sure that the returned value is properly increased.
+ - timestamp unwraparound making sure that the returned value is properly decreased.
+
+ The extended timestamp of @timestamp or 0 if the result can't go anywhere backwards.
+
+
+
+
+ a previous extended timestamp
+
+
+
+ a new timestamp
+
+
+
+
+
+ Map the contents of @buffer into @rtp.
+
+ %TRUE if @buffer could be mapped.
+
+
+
+
+ a #GstBuffer
+
+
+
+ #GstMapFlags
+
+
+
+ a #GstRTPBuffer
+
+
+
+
+
+ Allocate a new #GstBuffer with enough data to hold an RTP packet with
+@csrc_count CSRCs, a payload length of @payload_len and padding of @pad_len.
+All other RTP header fields will be set to 0/FALSE.
+
+ A newly allocated buffer that can hold an RTP packet with given
+parameters.
+
+
+
+
+ the length of the payload
+
+
+
+ the amount of padding
+
+
+
+ the number of CSRC entries
+
+
+
+
+
+ Create a new #GstBuffer that can hold an RTP packet that is exactly
+@packet_len long. The length of the payload depends on @pad_len and
+@csrc_count and can be calculated with gst_rtp_buffer_calc_payload_len().
+All RTP header fields will be set to 0/FALSE.
+
+ A newly allocated buffer that can hold an RTP packet of @packet_len.
+
+
+
+
+ the total length of the packet
+
+
+
+ the amount of padding
+
+
+
+ the number of CSRC entries
+
+
+
+
+
+ Create a new buffer and set the data to a copy of @len
+bytes of @data and the size to @len. The data will be freed when the buffer
+is freed.
+
+ A newly allocated buffer with a copy of @data and of size @len.
+
+
+
+
+ data for the new
+ buffer
+
+
+
+
+
+ the length of data
+
+
+
+
+
+ Create a new buffer and set the data and size of the buffer to @data and @len
+respectively. @data will be freed when the buffer is unreffed, so this
+function transfers ownership of @data to the new buffer.
+
+ A newly allocated buffer with @data and of size @len.
+
+
+
+
+
+ data for the new buffer
+
+
+
+
+
+ the length of data
+
+
+
+
+
+ Reads the NTP time from the @size NTP-56 extension bytes in @data and store the
+result in @ntptime.
+
+ %TRUE on success.
+
+
+
+
+ the data to read from
+
+
+
+
+
+ the size of @data
+
+
+
+ the result NTP time
+
+
+
+
+
+ Reads the NTP time from the @size NTP-64 extension bytes in @data and store the
+result in @ntptime.
+
+ %TRUE on success.
+
+
+
+
+ the data to read from
+
+
+
+
+
+ the size of @data
+
+
+
+ the result NTP time
+
+
+
+
+
+ Writes the NTP time in @ntptime to the format required for the NTP-56 header
+extension. @data must hold at least #GST_RTP_HDREXT_NTP_56_SIZE bytes.
+
+ %TRUE on success.
+
+
+
+
+ the data to write to
+
+
+
+ the size of @data
+
+
+
+ the NTP time
+
+
+
+
+
+ Writes the NTP time in @ntptime to the format required for the NTP-64 header
+extension. @data must hold at least #GST_RTP_HDREXT_NTP_64_SIZE bytes.
+
+ %TRUE on success.
+
+
+
+
+ the data to write to
+
+
+
+ the size of @data
+
+
+
+ the NTP time
+
+
+
+
+
+ Get the #GstRTPPayloadInfo for @media and @encoding_name. This function is
+mostly used to get the default clock-rate and bandwidth for dynamic payload
+types specified with @media and @encoding name.
+
+The search for @encoding_name will be performed in a case insensitve way.
+
+ a #GstRTPPayloadInfo or NULL when no info could be found.
+
+
+
+
+ the media to find
+
+
+
+ the encoding name to find
+
+
+
+
+
+ Get the #GstRTPPayloadInfo for @payload_type. This function is
+mostly used to get the default clock-rate and bandwidth for static payload
+types specified with @payload_type.
+
+ a #GstRTPPayloadInfo or NULL when no info could be found.
+
+
+
+
+ the payload_type to find
+
+
+
+
+
+
diff --git a/girs/GstRtsp-1.0.gir b/girs/GstRtsp-1.0.gir
new file mode 100644
index 0000000000..19dcbf6a04
--- /dev/null
+++ b/girs/GstRtsp-1.0.gir
@@ -0,0 +1,5058 @@
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Authentication methods, ordered by strength
+
+ no authentication
+
+
+ basic authentication
+
+
+ digest authentication
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ This object manages the RTSP connection to the server. It provides function
+to receive and send bytes and messages.
+
+ Clear the list of authentication directives stored in @conn.
+
+
+
+
+
+ a #GstRTSPConnection
+
+
+
+
+
+ Close the connected @conn. After this call, the connection is in the same
+state as when it was first created.
+
+ #GST_RTSP_OK on success.
+
+
+
+
+ a #GstRTSPConnection
+
+
+
+
+
+ Attempt to connect to the url of @conn made with
+gst_rtsp_connection_create(). If @timeout is %NULL this function can block
+forever. If @timeout contains a valid timeout, this function will return
+#GST_RTSP_ETIMEOUT after the timeout expired.
+
+This function can be cancelled with gst_rtsp_connection_flush().
+
+ #GST_RTSP_OK when a connection could be made.
+
+
+
+
+ a #GstRTSPConnection
+
+
+
+ a #GTimeVal timeout
+
+
+
+
+
+ Attempt to connect to the url of @conn made with
+gst_rtsp_connection_create(). If @timeout is %NULL this function can block
+forever. If @timeout contains a valid timeout, this function will return
+#GST_RTSP_ETIMEOUT after the timeout expired. If @conn is set to tunneled,
+@response will contain a response to the tunneling request messages.
+
+This function can be cancelled with gst_rtsp_connection_flush().
+
+ #GST_RTSP_OK when a connection could be made.
+
+Since 1.8
+
+
+
+
+ a #GstRTSPConnection
+
+
+
+ a #GTimeVal timeout
+
+
+
+ a #GstRTSPMessage
+
+
+
+
+
+ If @conn received the first tunnel connection and @conn2 received
+the second tunnel connection, link the two connections together so that
+@conn manages the tunneled connection.
+
+After this call, @conn2 cannot be used anymore and must be freed with
+gst_rtsp_connection_free().
+
+If @conn2 is %NULL then only the base64 decoding context will be setup for
+@conn.
+
+ return GST_RTSP_OK on success.
+
+
+
+
+ a #GstRTSPConnection
+
+
+
+ a #GstRTSPConnection or %NULL
+
+
+
+
+
+ Start or stop the flushing action on @conn. When flushing, all current
+and future actions on @conn will return #GST_RTSP_EINTR until the connection
+is set to non-flushing mode again.
+
+ #GST_RTSP_OK.
+
+
+
+
+ a #GstRTSPConnection
+
+
+
+ start or stop the flush
+
+
+
+
+
+ Close and free @conn.
+
+ #GST_RTSP_OK on success.
+
+
+
+
+ a #GstRTSPConnection
+
+
+
+
+
+ Retrieve the IP address of the other end of @conn.
+
+ The IP address as a string. this value remains valid until the
+connection is closed.
+
+
+
+
+ a #GstRTSPConnection
+
+
+
+
+
+ Get the file descriptor for reading.
+
+ the file descriptor used for reading or %NULL on
+error. The file descriptor remains valid until the connection is closed.
+
+
+
+
+ a #GstRTSPConnection
+
+
+
+
+
+
+ %TRUE if the #GstRTSPConnection remembers the session id in the
+last response to set it on any further request.
+
+
+
+
+ a #GstRTSPConnection
+
+
+
+
+
+ Get the TLS connection of @conn.
+
+For client side this will return the #GTlsClientConnection when connected
+over TLS.
+
+For server side connections, this function will create a GTlsServerConnection
+when called the first time and will return that same connection on subsequent
+calls. The server is then responsible for configuring the TLS connection.
+
+ the TLS connection for @conn.
+
+
+
+
+ a #GstRTSPConnection
+
+
+
+
+
+ Gets the anchor certificate authorities database that will be used
+after a server certificate can't be verified with the default
+certificate database.
+
+ the anchor certificate authorities database, or NULL if no
+database has been previously set. Use g_object_unref() to release the
+certificate database.
+
+
+
+
+ a #GstRTSPConnection
+
+
+
+
+
+ Gets a #GTlsInteraction object to be used when the connection or certificate
+database need to interact with the user. This will be used to prompt the
+user for passwords where necessary.
+
+ a reference on the #GTlsInteraction. Use
+g_object_unref() to release.
+
+
+
+
+ a #GstRTSPConnection
+
+
+
+
+
+ Gets the TLS validation flags used to verify the peer certificate
+when a TLS connection is established.
+
+ the validationg flags.
+
+
+
+
+ a #GstRTSPConnection
+
+
+
+
+
+ Get the tunnel session id the connection.
+
+ returns a non-empty string if @conn is being tunneled over HTTP.
+
+
+
+
+ a #GstRTSPConnection
+
+
+
+
+
+ Retrieve the URL of the other end of @conn.
+
+ The URL. This value remains valid until the
+connection is freed.
+
+
+
+
+ a #GstRTSPConnection
+
+
+
+
+
+ Get the file descriptor for writing.
+
+ the file descriptor used for writing or NULL on
+error. The file descriptor remains valid until the connection is closed.
+
+
+
+
+ a #GstRTSPConnection
+
+
+
+
+
+ Get the tunneling state of the connection.
+
+ if @conn is using HTTP tunneling.
+
+
+
+
+ a #GstRTSPConnection
+
+
+
+
+
+ Calculate the next timeout for @conn, storing the result in @timeout.
+
+ #GST_RTSP_OK.
+
+
+
+
+ a #GstRTSPConnection
+
+
+
+ a timeout
+
+
+
+
+
+ Wait up to the specified @timeout for the connection to become available for
+at least one of the operations specified in @events. When the function returns
+with #GST_RTSP_OK, @revents will contain a bitmask of available operations on
+@conn.
+
+@timeout can be %NULL, in which case this function might block forever.
+
+This function can be cancelled with gst_rtsp_connection_flush().
+
+ #GST_RTSP_OK on success.
+
+
+
+
+ a #GstRTSPConnection
+
+
+
+ a bitmask of #GstRTSPEvent flags to check
+
+
+
+ location for result flags
+
+
+
+ a timeout
+
+
+
+
+
+ Attempt to read @size bytes into @data from the connected @conn, blocking up to
+the specified @timeout. @timeout can be %NULL, in which case this function
+might block forever.
+
+This function can be cancelled with gst_rtsp_connection_flush().
+
+ #GST_RTSP_OK on success.
+
+
+
+
+ a #GstRTSPConnection
+
+
+
+ the data to read
+
+
+
+ the size of @data
+
+
+
+ a timeout value or %NULL
+
+
+
+
+
+ Attempt to read into @message from the connected @conn, blocking up to
+the specified @timeout. @timeout can be %NULL, in which case this function
+might block forever.
+
+This function can be cancelled with gst_rtsp_connection_flush().
+
+ #GST_RTSP_OK on success.
+
+
+
+
+ a #GstRTSPConnection
+
+
+
+ the message to read
+
+
+
+ a timeout value or %NULL
+
+
+
+
+
+ Reset the timeout of @conn.
+
+ #GST_RTSP_OK.
+
+
+
+
+ a #GstRTSPConnection
+
+
+
+
+
+ Attempt to send @message to the connected @conn, blocking up to
+the specified @timeout. @timeout can be %NULL, in which case this function
+might block forever.
+
+This function can be cancelled with gst_rtsp_connection_flush().
+
+ #GST_RTSP_OK on success.
+
+
+
+
+ a #GstRTSPConnection
+
+
+
+ the message to send
+
+
+
+ a timeout value or %NULL
+
+
+
+
+
+ Sets a custom accept-certificate function for checking certificates for
+validity. This will directly map to #GTlsConnection 's "accept-certificate"
+signal and be performed after the default checks of #GstRTSPConnection
+(checking against the #GTlsDatabase with the given #GTlsCertificateFlags)
+have failed. If no #GTlsDatabase is set on this connection, only @func will
+be called.
+
+
+
+
+
+ a #GstRTSPConnection
+
+
+
+ a #GstRTSPConnectionAcceptCertificateFunc to check certificates
+
+
+
+ User data passed to @func
+
+
+
+ #GDestroyNotify for @user_data
+
+
+
+
+
+ Configure @conn for authentication mode @method with @user and @pass as the
+user and password respectively.
+
+ #GST_RTSP_OK.
+
+
+
+
+ a #GstRTSPConnection
+
+
+
+ authentication method
+
+
+
+ the user
+
+
+
+ the password
+
+
+
+
+
+ Setup @conn with authentication directives. This is not necesary for
+methods #GST_RTSP_AUTH_NONE and #GST_RTSP_AUTH_BASIC. For
+#GST_RTSP_AUTH_DIGEST, directives should be taken from the digest challenge
+in the WWW-Authenticate response header and can include realm, domain,
+nonce, opaque, stale, algorithm, qop as per RFC2617.
+
+
+
+
+
+ a #GstRTSPConnection
+
+
+
+ authentication directive
+
+
+
+ value
+
+
+
+
+
+ By setting the HTTP mode to %TRUE the message parsing will support HTTP
+messages in addition to the RTSP messages. It will also disable the
+automatic handling of setting up an HTTP tunnel.
+
+
+
+
+
+ a #GstRTSPConnection
+
+
+
+ %TRUE to enable manual HTTP mode
+
+
+
+
+
+ Set the IP address of the server.
+
+
+
+
+
+ a #GstRTSPConnection
+
+
+
+ an ip address
+
+
+
+
+
+ Set the proxy host and port.
+
+ #GST_RTSP_OK.
+
+
+
+
+ a #GstRTSPConnection
+
+
+
+ the proxy host
+
+
+
+ the proxy port
+
+
+
+
+
+ Configure @conn to use the specified DSCP value.
+
+ #GST_RTSP_OK on success.
+
+
+
+
+ a #GstRTSPConnection
+
+
+
+ DSCP value
+
+
+
+
+
+ Sets if the #GstRTSPConnection should remember the session id from the last
+response received and force it onto any further requests.
+
+The default value is %TRUE
+
+
+
+
+
+ a #GstRTSPConnection
+
+
+
+ %TRUE if the connection should remember the session id
+
+
+
+
+
+ Sets the anchor certificate authorities database. This certificate
+database will be used to verify the server's certificate in case it
+can't be verified with the default certificate database first.
+
+
+
+
+
+ a #GstRTSPConnection
+
+
+
+ a #GTlsDatabase
+
+
+
+
+
+ Sets a #GTlsInteraction object to be used when the connection or certificate
+database need to interact with the user. This will be used to prompt the
+user for passwords where necessary.
+
+
+
+
+
+ a #GstRTSPConnection
+
+
+
+ a #GTlsInteraction
+
+
+
+
+
+ Sets the TLS validation flags to be used to verify the peer
+certificate when a TLS connection is established.
+
+ TRUE if the validation flags are set correctly, or FALSE if
+@conn is NULL or is not a TLS connection.
+
+
+
+
+ a #GstRTSPConnection
+
+
+
+ the validation flags.
+
+
+
+
+
+ Set the HTTP tunneling state of the connection. This must be configured before
+the @conn is connected.
+
+
+
+
+
+ a #GstRTSPConnection
+
+
+
+ the new state
+
+
+
+
+
+ Attempt to write @size bytes of @data to the connected @conn, blocking up to
+the specified @timeout. @timeout can be %NULL, in which case this function
+might block forever.
+
+This function can be cancelled with gst_rtsp_connection_flush().
+
+ #GST_RTSP_OK on success.
+
+
+
+
+ a #GstRTSPConnection
+
+
+
+ the data to write
+
+
+
+ the size of @data
+
+
+
+ a timeout value or %NULL
+
+
+
+
+
+ Accept a new connection on @socket and create a new #GstRTSPConnection for
+handling communication on new socket.
+
+ #GST_RTSP_OK when @conn contains a valid connection.
+
+
+
+
+ a socket
+
+
+
+ storage for a #GstRTSPConnection
+
+
+
+ a #GCancellable to cancel the operation
+
+
+
+
+
+ Create a newly allocated #GstRTSPConnection from @url and store it in @conn.
+The connection will not yet attempt to connect to @url, use
+gst_rtsp_connection_connect().
+
+A copy of @url will be made.
+
+ #GST_RTSP_OK when @conn contains a valid connection.
+
+
+
+
+ a #GstRTSPUrl
+
+
+
+ storage for a #GstRTSPConnection
+
+
+
+
+
+ Create a new #GstRTSPConnection for handling communication on the existing
+socket @socket. The @initial_buffer contains zero terminated data already
+read from @socket which should be used before starting to read new data.
+
+ #GST_RTSP_OK when @conn contains a valid connection.
+
+
+
+
+ a #GSocket
+
+
+
+ the IP address of the other end
+
+
+
+ the port used by the other end
+
+
+
+ data already read from @fd
+
+
+
+ storage for a #GstRTSPConnection
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ The possible events for the connection.
+
+ connection is readable
+
+
+ connection is writable
+
+
+
+ This interface is implemented e.g. by the Windows Media Streaming RTSP
+ exentension (rtspwms) and the RealMedia RTSP extension (rtspreal).
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ The possible network families.
+
+ unknown network family
+
+
+ internet
+
+
+ internet V6
+
+
+
+ Enumeration of rtsp header fields
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ The different transport methods.
+
+ invalid transport flag
+
+
+ stream data over UDP
+
+
+ stream data over UDP multicast
+
+
+ stream data over TCP
+
+
+ stream data tunneled over HTTP.
+
+
+ encrypt TCP and HTTP with TLS
+
+
+
+ Provides methods for creating and parsing request, response and data messages.
+
+ the message type
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Add a header with key @field and @value to @msg. This function takes a copy
+of @value.
+
+ a #GstRTSPResult.
+
+
+
+
+ a #GstRTSPMessage
+
+
+
+ a #GstRTSPHeaderField
+
+
+
+ the value of the header
+
+
+
+
+
+ Add a header with key @header and @value to @msg. This function takes a copy
+of @value.
+
+ a #GstRTSPResult.
+
+
+
+
+ a #GstRTSPMessage
+
+
+
+ header string
+
+
+
+ the value of the header
+
+
+
+
+
+ Append the currently configured headers in @msg to the #GString @str suitable
+for transmission.
+
+ #GST_RTSP_OK.
+
+
+
+
+ a #GstRTSPMessage
+
+
+
+ a string
+
+
+
+
+
+ Allocate a new copy of @msg and store the result in @copy. The value in
+@copy should be release with gst_rtsp_message_free function.
+
+ a #GstRTSPResult
+
+
+
+
+ a #GstRTSPMessage
+
+
+
+ pointer to new #GstRTSPMessage
+
+
+
+
+
+ Dump the contents of @msg to stdout.
+
+ #GST_RTSP_OK.
+
+
+
+
+ a #GstRTSPMessage
+
+
+
+
+
+ Free the memory used by @msg.
+
+ a #GstRTSPResult.
+
+
+
+
+ a #GstRTSPMessage
+
+
+
+
+
+ Get the body of @msg. @data remains valid for as long as @msg is valid and
+unchanged.
+
+ #GST_RTSP_OK.
+
+
+
+
+ a #GstRTSPMessage
+
+
+
+ location for the data
+
+
+
+
+
+ location for the size of @data
+
+
+
+
+
+ Get the @indx header value with key @field from @msg. The result in @value
+stays valid as long as it remains present in @msg.
+
+ #GST_RTSP_OK when @field was found, #GST_RTSP_ENOTIMPL if the key
+was not found.
+
+
+
+
+ a #GstRTSPMessage
+
+
+
+ a #GstRTSPHeaderField
+
+
+
+ pointer to hold the result
+
+
+
+ the index of the header
+
+
+
+
+
+ Get the @index header value with key @header from @msg. The result in @value
+stays valid as long as it remains present in @msg.
+
+ #GST_RTSP_OK when @field was found, #GST_RTSP_ENOTIMPL if the key
+was not found.
+
+
+
+
+ a #GstRTSPMessage
+
+
+
+ a #GstRTSPHeaderField
+
+
+
+ pointer to hold the result
+
+
+
+ the index of the header
+
+
+
+
+
+ Get the message type of @msg.
+
+ the message type.
+
+
+
+
+ a #GstRTSPMessage
+
+
+
+
+
+ Initialize @msg. This function is mostly used when @msg is allocated on the
+stack. The reverse operation of this is gst_rtsp_message_unset().
+
+ a #GstRTSPResult.
+
+
+
+
+ a #GstRTSPMessage
+
+
+
+
+
+ Initialize a new data #GstRTSPMessage for @channel.
+
+ a #GstRTSPResult.
+
+
+
+
+ a #GstRTSPMessage
+
+
+
+ a channel
+
+
+
+
+
+ Initialize @msg as a request message with @method and @uri. To clear @msg
+again, use gst_rtsp_message_unset().
+
+ a #GstRTSPResult.
+
+
+
+
+ a #GstRTSPMessage
+
+
+
+ the request method to use
+
+
+
+ the uri of the request
+
+
+
+
+
+ Initialize @msg with @code and @reason.
+
+When @reason is %NULL, the default reason for @code will be used.
+
+When @request is not %NULL, the relevant headers will be copied to the new
+response message.
+
+ a #GstRTSPResult.
+
+
+
+
+ a #GstRTSPMessage
+
+
+
+ the status code
+
+
+
+ the status reason or %NULL
+
+
+
+ the request that triggered the response or %NULL
+
+
+
+
+
+ Parses the credentials given in a WWW-Authenticate or Authorization header.
+
+
+ %NULL-terminated array of GstRTSPAuthCredential or %NULL.
+
+
+
+
+
+
+ a #GstRTSPMessage
+
+
+
+ a #GstRTSPHeaderField
+
+
+
+
+
+ Parse the data message @msg and store the channel in @channel.
+
+ a #GstRTSPResult.
+
+
+
+
+ a #GstRTSPMessage
+
+
+
+ location to hold the channel
+
+
+
+
+
+ Parse the request message @msg and store the values @method, @uri and
+@version. The result locations can be %NULL if one is not interested in its
+value.
+
+@uri remains valid for as long as @msg is valid and unchanged.
+
+ a #GstRTSPResult.
+
+
+
+
+ a #GstRTSPMessage
+
+
+
+ location to hold the method
+
+
+
+ location to hold the uri
+
+
+
+ location to hold the version
+
+
+
+
+
+ Parse the response message @msg and store the values @code, @reason and
+@version. The result locations can be %NULL if one is not interested in its
+value.
+
+@reason remains valid for as long as @msg is valid and unchanged.
+
+ a #GstRTSPResult.
+
+
+
+
+ a #GstRTSPMessage
+
+
+
+ location to hold the status code
+
+
+
+ location to hold the status reason
+
+
+
+ location to hold the version
+
+
+
+
+
+ Remove the @indx header with key @field from @msg. If @indx equals -1, all
+headers will be removed.
+
+ a #GstRTSPResult.
+
+
+
+
+ a #GstRTSPMessage
+
+
+
+ a #GstRTSPHeaderField
+
+
+
+ the index of the header
+
+
+
+
+
+ Remove the @index header with key @header from @msg. If @index equals -1,
+all matching headers will be removed.
+
+ a #GstRTSPResult
+
+
+
+
+ a #GstRTSPMessage
+
+
+
+ the header string
+
+
+
+ the index of the header
+
+
+
+
+
+ Set the body of @msg to a copy of @data.
+
+ #GST_RTSP_OK.
+
+
+
+
+ a #GstRTSPMessage
+
+
+
+ the data
+
+
+
+
+
+ the size of @data
+
+
+
+
+
+ Take the body of @msg and store it in @data and @size. After this method,
+the body and size of @msg will be set to %NULL and 0 respectively.
+
+ #GST_RTSP_OK.
+
+
+
+
+ a #GstRTSPMessage
+
+
+
+ location for the data
+
+
+
+
+
+ location for the size of @data
+
+
+
+
+
+ Set the body of @msg to @data and @size. This method takes ownership of
+@data.
+
+ #GST_RTSP_OK.
+
+
+
+
+ a #GstRTSPMessage
+
+
+
+ the data
+
+
+
+
+
+ the size of @data
+
+
+
+
+
+ Add a header with key @field and @value to @msg. This function takes
+ownership of @value.
+
+ a #GstRTSPResult.
+
+
+
+
+ a #GstRTSPMessage
+
+
+
+ a #GstRTSPHeaderField
+
+
+
+ the value of the header
+
+
+
+
+
+ Add a header with key @header and @value to @msg. This function takes
+ownership of @value, but not of @header.
+
+ a #GstRTSPResult.
+
+
+
+
+ a #GstRTSPMessage
+
+
+
+ a header string
+
+
+
+ the value of the header
+
+
+
+
+
+ Unset the contents of @msg so that it becomes an uninitialized
+#GstRTSPMessage again. This function is mostly used in combination with
+gst_rtsp_message_init_request(), gst_rtsp_message_init_response() and
+gst_rtsp_message_init_data() on stack allocated #GstRTSPMessage structures.
+
+ #GST_RTSP_OK.
+
+
+
+
+ a #GstRTSPMessage
+
+
+
+
+
+
+ The different supported RTSP methods.
+
+ invalid method
+
+
+ the DESCRIBE method
+
+
+ the ANNOUNCE method
+
+
+ the GET_PARAMETER method
+
+
+ the OPTIONS method
+
+
+ the PAUSE method
+
+
+ the PLAY method
+
+
+ the RECORD method
+
+
+ the REDIRECT method
+
+
+ the SETUP method
+
+
+ the SET_PARAMETER method
+
+
+ the TEARDOWN method
+
+
+ the GET method (HTTP).
+
+
+ the POST method (HTTP).
+
+
+ Convert @method to a string.
+
+ a string representation of @method.
+
+
+
+
+ a #GstRTSPMethod
+
+
+
+
+
+
+ The type of a message.
+
+ invalid message type
+
+
+ RTSP request message
+
+
+ RTSP response message
+
+
+ HTTP request message.
+
+
+ HTTP response message.
+
+
+ data message
+
+
+
+ The transfer profile to use.
+
+ invalid profile
+
+
+ the Audio/Visual profile (RFC 3551)
+
+
+ the secure Audio/Visual profile (RFC 3711)
+
+
+ the Audio/Visual profile with feedback (RFC 4585)
+
+
+ the secure Audio/Visual profile with feedback (RFC 5124)
+
+
+
+ Provides helper functions to deal with time ranges.
+
+ minimum value of the range
+
+
+
+ maximum value of the range
+
+
+
+ Converts the range in-place between different types of units.
+Ranges containing the special value #GST_RTSP_TIME_NOW can not be
+converted as these are only valid for #GST_RTSP_RANGE_NPT.
+
+ %TRUE if the range could be converted
+
+
+
+
+ a #GstRTSPTimeRange
+
+
+
+ the unit to convert the range into
+
+
+
+
+
+ Free the memory allocated by @range.
+
+
+
+
+
+ a #GstRTSPTimeRange
+
+
+
+
+
+ Retrieve the minimum and maximum values from @range converted to
+#GstClockTime in @min and @max.
+
+A value of %GST_CLOCK_TIME_NONE will be used to signal #GST_RTSP_TIME_NOW
+and #GST_RTSP_TIME_END for @min and @max respectively.
+
+UTC times will be converted to nanoseconds since 1900.
+
+ %TRUE on success.
+
+
+
+
+ a #GstRTSPTimeRange
+
+
+
+ result minimum #GstClockTime
+
+
+
+ result maximum #GstClockTime
+
+
+
+
+
+ Parse @rangestr to a #GstRTSPTimeRange.
+
+ #GST_RTSP_OK on success.
+
+
+
+
+ a range string to parse
+
+
+
+ location to hold the #GstRTSPTimeRange result
+
+
+
+
+
+ Convert @range into a string representation.
+
+ The string representation of @range. g_free() after usage.
+
+
+
+
+ a #GstRTSPTimeRange
+
+
+
+
+
+
+ Different possible time range units.
+
+ SMPTE timecode
+
+
+ 29.97 frames per second
+
+
+ 25 frames per second
+
+
+ Normal play time
+
+
+ Absolute time expressed as ISO 8601 timestamps
+
+
+
+ Result codes from the RTSP functions.
+
+ no error
+
+
+ some unspecified error occured
+
+
+ invalid arguments were provided to a function
+
+
+ an operation was canceled
+
+
+ no memory was available for the operation
+
+
+ a host resolve error occured
+
+
+ function not implemented
+
+
+ a system error occured, errno contains more details
+
+
+ a parsing error occured
+
+
+ windows networking could not start
+
+
+ windows networking stack has wrong version
+
+
+ end-of-file was reached
+
+
+ a network problem occured, h_errno contains more details
+
+
+ the host is not an IP host
+
+
+ a timeout occured
+
+
+ the tunnel GET request has been performed
+
+
+ the tunnel POST request has been performed
+
+
+ last error
+
+
+
+ The different RTSP states.
+
+ invalid state
+
+
+ initializing
+
+
+ ready for operation
+
+
+ seeking in progress
+
+
+ playing
+
+
+ recording
+
+
+
+ Enumeration of rtsp status codes
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ A time indication.
+
+ the time of the time
+
+
+
+ seconds when @type is GST_RTSP_TIME_SECONDS,
+ GST_RTSP_TIME_UTC and GST_RTSP_TIME_FRAMES
+
+
+
+
+ Extra fields for a time indication.
+
+ frames and subframes when type in GstRTSPTime is
+ GST_RTSP_TIME_FRAMES
+
+
+
+ year when type is GST_RTSP_TIME_UTC
+
+
+
+ month when type is GST_RTSP_TIME_UTC
+
+
+
+ day when type is GST_RTSP_TIME_UTC
+
+
+
+
+ A time range.
+
+ the time units used
+
+
+
+ the minimum interval
+
+
+
+ the maximum interval
+
+
+
+ extra fields in the minimum interval (Since: 1.2)
+
+
+
+ extra fields in the maximum interval (Since: 1.2)
+
+
+
+
+ Possible time types.
+
+ seconds
+
+
+ now
+
+
+ end
+
+
+ frames and subframes
+
+
+ UTC time
+
+
+
+ The transfer mode to use.
+
+ invalid tansport mode
+
+
+ transfer RTP data
+
+
+ transfer RDT (RealMedia) data
+
+
+
+ Provides helper functions to deal with RTSP transport strings.
+
+ the transport mode
+
+
+
+ the tansport profile
+
+
+
+ the lower transport
+
+
+
+ the destination ip/hostname
+
+
+
+ the source ip/hostname
+
+
+
+ the number of layers
+
+
+
+ if play mode was selected
+
+
+
+ if record mode was selected
+
+
+
+ is append mode was selected
+
+
+
+ the interleave range
+
+
+
+ the time to live for multicast UDP
+
+
+
+ the port pair for multicast sessions
+
+
+
+ the client port pair for receiving data. For TCP
+ based transports, applications can use this field to store the
+ sender and receiver ports of the client.
+
+
+
+ the server port pair for receiving data. For TCP
+ based transports, applications can use this field to store the
+ sender and receiver ports of the server.
+
+
+
+ the ssrc that the sender/receiver will use
+
+
+
+
+
+
+
+
+ Convert @transport into a string that can be used to signal the transport in
+an RTSP SETUP response.
+
+ a string describing the RTSP transport or %NULL when the transport
+is invalid.
+
+
+
+
+ a #GstRTSPTransport
+
+
+
+
+
+ Free the memory used by @transport.
+
+ #GST_RTSP_OK.
+
+
+
+
+ a #GstRTSPTransport
+
+
+
+
+
+ Get the media type of @transport. This media type is typically
+used to generate #GstCaps events.
+
+ #GST_RTSP_OK.
+
+
+
+
+ a #GstRTSPTransport
+
+
+
+ media type of @transport
+
+
+
+
+
+ Initialize @transport so that it can be used.
+
+ #GST_RTSP_OK.
+
+
+
+
+ a #GstRTSPTransport
+
+
+
+
+
+ Get the #GstElement that can handle the buffers transported over @trans.
+
+It is possible that there are several managers available, use @option to
+selected one.
+
+@manager will contain an element name or %NULL when no manager is
+needed/available for @trans.
+
+ #GST_RTSP_OK.
+
+
+
+
+ a #GstRTSPTransMode
+
+
+
+ location to hold the result
+
+
+
+ option index.
+
+
+
+
+
+ Get the mime type of the transport mode @trans. This mime type is typically
+used to generate #GstCaps events.
+ This functions only deals with the GstRTSPTransMode and only
+ returns the mime type for #GST_RTSP_PROFILE_AVP. Use
+ gst_rtsp_transport_get_media_type() instead.
+
+ #GST_RTSP_OK.
+
+
+
+
+ a #GstRTSPTransMode
+
+
+
+ location to hold the result
+
+
+
+
+
+ Allocate a new initialized #GstRTSPTransport. Use gst_rtsp_transport_free()
+after usage.
+
+ a #GstRTSPResult.
+
+
+
+
+ location to hold the new #GstRTSPTransport
+
+
+
+
+
+ Parse the RTSP transport string @str into @transport.
+
+ a #GstRTSPResult.
+
+
+
+
+ a transport string
+
+
+
+ a #GstRTSPTransport
+
+
+
+
+
+
+ Provides helper functions to handle RTSP urls.
+
+ the transports allowed
+
+
+
+ the family
+
+
+
+ the user
+
+
+
+ the password
+
+
+
+ the host
+
+
+
+ the port
+
+
+
+ the absolute path
+
+
+
+ additional query parameters
+
+
+
+ Make a copy of @url.
+
+ a copy of @url. Free with gst_rtsp_url_free () after usage.
+
+
+
+
+ a #GstRTSPUrl
+
+
+
+
+
+ Splits the path of @url on '/' boundaries, decoding the resulting components,
+
+The decoding performed by this routine is "URI decoding", as defined in RFC
+3986, commonly known as percent-decoding. For example, a string "foo\%2fbar"
+will decode to "foo/bar" -- the \%2f being replaced by the corresponding byte
+with hex value 0x2f. Note that there is no guarantee that the resulting byte
+sequence is valid in any given encoding. As a special case, \%00 is not
+unescaped to NUL, as that would prematurely terminate the string.
+
+Also note that since paths usually start with a slash, the first component
+will usually be the empty string.
+
+ %NULL-terminated array of URL components. Free with
+g_strfreev() when no longer needed.
+
+
+
+
+
+
+ a #GstRTSPUrl
+
+
+
+
+
+ Free the memory used by @url.
+
+
+
+
+
+ a #GstRTSPUrl
+
+
+
+
+
+ Get the port number of @url.
+
+ #GST_RTSP_OK.
+
+
+
+
+ a #GstRTSPUrl
+
+
+
+ location to hold the port
+
+
+
+
+
+ Get a newly allocated string describing the request URI for @url.
+
+ a string with the request URI. g_free() after usage.
+
+
+
+
+ a #GstRTSPUrl
+
+
+
+
+
+ Set the port number in @url to @port.
+
+ #GST_RTSP_OK.
+
+
+
+
+ a #GstRTSPUrl
+
+
+
+ the port
+
+
+
+
+
+ Parse the RTSP @urlstr into a newly allocated #GstRTSPUrl. Free after usage
+with gst_rtsp_url_free().
+
+ a #GstRTSPResult.
+
+
+
+
+ the url string to parse
+
+
+
+ location to hold the result.
+
+
+
+
+
+
+ The supported RTSP versions.
+
+ unknown/invalid version
+
+
+ version 1.0
+
+
+ version 1.1.
+
+
+ version 2.0.
+
+
+ Convert @version to a string.
+
+ a string representation of @version.
+
+
+
+
+ a #GstRTSPVersion
+
+
+
+
+
+
+ Opaque RTSP watch object that can be used for asynchronous RTSP
+operations.
+
+ Adds a #GstRTSPWatch to a context so that it will be executed within that context.
+
+ the ID (greater than 0) for the watch within the GMainContext.
+
+
+
+
+ a #GstRTSPWatch
+
+
+
+ a GMainContext (if NULL, the default context will be used)
+
+
+
+
+
+ Get the maximum amount of bytes and messages that will be queued in @watch.
+See gst_rtsp_watch_set_send_backlog().
+
+
+
+
+
+ a #GstRTSPWatch
+
+
+
+ maximum bytes
+
+
+
+ maximum messages
+
+
+
+
+
+ Reset @watch, this is usually called after gst_rtsp_connection_do_tunnel()
+when the file descriptors of the connection might have changed.
+
+
+
+
+
+ a #GstRTSPWatch
+
+
+
+
+
+ Send a @message using the connection of the @watch. If it cannot be sent
+immediately, it will be queued for transmission in @watch. The contents of
+@message will then be serialized and transmitted when the connection of the
+@watch becomes writable. In case the @message is queued, the ID returned in
+@id will be non-zero and used as the ID argument in the message_sent
+callback.
+
+ #GST_RTSP_OK on success.
+
+
+
+
+ a #GstRTSPWatch
+
+
+
+ a #GstRTSPMessage
+
+
+
+ location for a message ID or %NULL
+
+
+
+
+
+ When @flushing is %TRUE, abort a call to gst_rtsp_watch_wait_backlog()
+and make sure gst_rtsp_watch_write_data() returns immediately with
+#GST_RTSP_EINTR. And empty the queue.
+
+
+
+
+
+ a #GstRTSPWatch
+
+
+
+ new flushing state
+
+
+
+
+
+ Set the maximum amount of bytes and messages that will be queued in @watch.
+When the maximum amounts are exceeded, gst_rtsp_watch_write_data() and
+gst_rtsp_watch_send_message() will return #GST_RTSP_ENOMEM.
+
+A value of 0 for @bytes or @messages means no limits.
+
+
+
+
+
+ a #GstRTSPWatch
+
+
+
+ maximum bytes
+
+
+
+ maximum messages
+
+
+
+
+
+ Decreases the reference count of @watch by one. If the resulting reference
+count is zero the watch and associated memory will be destroyed.
+
+
+
+
+
+ a #GstRTSPWatch
+
+
+
+
+
+ Wait until there is place in the backlog queue, @timeout is reached
+or @watch is set to flushing.
+
+If @timeout is %NULL this function can block forever. If @timeout
+contains a valid timeout, this function will return %GST_RTSP_ETIMEOUT
+after the timeout expired.
+
+The typically use of this function is when gst_rtsp_watch_write_data
+returns %GST_RTSP_ENOMEM. The caller then calls this function to wait for
+free space in the backlog queue and try again.
+
+ %GST_RTSP_OK when if there is room in queue.
+ %GST_RTSP_ETIMEOUT when @timeout was reached.
+ %GST_RTSP_EINTR when @watch is flushing
+ %GST_RTSP_EINVAL when called with invalid parameters.
+
+
+
+
+ a #GstRTSPWatch
+
+
+
+ a #GTimeVal timeout
+
+
+
+
+
+ Write @data using the connection of the @watch. If it cannot be sent
+immediately, it will be queued for transmission in @watch. The contents of
+@message will then be serialized and transmitted when the connection of the
+@watch becomes writable. In case the @message is queued, the ID returned in
+@id will be non-zero and used as the ID argument in the message_sent
+callback.
+
+This function will take ownership of @data and g_free() it after use.
+
+If the amount of queued data exceeds the limits set with
+gst_rtsp_watch_set_send_backlog(), this function will return
+#GST_RTSP_ENOMEM.
+
+ #GST_RTSP_OK on success. #GST_RTSP_ENOMEM when the backlog limits
+are reached. #GST_RTSP_EINTR when @watch was flushing.
+
+
+
+
+ a #GstRTSPWatch
+
+
+
+ the data to queue
+
+
+
+
+
+ the size of @data
+
+
+
+ location for a message ID or %NULL
+
+
+
+
+
+ Create a watch object for @conn. The functions provided in @funcs will be
+called with @user_data when activity happened on the watch.
+
+The new watch is usually created so that it can be attached to a
+maincontext with gst_rtsp_watch_attach().
+
+@conn must exist for the entire lifetime of the watch.
+
+ a #GstRTSPWatch that can be used for asynchronous RTSP
+communication. Free with gst_rtsp_watch_unref () after usage.
+
+
+
+
+ a #GstRTSPConnection
+
+
+
+ watch functions
+
+
+
+ user data to pass to @funcs
+
+
+
+ notify when @user_data is not referenced anymore
+
+
+
+
+
+
+ Callback functions from a #GstRTSPWatch.
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ The default RTSP port to connect to.
+
+
+
+ Free a %NULL-terminated array of credentials returned from
+gst_rtsp_message_parse_auth_credentials().
+
+
+
+
+
+ a %NULL-terminated array of #GstRTSPAuthCredential
+
+
+
+
+
+ Accept a new connection on @socket and create a new #GstRTSPConnection for
+handling communication on new socket.
+
+ #GST_RTSP_OK when @conn contains a valid connection.
+
+
+
+
+ a socket
+
+
+
+ storage for a #GstRTSPConnection
+
+
+
+ a #GCancellable to cancel the operation
+
+
+
+
+
+ Create a newly allocated #GstRTSPConnection from @url and store it in @conn.
+The connection will not yet attempt to connect to @url, use
+gst_rtsp_connection_connect().
+
+A copy of @url will be made.
+
+ #GST_RTSP_OK when @conn contains a valid connection.
+
+
+
+
+ a #GstRTSPUrl
+
+
+
+ storage for a #GstRTSPConnection
+
+
+
+
+
+ Create a new #GstRTSPConnection for handling communication on the existing
+socket @socket. The @initial_buffer contains zero terminated data already
+read from @socket which should be used before starting to read new data.
+
+ #GST_RTSP_OK when @conn contains a valid connection.
+
+
+
+
+ a #GSocket
+
+
+
+ the IP address of the other end
+
+
+
+ the port used by the other end
+
+
+
+ data already read from @fd
+
+
+
+ storage for a #GstRTSPConnection
+
+
+
+
+
+ Convert @header to a #GstRTSPHeaderField.
+
+ a #GstRTSPHeaderField for @header or #GST_RTSP_HDR_INVALID if the
+header field is unknown.
+
+
+
+
+ a header string
+
+
+
+
+
+ Convert @method to a #GstRTSPMethod.
+
+ a #GstRTSPMethod for @method or #GST_RTSP_INVALID if the
+method is unknown.
+
+
+
+
+ a method
+
+
+
+
+
+ Calculates the digest auth response from the values given by the server and
+the username and password. See RFC2069 for details.
+
+Currently only supported algorithm "md5".
+
+ Authentication response or %NULL if unsupported
+
+
+
+
+ Hash algorithm to use, or %NULL for MD5
+
+
+
+ Request method, e.g. PLAY
+
+
+
+ Realm
+
+
+
+ Username
+
+
+
+ Password
+
+
+
+ Original request URI
+
+
+
+ Nonce
+
+
+
+
+
+ Check whether @field may appear multiple times in a message.
+
+ %TRUE if multiple headers are allowed.
+
+
+
+
+ a #GstRTSPHeaderField
+
+
+
+
+
+ Convert @field to a string.
+
+ a string representation of @field.
+
+
+
+
+ a #GstRTSPHeaderField
+
+
+
+
+
+ Create a new initialized #GstRTSPMessage. Free with gst_rtsp_message_free().
+
+ a #GstRTSPResult.
+
+
+
+
+ a location for the new #GstRTSPMessage
+
+
+
+
+
+ Create a new data #GstRTSPMessage with @channel and store the
+result message in @msg. Free with gst_rtsp_message_free().
+
+ a #GstRTSPResult.
+
+
+
+
+ a location for the new #GstRTSPMessage
+
+
+
+ the channel
+
+
+
+
+
+ Create a new #GstRTSPMessage with @method and @uri and store the result
+request message in @msg. Free with gst_rtsp_message_free().
+
+ a #GstRTSPResult.
+
+
+
+
+ a location for the new #GstRTSPMessage
+
+
+
+ the request method to use
+
+
+
+ the uri of the request
+
+
+
+
+
+ Create a new response #GstRTSPMessage with @code and @reason and store the
+result message in @msg. Free with gst_rtsp_message_free().
+
+When @reason is %NULL, the default reason for @code will be used.
+
+When @request is not %NULL, the relevant headers will be copied to the new
+response message.
+
+ a #GstRTSPResult.
+
+
+
+
+ a location for the new #GstRTSPMessage
+
+
+
+ the status code
+
+
+
+ the status reason or %NULL
+
+
+
+ the request that triggered the response or %NULL
+
+
+
+
+
+ Convert @method to a string.
+
+ a string representation of @method.
+
+
+
+
+ a #GstRTSPMethod
+
+
+
+
+
+ Convert @options to a string.
+
+ a new string of @options. g_free() after usage.
+
+
+
+
+ one or more #GstRTSPMethod
+
+
+
+
+
+ Convert the comma separated list @options to a #GstRTSPMethod bitwise or
+of methods. This functions is the reverse of gst_rtsp_options_as_text().
+
+ a #GstRTSPMethod
+
+
+
+
+ a comma separated list of options
+
+
+
+
+
+ Converts the range in-place between different types of units.
+Ranges containing the special value #GST_RTSP_TIME_NOW can not be
+converted as these are only valid for #GST_RTSP_RANGE_NPT.
+
+ %TRUE if the range could be converted
+
+
+
+
+ a #GstRTSPTimeRange
+
+
+
+ the unit to convert the range into
+
+
+
+
+
+ Free the memory allocated by @range.
+
+
+
+
+
+ a #GstRTSPTimeRange
+
+
+
+
+
+ Retrieve the minimum and maximum values from @range converted to
+#GstClockTime in @min and @max.
+
+A value of %GST_CLOCK_TIME_NONE will be used to signal #GST_RTSP_TIME_NOW
+and #GST_RTSP_TIME_END for @min and @max respectively.
+
+UTC times will be converted to nanoseconds since 1900.
+
+ %TRUE on success.
+
+
+
+
+ a #GstRTSPTimeRange
+
+
+
+ result minimum #GstClockTime
+
+
+
+ result maximum #GstClockTime
+
+
+
+
+
+ Parse @rangestr to a #GstRTSPTimeRange.
+
+ #GST_RTSP_OK on success.
+
+
+
+
+ a range string to parse
+
+
+
+ location to hold the #GstRTSPTimeRange result
+
+
+
+
+
+ Convert @range into a string representation.
+
+ The string representation of @range. g_free() after usage.
+
+
+
+
+ a #GstRTSPTimeRange
+
+
+
+
+
+ Convert @code to a string.
+
+ a string representation of @code.
+
+
+
+
+ a #GstRTSPStatusCode
+
+
+
+
+
+ Convert @result in a human readable string.
+
+ a newly allocated string. g_free() after usage.
+
+
+
+
+ a #GstRTSPResult
+
+
+
+
+
+ Get the #GstElement that can handle the buffers transported over @trans.
+
+It is possible that there are several managers available, use @option to
+selected one.
+
+@manager will contain an element name or %NULL when no manager is
+needed/available for @trans.
+
+ #GST_RTSP_OK.
+
+
+
+
+ a #GstRTSPTransMode
+
+
+
+ location to hold the result
+
+
+
+ option index.
+
+
+
+
+
+ Get the mime type of the transport mode @trans. This mime type is typically
+used to generate #GstCaps events.
+ This functions only deals with the GstRTSPTransMode and only
+ returns the mime type for #GST_RTSP_PROFILE_AVP. Use
+ gst_rtsp_transport_get_media_type() instead.
+
+ #GST_RTSP_OK.
+
+
+
+
+ a #GstRTSPTransMode
+
+
+
+ location to hold the result
+
+
+
+
+
+ Allocate a new initialized #GstRTSPTransport. Use gst_rtsp_transport_free()
+after usage.
+
+ a #GstRTSPResult.
+
+
+
+
+ location to hold the new #GstRTSPTransport
+
+
+
+
+
+ Parse the RTSP transport string @str into @transport.
+
+ a #GstRTSPResult.
+
+
+
+
+ a transport string
+
+
+
+ a #GstRTSPTransport
+
+
+
+
+
+ Parse the RTSP @urlstr into a newly allocated #GstRTSPUrl. Free after usage
+with gst_rtsp_url_free().
+
+ a #GstRTSPResult.
+
+
+
+
+ the url string to parse
+
+
+
+ location to hold the result.
+
+
+
+
+
+ Convert @version to a string.
+
+ a string representation of @version.
+
+
+
+
+ a #GstRTSPVersion
+
+
+
+
+
+ Create a watch object for @conn. The functions provided in @funcs will be
+called with @user_data when activity happened on the watch.
+
+The new watch is usually created so that it can be attached to a
+maincontext with gst_rtsp_watch_attach().
+
+@conn must exist for the entire lifetime of the watch.
+
+ a #GstRTSPWatch that can be used for asynchronous RTSP
+communication. Free with gst_rtsp_watch_unref () after usage.
+
+
+
+
+ a #GstRTSPConnection
+
+
+
+ watch functions
+
+
+
+ user data to pass to @funcs
+
+
+
+ notify when @user_data is not referenced anymore
+
+
+
+
+
+
diff --git a/girs/GstSdp-1.0.gir b/girs/GstSdp-1.0.gir
new file mode 100644
index 0000000000..b877d90067
--- /dev/null
+++ b/girs/GstSdp-1.0.gir
@@ -0,0 +1,4318 @@
+
+
+
+
+
+
+
+
+ The different cache types
+
+ The envelope key MUST NOT be cached
+
+
+ The envelope key MUST be cached
+
+
+ The envelope key MUST be cached, but only
+ to be used for the specific CSB.
+
+
+
+
+
+ The encryption algorithm used to encrypt the Encr data field
+
+ no encryption
+
+
+ AES-CM using a 128-bit key
+
+
+ AES Key Wrap using a 128-bit key
+
+
+
+
+
+ The key validity type
+
+ No specific usage rule
+
+
+ The key is associated with the SPI/MKI
+
+
+ The key has a start and expiration time
+
+
+
+ The type of key.
+
+ a TEK Generation Key
+
+
+ Traffic-Encrypting Key
+
+
+
+ Specifies the authentication algorithm used
+
+ no authentication
+
+
+ HMAC-SHA-1-160
+
+
+
+ The Security policy Map item for SRTP
+
+ The security policy applied for the stream with @ssrc
+
+
+
+ the SSRC that must be used for the stream
+
+
+
+ current rollover counter
+
+
+
+
+ Specifies the method of uniquely mapping Crypto Sessions to the security
+protocol sessions.
+
+
+
+
+ Structure holding the information of the MIKEY message
+
+
+
+
+ the version
+
+
+
+ the #GstMIKEYType message type
+
+
+
+ verify flag
+
+
+
+ a #GstMIKEYPRFFunc
+
+
+
+ Identifies the Crypto Session Bundle
+
+
+
+ a #GstMIKEYMapType
+
+
+
+ map info array of type depending on @map_type
+
+
+
+
+
+ the payload array of #GstMIKEYPayload
+
+
+
+
+
+ Make a new MIKEY message.
+
+ a new #GstMIKEYMessage on success
+
+
+
+
+ Make a new #GstMIKEYMessage from @bytes.
+
+ a new #GstMIKEYMessage
+
+
+
+
+ a #GBytes
+
+
+
+ a #GstMIKEYDecryptInfo
+
+
+
+
+
+ Makes mikey message including:
+ - Security Policy Payload
+ - Key Data Transport Payload
+ - Key Data Sub-Payload
+
+ a #GstMIKEYMessage,
+or %NULL if there is no srtp information in the caps.
+
+
+
+
+ a #GstCaps, including SRTP parameters (srtp/srtcp cipher, authorization, key data)
+
+
+
+
+
+ Parse @size bytes from @data into a #GstMIKEYMessage. @info contains the
+parameters to decrypt and verify the data.
+
+ a #GstMIKEYMessage on success or %NULL when parsing failed and
+@error will be set.
+
+
+
+
+ bytes to read
+
+
+
+
+
+ length of @data
+
+
+
+ #GstMIKEYDecryptInfo
+
+
+
+
+
+ Add a Crypto policy for SRTP to @msg.
+
+ %TRUE on success
+
+
+
+
+ a #GstMIKEYMessage
+
+
+
+ The security policy applied for the stream with @ssrc
+
+
+
+ the SSRC that must be used for the stream
+
+
+
+ current rollover counter
+
+
+
+
+
+ Add a new payload to @msg.
+
+ %TRUE on success
+
+
+
+
+ a #GstMIKEYMessage
+
+
+
+ a #GstMIKEYPayload
+
+
+
+
+
+ Add a new PKE payload to @msg with the given parameters.
+
+ %TRUE on success
+
+
+
+
+ a #GstMIKEYMessage
+
+
+
+ envelope key cache indicator
+
+
+
+ the length of @data
+
+
+
+ the encrypted envelope key
+
+
+
+
+
+
+
+ Add a new RAND payload to @msg with the given parameters.
+
+ %TRUE on success
+
+
+
+
+ a #GstMIKEYMessage
+
+
+
+ the length of @rand
+
+
+
+ random data
+
+
+
+
+
+
+
+ Add a new RAND payload to @msg with @len random bytes.
+
+ %TRUE on success
+
+
+
+
+ a #GstMIKEYMessage
+
+
+
+ length
+
+
+
+
+
+ Add a new T payload to @msg with the given parameters.
+
+ %TRUE on success
+
+
+
+
+ a #GstMIKEYMessage
+
+
+
+ specifies the timestamp type used
+
+
+
+ The timestamp value of the specified @type
+
+
+
+
+
+
+
+ Add a new T payload to @msg that contains the current time
+in NTP-UTC format.
+
+ %TRUE on success
+
+
+
+
+ a #GstMIKEYMessage
+
+
+
+
+
+
+ a #gchar, base64-encoded data
+
+
+
+
+ a #GstMIKEYMessage
+
+
+
+
+
+ Find the @nth occurence of the payload with @type in @msg.
+
+ the @nth #GstMIKEYPayload of @type.
+
+
+
+
+ a #GstMIKEYMessage
+
+
+
+ a #GstMIKEYPayloadType
+
+
+
+ payload to find
+
+
+
+
+
+ Get the policy information of @msg at @idx.
+
+ a #GstMIKEYMapSRTP
+
+
+
+
+ a #GstMIKEYMessage
+
+
+
+ an index
+
+
+
+
+
+ Get the number of crypto sessions in @msg.
+
+ the number of crypto sessions
+
+
+
+
+ a #GstMIKEYMessage
+
+
+
+
+
+ Get the number of payloads in @msg.
+
+ the number of payloads in @msg
+
+
+
+
+ a #GstMIKEYMessage
+
+
+
+
+
+ Get the #GstMIKEYPayload at @idx in @msg
+
+ the #GstMIKEYPayload at @idx. The payload
+remains valid for as long as it is part of @msg.
+
+
+
+
+ a #GstMIKEYMessage
+
+
+
+ an index
+
+
+
+
+
+ Insert a Crypto Session map for SRTP in @msg at @idx
+
+When @idx is -1, the policy will be appended.
+
+ %TRUE on success
+
+
+
+
+ a #GstMIKEYMessage
+
+
+
+ the index to insert at
+
+
+
+ the map info
+
+
+
+
+
+ Insert the @payload at index @idx in @msg. If @idx is -1, the payload
+will be appended to @msg.
+
+ %TRUE on success
+
+
+
+
+ a #GstMIKEYMessage
+
+
+
+ an index
+
+
+
+ a #GstMIKEYPayload
+
+
+
+
+
+ Remove the SRTP policy at @idx.
+
+ %TRUE on success
+
+
+
+
+ a #GstMIKEYMessage
+
+
+
+ the index to remove
+
+
+
+
+
+ Remove the payload in @msg at @idx
+
+ %TRUE on success
+
+
+
+
+ a #GstMIKEYMessage
+
+
+
+ an index
+
+
+
+
+
+ Replace a Crypto Session map for SRTP in @msg at @idx with @map.
+
+ %TRUE on success
+
+
+
+
+ a #GstMIKEYMessage
+
+
+
+ the index to insert at
+
+
+
+ the map info
+
+
+
+
+
+ Replace the payload at @idx in @msg with @payload.
+
+ %TRUE on success
+
+
+
+
+ a #GstMIKEYMessage
+
+
+
+ an index
+
+
+
+ a #GstMIKEYPayload
+
+
+
+
+
+ Set the information in @msg.
+
+ %TRUE on success
+
+
+
+
+ a #GstMIKEYMessage
+
+
+
+ a version
+
+
+
+ a #GstMIKEYType
+
+
+
+ verify flag
+
+
+
+ the #GstMIKEYPRFFunc function to use
+
+
+
+ the Crypto Session Bundle id
+
+
+
+ the #GstMIKEYCSIDMapType
+
+
+
+
+
+ Convert @msg to a #GBytes.
+
+ a new #GBytes for @msg.
+
+
+
+
+ a #GstMIKEYMessage
+
+
+
+ a #GstMIKEYEncryptInfo
+
+
+
+
+
+
+ %TRUE on success
+
+
+
+
+ a #GstMIKEYMessage
+
+
+
+ a #GstCaps to be filled with SRTP parameters (srtp/srtcp cipher, authorization, key data)
+
+
+
+
+
+
+ The PRF function that has been/will be used for key derivation
+
+ MIKEY-1 PRF function
+
+
+
+ Hold the common fields for all payloads
+
+
+
+
+ the payload type
+
+
+
+ length of the payload
+
+
+
+ Make a new #GstMIKEYPayload with @type.
+
+ a new #GstMIKEYPayload or %NULL on failure.
+
+
+
+
+ a #GstMIKEYPayloadType
+
+
+
+
+
+ Add a new sub payload to @payload.
+
+ %TRUE on success.
+
+
+
+
+ a #GstMIKEYPayload
+
+
+
+ a #GstMIKEYPayload to add
+
+
+
+
+
+ Get the number of sub payloads of @payload. @payload should be of type
+%GST_MIKEY_PT_KEMAC.
+
+ the number of sub payloads in @payload
+
+
+
+
+ a #GstMIKEYPayload
+
+
+
+
+
+ Get the sub payload of @payload at @idx. @payload should be of type
+%GST_MIKEY_PT_KEMAC.
+
+ the #GstMIKEYPayload at @idx.
+
+
+
+
+ a #GstMIKEYPayload
+
+
+
+ an index
+
+
+
+
+
+ Remove the sub payload at @idx in @payload.
+
+ %TRUE on success.
+
+
+
+
+ a #GstMIKEYPayload
+
+
+
+ the index to remove
+
+
+
+
+
+ Set the KEMAC parameters. @payload should point to a %GST_MIKEY_PT_KEMAC
+payload.
+
+ %TRUE on success
+
+
+
+
+ a #GstMIKEYPayload
+
+
+
+ the #GstMIKEYEncAlg
+
+
+
+ a #GstMIKEYMacAlg
+
+
+
+
+
+ Set the key validity period in the %GST_MIKEY_PT_KEY_DATA @payload.
+
+ %TRUE on success
+
+
+
+
+ a #GstMIKEYPayload
+
+
+
+ the length of @vf_data
+
+
+
+ the Valid From data
+
+
+
+
+
+ the length of @vt_data
+
+
+
+ the Valid To data
+
+
+
+
+
+
+
+ Set @key_len bytes of @key_data of type @key_type as the key for the
+%GST_MIKEY_PT_KEY_DATA @payload.
+
+ %TRUE on success
+
+
+
+
+ a #GstMIKEYPayload
+
+
+
+ a #GstMIKEYKeyDataType
+
+
+
+ the length of @key_data
+
+
+
+ the key of type @key_type
+
+
+
+
+
+
+
+ Set the salt key data. If @salt_len is 0 and @salt_data is %NULL, the
+salt data will be removed.
+
+ %TRUE on success
+
+
+
+
+ a #GstMIKEYPayload
+
+
+
+ the length of @salt_data
+
+
+
+ the salt
+
+
+
+
+
+
+
+ Set the SPI/MKI validity in the %GST_MIKEY_PT_KEY_DATA @payload.
+
+ %TRUE on success
+
+
+
+
+ a #GstMIKEYPayload
+
+
+
+ the length of @spi_data
+
+
+
+ the SPI/MKI data
+
+
+
+
+
+
+
+ Set the PKE values in @payload. @payload must be of type
+%GST_MIKEY_PT_PKE.
+
+ %TRUE on success
+
+
+
+
+ a #GstMIKEYPayload
+
+
+
+ envelope key cache indicator
+
+
+
+ the length of @data
+
+
+
+ the encrypted envelope key
+
+
+
+
+
+
+
+ Set the random values in a %GST_MIKEY_PT_RAND @payload.
+
+ %TRUE on success
+
+
+
+
+ a #GstMIKEYPayload
+
+
+
+ the length of @rand
+
+
+
+ random values
+
+
+
+
+
+
+
+ Add a new parameter to the %GST_MIKEY_PT_SP @payload with @type, @len
+and @val.
+
+ %TRUE on success
+
+
+
+
+ a #GstMIKEYPayload
+
+
+
+ a type
+
+
+
+ a length
+
+
+
+ @len bytes of data
+
+
+
+
+
+
+
+ Get the number of security policy parameters in a %GST_MIKEY_PT_SP
+@payload.
+
+ the number of parameters in @payload
+
+
+
+
+ a #GstMIKEYPayload
+
+
+
+
+
+ Get the Security Policy parameter in a %GST_MIKEY_PT_SP @payload
+at @idx.
+
+ the #GstMIKEYPayloadSPParam at @idx in @payload
+
+
+
+
+ a #GstMIKEYPayload
+
+
+
+ an index
+
+
+
+
+
+ Remove the Security Policy parameters from a %GST_MIKEY_PT_SP
+@payload at @idx.
+
+ %TRUE on success
+
+
+
+
+ a #GstMIKEYPayload
+
+
+
+ an index
+
+
+
+
+
+ Set the Security Policy parameters for @payload.
+
+ %TRUE on success
+
+
+
+
+ a #GstMIKEYPayload
+
+
+
+ the policy number
+
+
+
+ a #GstMIKEYSecProto
+
+
+
+
+
+ Set the timestamp in a %GST_MIKEY_PT_T @payload.
+
+ %TRUE on success
+
+
+
+
+ a #GstMIKEYPayload
+
+
+
+ the #GstMIKEYTSType
+
+
+
+ the timestamp value
+
+
+
+
+
+
+
+
+ A structure holding the KEMAC payload
+
+ the common #GstMIKEYPayload
+
+
+
+ the #GstMIKEYEncAlg
+
+
+
+ the #GstMIKEYMacAlg
+
+
+
+ the subpayloads
+
+
+
+
+
+
+ The Key data payload contains key material. It should be added as sub
+payload to the KEMAC.
+
+ the payload header
+
+
+
+
+
+
+ length of @key_data
+
+
+
+
+
+
+ the length of @salt_data, can be 0
+
+
+
+ salt data
+
+
+
+ the Key Validity type
+
+
+
+ length of @kv_data
+
+
+
+
+
+ key validity data
+
+
+
+
+
+
+ The Envelope data payload contains the encrypted envelope key that is
+used in the public-key transport to protect the data in the Key data
+transport payload. The encryption algorithm used is implicit from
+the certificate/public key used.
+
+ the common #GstMIKEYPayload
+
+
+
+ envelope key cache indicator
+
+
+
+ length of @data
+
+
+
+ the encrypted envelope key
+
+
+
+
+ The RAND payload consists of a (pseudo-)random bit-string
+
+ the payload header
+
+
+
+ the length of @rand
+
+
+
+ random values
+
+
+
+
+ The Security Policy payload defines a set of policies that apply to a
+specific security protocol
+
+ the payload header
+
+
+
+ the policy number
+
+
+
+ the security protocol
+
+
+
+ array of #GstMIKEYPayloadPSParam
+
+
+
+
+
+
+ A Type/Length/Value field for security paramaters
+
+ specifies the type of the parameter
+
+
+
+ specifies the length of @val
+
+
+
+ specifies the value of the parameter
+
+
+
+
+ The timestamp payload carries the timestamp information
+
+ the payload header
+
+
+
+ a #GstMIKEYTSType
+
+
+
+ the timestamp value
+
+
+
+
+ Different MIKEY Payload types.
+
+ Last payload
+
+
+ Key data transport payload
+
+
+ Envelope data payload
+
+
+ DH data payload
+
+
+ Signature payload
+
+
+ Timestamp payload
+
+
+ ID payload
+
+
+ Certificate Payload
+
+
+ Cert hash payload
+
+
+ Verfication message payload
+
+
+ Security Policy payload
+
+
+ RAND payload
+
+
+ Error payload
+
+
+ Key data sub-payload
+
+
+ General Extension Payload
+
+
+
+ Specifies the security protocol
+
+
+
+
+ This policy specifies the parameters for SRTP and SRTCP
+
+ Encryption algorithm
+
+
+ Session Encr. key length
+
+
+ Authentication algorithm
+
+
+ Session Auth. key length
+
+
+ Session Salt key length
+
+
+ SRTP Pseudo Random Function
+
+
+ Key derivation rate
+
+
+ SRTP encryption off/on, 0 if off, 1 if on
+
+
+ SRTCP encryption off/on, 0 if off, 1 if on
+
+
+ sender's FEC order
+
+
+ SRTP authentication off/on, 0 if off, 1 if on
+
+
+ Authentication tag length
+
+
+ SRTP prefix length
+
+
+
+ Specifies the timestamp type.
+
+ an NTP time in UTC timezone
+
+
+ an NTP time
+
+
+ a counter
+
+
+
+ Different MIKEY data types.
+
+ Invalid type
+
+
+ Initiator's pre-shared key message
+
+
+ Verification message of a Pre-shared key message
+
+
+ Initiator's public-key transport message
+
+
+ Verification message of a public-key message
+
+
+ Initiator's DH exchange message
+
+
+ Responder's DH exchange message
+
+
+ Error message
+
+
+
+ The supported MIKEY version 1.
+
+
+
+ The contents of the SDP "a=" field which contains a key/value pair.
+
+ the attribute key
+
+
+
+ the attribute value or NULL when it was a property attribute
+
+
+
+ Clear the attribute.
+
+ @GST_SDP_OK.
+
+
+
+
+ a #GstSDPAttribute
+
+
+
+
+
+ Set the attribute with @key and @value.
+
+ @GST_SDP_OK.
+
+
+
+
+ a #GstSDPAttribute
+
+
+
+ the key
+
+
+
+ the value
+
+
+
+
+
+
+ The contents of the SDP "b=" field which specifies the proposed bandwidth to
+be used by the session or media.
+
+ the bandwidth modifier type
+
+
+
+ the bandwidth in kilobits per second
+
+
+
+ Reset the bandwidth information in @bw.
+
+ a #GstSDPResult.
+
+
+
+
+ a #GstSDPBandwidth
+
+
+
+
+
+ Set bandwidth information in @bw.
+
+ a #GstSDPResult.
+
+
+
+
+ a #GstSDPBandwidth
+
+
+
+ the bandwidth modifier type
+
+
+
+ the bandwidth in kilobits per second
+
+
+
+
+
+
+ The contents of the SDP "c=" field which contains connection data.
+
+ the type of network. "IN" is defined to have the meaning
+ "Internet".
+
+
+
+ the type of @address.
+
+
+
+ the address
+
+
+
+ the time to live of the address
+
+
+
+ the number of layers
+
+
+
+ Clear the connection.
+
+ @GST_SDP_OK.
+
+
+
+
+ a #GstSDPConnection
+
+
+
+
+
+ Set the connection with the given parameters.
+
+ @GST_SDP_OK.
+
+
+
+
+ a #GstSDPConnection
+
+
+
+ the type of network. "IN" is defined to have the meaning
+"Internet".
+
+
+
+ the type of address.
+
+
+
+ the address
+
+
+
+ the time to live of the address
+
+
+
+ the number of layers
+
+
+
+
+
+
+ The contents of the SDP "k=" field which is used to convey encryption
+keys.
+
+ the encryption type
+
+
+
+ the encryption data
+
+
+
+
+ The contents of the SDP "m=" field with all related fields.
+
+ the media type
+
+
+
+ the transport port to which the media stream will be sent
+
+
+
+ the number of ports or -1 if only one port was specified
+
+
+
+ the transport protocol
+
+
+
+ an array of #gchar formats
+
+
+
+
+
+ the media title
+
+
+
+ array of #GstSDPConnection with media connection information
+
+
+
+
+
+ array of #GstSDPBandwidth with media bandwidth information
+
+
+
+
+
+ the encryption key
+
+
+
+ array of #GstSDPAttribute with the additional media attributes
+
+
+
+
+
+ Add the attribute with @key and @value to @media.
+
+ #GST_SDP_OK.
+
+
+
+
+ a #GstSDPMedia
+
+
+
+ a key
+
+
+
+ a value
+
+
+
+
+
+ Add the bandwidth information with @bwtype and @bandwidth to @media.
+
+ #GST_SDP_OK.
+
+
+
+
+ a #GstSDPMedia
+
+
+
+ the bandwidth modifier type
+
+
+
+ the bandwidth in kilobits per second
+
+
+
+
+
+ Add the given connection parameters to @media.
+
+ a #GstSDPResult.
+
+
+
+
+ a #GstSDPMedia
+
+
+
+ the type of network. "IN" is defined to have the meaning
+"Internet".
+
+
+
+ the type of address.
+
+
+
+ the address
+
+
+
+ the time to live of the address
+
+
+
+ the number of layers
+
+
+
+
+
+ Add the format information to @media.
+
+ #GST_SDP_OK.
+
+
+
+
+ a #GstSDPMedia
+
+
+
+ the format
+
+
+
+
+
+ Convert the contents of @media to a text string.
+
+ A dynamically allocated string representing the media.
+
+
+
+
+ a #GstSDPMedia
+
+
+
+
+
+ Get the number of attribute fields in @media.
+
+ the number of attributes in @media.
+
+
+
+
+ a #GstSDPMedia
+
+
+
+
+
+ Mapping of attributes of #GstSDPMedia to #GstCaps
+
+ a #GstSDPResult.
+
+
+
+
+ a #GstSDPMedia
+
+
+
+ a #GstCaps
+
+
+
+
+
+ Get the number of bandwidth fields in @media.
+
+ the number of bandwidths in @media.
+
+
+
+
+ a #GstSDPMedia
+
+
+
+
+
+ Get the number of connection fields in @media.
+
+ the number of connections in @media.
+
+
+
+
+ a #GstSDPMedia
+
+
+
+
+
+ Allocate a new copy of @media and store the result in @copy. The value in
+@copy should be release with gst_sdp_media_free function.
+
+ a #GstSDPResult
+
+
+
+
+ a #GstSDPMedia
+
+
+
+ pointer to new #GstSDPMedia
+
+
+
+
+
+ Get the number of formats in @media.
+
+ the number of formats in @media.
+
+
+
+
+ a #GstSDPMedia
+
+
+
+
+
+ Free all resources allocated by @media. @media should not be used anymore after
+this function. This function should be used when @media was dynamically
+allocated with gst_sdp_media_new().
+
+ a #GstSDPResult.
+
+
+
+
+ a #GstSDPMedia
+
+
+
+
+
+ Get the attribute at position @idx in @media.
+
+ the #GstSDPAttribute at position @idx.
+
+
+
+
+ a #GstSDPMedia
+
+
+
+ an index
+
+
+
+
+
+ Get the first attribute value for @key in @media.
+
+ the first attribute value for @key.
+
+
+
+
+ a #GstSDPMedia
+
+
+
+ a key
+
+
+
+
+
+ Get the @nth attribute value for @key in @media.
+
+ the @nth attribute value.
+
+
+
+
+ a #GstSDPMedia
+
+
+
+ a key
+
+
+
+ an index
+
+
+
+
+
+ Get the bandwidth at position @idx in @media.
+
+ the #GstSDPBandwidth at position @idx.
+
+
+
+
+ a #GstSDPMedia
+
+
+
+ an index
+
+
+
+
+
+ Mapping of caps from SDP fields:
+
+a=rtpmap:(payload) (encoding_name)/(clock_rate)[/(encoding_params)]
+
+a=framesize:(payload) (width)-(height)
+
+a=fmtp:(payload) (param)[=(value)];...
+
+ a #GstCaps, or %NULL if an error happened
+
+
+
+
+ a #GstSDPMedia
+
+
+
+ a payload type
+
+
+
+
+
+ Get the connection at position @idx in @media.
+
+ the #GstSDPConnection at position @idx.
+
+
+
+
+ a #GstSDPMedia
+
+
+
+ an index
+
+
+
+
+
+ Get the format information at position @idx in @media.
+
+ the format at position @idx.
+
+
+
+
+ a #GstSDPMedia
+
+
+
+ an index
+
+
+
+
+
+ Get the information of @media
+
+ the information of @media.
+
+
+
+
+ a #GstSDPMedia
+
+
+
+
+
+ Get the encryption information from @media.
+
+ a #GstSDPKey.
+
+
+
+
+ a #GstSDPMedia
+
+
+
+
+
+ Get the media description of @media.
+
+ the media description.
+
+
+
+
+ a #GstSDPMedia
+
+
+
+
+
+ Get the number of ports for @media.
+
+ the number of ports for @media.
+
+
+
+
+ a #GstSDPMedia
+
+
+
+
+
+ Get the port number for @media.
+
+ the port number of @media.
+
+
+
+
+ a #GstSDPMedia
+
+
+
+
+
+ Get the transport protocol of @media
+
+ the transport protocol of @media.
+
+
+
+
+ a #GstSDPMedia
+
+
+
+
+
+ Initialize @media so that its contents are as if it was freshly allocated
+with gst_sdp_media_new(). This function is mostly used to initialize a media
+allocated on the stack. gst_sdp_media_uninit() undoes this operation.
+
+When this function is invoked on newly allocated data (with malloc or on the
+stack), its contents should be set to 0 before calling this function.
+
+ a #GstSDPResult.
+
+
+
+
+ a #GstSDPMedia
+
+
+
+
+
+ Insert the attribute to @media at @idx. When @idx is -1,
+the attribute is appended.
+
+ #GST_SDP_OK.
+
+
+
+
+ a #GstSDPMedia
+
+
+
+ an index
+
+
+
+ a #GstSDPAttribute
+
+
+
+
+
+ Insert the bandwidth information to @media at @idx. When @idx is -1,
+the bandwidth is appended.
+
+ #GST_SDP_OK.
+
+
+
+
+ a #GstSDPMedia
+
+
+
+ an index
+
+
+
+ a #GstSDPBandwidth
+
+
+
+
+
+ Insert the connection information to @media at @idx. When @idx is -1,
+the connection is appended.
+
+ #GST_SDP_OK.
+
+
+
+
+ a #GstSDPMedia
+
+
+
+ an index
+
+
+
+ a #GstSDPConnection
+
+
+
+
+
+ Insert the format information to @media at @idx. When @idx is -1,
+the format is appended.
+
+ #GST_SDP_OK.
+
+
+
+
+ a #GstSDPMedia
+
+
+
+ an index
+
+
+
+ the format
+
+
+
+
+
+ Creates a new #GstMIKEYMessage after parsing the key-mgmt attribute
+from a #GstSDPMedia.
+
+ a #GstSDPResult.
+
+
+
+
+ a #GstSDPMedia
+
+
+
+ pointer to new #GstMIKEYMessage
+
+
+
+
+
+ Remove the attribute in @media at @idx.
+
+ #GST_SDP_OK.
+
+
+
+
+ a #GstSDPMedia
+
+
+
+ an index
+
+
+
+
+
+ Remove the bandwidth information in @media at @idx.
+
+ #GST_SDP_OK.
+
+
+
+
+ a #GstSDPMedia
+
+
+
+ an index
+
+
+
+
+
+ Remove the connection information in @media at @idx.
+
+ #GST_SDP_OK.
+
+
+
+
+ a #GstSDPMedia
+
+
+
+ an index
+
+
+
+
+
+ Remove the format information in @media at @idx.
+
+ #GST_SDP_OK.
+
+
+
+
+ a #GstSDPMedia
+
+
+
+ an index
+
+
+
+
+
+ Replace the attribute in @media at @idx with @attr.
+
+ #GST_SDP_OK.
+
+
+
+
+ a #GstSDPMedia
+
+
+
+ an index
+
+
+
+ a #GstSDPAttribute
+
+
+
+
+
+ Replace the bandwidth information in @media at @idx with @bw.
+
+ #GST_SDP_OK.
+
+
+
+
+ a #GstSDPMedia
+
+
+
+ an index
+
+
+
+ a #GstSDPBandwidth
+
+
+
+
+
+ Replace the connection information in @media at @idx with @conn.
+
+ #GST_SDP_OK.
+
+
+
+
+ a #GstSDPMedia
+
+
+
+ an index
+
+
+
+ a #GstSDPConnection
+
+
+
+
+
+ Replace the format information in @media at @idx with @format.
+
+ #GST_SDP_OK.
+
+
+
+
+ a #GstSDPMedia
+
+
+
+ an index
+
+
+
+ the format
+
+
+
+
+
+ Set the media information of @media to @information.
+
+ #GST_SDP_OK.
+
+
+
+
+ a #GstSDPMedia
+
+
+
+ the media information
+
+
+
+
+
+ Adds the encryption information to @media.
+
+ a #GstSDPResult.
+
+
+
+
+ a #GstSDPMedia
+
+
+
+ the encryption type
+
+
+
+ the encryption data
+
+
+
+
+
+ Set the media description of @media to @med.
+
+ #GST_SDP_OK.
+
+
+
+
+ a #GstSDPMedia
+
+
+
+ the media description
+
+
+
+
+
+ Set the port information in @media.
+
+ #GST_SDP_OK.
+
+
+
+
+ a #GstSDPMedia
+
+
+
+ the port number
+
+
+
+ the number of ports
+
+
+
+
+
+ Set the media transport protocol of @media to @proto.
+
+ #GST_SDP_OK.
+
+
+
+
+ a #GstSDPMedia
+
+
+
+ the media transport protocol
+
+
+
+
+
+ Free all resources allocated in @media. @media should not be used anymore after
+this function. This function should be used when @media was allocated on the
+stack and initialized with gst_sdp_media_init().
+
+ a #GstSDPResult.
+
+
+
+
+ a #GstSDPMedia
+
+
+
+
+
+ Allocate a new GstSDPMedia and store the result in @media.
+
+ a #GstSDPResult.
+
+
+
+
+ pointer to new #GstSDPMedia
+
+
+
+
+
+ Mapping of caps to SDP fields:
+
+a=rtpmap:(payload) (encoding_name) or (clock_rate)[or (encoding_params)]
+
+a=framesize:(payload) (width)-(height)
+
+a=fmtp:(payload) (param)[=(value)];...
+
+a=rtcp-fb:(payload) (param1) [param2]...
+
+ a #GstSDPResult.
+
+
+
+
+ a #GstCaps
+
+
+
+ a #GstSDPMedia
+
+
+
+
+
+
+ The GstSDPMessage helper functions makes it easy to parse and create SDP
+messages.
+
+ the protocol version
+
+
+
+ owner/creator and session identifier
+
+
+
+ session name
+
+
+
+ session information
+
+
+
+ URI of description
+
+
+
+ array of #gchar with email addresses
+
+
+
+
+
+ array of #gchar with phone numbers
+
+
+
+
+
+ connection information for the session
+
+
+
+ array of #GstSDPBandwidth with bandwidth information
+
+
+
+
+
+ array of #GstSDPTime with time descriptions
+
+
+
+
+
+ array of #GstSDPZone with time zone adjustments
+
+
+
+
+
+ encryption key
+
+
+
+ array of #GstSDPAttribute with session attributes
+
+
+
+
+
+ array of #GstSDPMedia with media descriptions
+
+
+
+
+
+ Add the attribute with @key and @value to @msg.
+
+ @GST_SDP_OK.
+
+
+
+
+ a #GstSDPMessage
+
+
+
+ the key
+
+
+
+ the value
+
+
+
+
+
+ Add the specified bandwidth information to @msg.
+
+ a #GstSDPResult.
+
+
+
+
+ a #GstSDPMessage
+
+
+
+ the bandwidth modifier type
+
+
+
+ the bandwidth in kilobits per second
+
+
+
+
+
+ Add @email to the list of emails in @msg.
+
+ a #GstSDPResult.
+
+
+
+
+ a #GstSDPMessage
+
+
+
+ an email
+
+
+
+
+
+ Adds @media to the array of medias in @msg. This function takes ownership of
+the contents of @media so that @media will have to be reinitialized with
+gst_sdp_media_init() before it can be used again.
+
+ a #GstSDPResult.
+
+
+
+
+ a #GstSDPMessage
+
+
+
+ a #GstSDPMedia to add
+
+
+
+
+
+ Add @phone to the list of phones in @msg.
+
+ a #GstSDPResult.
+
+
+
+
+ a #GstSDPMessage
+
+
+
+ a phone
+
+
+
+
+
+ Add time information @start and @stop to @msg.
+
+ a #GstSDPResult.
+
+
+
+
+ a #GstSDPMessage
+
+
+
+ the start time
+
+
+
+ the stop time
+
+
+
+ the repeat times
+
+
+
+
+
+
+
+ Add time zone information to @msg.
+
+ a #GstSDPResult.
+
+
+
+
+ a #GstSDPMessage
+
+
+
+ the NTP time that a time zone adjustment happens
+
+
+
+ the offset from the time when the session was first scheduled
+
+
+
+
+
+ Convert the contents of @msg to a text string.
+
+ A dynamically allocated string representing the SDP description.
+
+
+
+
+ a #GstSDPMessage
+
+
+
+
+
+ Get the number of attributes in @msg.
+
+ the number of attributes in @msg.
+
+
+
+
+ a #GstSDPMessage
+
+
+
+
+
+ Mapping of attributes of #GstSDPMessage to #GstCaps
+
+ a #GstSDPResult.
+
+
+
+
+ a #GstSDPMessage
+
+
+
+ a #GstCaps
+
+
+
+
+
+ Get the number of bandwidth information in @msg.
+
+ the number of bandwidth information in @msg.
+
+
+
+
+ a #GstSDPMessage
+
+
+
+
+
+ Allocate a new copy of @msg and store the result in @copy. The value in
+@copy should be release with gst_sdp_message_free function.
+
+ a #GstSDPResult
+
+
+
+
+ a #GstSDPMessage
+
+
+
+ pointer to new #GstSDPMessage
+
+
+
+
+
+ Dump the parsed contents of @msg to stdout.
+
+ a #GstSDPResult.
+
+
+
+
+ a #GstSDPMessage
+
+
+
+
+
+ Get the number of emails in @msg.
+
+ the number of emails in @msg.
+
+
+
+
+ a #GstSDPMessage
+
+
+
+
+
+ Free all resources allocated by @msg. @msg should not be used anymore after
+this function. This function should be used when @msg was dynamically
+allocated with gst_sdp_message_new().
+
+ a #GstSDPResult.
+
+
+
+
+ a #GstSDPMessage
+
+
+
+
+
+ Get the attribute at position @idx in @msg.
+
+ the #GstSDPAttribute at position @idx.
+
+
+
+
+ a #GstSDPMessage
+
+
+
+ the index
+
+
+
+
+
+ Get the first attribute with key @key in @msg.
+
+ the attribute value of the first attribute with @key.
+
+
+
+
+ a #GstSDPMessage
+
+
+
+ the key
+
+
+
+
+
+ Get the @nth attribute with key @key in @msg.
+
+ the attribute value of the @nth attribute with @key.
+
+
+
+
+ a #GstSDPMessage
+
+
+
+ the key
+
+
+
+ the index
+
+
+
+
+
+ Get the bandwidth at index @idx from @msg.
+
+ a #GstSDPBandwidth.
+
+
+
+
+ a #GstSDPMessage
+
+
+
+ the bandwidth index
+
+
+
+
+
+ Get the connection of @msg.
+
+ a #GstSDPConnection. The result remains valid as long as @msg is valid.
+
+
+
+
+ a #GstSDPMessage
+
+
+
+
+
+ Get the email with number @idx from @msg.
+
+ the email at position @idx.
+
+
+
+
+ a #GstSDPMessage
+
+
+
+ an email index
+
+
+
+
+
+ Get the information in @msg.
+
+ a #GstSDPResult.
+
+
+
+
+ a #GstSDPMessage
+
+
+
+
+
+ Get the encryption information from @msg.
+
+ a #GstSDPKey.
+
+
+
+
+ a #GstSDPMessage
+
+
+
+
+
+ Get the media description at index @idx in @msg.
+
+ a #GstSDPMedia.
+
+
+
+
+ a #GstSDPMessage
+
+
+
+ the index
+
+
+
+
+
+ Get the origin of @msg.
+
+ a #GstSDPOrigin. The result remains valid as long as @msg is valid.
+
+
+
+
+ a #GstSDPMessage
+
+
+
+
+
+ Get the phone with number @idx from @msg.
+
+ the phone at position @idx.
+
+
+
+
+ a #GstSDPMessage
+
+
+
+ a phone index
+
+
+
+
+
+ Get the session name in @msg.
+
+ a #GstSDPResult.
+
+
+
+
+ a #GstSDPMessage
+
+
+
+
+
+ Get time information with index @idx from @msg.
+
+ a #GstSDPTime.
+
+
+
+
+ a #GstSDPMessage
+
+
+
+ the time index
+
+
+
+
+
+ Get the URI in @msg.
+
+ a #GstSDPResult.
+
+
+
+
+ a #GstSDPMessage
+
+
+
+
+
+ Get the version in @msg.
+
+ a #GstSDPResult.
+
+
+
+
+ a #GstSDPMessage
+
+
+
+
+
+ Get time zone information with index @idx from @msg.
+
+ a #GstSDPZone.
+
+
+
+
+ a #GstSDPMessage
+
+
+
+ the zone index
+
+
+
+
+
+ Initialize @msg so that its contents are as if it was freshly allocated
+with gst_sdp_message_new(). This function is mostly used to initialize a message
+allocated on the stack. gst_sdp_message_uninit() undoes this operation.
+
+When this function is invoked on newly allocated data (with malloc or on the
+stack), its contents should be set to 0 before calling this function.
+
+ a #GstSDPResult.
+
+
+
+
+ a #GstSDPMessage
+
+
+
+
+
+ Insert attribute into the array of attributes in @msg
+at index @idx.
+When -1 is given as @idx, the attribute is inserted at the end.
+
+ a #GstSDPResult.
+
+
+
+
+ a #GstSDPMessage
+
+
+
+ an index
+
+
+
+ a #GstSDPAttribute
+
+
+
+
+
+ Insert bandwidth parameters into the array of bandwidths in @msg
+at index @idx.
+When -1 is given as @idx, the bandwidth is inserted at the end.
+
+ a #GstSDPResult.
+
+
+
+
+ a #GstSDPMessage
+
+
+
+ an index
+
+
+
+ the bandwidth
+
+
+
+
+
+ Insert @email into the array of emails in @msg at index @idx.
+When -1 is given as @idx, the email is inserted at the end.
+
+ a #GstSDPResult.
+
+
+
+
+ a #GstSDPMessage
+
+
+
+ an index
+
+
+
+ an email
+
+
+
+
+
+ Insert @phone into the array of phone numbers in @msg at index @idx.
+When -1 is given as @idx, the phone is inserted at the end.
+
+ a #GstSDPResult.
+
+
+
+
+ a #GstSDPMessage
+
+
+
+ a phone index
+
+
+
+ a phone
+
+
+
+
+
+ Insert time parameters into the array of times in @msg
+at index @idx.
+When -1 is given as @idx, the times are inserted at the end.
+
+ a #GstSDPResult.
+
+
+
+
+ a #GstSDPMessage
+
+
+
+ an index
+
+
+
+ a #GstSDPTime
+
+
+
+
+
+ Insert zone parameters into the array of zones in @msg
+at index @idx.
+When -1 is given as @idx, the zone is inserted at the end.
+
+ a #GstSDPResult.
+
+
+
+
+ a #GstSDPMessage
+
+
+
+ an index
+@zone a #GstSDPZone
+
+
+
+
+
+
+
+
+ Get the number of media descriptions in @msg.
+
+ the number of media descriptions in @msg.
+
+
+
+
+ a #GstSDPMessage
+
+
+
+
+
+ Creates a new #GstMIKEYMessage after parsing the key-mgmt attribute
+from a #GstSDPMessage.
+
+ a #GstSDPResult.
+
+
+
+
+ a #GstSDPMessage
+
+
+
+ pointer to new #GstMIKEYMessage
+
+
+
+
+
+ Get the number of phones in @msg.
+
+ the number of phones in @msg.
+
+
+
+
+ a #GstSDPMessage
+
+
+
+
+
+ Remove the attribute in @msg at index @idx.
+
+ a #GstSDPResult.
+
+
+
+
+ a #GstSDPMessage
+
+
+
+ the index
+
+
+
+
+
+ Remove the bandwidth information in @msg at index @idx.
+
+ a #GstSDPResult.
+
+
+
+
+ a #GstSDPMessage
+
+
+
+ the bandwidth index
+
+
+
+
+
+ Remove the email in @msg at index @idx.
+
+ a #GstSDPResult.
+
+
+
+
+ a #GstSDPMessage
+
+
+
+ an email index
+
+
+
+
+
+ Remove the phone number in @msg at index @idx.
+
+ a #GstSDPResult.
+
+
+
+
+ a #GstSDPMessage
+
+
+
+ a phone index
+
+
+
+
+
+ Remove the time information in @msg at index @idx.
+
+ a #GstSDPResult.
+
+
+
+
+ a #GstSDPMessage
+
+
+
+ the index
+
+
+
+
+
+ Remove the zone information in @msg at index @idx.
+
+ a #GstSDPResult.
+
+
+
+
+ a #GstSDPMessage
+
+
+
+ the index
+
+
+
+
+
+ Replace the attribute in @msg at index @idx with @attr.
+
+ a #GstSDPResult.
+
+
+
+
+ a #GstSDPMessage
+
+
+
+ the index
+
+
+
+ a #GstSDPAttribute
+
+
+
+
+
+ Replace the bandwidth information in @msg at index @idx with @bw.
+
+ a #GstSDPResult.
+
+
+
+
+ a #GstSDPMessage
+
+
+
+ the bandwidth index
+
+
+
+ the bandwidth
+
+
+
+
+
+ Replace the email in @msg at index @idx with @email.
+
+ a #GstSDPResult.
+
+
+
+
+ a #GstSDPMessage
+
+
+
+ an email index
+
+
+
+ an email
+
+
+
+
+
+ Replace the phone number in @msg at index @idx with @phone.
+
+ a #GstSDPResult.
+
+
+
+
+ a #GstSDPMessage
+
+
+
+ a phone index
+
+
+
+ a phone
+
+
+
+
+
+ Replace the time information in @msg at index @idx with @t.
+
+ a #GstSDPResult.
+
+
+
+
+ a #GstSDPMessage
+
+
+
+ the index
+
+
+
+ a #GstSDPTime
+
+
+
+
+
+ Replace the zone information in @msg at index @idx with @zone.
+
+ a #GstSDPResult.
+
+
+
+
+ a #GstSDPMessage
+
+
+
+ the index
+
+
+
+ a #GstSDPZone
+
+
+
+
+
+ Configure the SDP connection in @msg with the given parameters.
+
+ a #GstSDPResult.
+
+
+
+
+ a #GstSDPMessage
+
+
+
+ the type of network. "IN" is defined to have the meaning
+"Internet".
+
+
+
+ the type of address.
+
+
+
+ the address
+
+
+
+ the time to live of the address
+
+
+
+ the number of layers
+
+
+
+
+
+ Set the information in @msg.
+
+ a #GstSDPResult.
+
+
+
+
+ a #GstSDPMessage
+
+
+
+ the information
+
+
+
+
+
+ Adds the encryption information to @msg.
+
+ a #GstSDPResult.
+
+
+
+
+ a #GstSDPMessage
+
+
+
+ the encryption type
+
+
+
+ the encryption data
+
+
+
+
+
+ Configure the SDP origin in @msg with the given parameters.
+
+ #GST_SDP_OK.
+
+
+
+
+ a #GstSDPMessage
+
+
+
+ the user name
+
+
+
+ a session id
+
+
+
+ a session version
+
+
+
+ a network type
+
+
+
+ an address type
+
+
+
+ an address
+
+
+
+
+
+ Set the session name in @msg.
+
+ a #GstSDPResult.
+
+
+
+
+ a #GstSDPMessage
+
+
+
+ the session name
+
+
+
+
+
+ Set the URI in @msg.
+
+ a #GstSDPResult.
+
+
+
+
+ a #GstSDPMessage
+
+
+
+ the URI
+
+
+
+
+
+ Set the version in @msg.
+
+ a #GstSDPResult.
+
+
+
+
+ a #GstSDPMessage
+
+
+
+ the version
+
+
+
+
+
+ Get the number of time information entries in @msg.
+
+ the number of time information entries in @msg.
+
+
+
+
+ a #GstSDPMessage
+
+
+
+
+
+ Free all resources allocated in @msg. @msg should not be used anymore after
+this function. This function should be used when @msg was allocated on the
+stack and initialized with gst_sdp_message_init().
+
+ a #GstSDPResult.
+
+
+
+
+ a #GstSDPMessage
+
+
+
+
+
+ Get the number of time zone information entries in @msg.
+
+ the number of time zone information entries in @msg.
+
+
+
+
+ a #GstSDPMessage
+
+
+
+
+
+ Creates a uri from @msg with the given @scheme. The uri has the format:
+
+ \@scheme:///[#type=value *[&type=value]]
+
+ Where each value is url encoded.
+
+ a uri for @msg.
+
+
+
+
+ the uri scheme
+
+
+
+ the #GstSDPMessage
+
+
+
+
+
+ Allocate a new GstSDPMessage and store the result in @msg.
+
+ a #GstSDPResult.
+
+
+
+
+ pointer to new #GstSDPMessage
+
+
+
+
+
+ Parse the contents of @size bytes pointed to by @data and store the result in
+@msg.
+
+ #GST_SDP_OK on success.
+
+
+
+
+ the start of the buffer
+
+
+
+
+
+ the size of the buffer
+
+
+
+ the result #GstSDPMessage
+
+
+
+
+
+ Parse the null-terminated @uri and store the result in @msg.
+
+The uri should be of the form:
+
+ scheme://[address[:ttl=ttl][:noa=noa]]/[sessionname]
+ [#type=value *[&type=value]]
+
+ where value is url encoded. This looslely resembles
+ http://tools.ietf.org/html/draft-fujikawa-sdp-url-01
+
+ #GST_SDP_OK on success.
+
+
+
+
+ the start of the uri
+
+
+
+ the result #GstSDPMessage
+
+
+
+
+
+
+ The contents of the SDP "o=" field which gives the originator of the session
+(their username and the address of the user's host) plus a session id and
+session version number.
+
+ the user's login on the originating host, or it is "-"
+ if the originating host does not support the concept of user ids.
+
+
+
+ is a numeric string such that the tuple of @username, @sess_id,
+ @nettype, @addrtype and @addr form a globally unique identifier for the
+ session.
+
+
+
+ a version number for this announcement
+
+
+
+ the type of network. "IN" is defined to have the meaning
+ "Internet".
+
+
+
+ the type of @addr.
+
+
+
+ the globally unique address of the machine from which the session was
+ created.
+
+
+
+
+ Return values for the SDP functions.
+
+ A successful return value
+
+
+ a function was given invalid parameters
+
+
+
+ The contents of the SDP "t=" field which specify the start and stop times for
+a conference session.
+
+ start time for the conference. The value is the decimal
+ representation of Network Time Protocol (NTP) time values in seconds
+
+
+
+ stop time for the conference. The value is the decimal
+ representation of Network Time Protocol (NTP) time values in seconds
+
+
+
+ repeat times for a session
+
+
+
+
+
+ Reset the time information in @t.
+
+ a #GstSDPResult.
+
+
+
+
+ a #GstSDPTime
+
+
+
+
+
+ Set time information @start, @stop and @repeat in @t.
+
+ a #GstSDPResult.
+
+
+
+
+ a #GstSDPTime
+
+
+
+ the start time
+
+
+
+ the stop time
+
+
+
+ the repeat times
+
+
+
+
+
+
+
+
+ The contents of the SDP "z=" field which allows the sender to
+specify a list of time zone adjustments and offsets from the base
+time.
+
+ the NTP time that a time zone adjustment happens
+
+
+
+ the offset from the time when the session was first scheduled
+
+
+
+ Reset the zone information in @zone.
+
+ a #GstSDPResult.
+
+
+
+
+ a #GstSDPZone
+
+
+
+
+
+ Set zone information in @zone.
+
+ a #GstSDPResult.
+
+
+
+
+ a #GstSDPZone
+
+
+
+ the NTP time that a time zone adjustment happens
+
+
+
+ the offset from the time when the session was first scheduled
+
+
+
+
+
+
+ The Application-Specific Maximum bandwidth modifier.
+
+
+
+ The Conference Total bandwidth modifier.
+
+
+
+ The extension prefix bandwidth modifier.
+
+
+
+ RTCP bandwidth allocated to data receivers (RFC 3556).
+
+
+
+ RTCP bandwidth allocated to active data senders (RFC 3556).
+
+
+
+ Transport Independent Application Specific Maximum bandwidth (RFC 3890).
+
+
+
+ Check if the given @addr is a multicast address.
+
+ TRUE when @addr is multicast.
+
+
+
+
+ a network type
+
+
+
+ an address type
+
+
+
+ an address
+
+
+
+
+
+ Makes key management data
+
+ a #gchar key-mgmt data,
+
+
+
+
+ a #gchar URI
+
+
+
+ a #gchar base64-encoded key data
+
+
+
+
+
+ Allocate a new GstSDPMedia and store the result in @media.
+
+ a #GstSDPResult.
+
+
+
+
+ pointer to new #GstSDPMedia
+
+
+
+
+
+ Mapping of caps to SDP fields:
+
+a=rtpmap:(payload) (encoding_name) or (clock_rate)[or (encoding_params)]
+
+a=framesize:(payload) (width)-(height)
+
+a=fmtp:(payload) (param)[=(value)];...
+
+a=rtcp-fb:(payload) (param1) [param2]...
+
+ a #GstSDPResult.
+
+
+
+
+ a #GstCaps
+
+
+
+ a #GstSDPMedia
+
+
+
+
+
+ Creates a uri from @msg with the given @scheme. The uri has the format:
+
+ \@scheme:///[#type=value *[&type=value]]
+
+ Where each value is url encoded.
+
+ a uri for @msg.
+
+
+
+
+ the uri scheme
+
+
+
+ the #GstSDPMessage
+
+
+
+
+
+ Allocate a new GstSDPMessage and store the result in @msg.
+
+ a #GstSDPResult.
+
+
+
+
+ pointer to new #GstSDPMessage
+
+
+
+
+
+ Parse the contents of @size bytes pointed to by @data and store the result in
+@msg.
+
+ #GST_SDP_OK on success.
+
+
+
+
+ the start of the buffer
+
+
+
+
+
+ the size of the buffer
+
+
+
+ the result #GstSDPMessage
+
+
+
+
+
+ Parse the null-terminated @uri and store the result in @msg.
+
+The uri should be of the form:
+
+ scheme://[address[:ttl=ttl][:noa=noa]]/[sessionname]
+ [#type=value *[&type=value]]
+
+ where value is url encoded. This looslely resembles
+ http://tools.ietf.org/html/draft-fujikawa-sdp-url-01
+
+ #GST_SDP_OK on success.
+
+
+
+
+ the start of the uri
+
+
+
+ the result #GstSDPMessage
+
+
+
+
+
+
diff --git a/girs/GstTag-1.0.gir b/girs/GstTag-1.0.gir
new file mode 100644
index 0000000000..4307b88cdd
--- /dev/null
+++ b/girs/GstTag-1.0.gir
@@ -0,0 +1,1846 @@
+
+
+
+
+
+
+
+
+
+ Direction of contrast processing applied when capturing an image. (string)
+
+The allowed values are:
+ "normal"
+ "soft"
+ "hard"
+
+
+
+ Digital zoom ratio used when capturing an image. (double)
+
+
+
+ Exposure compensation using when capturing an image in EV. (double)
+
+
+
+ Exposure mode used when capturing an image. (string)
+
+The allowed values are:
+ "auto-exposure"
+ "manual-exposure"
+ "auto-bracket"
+
+
+
+ Type of exposure control used when capturing an image. (string)
+
+The allowed values are:
+ "undefined"
+ "manual"
+ "normal" - automatically controlled
+ "aperture-priority" - user selects aperture value
+ "shutter-priority" - user selects shutter speed
+ "creative" - biased towards depth of field
+ "action" - biased towards fast shutter speed
+ "portrait" - closeup, leaving background out of focus
+ "landscape" - landscape photos, background in focus
+
+
+
+ If flash was fired during the capture of an image. (boolean)
+
+Note that if this tag isn't present, it should not be assumed that
+the flash did not fire. It should be treated as unknown.
+
+
+
+ The flash mode selected during the capture of an image. (string)
+
+The allowed values are:
+ "auto"
+ "always"
+ "never"
+
+
+
+ Focal length used when capturing an image, in mm. (double)
+
+
+
+ 35 mm equivalent focal length used when capturing an image, in mm. (double)
+
+
+
+ Focal ratio (f-number) used when capturing an image. (double)
+
+The value stored is the denominator of the focal ratio (f-number).
+For example, if this tag value is 2, the focal ratio is f/2.
+
+
+
+ Gain adjustment applied to an image. (string)
+
+The allowed values are:
+ "none"
+ "low-gain-up"
+ "high-gain-up"
+ "low-gain-down"
+ "high-gain-down"
+
+
+
+ ISO speed used when capturing an image. (integer)
+
+
+
+ Defines the way a camera determines the exposure. (string)
+
+The allowed values are:
+ "unknown"
+ "average"
+ "center-weighted-average"
+ "spot"
+ "multi-spot"
+ "pattern"
+ "partial"
+ "other"
+
+
+
+ Direction of saturation processing applied when capturing an image. (string)
+
+The allowed values are:
+ "normal"
+ "low-saturation"
+ "high-saturation"
+
+
+
+ Scene mode used when capturing an image. (string)
+
+The allowed values are:
+ "standard"
+ "landscape"
+ "portrait"
+ "night-scene"
+
+
+
+ Direction of sharpness processing applied when capturing an image. (string)
+
+The allowed values are:
+ "normal"
+ "soft"
+ "hard"
+
+
+
+ Shutter speed used when capturing an image, in seconds. (fraction)
+
+
+
+ Indicates the source of capture. The device/medium used to do the
+capture. (string)
+
+Allowed values are:
+ "dsc" (= digital still camera)
+ "transparent-scanner"
+ "reflex-scanner"
+ "other"
+
+
+
+ White balance mode used when capturing an image. (string)
+
+The allowed values are:
+ "auto"
+ "manual"
+ "daylight"
+ "cloudy"
+ "tungsten"
+ "fluorescent"
+ "fluorescent h" (newer daylight-calibrated fluorescents)
+ "flash"
+
+
+
+ CDDB disc id in its short form (e.g. 'aa063d0f')
+
+
+
+ CDDB disc id including all details
+
+
+
+ Musicbrainz disc id (e.g. 'ahg7JUcfR3vCYBphSDIogOOWrr0-')
+
+
+
+ Musicbrainz disc id details
+
+
+
+ Annodex CMML clip element tag
+
+
+
+ Annodex CMML head element tag
+
+
+
+ Annodex CMML stream element tag
+
+
+
+ ID3V2 header size considered minimum input for some functions such as
+gst_tag_list_from_id3v2_tag() and gst_tag_get_id3v2_tag_size() for example.
+
+
+
+ Media (image/video) intended horizontal pixel density in ppi. (double)
+
+
+
+ Media (image/video) intended vertical pixel density in ppi. (double)
+
+
+
+ Musical key in which the sound starts. It is represented as a string
+with a maximum length of three characters. The ground keys are
+represented with "A","B","C","D","E", "F" and "G" and halfkeys
+represented with "b" and "#". Minor is represented as "m" (e.g. "Dbm").
+Off key is represented with an "o" only.
+This notation might be extended in the future to support non-minor/major
+keys.
+
+
+
+ MusicBrainz album artist ID
+
+
+
+ MusicBrainz album ID
+
+
+
+ MusicBrainz artist ID
+
+
+
+ MusicBrainz track ID
+
+
+
+ MusicBrainz track TRM ID
+
+
+
+ Provides a base class for demuxing tags at the beginning or end of a
+stream and handles things like typefinding, querying, seeking, and
+different modes of operation (chain-based, pull_range-based, and providing
+downstream elements with random access if upstream supports that). The tag
+is stripped from the output, and all offsets are adjusted for the tag
+sizes, so that to the downstream element the stream will appear as if
+there was no tag at all. Also, once the tag has been parsed, GstTagDemux
+will try to determine the media type of the resulting stream and add a
+source pad with the appropriate caps in order to facilitate auto-plugging.
+
+## Deriving from GstTagDemux
+
+Subclasses have to do four things:
+
+ * In their base init function, they must add a pad template for the sink
+ pad to the element class, describing the media type they can parse in
+ the caps of the pad template.
+ * In their class init function, they must override
+ GST_TAG_DEMUX_CLASS(demux_klass)->identify_tag with their own identify
+ function.
+ * In their class init function, they must override
+ GST_TAG_DEMUX_CLASS(demux_klass)->parse_tag with their own parse
+ function.
+ * In their class init function, they must also set
+ GST_TAG_DEMUX_CLASS(demux_klass)->min_start_size and/or
+ GST_TAG_DEMUX_CLASS(demux_klass)->min_end_size to the minimum size required
+ for the identify function to decide whether the stream has a supported tag
+ or not. A class parsing ID3v1 tags, for example, would set min_end_size to
+ 128 bytes.
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ parent element
+
+
+
+
+
+
+
+
+
+
+
+
+ The #GstTagDemuxClass structure. See documentation at beginning of section
+for details about what subclasses need to override and do.
+
+ the parent class.
+
+
+
+ minimum size required to identify a tag at the start and
+determine its total size. Set to 0 if not interested in start tags.
+Subclasses should set this in their class_init function.
+
+
+
+ minimum size required to identify a tag at the end and
+determine its total size. Set to 0 if not interested in end tags.
+Subclasses should set this in their class_init function.
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Result values from the parse_tag virtual function.
+
+ cannot parse tag, just skip it
+
+
+ call again with less or more data
+
+
+ parsed tag successfully
+
+
+
+ Type of image contained in an image tag (specified as "image-type" field in
+the info structure in the image's #GstSample)
+
+ No image type. Can be used to
+ tell functions such as gst_tag_image_data_to_image_sample() that no
+ image type should be set.
+
+
+ Undefined/other image type
+
+
+ Cover (front)
+
+
+ Cover (back)
+
+
+ Leaflet page
+
+
+ Medium (e.g. label side of CD)
+
+
+ Lead artist/lead performer/soloist
+
+
+ Artist/performer
+
+
+ Conductor
+
+
+ Band/orchestra
+
+
+ Composer
+
+
+ Lyricist/text writer
+
+
+ Recording location
+
+
+ During recording
+
+
+ During performance
+
+
+ Movie/video screen capture
+
+
+ A fish as funny as the ID3v2 spec
+
+
+ Illustration
+
+
+ Band/artist logotype
+
+
+ Publisher/studio logotype
+
+
+
+ See http://creativecommons.org/ns for more information.
+
+ making multiple copies
+ is allowed
+
+
+ distribution, public display
+ and public performance are allowed
+
+
+ distribution of derivative
+ works is allowed
+
+
+ commercial derivatives are allowed,
+ but only non-commercial distribution is allowed
+
+
+ copyright and license notices
+ must be kept intact
+
+
+ credit must be given to
+ copyright holder and/or author
+
+
+ derivative works must be
+ licensed under the same terms or compatible terms as the original work
+
+
+ source code (the preferred
+ form for making modifications) must be provided when exercising some
+ rights granted by the license
+
+
+ derivative and combined works
+ must be licensed under specified terms, similar to those of the original
+ work
+
+
+ derivative works must be
+ licensed under specified terms, with at least the same conditions as
+ the original work; combinations with the work may be licensed under
+ different terms
+
+
+ exercising rights for
+ commercial purposes is prohibited
+
+
+ use in a
+ non-developing country is prohibited
+
+
+ this license was created
+ by the Creative Commons project
+
+
+ this license was
+ created by the Free Software Foundation (FSF)
+
+
+
+ Provides a base class for adding tags at the beginning or end of a
+stream.
+
+## Deriving from GstTagMux
+
+Subclasses have to do the following things:
+
+ * In their base init function, they must add pad templates for the sink
+ pad and the source pad to the element class, describing the media type
+ they accept and output in the caps of the pad template.
+ * In their class init function, they must override the
+ GST_TAG_MUX_CLASS(mux_klass)->render_start_tag and/or
+ GST_TAG_MUX_CLASS(mux_klass)->render_end_tag vfuncs and set up a render
+ function.
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ parent element
+
+
+
+
+
+
+
+
+
+
+
+
+ The #GstTagMuxClass structure. Subclasses need to override at least one
+of the two render vfuncs.
+
+ the parent class.
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ This interface is implemented by elements that are able to do XMP serialization. Examples for
+such elements are #jifmux and #qtmux.
+
+Applications can use this interface to configure which XMP schemas should be used when serializing
+tags into XMP. Schemas are represented by their names, a full list of the supported schemas can be
+obtained from gst_tag_xmp_list_schemas(). By default, all schemas are used.
+
+
+ Adds all available XMP schemas to the configuration. Meaning that
+all will be used.
+
+
+
+
+
+ a #GstTagXmpWriter
+
+
+
+
+
+ Adds @schema to the list schemas
+
+
+
+
+
+ a #GstTagXmpWriter
+
+
+
+ the schema to be added
+
+
+
+
+
+ Checks if @schema is going to be used
+
+ %TRUE if it is going to be used
+
+
+
+
+ a #GstTagXmpWriter
+
+
+
+ the schema to test
+
+
+
+
+
+ Removes all schemas from the list of schemas to use. Meaning that no
+XMP will be generated.
+
+
+
+
+
+ a #GstTagXmpWriter
+
+
+
+
+
+ Removes a schema from the list of schemas to use. Nothing is done if
+the schema wasn't in the list
+
+
+
+
+
+ a #GstTagXmpWriter
+
+
+
+ the schema to remove
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Check if a given string contains a known ISO 639 language code.
+
+This is useful in situations where it's not clear whether a given
+string is a language code (which should be put into a #GST_TAG_LANGUAGE_CODE
+tag) or a free-form language name descriptor (which should be put into a
+#GST_TAG_LANGUAGE_NAME tag instead).
+
+ TRUE if the two- or three-letter language code in @lang_code
+ is a valid ISO-639 language code.
+
+
+
+
+ ISO-639 language code (e.g. "deu" or "ger" or "de")
+
+
+
+
+
+ Convenience function to read a string with unknown character encoding. If
+the string is already in UTF-8 encoding, it will be returned right away.
+If not it tries to detect byte-order-mark for UTF-16/32 cases and use that.
+Otherwise, the environment will be searched for a number of environment
+variables (whose names are specified in the NULL-terminated string array
+@env_vars) containing a list of character encodings to try/use. If none
+are specified, the current locale will be tried. If that also doesn't work,
+WINDOWS-1252/ISO-8859-1 is assumed (which will almost always succeed).
+
+ a newly-allocated string in UTF-8 encoding, or NULL
+
+
+
+
+ string data
+
+
+
+
+
+ length of string data, or -1 if the string is NUL-terminated
+
+
+
+
+ a NULL-terminated string array of environment variable names, or NULL
+
+
+
+
+
+
+
+ Looks up the GStreamer tag for a ID3v2 tag.
+
+ The corresponding GStreamer tag or NULL if none exists.
+
+
+
+
+ ID3v2 tag to convert to GStreamer tag
+
+
+
+
+
+ Looks up the GStreamer tag for an ID3v2 user tag (e.g. description in
+TXXX frame or owner in UFID frame).
+
+ The corresponding GStreamer tag or NULL if none exists.
+
+
+
+
+ the type of ID3v2 user tag (e.g. "TXXX" or "UDIF")
+
+
+
+ ID3v2 user tag to convert to GStreamer tag
+
+
+
+
+
+ Looks up the GStreamer tag for a vorbiscomment tag.
+
+ The corresponding GStreamer tag or NULL if none exists.
+
+
+
+
+ vorbiscomment tag to convert to GStreamer tag
+
+
+
+
+
+ Determines size of an ID3v2 tag on buffer containing at least ID3v2 header,
+i.e. at least #GST_TAG_ID3V2_HEADER_SIZE (10) bytes;
+
+ Size of tag, or 0 if header is invalid or too small.
+
+
+
+
+ buffer holding ID3v2 tag (or at least the start of one)
+
+
+
+
+
+ Returns two-letter ISO-639-1 language code given a three-letter ISO-639-2
+language code or two-letter ISO-639-1 language code (both are accepted for
+convenience).
+
+Language codes are case-sensitive and expected to be lower case.
+
+ two-letter ISO-639-1 language code string that maps to @lang_code,
+ or NULL if no mapping is known. The returned string must not be
+ modified or freed.
+
+
+
+
+ ISO-639 language code (e.g. "deu" or "ger" or "de")
+
+
+
+
+
+ Returns three-letter ISO-639-2 "bibliographic" language code given a
+two-letter ISO-639-1 language code or a three-letter ISO-639-2 language
+code (both are accepted for convenience).
+
+The "bibliographic" code is derived from the English name of the language
+(e.g. "ger" for German instead of "de" or "deu"). In most scenarios, the
+"terminological" codes are prefered.
+
+Language codes are case-sensitive and expected to be lower case.
+
+ three-letter ISO-639-2 language code string that maps to @lang_code,
+ or NULL if no mapping is known. The returned string must not be
+ modified or freed.
+
+
+
+
+ ISO-639 language code (e.g. "deu" or "ger" or "de")
+
+
+
+
+
+ Returns three-letter ISO-639-2 "terminological" language code given a
+two-letter ISO-639-1 language code or a three-letter ISO-639-2 language
+code (both are accepted for convenience).
+
+The "terminological" code is derived from the local name of the language
+(e.g. "deu" for German instead of "ger"). In most scenarios, the
+"terminological" codes are prefered over the "bibliographic" ones.
+
+Language codes are case-sensitive and expected to be lower case.
+
+ three-letter ISO-639-2 language code string that maps to @lang_code,
+ or NULL if no mapping is known. The returned string must not be
+ modified or freed.
+
+
+
+
+ ISO-639 language code (e.g. "deu" or "ger" or "de")
+
+
+
+
+
+ Returns a list of known language codes (in form of two-letter ISO-639-1
+codes). This is useful for UIs to build a list of available languages for
+tagging purposes (e.g. to tag an audio track appropriately in a video or
+audio editor).
+
+ NULL-terminated string array with two-letter
+ language codes. Free with g_strfreev() when no longer needed.
+
+
+
+
+
+
+ Returns the name of the language given an ISO-639 language code as
+found in a GST_TAG_LANGUAGE_CODE tag. The name will be translated
+according to the current locale (if the library was built against the
+iso-codes package, otherwise the English name will be returned).
+
+Language codes are case-sensitive and expected to be lower case.
+
+ language name in UTF-8 format, or NULL if @language_code could
+ not be mapped to a language name. The returned string must not be
+ modified and does not need to freed; it will stay valid until the
+ application is terminated.
+
+
+
+
+ two or three-letter ISO-639 language code
+
+
+
+
+
+ Get the description of a license, which is a translated description
+of the license's main features.
+
+ the description of the license, or NULL if the license is unknown
+ or a description is not available.
+
+
+
+
+ a license reference string in form of a URI,
+ e.g. "http://creativecommons.org/licenses/by-nc-nd/2.0/"
+
+
+
+
+
+ Get the flags of a license, which describe most of the features of
+a license in their most general form.
+
+ the flags of the license, or 0 if the license is unknown
+
+
+
+
+ a license reference string in form of a URI,
+ e.g. "http://creativecommons.org/licenses/by-nc-nd/2.0/"
+
+
+
+
+
+ Get the jurisdiction code of a license. This is usually a two-letter
+ISO 3166-1 alpha-2 code, but there is also the special case of Scotland,
+for which no code exists and which is thus represented as "scotland".
+
+Known jurisdictions: ar, at, au, be, bg, br, ca, ch, cl, cn, co, de,
+dk, es, fi, fr, hr, hu, il, in, it, jp, kr, mk, mt, mx, my, nl, pe, pl,
+pt, scotland, se, si, tw, uk, us, za.
+
+ the jurisdiction code of the license, or NULL if the license is
+ unknown or is not specific to a particular jurisdiction.
+
+
+
+
+ a license reference string in form of a URI,
+ e.g. "http://creativecommons.org/licenses/by-nc-nd/2.0/"
+
+
+
+
+
+ Get the nick name of a license, which is a short (untranslated) string
+such as e.g. "CC BY-NC-ND 2.0 UK".
+
+ the nick name of the license, or NULL if the license is unknown
+
+
+
+
+ a license reference string in form of a URI,
+ e.g. "http://creativecommons.org/licenses/by-nc-nd/2.0/"
+
+
+
+
+
+ Get the title of a license, which is a short translated description
+of the license's features (generally not very pretty though).
+
+ the title of the license, or NULL if the license is unknown or
+ no title is available.
+
+
+
+
+ a license reference string in form of a URI,
+ e.g. "http://creativecommons.org/licenses/by-nc-nd/2.0/"
+
+
+
+
+
+ Get the version of a license.
+
+ the version of the license, or NULL if the license is not known or
+ has no version
+
+
+
+
+ a license reference string in form of a URI,
+ e.g. "http://creativecommons.org/licenses/by-nc-nd/2.0/"
+
+
+
+
+
+ Returns a list of known license references (in form of URIs). This is
+useful for UIs to build a list of available licenses for tagging purposes
+(e.g. to tag an audio track appropriately in a video or audio editor, or
+an image in a camera application).
+
+ NULL-terminated array of license strings. Free
+ with g_strfreev() when no longer needed.
+
+
+
+
+
+
+ Gets the number of ID3v1 genres that can be identified. Winamp genres are
+included.
+
+ the number of ID3v1 genres that can be identified
+
+
+
+
+ Gets the ID3v1 genre name for a given ID.
+
+ the genre or NULL if no genre is associated with that ID.
+
+
+
+
+ ID of genre to query
+
+
+
+
+
+ Helper function for tag-reading plugins to create a #GstSample suitable to
+add to a #GstTagList as an image tag (such as #GST_TAG_IMAGE or
+#GST_TAG_PREVIEW_IMAGE) from the encoded image data and an (optional) image
+type.
+
+Background: cover art and other images in tags are usually stored as a
+blob of binary image data, often accompanied by a MIME type or some other
+content type string (e.g. 'png', 'jpeg', 'jpg'). Sometimes there is also an
+'image type' to indicate what kind of image this is (e.g. front cover,
+back cover, artist, etc.). The image data may also be an URI to the image
+rather than the image itself.
+
+In GStreamer, image tags are #GstSample<!-- -->s containing the raw image
+data, with the sample caps describing the content type of the image
+(e.g. image/jpeg, image/png, text/uri-list). The sample info may contain
+an additional 'image-type' field of #GST_TYPE_TAG_IMAGE_TYPE to describe
+the type of image (front cover, back cover etc.). #GST_TAG_PREVIEW_IMAGE
+tags should not carry an image type, their type is already indicated via
+the special tag name.
+
+This function will do various checks and typefind the encoded image
+data (we can't trust the declared mime type).
+
+ a newly-allocated image sample for use in tag lists, or NULL
+
+
+
+
+ the (encoded) image
+
+
+
+
+
+ the length of the encoded image data at @image_data
+
+
+
+ type of the image, or #GST_TAG_IMAGE_TYPE_UNDEFINED. Pass
+ #GST_TAG_IMAGE_TYPE_NONE if no image type should be set at all (e.g.
+ for preview images)
+
+
+
+
+
+ Adds an image from an ID3 APIC frame (or similar, such as used in FLAC)
+to the given tag list. Also see gst_tag_image_data_to_image_sample() for
+more information on image tags in GStreamer.
+
+ %TRUE if the image was processed, otherwise %FALSE
+
+
+
+
+ a tag list
+
+
+
+ the (encoded) image
+
+
+
+
+
+ the length of the encoded image data at @image_data
+
+
+
+ picture type as per the ID3 (v2.4.0) specification for
+ the APIC frame (0 = unknown/other)
+
+
+
+
+
+ Parses the IFD and IFD tags data contained in the buffer and puts it
+on a taglist. The base_offset is used to subtract from the offset in
+the tag entries and be able to get the offset relative to the buffer
+start
+
+ The parsed taglist
+
+
+
+
+ The exif buffer
+
+
+
+ byte order of the data
+
+
+
+ Offset from the tiff header to this buffer
+
+
+
+
+
+ Parses the exif tags starting with a tiff header structure.
+
+ The taglist
+
+
+
+
+ The exif buffer
+
+
+
+
+
+ Creates a new tag list that contains the information parsed out of a
+ID3 tag.
+
+ A new #GstTagList with all tags that could be extracted from the
+ given vorbiscomment buffer or NULL on error.
+
+
+
+
+ buffer to convert
+
+
+
+
+
+ Creates a new tag list that contains the information parsed out of a
+vorbiscomment packet.
+
+ A new #GstTagList with all tags that could be extracted from the
+ given vorbiscomment buffer or NULL on error.
+
+
+
+
+ data to convert
+
+
+
+
+
+ size of @data
+
+
+
+ identification data at start of stream
+
+
+
+
+
+ length of identification data
+
+
+
+ pointer to a string that should take the
+ vendor string of this vorbis comment or NULL if you don't need it.
+
+
+
+
+
+ Creates a new tag list that contains the information parsed out of a
+vorbiscomment packet.
+
+ A new #GstTagList with all tags that could be extracted from the
+ given vorbiscomment buffer or NULL on error.
+
+
+
+
+ buffer to convert
+
+
+
+ identification data at start of stream
+
+
+
+
+
+ length of identification data
+
+
+
+ pointer to a string that should take the
+ vendor string of this vorbis comment or NULL if you don't need it.
+
+
+
+
+
+ Parse a xmp packet into a taglist.
+
+ new taglist or %NULL, free the list when done
+
+
+
+
+ buffer
+
+
+
+
+
+ Parses the data containing an ID3v1 tag and returns a #GstTagList from the
+parsed data.
+
+ A new tag list or NULL if the data was not an ID3v1 tag.
+
+
+
+
+ 128 bytes of data containing the ID3v1 tag
+
+
+
+
+
+
+
+ Formats the tags in taglist on exif format. The resulting buffer contains
+the tags IFD and is followed by the data pointed by the tag entries.
+
+ A GstBuffer containing the tag entries followed by the tag data
+
+
+
+
+ The taglist
+
+
+
+ byte order used in writing (G_LITTLE_ENDIAN or G_BIG_ENDIAN)
+
+
+
+ Offset from the tiff header first byte
+
+
+
+
+
+ Formats the tags in taglist into exif structure, a tiff header
+is put in the beginning of the buffer.
+
+ A GstBuffer containing the data
+
+
+
+
+ The taglist
+
+
+
+
+
+ Creates a new vorbiscomment buffer from a tag list.
+
+ A new #GstBuffer containing a vorbiscomment buffer with all tags
+ that could be converted from the given tag list.
+
+
+
+
+ tag list to convert
+
+
+
+ identification data at start of stream
+
+
+
+
+
+ length of identification data, may be 0 if @id_data is NULL
+
+
+
+ string that describes the vendor string or NULL
+
+
+
+
+
+ Formats a taglist as a xmp packet using only the selected
+schemas. An empty list (%NULL) means that all schemas should
+be used
+
+ new buffer or %NULL, unref the buffer when done
+
+
+
+
+ tags
+
+
+
+ does the container forbid inplace editing
+
+
+
+
+ %NULL terminated array of schemas to be used on serialization
+
+
+
+
+
+
+
+ Convenience function to parse a GST_TAG_EXTENDED_COMMENT string and
+separate it into its components.
+
+If successful, @key, @lang and/or @value will be set to newly allocated
+strings that you need to free with g_free() when done. @key and @lang
+may also be set to NULL by this function if there is no key or no language
+code in the extended comment string.
+
+ TRUE if the string could be parsed, otherwise FALSE
+
+
+
+
+ an extended comment string, see #GST_TAG_EXTENDED_COMMENT
+
+
+
+
+ return location for the comment description key, or NULL
+
+
+
+
+ return location for the comment ISO-639 language code, or NULL
+
+
+
+ return location for the actual comment string, or NULL
+
+
+
+ whether to fail if strings are not in key=value form
+
+
+
+
+
+ Registers additional musicbrainz-specific tags with the GStreamer tag
+system. Plugins and applications that use these tags should call this
+function before using them. Can be called multiple times.
+
+
+
+
+
+ Looks up the ID3v2 tag for a GStreamer tag.
+
+ The corresponding ID3v2 tag or NULL if none exists.
+
+
+
+
+ GStreamer tag to convert to vorbiscomment tag
+
+
+
+
+
+ Creates a new tag list that contains the information parsed out of a
+vorbiscomment packet.
+
+ A #GList of newly-allocated
+ key=value strings. Free with g_list_foreach (list, (GFunc) g_free, NULL)
+ plus g_list_free (list)
+
+
+
+
+
+
+ a #GstTagList
+
+
+
+ a GStreamer tag identifier, such as #GST_TAG_ARTIST
+
+
+
+
+
+ Looks up the vorbiscomment tag for a GStreamer tag.
+
+ The corresponding vorbiscomment tag or NULL if none exists.
+
+
+
+
+ GStreamer tag to convert to vorbiscomment tag
+
+
+
+
+
+ Gets the list of supported schemas in the xmp lib
+
+ a %NULL terminated array of strings with the
+ schema names
+
+
+
+
+
+
+ Convenience function using gst_tag_from_vorbis_tag(), parsing
+a vorbis comment string into the right type and adding it to the
+given taglist @list.
+
+Unknown vorbiscomment tags will be added to the tag list in form
+of a #GST_TAG_EXTENDED_COMMENT.
+
+
+
+
+
+ a #GstTagList
+
+
+
+ a vorbiscomment tag string (key in key=value), must be valid UTF-8
+
+
+
+ a vorbiscomment value string (value in key=value), must be valid UTF-8
+
+
+
+
+
+
diff --git a/girs/GstVideo-1.0.gir b/girs/GstVideo-1.0.gir
new file mode 100644
index 0000000000..7deaa9a541
--- /dev/null
+++ b/girs/GstVideo-1.0.gir
@@ -0,0 +1,13807 @@
+
+
+
+
+
+
+
+
+
+
+
+
+ A bufferpool option to enable extra padding. When a bufferpool supports this
+option, gst_buffer_pool_config_set_video_alignment() can be called.
+
+When this option is enabled on the bufferpool,
+#GST_BUFFER_POOL_OPTION_VIDEO_META should also be enabled.
+
+
+
+ An option that can be activated on a bufferpool to request gl texture upload
+meta on buffers from the pool.
+
+When this option is enabled on the bufferpool,
+@GST_BUFFER_POOL_OPTION_VIDEO_META should also be enabled.
+
+
+
+ An option that can be activated on bufferpool to request video metadata
+on buffers from the pool.
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ This interface is implemented by elements which can perform some color
+balance operation on video frames they process. For example, modifying
+the brightness, contrast, hue or saturation.
+
+Example elements are 'xvimagesink' and 'colorbalance'
+
+ Get the #GstColorBalanceType of this implementation.
+
+ A the #GstColorBalanceType.
+
+
+
+
+ The #GstColorBalance implementation
+
+
+
+
+
+ Retrieve the current value of the indicated channel, between min_value
+and max_value.
+
+See Also: The #GstColorBalanceChannel.min_value and
+ #GstColorBalanceChannel.max_value members of the
+ #GstColorBalanceChannel object.
+
+ The current value of the channel.
+
+
+
+
+ A #GstColorBalance instance
+
+
+
+ A #GstColorBalanceChannel instance
+
+
+
+
+
+ Retrieve a list of the available channels.
+
+ A
+ GList containing pointers to #GstColorBalanceChannel
+ objects. The list is owned by the #GstColorBalance
+ instance and must not be freed.
+
+
+
+
+
+
+ A #GstColorBalance instance
+
+
+
+
+
+ Sets the current value of the channel to the passed value, which must
+be between min_value and max_value.
+
+See Also: The #GstColorBalanceChannel.min_value and
+ #GstColorBalanceChannel.max_value members of the
+ #GstColorBalanceChannel object.
+
+
+
+
+
+ A #GstColorBalance instance
+
+
+
+ A #GstColorBalanceChannel instance
+
+
+
+ The new value for the channel.
+
+
+
+
+
+ A helper function called by implementations of the GstColorBalance
+interface. It fires the #GstColorBalance::value-changed signal on the
+instance, and the #GstColorBalanceChannel::value-changed signal on the
+channel object.
+
+
+
+
+
+ A #GstColorBalance instance
+
+
+
+ A #GstColorBalanceChannel whose value has changed
+
+
+
+ The new value of the channel
+
+
+
+
+
+ Get the #GstColorBalanceType of this implementation.
+
+ A the #GstColorBalanceType.
+
+
+
+
+ The #GstColorBalance implementation
+
+
+
+
+
+ Retrieve the current value of the indicated channel, between min_value
+and max_value.
+
+See Also: The #GstColorBalanceChannel.min_value and
+ #GstColorBalanceChannel.max_value members of the
+ #GstColorBalanceChannel object.
+
+ The current value of the channel.
+
+
+
+
+ A #GstColorBalance instance
+
+
+
+ A #GstColorBalanceChannel instance
+
+
+
+
+
+ Retrieve a list of the available channels.
+
+ A
+ GList containing pointers to #GstColorBalanceChannel
+ objects. The list is owned by the #GstColorBalance
+ instance and must not be freed.
+
+
+
+
+
+
+ A #GstColorBalance instance
+
+
+
+
+
+ Sets the current value of the channel to the passed value, which must
+be between min_value and max_value.
+
+See Also: The #GstColorBalanceChannel.min_value and
+ #GstColorBalanceChannel.max_value members of the
+ #GstColorBalanceChannel object.
+
+
+
+
+
+ A #GstColorBalance instance
+
+
+
+ A #GstColorBalanceChannel instance
+
+
+
+ The new value for the channel.
+
+
+
+
+
+ A helper function called by implementations of the GstColorBalance
+interface. It fires the #GstColorBalance::value-changed signal on the
+instance, and the #GstColorBalanceChannel::value-changed signal on the
+channel object.
+
+
+
+
+
+ A #GstColorBalance instance
+
+
+
+ A #GstColorBalanceChannel whose value has changed
+
+
+
+ The new value of the channel
+
+
+
+
+
+ Fired when the value of the indicated channel has changed.
+
+
+
+
+
+ The #GstColorBalanceChannel
+
+
+
+ The new value
+
+
+
+
+
+
+ The #GstColorBalanceChannel object represents a parameter
+for modifying the color balance implemented by an element providing the
+#GstColorBalance interface. For example, Hue or Saturation.
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ A string containing a descriptive name for this channel
+
+
+
+ The minimum valid value for this channel.
+
+
+
+ The maximum valid value for this channel.
+
+
+
+
+
+
+
+
+ Fired when the value of the indicated channel has changed.
+
+
+
+
+
+ The new value
+
+
+
+
+
+
+ Color-balance channel class.
+
+ the parent class
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Color-balance interface.
+
+ the parent interface
+
+
+
+
+
+ A
+ GList containing pointers to #GstColorBalanceChannel
+ objects. The list is owned by the #GstColorBalance
+ instance and must not be freed.
+
+
+
+
+
+
+ A #GstColorBalance instance
+
+
+
+
+
+
+
+
+
+
+
+
+ A #GstColorBalance instance
+
+
+
+ A #GstColorBalanceChannel instance
+
+
+
+ The new value for the channel.
+
+
+
+
+
+
+
+
+ The current value of the channel.
+
+
+
+
+ A #GstColorBalance instance
+
+
+
+ A #GstColorBalanceChannel instance
+
+
+
+
+
+
+
+
+ A the #GstColorBalanceType.
+
+
+
+
+ The #GstColorBalance implementation
+
+
+
+
+
+
+
+
+
+
+
+
+ A #GstColorBalance instance
+
+
+
+ A #GstColorBalanceChannel whose value has changed
+
+
+
+ The new value of the channel
+
+
+
+
+
+
+
+
+
+
+
+
+ An enumeration indicating whether an element implements color balancing
+operations in software or in dedicated hardware. In general, dedicated
+hardware implementations (such as those provided by xvimagesink) are
+preferred.
+
+ Color balance is implemented with dedicated
+ hardware.
+
+
+ Color balance is implemented via software
+ processing.
+
+
+
+ This metadata stays relevant as long as video colorspace is unchanged.
+
+
+
+ This metadata stays relevant as long as video orientation is unchanged.
+
+
+
+ This metadata stays relevant as long as video size is unchanged.
+
+
+
+ This metadata is relevant for video streams.
+
+
+
+ The Navigation interface is used for creating and injecting navigation related
+events such as mouse button presses, cursor motion and key presses. The associated
+library also provides methods for parsing received events, and for sending and
+receiving navigation related bus events. One main usecase is DVD menu navigation.
+
+The main parts of the API are:
+
+* The GstNavigation interface, implemented by elements which provide an application
+ with the ability to create and inject navigation events into the pipeline.
+* GstNavigation event handling API. GstNavigation events are created in response to
+ calls on a GstNavigation interface implementation, and sent in the pipeline. Upstream
+ elements can use the navigation event API functions to parse the contents of received
+ messages.
+
+* GstNavigation message handling API. GstNavigation messages may be sent on the message
+ bus to inform applications of navigation related changes in the pipeline, such as the
+ mouse moving over a clickable region, or the set of available angles changing.
+
+The GstNavigation message functions provide functions for creating and parsing
+custom bus messages for signaling GstNavigation changes.
+
+ Inspect a #GstEvent and return the #GstNavigationEventType of the event, or
+#GST_NAVIGATION_EVENT_INVALID if the event is not a #GstNavigation event.
+
+
+
+
+
+ A #GstEvent to inspect.
+
+
+
+
+
+ Inspect a #GstNavigation command event and retrieve the enum value of the
+associated command.
+
+ TRUE if the navigation command could be extracted, otherwise FALSE.
+
+
+
+
+ A #GstEvent to inspect.
+
+
+
+ Pointer to GstNavigationCommand to receive the
+ type of the navigation event.
+
+
+
+
+
+
+
+
+
+
+ A #GstEvent to inspect.
+
+
+
+ A pointer to a location to receive
+ the string identifying the key press. The returned string is owned by the
+ event, and valid only until the event is unreffed.
+
+
+
+
+
+ Retrieve the details of either a #GstNavigation mouse button press event or
+a mouse button release event. Determine which type the event is using
+gst_navigation_event_get_type() to retrieve the #GstNavigationEventType.
+
+ TRUE if the button number and both coordinates could be extracted,
+ otherwise FALSE.
+
+
+
+
+ A #GstEvent to inspect.
+
+
+
+ Pointer to a gint that will receive the button
+ number associated with the event.
+
+
+
+ Pointer to a gdouble to receive the x coordinate of the
+ mouse button event.
+
+
+
+ Pointer to a gdouble to receive the y coordinate of the
+ mouse button event.
+
+
+
+
+
+ Inspect a #GstNavigation mouse movement event and extract the coordinates
+of the event.
+
+ TRUE if both coordinates could be extracted, otherwise FALSE.
+
+
+
+
+ A #GstEvent to inspect.
+
+
+
+ Pointer to a gdouble to receive the x coordinate of the
+ mouse movement.
+
+
+
+ Pointer to a gdouble to receive the y coordinate of the
+ mouse movement.
+
+
+
+
+
+ Check a bus message to see if it is a #GstNavigation event, and return
+the #GstNavigationMessageType identifying the type of the message if so.
+
+ The type of the #GstMessage, or
+#GST_NAVIGATION_MESSAGE_INVALID if the message is not a #GstNavigation
+notification.
+
+
+
+
+ A #GstMessage to inspect.
+
+
+
+
+
+ Creates a new #GstNavigation message with type
+#GST_NAVIGATION_MESSAGE_ANGLES_CHANGED for notifying an application
+that the current angle, or current number of angles available in a
+multiangle video has changed.
+
+ The new #GstMessage.
+
+
+
+
+ A #GstObject to set as source of the new message.
+
+
+
+ The currently selected angle.
+
+
+
+ The number of viewing angles now available.
+
+
+
+
+
+ Creates a new #GstNavigation message with type
+#GST_NAVIGATION_MESSAGE_COMMANDS_CHANGED
+
+ The new #GstMessage.
+
+
+
+
+ A #GstObject to set as source of the new message.
+
+
+
+
+
+ Creates a new #GstNavigation message with type
+#GST_NAVIGATION_MESSAGE_EVENT.
+
+ The new #GstMessage.
+
+
+
+
+ A #GstObject to set as source of the new message.
+
+
+
+ A navigation #GstEvent
+
+
+
+
+
+ Creates a new #GstNavigation message with type
+#GST_NAVIGATION_MESSAGE_MOUSE_OVER.
+
+ The new #GstMessage.
+
+
+
+
+ A #GstObject to set as source of the new message.
+
+
+
+ %TRUE if the mouse has entered a clickable area of the display.
+%FALSE if it over a non-clickable area.
+
+
+
+
+
+ Parse a #GstNavigation message of type GST_NAVIGATION_MESSAGE_ANGLES_CHANGED
+and extract the @cur_angle and @n_angles parameters.
+
+ %TRUE if the message could be successfully parsed. %FALSE if not.
+
+
+
+
+ A #GstMessage to inspect.
+
+
+
+ A pointer to a #guint to receive the new
+ current angle number, or NULL
+
+
+
+ A pointer to a #guint to receive the new angle
+ count, or NULL.
+
+
+
+
+
+ Parse a #GstNavigation message of type #GST_NAVIGATION_MESSAGE_EVENT
+and extract contained #GstEvent. The caller must unref the @event when done
+with it.
+
+ %TRUE if the message could be successfully parsed. %FALSE if not.
+
+
+
+
+ A #GstMessage to inspect.
+
+
+
+ a pointer to a #GstEvent to receive
+ the contained navigation event.
+
+
+
+
+
+ Parse a #GstNavigation message of type #GST_NAVIGATION_MESSAGE_MOUSE_OVER
+and extract the active/inactive flag. If the mouse over event is marked
+active, it indicates that the mouse is over a clickable area.
+
+ %TRUE if the message could be successfully parsed. %FALSE if not.
+
+
+
+
+ A #GstMessage to inspect.
+
+
+
+ A pointer to a gboolean to receive the
+ active/inactive state, or NULL.
+
+
+
+
+
+ Inspect a #GstQuery and return the #GstNavigationQueryType associated with
+it if it is a #GstNavigation query.
+
+ The #GstNavigationQueryType of the query, or
+#GST_NAVIGATION_QUERY_INVALID
+
+
+
+
+ The query to inspect
+
+
+
+
+
+ Create a new #GstNavigation angles query. When executed, it will
+query the pipeline for the set of currently available angles, which may be
+greater than one in a multiangle video.
+
+ The new query.
+
+
+
+
+ Create a new #GstNavigation commands query. When executed, it will
+query the pipeline for the set of currently available commands.
+
+ The new query.
+
+
+
+
+ Parse the current angle number in the #GstNavigation angles @query into the
+#guint pointed to by the @cur_angle variable, and the number of available
+angles into the #guint pointed to by the @n_angles variable.
+
+ %TRUE if the query could be successfully parsed. %FALSE if not.
+
+
+
+
+ a #GstQuery
+
+
+
+ Pointer to a #guint into which to store the
+ currently selected angle value from the query, or NULL
+
+
+
+ Pointer to a #guint into which to store the
+ number of angles value from the query, or NULL
+
+
+
+
+
+ Parse the number of commands in the #GstNavigation commands @query.
+
+ %TRUE if the query could be successfully parsed. %FALSE if not.
+
+
+
+
+ a #GstQuery
+
+
+
+ the number of commands in this query.
+
+
+
+
+
+ Parse the #GstNavigation command query and retrieve the @nth command from
+it into @cmd. If the list contains less elements than @nth, @cmd will be
+set to #GST_NAVIGATION_COMMAND_INVALID.
+
+ %TRUE if the query could be successfully parsed. %FALSE if not.
+
+
+
+
+ a #GstQuery
+
+
+
+ the nth command to retrieve.
+
+
+
+ a pointer to store the nth command into.
+
+
+
+
+
+ Set the #GstNavigation angles query result field in @query.
+
+
+
+
+
+ a #GstQuery
+
+
+
+ the current viewing angle to set.
+
+
+
+ the number of viewing angles to set.
+
+
+
+
+
+ Set the #GstNavigation command query result fields in @query. The number
+of commands passed must be equal to @n_commands.
+
+
+
+
+
+ a #GstQuery
+
+
+
+ the number of commands to set.
+
+
+
+ A list of @GstNavigationCommand values, @n_cmds entries long.
+
+
+
+
+
+ Set the #GstNavigation command query result fields in @query. The number
+of commands passed must be equal to @n_commands.
+
+
+
+
+
+ a #GstQuery
+
+
+
+ the number of commands to set.
+
+
+
+ An array containing @n_cmds
+ @GstNavigationCommand values.
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Sends the indicated command to the navigation interface.
+
+
+
+
+
+ The navigation interface instance
+
+
+
+ The command to issue
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ The navigation interface instance
+
+
+
+ The type of the key event. Recognised values are "key-press" and
+"key-release"
+
+
+
+ Character representation of the key. This is typically as produced
+by XKeysymToString.
+
+
+
+
+
+ Sends a mouse event to the navigation interface. Mouse event coordinates
+are sent relative to the display space of the related output area. This is
+usually the size in pixels of the window associated with the element
+implementing the #GstNavigation interface.
+
+
+
+
+
+ The navigation interface instance
+
+
+
+ The type of mouse event, as a text string. Recognised values are
+"mouse-button-press", "mouse-button-release" and "mouse-move".
+
+
+
+ The button number of the button being pressed or released. Pass 0
+for mouse-move events.
+
+
+
+ The x coordinate of the mouse event.
+
+
+
+ The y coordinate of the mouse event.
+
+
+
+
+
+
+ A set of commands that may be issued to an element providing the
+#GstNavigation interface. The available commands can be queried via
+the gst_navigation_query_new_commands() query.
+
+For convenience in handling DVD navigation, the MENU commands are aliased as:
+ GST_NAVIGATION_COMMAND_DVD_MENU = @GST_NAVIGATION_COMMAND_MENU1
+ GST_NAVIGATION_COMMAND_DVD_TITLE_MENU = @GST_NAVIGATION_COMMAND_MENU2
+ GST_NAVIGATION_COMMAND_DVD_ROOT_MENU = @GST_NAVIGATION_COMMAND_MENU3
+ GST_NAVIGATION_COMMAND_DVD_SUBPICTURE_MENU = @GST_NAVIGATION_COMMAND_MENU4
+ GST_NAVIGATION_COMMAND_DVD_AUDIO_MENU = @GST_NAVIGATION_COMMAND_MENU5
+ GST_NAVIGATION_COMMAND_DVD_ANGLE_MENU = @GST_NAVIGATION_COMMAND_MENU6
+ GST_NAVIGATION_COMMAND_DVD_CHAPTER_MENU = @GST_NAVIGATION_COMMAND_MENU7
+
+ An invalid command entry
+
+
+ Execute navigation menu command 1. For DVD,
+this enters the DVD root menu, or exits back to the title from the menu.
+
+
+ Execute navigation menu command 2. For DVD,
+this jumps to the DVD title menu.
+
+
+ Execute navigation menu command 3. For DVD,
+this jumps into the DVD root menu.
+
+
+ Execute navigation menu command 4. For DVD,
+this jumps to the Subpicture menu.
+
+
+ Execute navigation menu command 5. For DVD,
+the jumps to the audio menu.
+
+
+ Execute navigation menu command 6. For DVD,
+this jumps to the angles menu.
+
+
+ Execute navigation menu command 7. For DVD,
+this jumps to the chapter menu.
+
+
+ Select the next button to the left in a menu,
+if such a button exists.
+
+
+ Select the next button to the right in a menu,
+if such a button exists.
+
+
+ Select the button above the current one in a
+menu, if such a button exists.
+
+
+ Select the button below the current one in a
+menu, if such a button exists.
+
+
+ Activate (click) the currently selected
+button in a menu, if such a button exists.
+
+
+ Switch to the previous angle in a
+multiangle feature.
+
+
+ Switch to the next angle in a multiangle
+feature.
+
+
+
+ Enum values for the various events that an element implementing the
+GstNavigation interface might send up the pipeline.
+
+ Returned from
+gst_navigation_event_get_type() when the passed event is not a navigation event.
+
+
+ A key press event. Use
+gst_navigation_event_parse_key_event() to extract the details from the event.
+
+
+ A key release event. Use
+gst_navigation_event_parse_key_event() to extract the details from the event.
+
+
+ A mouse button press event. Use
+gst_navigation_event_parse_mouse_button_event() to extract the details from the
+event.
+
+
+ A mouse button release event. Use
+gst_navigation_event_parse_mouse_button_event() to extract the details from the
+event.
+
+
+ A mouse movement event. Use
+gst_navigation_event_parse_mouse_move_event() to extract the details from the
+event.
+
+
+ A navigation command event. Use
+gst_navigation_event_parse_command() to extract the details from the event.
+
+
+
+ Navigation interface.
+
+ the parent interface
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ A set of notifications that may be received on the bus when navigation
+related status changes.
+
+ Returned from
+gst_navigation_message_get_type() when the passed message is not a
+navigation message.
+
+
+ Sent when the mouse moves over or leaves a
+clickable region of the output, such as a DVD menu button.
+
+
+ Sent when the set of available commands
+changes and should re-queried by interested applications.
+
+
+ Sent when display angles in a multi-angle
+feature (such as a multiangle DVD) change - either angles have appeared or
+disappeared.
+
+
+ Sent when a navigation event was not handled
+by any element in the pipeline (Since 1.6)
+
+
+
+ Tyoes of navigation interface queries.
+
+ invalid query
+
+
+ command query
+
+
+ viewing angle query
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ #GST_TYPE_VIDEO_ALPHA_MODE, the alpha mode to use.
+Default is #GST_VIDEO_ALPHA_MODE_COPY.
+
+
+
+ #G_TYPE_DOUBLE, the alpha color value to use.
+Default to 1.0
+
+
+
+ #G_TYPE_UINT, the border color to use if #GST_VIDEO_CONVERTER_OPT_FILL_BORDER
+is set to %TRUE. The color is in ARGB format.
+Default 0xff000000
+
+
+
+ #GST_TYPE_VIDEO_CHROMA_MODE, set the chroma resample mode subsampled
+formats. Default is #GST_VIDEO_CHROMA_MODE_FULL.
+
+
+
+ #GST_TYPE_RESAMPLER_METHOD, The resampler method to use for
+chroma resampling. Other options for the resampler can be used, see
+the #GstResampler. Default is #GST_RESAMPLER_METHOD_LINEAR
+
+
+
+ #G_TYPE_INT, height in the destination frame, default destination height
+
+
+
+ #G_TYPE_INT, width in the destination frame, default destination width
+
+
+
+ #G_TYPE_INT, x position in the destination frame, default 0
+
+
+
+ #G_TYPE_INT, y position in the destination frame, default 0
+
+
+
+ #GST_TYPE_VIDEO_DITHER_METHOD, The dither method to use when
+changing bit depth.
+Default is #GST_VIDEO_DITHER_BAYER.
+
+
+
+ #G_TYPE_UINT, The quantization amount to dither to. Components will be
+quantized to multiples of this value.
+Default is 1
+
+
+
+ #G_TYPE_BOOLEAN, if the destination rectangle does not fill the complete
+destination image, render a border with
+#GST_VIDEO_CONVERTER_OPT_BORDER_ARGB. Otherwise the unusded pixels in the
+destination are untouched. Default %TRUE.
+
+
+
+ #GST_TYPE_VIDEO_GAMMA_MODE, set the gamma mode.
+Default is #GST_VIDEO_GAMMA_MODE_NONE.
+
+
+
+ #GST_TYPE_VIDEO_MATRIX_MODE, set the color matrix conversion mode for
+converting between Y'PbPr and non-linear RGB (R'G'B').
+Default is #GST_VIDEO_MATRIX_MODE_FULL.
+
+
+
+ #GST_TYPE_VIDEO_PRIMARIES_MODE, set the primaries conversion mode.
+Default is #GST_VIDEO_PRIMARIES_MODE_NONE.
+
+
+
+ #GST_TYPE_RESAMPLER_METHOD, The resampler method to use for
+resampling. Other options for the resampler can be used, see
+the #GstResampler. Default is #GST_RESAMPLER_METHOD_CUBIC
+
+
+
+ #G_TYPE_UINT, The number of taps for the resampler.
+Default is 0: let the resampler choose a good value.
+
+
+
+ #G_TYPE_INT, source height to convert, default source height
+
+
+
+ #G_TYPE_INT, source width to convert, default source width
+
+
+
+ #G_TYPE_INT, source x position to start conversion, default 0
+
+
+
+ #G_TYPE_INT, source y position to start conversion, default 0
+
+
+
+ #G_TYPE_UINT, maximum number of threads to use. Default 1, 0 for the number
+of cores.
+
+
+
+ Default maximum number of errors tolerated before signaling error.
+
+
+
+ The name of the templates for the sink pad.
+
+
+
+ The name of the templates for the source pad.
+
+
+
+ The name of the templates for the sink pad.
+
+
+
+ The name of the templates for the source pad.
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Video formats supported by gst_video_overlay_composition_blend(), for
+use in overlay elements' pad template caps.
+
+
+
+ G_TYPE_DOUBLE, B parameter of the cubic filter. The B
+parameter controls the bluriness. Values between 0.0 and
+2.0 are accepted. 1/3 is the default.
+
+Below are some values of popular filters:
+ B C
+Hermite 0.0 0.0
+Spline 1.0 0.0
+Catmull-Rom 0.0 1/2
+Mitchell 1/3 1/3
+Robidoux 0.3782 0.3109
+Robidoux
+ Sharp 0.2620 0.3690
+Robidoux
+ Soft 0.6796 0.1602
+
+
+
+ G_TYPE_DOUBLE, C parameter of the cubic filter. The C
+parameter controls the Keys alpha value. Values between 0.0 and
+2.0 are accepted. 1/3 is the default.
+
+See #GST_VIDEO_RESAMPLER_OPT_CUBIC_B for some more common values
+
+
+
+ G_TYPE_DOUBLE, specifies the size of filter envelope for
+@GST_VIDEO_RESAMPLER_METHOD_LANCZOS. values are clamped between
+1.0 and 5.0. 2.0 is the default.
+
+
+
+ G_TYPE_INT, limits the maximum number of taps to use.
+16 is the default.
+
+
+
+ G_TYPE_DOUBLE, specifies sharpening of the filter for
+@GST_VIDEO_RESAMPLER_METHOD_LANCZOS. values are clamped between
+0.0 and 1.0. 0.0 is the default.
+
+
+
+ G_TYPE_DOUBLE, specifies sharpness of the filter for
+@GST_VIDEO_RESAMPLER_METHOD_LANCZOS. values are clamped between
+0.5 and 1.5. 1.0 is the default.
+
+
+
+ #GST_TYPE_VIDEO_DITHER_METHOD, The dither method to use for propagating
+quatization errors.
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Extra buffer metadata for performing an affine transformation using a 4x4
+matrix. The transformation matrix can be composed with
+gst_video_affine_transformation_meta_apply_matrix().
+
+The vertices operated on are all in the range 0 to 1, not in
+Normalized Device Coordinates (-1 to +1). Transforming points in this space
+are assumed to have an origin at (0.5, 0.5, 0.5) in a left-handed coordinate
+system with the x-axis moving horizontally (positive values to the right),
+the y-axis moving vertically (positive values up the screen) and the z-axis
+perpendicular to the screen (positive values into the screen).
+
+ parent #GstMeta
+
+
+
+ the column-major 4x4 transformation matrix
+
+
+
+
+
+ Apply a transformation using the given 4x4 transformation matrix.
+Performs the multiplication, meta->matrix X matrix.
+
+
+
+
+
+ a #GstVideoAffineTransformationMeta
+
+
+
+ a 4x4 transformation matrix to be applied
+
+
+
+
+
+
+
+
+
+
+
+ Extra alignment parameters for the memory of video buffers. This
+structure is usually used to configure the bufferpool if it supports the
+#GST_BUFFER_POOL_OPTION_VIDEO_ALIGNMENT.
+
+ extra pixels on the top
+
+
+
+ extra pixels on the bottom
+
+
+
+ extra pixels on the left side
+
+
+
+ extra pixels on the right side
+
+
+
+ array with extra alignment requirements for the strides
+
+
+
+
+
+ Set @align to its default values with no padding and no alignment.
+
+
+
+
+
+ a #GstVideoAlignment
+
+
+
+
+
+
+ Different alpha modes.
+
+ When input and output have alpha, it will be copied.
+ When the input has no alpha, alpha will be set to
+ #GST_VIDEO_CONVERTER_OPT_ALPHA_VALUE
+
+
+ set all alpha to
+ #GST_VIDEO_CONVERTER_OPT_ALPHA_VALUE
+
+
+ multiply all alpha with
+ #GST_VIDEO_CONVERTER_OPT_ALPHA_VALUE.
+ When the input format has no alpha but the output format has, the
+ alpha value will be set to #GST_VIDEO_CONVERTER_OPT_ALPHA_VALUE
+
+
+
+ Additional video buffer flags. These flags can potentially be used on any
+buffers carrying video data - even encoded data.
+
+Note that these are only valid for #GstCaps of type: video/...
+They can conflict with other extended buffer flags.
+
+ If the #GstBuffer is interlaced. In mixed
+ interlace-mode, this flags specifies if the frame is
+ interlaced or progressive.
+
+
+ If the #GstBuffer is interlaced, then the first field
+ in the video frame is the top field. If unset, the
+ bottom field is first.
+
+
+ If the #GstBuffer is interlaced, then the first field
+ (as defined by the %GST_VIDEO_BUFFER_TFF flag setting)
+ is repeated.
+
+
+ If the #GstBuffer is interlaced, then only the
+ first field (as defined by the %GST_VIDEO_BUFFER_TFF
+ flag setting) is to be displayed.
+
+
+ The #GstBuffer contains one or more specific views,
+ such as left or right eye view. This flags is set on
+ any buffer that contains non-mono content - even for
+ streams that contain only a single viewpoint. In mixed
+ mono / non-mono streams, the absense of the flag marks
+ mono buffers.
+
+
+ When conveying stereo/multiview content with
+ frame-by-frame methods, this flag marks the first buffer
+ in a bundle of frames that belong together.
+
+
+ Offset to define more flags
+
+
+
+
+ Create a new bufferpool that can allocate video frames. This bufferpool
+supports all the video bufferpool options.
+
+ a new #GstBufferPool to allocate video frames
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Extra flags that influence the result from gst_video_chroma_resample_new().
+
+ no flags
+
+
+ the input is interlaced
+
+
+
+ Different subsampling and upsampling methods
+
+ Duplicates the chroma samples when
+ upsampling and drops when subsampling
+
+
+ Uses linear interpolation to reconstruct
+ missing chroma and averaging to subsample
+
+
+
+ Different chroma downsampling and upsampling modes
+
+ do full chroma up and down sampling
+
+
+ only perform chroma upsampling
+
+
+ only perform chroma downsampling
+
+
+ disable chroma resampling
+
+
+
+
+ Perform resampling of @width chroma pixels in @lines.
+
+
+
+
+
+ a #GstVideoChromaResample
+
+
+
+ pixel lines
+
+
+
+ the number of pixels on one line
+
+
+
+
+
+ Free @resample
+
+
+
+
+
+ a #GstVideoChromaResample
+
+
+
+
+
+ The resampler must be fed @n_lines at a time. The first line should be
+at @offset.
+
+
+
+
+
+ a #GstVideoChromaResample
+
+
+
+ the number of input lines
+
+
+
+ the first line
+
+
+
+
+
+ Create a new resampler object for the given parameters. When @h_factor or
+@v_factor is > 0, upsampling will be used, otherwise subsampling is
+performed.
+
+ a new #GstVideoChromaResample that should be freed with
+ gst_video_chroma_resample_free() after usage.
+
+
+
+
+ a #GstVideoChromaMethod
+
+
+
+ a #GstVideoChromaSite
+
+
+
+ #GstVideoChromaFlags
+
+
+
+ the #GstVideoFormat
+
+
+
+ horizontal resampling factor
+
+
+
+ vertical resampling factor
+
+
+
+
+
+
+ Various Chroma sitings.
+
+ unknown cositing
+
+
+ no cositing
+
+
+ chroma is horizontally cosited
+
+
+ chroma is vertically cosited
+
+
+ choma samples are sited on alternate lines
+
+
+ chroma samples cosited with luma samples
+
+
+ jpeg style cositing, also for mpeg1 and mjpeg
+
+
+ mpeg2 style cositing
+
+
+ DV style cositing
+
+
+
+ A #GstVideoCodecFrame represents a video frame both in raw and
+encoded form.
+
+
+
+
+
+
+
+ Unique identifier for the frame. Use this if you need
+ to get hold of the frame later (like when data is being decoded).
+ Typical usage in decoders is to set this on the opaque value provided
+ to the library and get back the frame using gst_video_decoder_get_frame()
+
+
+
+
+
+
+
+
+
+ Decoding timestamp
+
+
+
+ Presentation timestamp
+
+
+
+ Duration of the frame
+
+
+
+ Distance in frames from the last synchronization point.
+
+
+
+ the input #GstBuffer that created this frame. The buffer is owned
+ by the frame and references to the frame instead of the buffer should
+ be kept.
+
+
+
+ the output #GstBuffer. Implementations should set this either
+ directly, or by using the
+ @gst_video_decoder_allocate_output_frame() or
+ @gst_video_decoder_allocate_output_buffer() methods. The buffer is
+ owned by the frame and references to the frame instead of the
+ buffer should be kept.
+
+
+
+ Running time when the frame will be used.
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Gets private data set on the frame by the subclass via
+gst_video_codec_frame_set_user_data() previously.
+
+ The previously set user_data
+
+
+
+
+ a #GstVideoCodecFrame
+
+
+
+
+
+ Increases the refcount of the given frame by one.
+
+ @buf
+
+
+
+
+ a #GstVideoCodecFrame
+
+
+
+
+
+ Sets @user_data on the frame and the #GDestroyNotify that will be called when
+the frame is freed. Allows to attach private data by the subclass to frames.
+
+If a @user_data was previously set, then the previous set @notify will be called
+before the @user_data is replaced.
+
+
+
+
+
+ a #GstVideoCodecFrame
+
+
+
+ private data
+
+
+
+ a #GDestroyNotify
+
+
+
+
+
+ Decreases the refcount of the frame. If the refcount reaches 0, the frame
+will be freed.
+
+
+
+
+
+ a #GstVideoCodecFrame
+
+
+
+
+
+
+ Flags for #GstVideoCodecFrame
+
+ is the frame only meant to be decoded
+
+
+ is the frame a synchronization point (keyframe)
+
+
+ should the output frame be made a keyframe
+
+
+ should the encoder output stream headers
+
+
+
+ Structure representing the state of an incoming or outgoing video
+stream for encoders and decoders.
+
+Decoders and encoders will receive such a state through their
+respective @set_format vmethods.
+
+Decoders and encoders can set the downstream state, by using the
+@gst_video_decoder_set_output_state() or
+@gst_video_encoder_set_output_state() methods.
+
+
+
+
+ The #GstVideoInfo describing the stream
+
+
+
+ The #GstCaps used in the caps negotiation of the pad.
+
+
+
+ a #GstBuffer corresponding to the
+ 'codec_data' field of a stream, or NULL.
+
+
+
+ The #GstCaps for allocation query and pool
+ negotiation. Since: 1.10
+
+
+
+
+
+
+
+
+ Increases the refcount of the given state by one.
+
+ @buf
+
+
+
+
+ a #GstVideoCodecState
+
+
+
+
+
+ Decreases the refcount of the state. If the refcount reaches 0, the state
+will be freed.
+
+
+
+
+
+ a #GstVideoCodecState
+
+
+
+
+
+
+ The color matrix is used to convert between Y'PbPr and
+non-linear RGB (R'G'B')
+
+ unknown matrix
+
+
+ identity matrix
+
+
+ FCC color matrix
+
+
+ ITU-R BT.709 color matrix
+
+
+ ITU-R BT.601 color matrix
+
+
+ SMPTE 240M color matrix
+
+
+ ITU-R BT.2020 color matrix. Since: 1.6
+
+
+ Get the coefficients used to convert between Y'PbPr and R'G'B' using @matrix.
+
+When:
+
+|[
+ 0.0 <= [Y',R',G',B'] <= 1.0)
+ (-0.5 <= [Pb,Pr] <= 0.5)
+]|
+
+the general conversion is given by:
+
+|[
+ Y' = Kr*R' + (1-Kr-Kb)*G' + Kb*B'
+ Pb = (B'-Y')/(2*(1-Kb))
+ Pr = (R'-Y')/(2*(1-Kr))
+]|
+
+and the other way around:
+
+|[
+ R' = Y' + Cr*2*(1-Kr)
+ G' = Y' - Cb*2*(1-Kb)*Kb/(1-Kr-Kb) - Cr*2*(1-Kr)*Kr/(1-Kr-Kb)
+ B' = Y' + Cb*2*(1-Kb)
+]|
+
+ TRUE if @matrix was a YUV color format and @Kr and @Kb contain valid
+ values.
+
+
+
+
+ a #GstVideoColorMatrix
+
+
+
+ result red channel coefficient
+
+
+
+ result blue channel coefficient
+
+
+
+
+
+
+ The color primaries define the how to transform linear RGB values to and from
+the CIE XYZ colorspace.
+
+ unknown color primaries
+
+
+ BT709 primaries
+
+
+ BT470M primaries
+
+
+ BT470BG primaries
+
+
+ SMPTE170M primaries
+
+
+ SMPTE240M primaries
+
+
+ Generic film
+
+
+ BT2020 primaries. Since: 1.6
+
+
+ Adobe RGB primaries. Since: 1.8
+
+
+ Get information about the chromaticity coordinates of @primaries.
+
+ a #GstVideoColorPrimariesInfo for @primaries.
+
+
+
+
+ a #GstVideoColorPrimaries
+
+
+
+
+
+
+ Structure describing the chromaticity coordinates of an RGB system. These
+values can be used to construct a matrix to transform RGB to and from the
+XYZ colorspace.
+
+ a #GstVideoColorPrimaries
+
+
+
+ reference white x coordinate
+
+
+
+ reference white y coordinate
+
+
+
+ red x coordinate
+
+
+
+ red y coordinate
+
+
+
+ green x coordinate
+
+
+
+ green y coordinate
+
+
+
+ blue x coordinate
+
+
+
+ blue y coordinate
+
+
+
+
+ Possible color range values. These constants are defined for 8 bit color
+values and can be scaled for other bit depths.
+
+ unknown range
+
+
+ [0..255] for 8 bit components
+
+
+ [16..235] for 8 bit components. Chroma has
+ [16..240] range.
+
+
+ Compute the offset and scale values for each component of @info. For each
+component, (c[i] - offset[i]) / scale[i] will scale the component c[i] to the
+range [0.0 .. 1.0].
+
+The reverse operation (c[i] * scale[i]) + offset[i] can be used to convert
+the component values in range [0.0 .. 1.0] back to their representation in
+@info and @range.
+
+
+
+
+
+ a #GstVideoColorRange
+
+
+
+ a #GstVideoFormatInfo
+
+
+
+ output offsets
+
+
+
+
+
+ output scale
+
+
+
+
+
+
+
+
+ Structure describing the color info.
+
+ the color range. This is the valid range for the samples.
+ It is used to convert the samples to Y'PbPr values.
+
+
+
+ the color matrix. Used to convert between Y'PbPr and
+ non-linear RGB (R'G'B')
+
+
+
+ the transfer function. used to convert between R'G'B' and RGB
+
+
+
+ color primaries. used to convert between R'G'B' and CIE XYZ
+
+
+
+ Parse the colorimetry string and update @cinfo with the parsed
+values.
+
+ %TRUE if @color points to valid colorimetry info.
+
+
+
+
+ a #GstVideoColorimetry
+
+
+
+ a colorimetry string
+
+
+
+
+
+ Compare the 2 colorimetry sets for equality
+
+ %TRUE if @cinfo and @other are equal.
+
+
+
+
+ a #GstVideoColorimetry
+
+
+
+ another #GstVideoColorimetry
+
+
+
+
+
+ Check if the colorimetry information in @info matches that of the
+string @color.
+
+ %TRUE if @color conveys the same colorimetry info as the color
+information in @info.
+
+
+
+
+ a #GstVideoInfo
+
+
+
+ a colorimetry string
+
+
+
+
+
+ Make a string representation of @cinfo.
+
+ a string representation of @cinfo.
+
+
+
+
+ a #GstVideoColorimetry
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Convert the pixels of @src into @dest using @convert.
+
+
+
+
+
+ a #GstVideoConverter
+
+
+
+ a #GstVideoFrame
+
+
+
+ a #GstVideoFrame
+
+
+
+
+
+ Free @convert
+
+
+
+
+
+ a #GstVideoConverter
+
+
+
+
+
+ Get the current configuration of @convert.
+
+ a #GstStructure that remains valid for as long as @convert is valid
+ or until gst_video_converter_set_config() is called.
+
+
+
+
+ a #GstVideoConverter
+
+
+
+
+
+ Set @config as extra configuraion for @convert.
+
+If the parameters in @config can not be set exactly, this function returns
+%FALSE and will try to update as much state as possible. The new state can
+then be retrieved and refined with gst_video_converter_get_config().
+
+Look at the #GST_VIDEO_CONVERTER_OPT_* fields to check valid configuration
+option and values.
+
+ %TRUE when @config could be set.
+
+
+
+
+ a #GstVideoConverter
+
+
+
+ a #GstStructure
+
+
+
+
+
+ Create a new converter object to convert between @in_info and @out_info
+with @config.
+
+ a #GstVideoConverter or %NULL if conversion is not possible.
+
+
+
+
+ a #GstVideoInfo
+
+
+
+ a #GstVideoInfo
+
+
+
+ a #GstStructure with configuration options
+
+
+
+
+
+
+ Extra buffer metadata describing image cropping.
+
+ parent #GstMeta
+
+
+
+ the horizontal offset
+
+
+
+ the vertical offset
+
+
+
+ the cropped width
+
+
+
+ the cropped height
+
+
+
+
+
+
+
+
+
+ This base class is for video decoders turning encoded data into raw video
+frames.
+
+The GstVideoDecoder base class and derived subclasses should cooperate as
+follows:
+
+## Configuration
+
+ * Initially, GstVideoDecoder calls @start when the decoder element
+ is activated, which allows the subclass to perform any global setup.
+
+ * GstVideoDecoder calls @set_format to inform the subclass of caps
+ describing input video data that it is about to receive, including
+ possibly configuration data.
+ While unlikely, it might be called more than once, if changing input
+ parameters require reconfiguration.
+
+ * Incoming data buffers are processed as needed, described in Data
+ Processing below.
+
+ * GstVideoDecoder calls @stop at end of all processing.
+
+## Data processing
+
+ * The base class gathers input data, and optionally allows subclass
+ to parse this into subsequently manageable chunks, typically
+ corresponding to and referred to as 'frames'.
+
+ * Each input frame is provided in turn to the subclass' @handle_frame
+ callback.
+ The ownership of the frame is given to the @handle_frame callback.
+
+ * If codec processing results in decoded data, the subclass should call
+ @gst_video_decoder_finish_frame to have decoded data pushed.
+ downstream. Otherwise, the subclass must call
+ @gst_video_decoder_drop_frame, to allow the base class to do timestamp
+ and offset tracking, and possibly to requeue the frame for a later
+ attempt in the case of reverse playback.
+
+## Shutdown phase
+
+ * The GstVideoDecoder class calls @stop to inform the subclass that data
+ parsing will be stopped.
+
+## Additional Notes
+
+ * Seeking/Flushing
+
+ * When the pipeline is seeked or otherwise flushed, the subclass is
+ informed via a call to its @reset callback, with the hard parameter
+ set to true. This indicates the subclass should drop any internal data
+ queues and timestamps and prepare for a fresh set of buffers to arrive
+ for parsing and decoding.
+
+ * End Of Stream
+
+ * At end-of-stream, the subclass @parse function may be called some final
+ times with the at_eos parameter set to true, indicating that the element
+ should not expect any more data to be arriving, and it should parse and
+ remaining frames and call gst_video_decoder_have_frame() if possible.
+
+The subclass is responsible for providing pad template caps for
+source and sink pads. The pads need to be named "sink" and "src". It also
+needs to provide information about the ouptput caps, when they are known.
+This may be when the base class calls the subclass' @set_format function,
+though it might be during decoding, before calling
+@gst_video_decoder_finish_frame. This is done via
+@gst_video_decoder_set_output_state
+
+The subclass is also responsible for providing (presentation) timestamps
+(likely based on corresponding input ones). If that is not applicable
+or possible, the base class provides limited framerate based interpolation.
+
+Similarly, the base class provides some limited (legacy) seeking support
+if specifically requested by the subclass, as full-fledged support
+should rather be left to upstream demuxer, parser or alike. This simple
+approach caters for seeking and duration reporting using estimated input
+bitrates. To enable it, a subclass should call
+@gst_video_decoder_set_estimate_rate to enable handling of incoming
+byte-streams.
+
+The base class provides some support for reverse playback, in particular
+in case incoming data is not packetized or upstream does not provide
+fragments on keyframe boundaries. However, the subclass should then be
+prepared for the parsing and frame processing stage to occur separately
+(in normal forward processing, the latter immediately follows the former),
+The subclass also needs to ensure the parsing stage properly marks
+keyframes, unless it knows the upstream elements will do so properly for
+incoming data.
+
+The bare minimum that a functional subclass needs to implement is:
+
+ * Provide pad templates
+ * Inform the base class of output caps via
+ @gst_video_decoder_set_output_state
+
+ * Parse input data, if it is not considered packetized from upstream
+ Data will be provided to @parse which should invoke
+ @gst_video_decoder_add_to_frame and @gst_video_decoder_have_frame to
+ separate the data belonging to each video frame.
+
+ * Accept data in @handle_frame and provide decoded results to
+ @gst_video_decoder_finish_frame, or call @gst_video_decoder_drop_frame.
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Negotiate with downstream elements to currently configured #GstVideoCodecState.
+Unmark GST_PAD_FLAG_NEED_RECONFIGURE in any case. But mark it again if
+negotiate fails.
+
+ %TRUE if the negotiation succeeded, else %FALSE.
+
+
+
+
+ a #GstVideoDecoder
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Removes next @n_bytes of input data and adds it to currently parsed frame.
+
+
+
+
+
+ a #GstVideoDecoder
+
+
+
+ the number of bytes to add
+
+
+
+
+
+ Helper function that allocates a buffer to hold a video frame for @decoder's
+current #GstVideoCodecState.
+
+You should use gst_video_decoder_allocate_output_frame() instead of this
+function, if possible at all.
+
+ allocated buffer, or NULL if no buffer could be
+ allocated (e.g. when downstream is flushing or shutting down)
+
+
+
+
+ a #GstVideoDecoder
+
+
+
+
+
+ Helper function that allocates a buffer to hold a video frame for @decoder's
+current #GstVideoCodecState. Subclass should already have configured video
+state and set src pad caps.
+
+The buffer allocated here is owned by the frame and you should only
+keep references to the frame, not the buffer.
+
+ %GST_FLOW_OK if an output buffer could be allocated
+
+
+
+
+ a #GstVideoDecoder
+
+
+
+ a #GstVideoCodecFrame
+
+
+
+
+
+ Same as #gst_video_decoder_allocate_output_frame except it allows passing
+#GstBufferPoolAcquireParams to the sub call gst_buffer_pool_acquire_buffer.
+
+ %GST_FLOW_OK if an output buffer could be allocated
+
+
+
+
+ a #GstVideoDecoder
+
+
+
+ a #GstVideoCodecFrame
+
+
+
+ a #GstBufferPoolAcquireParams
+
+
+
+
+
+ Similar to gst_video_decoder_finish_frame(), but drops @frame in any
+case and posts a QoS message with the frame's details on the bus.
+In any case, the frame is considered finished and released.
+
+ a #GstFlowReturn, usually GST_FLOW_OK.
+
+
+
+
+ a #GstVideoDecoder
+
+
+
+ the #GstVideoCodecFrame to drop
+
+
+
+
+
+ @frame should have a valid decoded data buffer, whose metadata fields
+are then appropriately set according to frame data and pushed downstream.
+If no output data is provided, @frame is considered skipped.
+In any case, the frame is considered finished and released.
+
+After calling this function the output buffer of the frame is to be
+considered read-only. This function will also change the metadata
+of the buffer.
+
+ a #GstFlowReturn resulting from sending data downstream
+
+
+
+
+ a #GstVideoDecoder
+
+
+
+ a decoded #GstVideoCodecFrame
+
+
+
+
+
+ Lets #GstVideoDecoder sub-classes to know the memory @allocator
+used by the base class and its @params.
+
+Unref the @allocator after use it.
+
+
+
+
+
+ a #GstVideoDecoder
+
+
+
+ the #GstAllocator
+used
+
+
+
+ the
+#GstAllocatorParams of @allocator
+
+
+
+
+
+
+ the instance of the #GstBufferPool used
+by the decoder; free it after use it
+
+
+
+
+ a #GstVideoDecoder
+
+
+
+
+
+
+ currently configured byte to time conversion setting
+
+
+
+
+ a #GstVideoDecoder
+
+
+
+
+
+ Get a pending unfinished #GstVideoCodecFrame
+
+ pending unfinished #GstVideoCodecFrame identified by @frame_number.
+
+
+
+
+ a #GstVideoDecoder
+
+
+
+ system_frame_number of a frame
+
+
+
+
+
+ Get all pending unfinished #GstVideoCodecFrame
+
+ pending unfinished #GstVideoCodecFrame.
+
+
+
+
+
+
+ a #GstVideoDecoder
+
+
+
+
+
+ Query the configured decoder latency. Results will be returned via
+@min_latency and @max_latency.
+
+
+
+
+
+ a #GstVideoDecoder
+
+
+
+ address of variable in which to store the
+ configured minimum latency, or %NULL
+
+
+
+ address of variable in which to store the
+ configured mximum latency, or %NULL
+
+
+
+
+
+ Determines maximum possible decoding time for @frame that will
+allow it to decode and arrive in time (as determined by QoS events).
+In particular, a negative result means decoding in time is no longer possible
+and should therefore occur as soon/skippy as possible.
+
+ max decoding time.
+
+
+
+
+ a #GstVideoDecoder
+
+
+
+ a #GstVideoCodecFrame
+
+
+
+
+
+
+ currently configured decoder tolerated error count.
+
+
+
+
+ a #GstVideoDecoder
+
+
+
+
+
+ Queries decoder required format handling.
+
+ %TRUE if required format handling is enabled.
+
+
+
+
+ a #GstVideoDecoder
+
+
+
+
+
+ Get the oldest pending unfinished #GstVideoCodecFrame
+
+ oldest pending unfinished #GstVideoCodecFrame.
+
+
+
+
+ a #GstVideoDecoder
+
+
+
+
+
+ Get the #GstVideoCodecState currently describing the output stream.
+
+ #GstVideoCodecState describing format of video data.
+
+
+
+
+ a #GstVideoDecoder
+
+
+
+
+
+ Queries whether input data is considered packetized or not by the
+base class.
+
+ TRUE if input data is considered packetized.
+
+
+
+
+ a #GstVideoDecoder
+
+
+
+
+
+ Returns the number of bytes previously added to the current frame
+by calling gst_video_decoder_add_to_frame().
+
+ The number of bytes pending for the current frame
+
+
+
+
+ a #GstVideoDecoder
+
+
+
+
+
+
+ The current QoS proportion.
+
+
+
+
+ a #GstVideoDecoder
+ current QoS proportion, or %NULL
+
+
+
+
+
+ Gathers all data collected for currently parsed frame, gathers corresponding
+metadata and passes it along for further processing, i.e. @handle_frame.
+
+ a #GstFlowReturn
+
+
+
+
+ a #GstVideoDecoder
+
+
+
+
+
+ Sets the audio decoder tags and how they should be merged with any
+upstream stream tags. This will override any tags previously-set
+with gst_audio_decoder_merge_tags().
+
+Note that this is provided for convenience, and the subclass is
+not required to use this and can still do tag handling on its own.
+
+MT safe.
+
+
+
+
+
+ a #GstVideoDecoder
+
+
+
+ a #GstTagList to merge, or NULL to unset
+ previously-set tags
+
+
+
+ the #GstTagMergeMode to use, usually #GST_TAG_MERGE_REPLACE
+
+
+
+
+
+ Negotiate with downstream elements to currently configured #GstVideoCodecState.
+Unmark GST_PAD_FLAG_NEED_RECONFIGURE in any case. But mark it again if
+negotiate fails.
+
+ %TRUE if the negotiation succeeded, else %FALSE.
+
+
+
+
+ a #GstVideoDecoder
+
+
+
+
+
+ Returns caps that express @caps (or sink template caps if @caps == NULL)
+restricted to resolution/format/... combinations supported by downstream
+elements.
+
+ a #GstCaps owned by caller
+
+
+
+
+ a #GstVideoDecoder
+
+
+
+ initial caps
+
+
+
+ filter caps
+
+
+
+
+
+ Similar to gst_video_decoder_drop_frame(), but simply releases @frame
+without any processing other than removing it from list of pending frames,
+after which it is considered finished and released.
+
+
+
+
+
+ a #GstVideoDecoder
+
+
+
+ the #GstVideoCodecFrame to release
+
+
+
+
+
+ Allows baseclass to perform byte to time estimated conversion.
+
+
+
+
+
+ a #GstVideoDecoder
+
+
+
+ whether to enable byte to time conversion
+
+
+
+
+
+ Lets #GstVideoDecoder sub-classes tell the baseclass what the decoder
+latency is. Will also post a LATENCY message on the bus so the pipeline
+can reconfigure its global latency.
+
+
+
+
+
+ a #GstVideoDecoder
+
+
+
+ minimum latency
+
+
+
+ maximum latency
+
+
+
+
+
+ Sets numbers of tolerated decoder errors, where a tolerated one is then only
+warned about, but more than tolerated will lead to fatal error. You can set
+-1 for never returning fatal errors. Default is set to
+GST_VIDEO_DECODER_MAX_ERRORS.
+
+The '-1' option was added in 1.4
+
+
+
+
+
+ a #GstVideoDecoder
+
+
+
+ max tolerated errors
+
+
+
+
+
+ Configures decoder format needs. If enabled, subclass needs to be
+negotiated with format caps before it can process any data. It will then
+never be handed any data before it has been configured.
+Otherwise, it might be handed data without having been configured and
+is then expected being able to do so either by default
+or based on the input data.
+
+
+
+
+
+ a #GstVideoDecoder
+
+
+
+ new state
+
+
+
+
+
+ Creates a new #GstVideoCodecState with the specified @fmt, @width and @height
+as the output state for the decoder.
+Any previously set output state on @decoder will be replaced by the newly
+created one.
+
+If the subclass wishes to copy over existing fields (like pixel aspec ratio,
+or framerate) from an existing #GstVideoCodecState, it can be provided as a
+@reference.
+
+If the subclass wishes to override some fields from the output state (like
+pixel-aspect-ratio or framerate) it can do so on the returned #GstVideoCodecState.
+
+The new output state will only take effect (set on pads and buffers) starting
+from the next call to #gst_video_decoder_finish_frame().
+
+ the newly configured output state.
+
+
+
+
+ a #GstVideoDecoder
+
+
+
+ a #GstVideoFormat
+
+
+
+ The width in pixels
+
+
+
+ The height in pixels
+
+
+
+ An optional reference #GstVideoCodecState
+
+
+
+
+
+ Allows baseclass to consider input data as packetized or not. If the
+input is packetized, then the @parse method will not be called.
+
+
+
+
+
+ a #GstVideoDecoder
+
+
+
+ whether the input data should be considered as packetized.
+
+
+
+
+
+ Lets #GstVideoDecoder sub-classes decide if they want the sink pad
+to use the default pad query handler to reply to accept-caps queries.
+
+By setting this to true it is possible to further customize the default
+handler with %GST_PAD_SET_ACCEPT_INTERSECT and
+%GST_PAD_SET_ACCEPT_TEMPLATE
+
+
+
+
+
+ a #GstVideoDecoder
+
+
+
+ if the default pad accept-caps query handling should be used
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Subclasses can override any of the available virtual methods or not, as
+needed. At minimum @handle_frame needs to be overridden, and @set_format
+and likely as well. If non-packetized input is supported or expected,
+@parse needs to be overridden as well.
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ %TRUE if the negotiation succeeded, else %FALSE.
+
+
+
+
+ a #GstVideoDecoder
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ The interface allows unified access to control flipping and rotation
+operations of video-sources or operators.
+
+
+
+
+
+ #GstVideoDirectionInterface interface.
+
+ parent interface type.
+
+
+
+
+ GstVideoDither provides implementations of several dithering algorithms
+that can be applied to lines of video pixels to quantize and dither them.
+
+ Free @dither
+
+
+
+
+
+ a #GstVideoDither
+
+
+
+
+
+ Dither @width pixels starting from offset @x in @line using @dither.
+
+@y is the line number of @line in the output image.
+
+
+
+
+
+ a #GstVideoDither
+
+
+
+ pointer to the pixels of the line
+
+
+
+ x coordinate
+
+
+
+ y coordinate
+
+
+
+ the width
+
+
+
+
+
+ Make a new dither object for dithering lines of @format using the
+algorithm described by @method.
+
+Each component will be quantized to a multiple of @quantizer. Better
+performance is achived when @quantizer is a power of 2.
+
+@width is the width of the lines that this ditherer will handle.
+
+ a new #GstVideoDither
+
+
+
+
+ a #GstVideoDitherMethod
+
+
+
+ a #GstVideoDitherFlags
+
+
+
+ a #GstVideoFormat
+
+
+
+ quantizer
+
+
+
+ the width of the lines
+
+
+
+
+
+
+ Extra flags that influence the result from gst_video_chroma_resample_new().
+
+ no flags
+
+
+ the input is interlaced
+
+
+ quantize values in addition to adding dither.
+
+
+
+ Different dithering methods to use.
+
+ no dithering
+
+
+ propagate rounding errors downwards
+
+
+ Dither with floyd-steinberg error diffusion
+
+
+ Dither with Sierra Lite error diffusion
+
+
+ ordered dither using a bayer pattern
+
+
+
+ This base class is for video encoders turning raw video into
+encoded video data.
+
+GstVideoEncoder and subclass should cooperate as follows.
+
+## Configuration
+
+ * Initially, GstVideoEncoder calls @start when the encoder element
+ is activated, which allows subclass to perform any global setup.
+ * GstVideoEncoder calls @set_format to inform subclass of the format
+ of input video data that it is about to receive. Subclass should
+ setup for encoding and configure base class as appropriate
+ (e.g. latency). While unlikely, it might be called more than once,
+ if changing input parameters require reconfiguration. Baseclass
+ will ensure that processing of current configuration is finished.
+ * GstVideoEncoder calls @stop at end of all processing.
+
+## Data processing
+
+ * Base class collects input data and metadata into a frame and hands
+ this to subclass' @handle_frame.
+
+ * If codec processing results in encoded data, subclass should call
+ @gst_video_encoder_finish_frame to have encoded data pushed
+ downstream.
+
+ * If implemented, baseclass calls subclass @pre_push just prior to
+ pushing to allow subclasses to modify some metadata on the buffer.
+ If it returns GST_FLOW_OK, the buffer is pushed downstream.
+
+ * GstVideoEncoderClass will handle both srcpad and sinkpad events.
+ Sink events will be passed to subclass if @event callback has been
+ provided.
+
+## Shutdown phase
+
+ * GstVideoEncoder class calls @stop to inform the subclass that data
+ parsing will be stopped.
+
+Subclass is responsible for providing pad template caps for
+source and sink pads. The pads need to be named "sink" and "src". It should
+also be able to provide fixed src pad caps in @getcaps by the time it calls
+@gst_video_encoder_finish_frame.
+
+Things that subclass need to take care of:
+
+ * Provide pad templates
+ * Provide source pad caps before pushing the first buffer
+ * Accept data in @handle_frame and provide encoded results to
+ @gst_video_encoder_finish_frame.
+
+
+The #GstVideoEncoder:qos property will enable the Quality-of-Service
+features of the encoder which gather statistics about the real-time
+performance of the downstream elements. If enabled, subclasses can
+use gst_video_encoder_get_max_encode_time() to check if input frames
+are already late and drop them right away to give a chance to the
+pipeline to catch up.
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Negotiate with downstream elements to currently configured #GstVideoCodecState.
+Unmark GST_PAD_FLAG_NEED_RECONFIGURE in any case. But mark it again if
+negotiate fails.
+
+ %TRUE if the negotiation succeeded, else %FALSE.
+
+
+
+
+ a #GstVideoEncoder
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Helper function that allocates a buffer to hold an encoded video frame
+for @encoder's current #GstVideoCodecState.
+
+ allocated buffer
+
+
+
+
+ a #GstVideoEncoder
+
+
+
+ size of the buffer
+
+
+
+
+
+ Helper function that allocates a buffer to hold an encoded video frame for @encoder's
+current #GstVideoCodecState. Subclass should already have configured video
+state and set src pad caps.
+
+The buffer allocated here is owned by the frame and you should only
+keep references to the frame, not the buffer.
+
+ %GST_FLOW_OK if an output buffer could be allocated
+
+
+
+
+ a #GstVideoEncoder
+
+
+
+ a #GstVideoCodecFrame
+
+
+
+ size of the buffer
+
+
+
+
+
+ @frame must have a valid encoded data buffer, whose metadata fields
+are then appropriately set according to frame data or no buffer at
+all if the frame should be dropped.
+It is subsequently pushed downstream or provided to @pre_push.
+In any case, the frame is considered finished and released.
+
+After calling this function the output buffer of the frame is to be
+considered read-only. This function will also change the metadata
+of the buffer.
+
+ a #GstFlowReturn resulting from sending data downstream
+
+
+
+
+ a #GstVideoEncoder
+
+
+
+ an encoded #GstVideoCodecFrame
+
+
+
+
+
+ Lets #GstVideoEncoder sub-classes to know the memory @allocator
+used by the base class and its @params.
+
+Unref the @allocator after use it.
+
+
+
+
+
+ a #GstVideoEncoder
+
+
+
+ the #GstAllocator
+used
+
+
+
+ the
+#GstAllocatorParams of @allocator
+
+
+
+
+
+ Get a pending unfinished #GstVideoCodecFrame
+
+ pending unfinished #GstVideoCodecFrame identified by @frame_number.
+
+
+
+
+ a #GstVideoEnccoder
+
+
+
+ system_frame_number of a frame
+
+
+
+
+
+ Get all pending unfinished #GstVideoCodecFrame
+
+ pending unfinished #GstVideoCodecFrame.
+
+
+
+
+
+
+ a #GstVideoEncoder
+
+
+
+
+
+ Query the configured encoding latency. Results will be returned via
+@min_latency and @max_latency.
+
+
+
+
+
+ a #GstVideoEncoder
+
+
+
+ address of variable in which to store the
+ configured minimum latency, or %NULL
+
+
+
+ address of variable in which to store the
+ configured maximum latency, or %NULL
+
+
+
+
+
+ Determines maximum possible encoding time for @frame that will
+allow it to encode and arrive in time (as determined by QoS events).
+In particular, a negative result means encoding in time is no longer possible
+and should therefore occur as soon/skippy as possible.
+
+If no QoS events have been received from downstream, or if
+#GstVideoEncoder:qos is disabled this function returns #G_MAXINT64.
+
+ max decoding time.
+
+
+
+
+ a #GstVideoEncoder
+
+
+
+ a #GstVideoCodecFrame
+
+
+
+
+
+ Get the oldest unfinished pending #GstVideoCodecFrame
+
+ oldest unfinished pending #GstVideoCodecFrame
+
+
+
+
+ a #GstVideoEncoder
+
+
+
+
+
+ Get the current #GstVideoCodecState
+
+ #GstVideoCodecState describing format of video data.
+
+
+
+
+ a #GstVideoEncoder
+
+
+
+
+
+ Checks if @encoder is currently configured to handle Quality-of-Service
+events from downstream.
+
+ %TRUE if the encoder is configured to perform Quality-of-Service.
+
+
+
+
+ the encoder
+
+
+
+
+
+ Sets the video encoder tags and how they should be merged with any
+upstream stream tags. This will override any tags previously-set
+with gst_video_encoder_merge_tags().
+
+Note that this is provided for convenience, and the subclass is
+not required to use this and can still do tag handling on its own.
+
+MT safe.
+
+
+
+
+
+ a #GstVideoEncoder
+
+
+
+ a #GstTagList to merge, or NULL to unset
+ previously-set tags
+
+
+
+ the #GstTagMergeMode to use, usually #GST_TAG_MERGE_REPLACE
+
+
+
+
+
+ Negotiate with downstream elements to currently configured #GstVideoCodecState.
+Unmark GST_PAD_FLAG_NEED_RECONFIGURE in any case. But mark it again if
+negotiate fails.
+
+ %TRUE if the negotiation succeeded, else %FALSE.
+
+
+
+
+ a #GstVideoEncoder
+
+
+
+
+
+ Returns caps that express @caps (or sink template caps if @caps == NULL)
+restricted to resolution/format/... combinations supported by downstream
+elements (e.g. muxers).
+
+ a #GstCaps owned by caller
+
+
+
+
+ a #GstVideoEncoder
+
+
+
+ initial caps
+
+
+
+ filter caps
+
+
+
+
+
+ Set the codec headers to be sent downstream whenever requested.
+
+
+
+
+
+ a #GstVideoEncoder
+
+
+
+ a list of #GstBuffer containing the codec header
+
+
+
+
+
+
+
+ Informs baseclass of encoding latency.
+
+
+
+
+
+ a #GstVideoEncoder
+
+
+
+ minimum latency
+
+
+
+ maximum latency
+
+
+
+
+
+ Request minimal value for PTS passed to handle_frame.
+
+For streams with reordered frames this can be used to ensure that there
+is enough time to accomodate first DTS, which may be less than first PTS
+
+Since 1.6
+
+
+
+
+
+ a #GstVideoEncoder
+
+
+
+ minimal PTS that will be passed to handle_frame
+
+
+
+
+
+ Creates a new #GstVideoCodecState with the specified caps as the output state
+for the encoder.
+Any previously set output state on @encoder will be replaced by the newly
+created one.
+
+The specified @caps should not contain any resolution, pixel-aspect-ratio,
+framerate, codec-data, .... Those should be specified instead in the returned
+#GstVideoCodecState.
+
+If the subclass wishes to copy over existing fields (like pixel aspect ratio,
+or framerate) from an existing #GstVideoCodecState, it can be provided as a
+@reference.
+
+If the subclass wishes to override some fields from the output state (like
+pixel-aspect-ratio or framerate) it can do so on the returned #GstVideoCodecState.
+
+The new output state will only take effect (set on pads and buffers) starting
+from the next call to #gst_video_encoder_finish_frame().
+
+ the newly configured output state.
+
+
+
+
+ a #GstVideoEncoder
+
+
+
+ the #GstCaps to use for the output
+
+
+
+ An optional reference @GstVideoCodecState
+
+
+
+
+
+ Configures @encoder to handle Quality-of-Service events from downstream.
+
+
+
+
+
+ the encoder
+
+
+
+ the new qos value.
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Subclasses can override any of the available virtual methods or not, as
+needed. At minimum @handle_frame needs to be overridden, and @set_format
+and @get_caps are likely needed as well.
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ %TRUE if the negotiation succeeded, else %FALSE.
+
+
+
+
+ a #GstVideoEncoder
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Field order of interlaced content. This is only valid for
+interlace-mode=interleaved and not interlace-mode=mixed. In the case of
+mixed or GST_VIDEO_FIELD_ORDER_UNKOWN, the field order is signalled via
+buffer flags.
+
+ unknown field order for interlaced content.
+ The actual field order is signalled via buffer flags.
+
+
+ top field is first
+
+
+ bottom field is first
+
+
+ Convert @order to a #GstVideoFieldOrder
+
+ the #GstVideoFieldOrder of @order or
+ #GST_VIDEO_FIELD_ORDER_UNKNOWN when @order is not a valid
+ string representation for a #GstVideoFieldOrder.
+
+
+
+
+ a field order
+
+
+
+
+
+ Convert @order to its string representation.
+
+ @order as a string or NULL if @order in invalid.
+
+
+
+
+ a #GstVideoFieldOrder
+
+
+
+
+
+
+ Provides useful functions and a base class for video filters.
+
+The videofilter will by default enable QoS on the parent GstBaseTransform
+to implement frame dropping.
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ The video filter class structure.
+
+ the parent class structure
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Extra video flags
+
+ no flags
+
+
+ a variable fps is selected, fps_n and fps_d
+ denote the maximum fps of the video
+
+
+ Each color has been scaled by the alpha
+ value.
+
+
+
+ Enum value describing the most common video formats.
+
+ Unknown or unset video format id
+
+
+ Encoded video format. Only ever use that in caps for
+ special video formats in combination with non-system
+ memory GstCapsFeatures where it does not make sense
+ to specify a real video format.
+
+
+ planar 4:2:0 YUV
+
+
+ planar 4:2:0 YVU (like I420 but UV planes swapped)
+
+
+ packed 4:2:2 YUV (Y0-U0-Y1-V0 Y2-U2-Y3-V2 Y4 ...)
+
+
+ packed 4:2:2 YUV (U0-Y0-V0-Y1 U2-Y2-V2-Y3 U4 ...)
+
+
+ packed 4:4:4 YUV with alpha channel (A0-Y0-U0-V0 ...)
+
+
+ sparse rgb packed into 32 bit, space last
+
+
+ sparse reverse rgb packed into 32 bit, space last
+
+
+ sparse rgb packed into 32 bit, space first
+
+
+ sparse reverse rgb packed into 32 bit, space first
+
+
+ rgb with alpha channel last
+
+
+ reverse rgb with alpha channel last
+
+
+ rgb with alpha channel first
+
+
+ reverse rgb with alpha channel first
+
+
+ rgb
+
+
+ reverse rgb
+
+
+ planar 4:1:1 YUV
+
+
+ planar 4:2:2 YUV
+
+
+ packed 4:2:2 YUV (Y0-V0-Y1-U0 Y2-V2-Y3-U2 Y4 ...)
+
+
+ planar 4:4:4 YUV
+
+
+ packed 4:2:2 10-bit YUV, complex format
+
+
+ packed 4:2:2 16-bit YUV, Y0-U0-Y1-V1 order
+
+
+ planar 4:2:0 YUV with interleaved UV plane
+
+
+ planar 4:2:0 YUV with interleaved VU plane
+
+
+ 8-bit grayscale
+
+
+ 16-bit grayscale, most significant byte first
+
+
+ 16-bit grayscale, least significant byte first
+
+
+ packed 4:4:4 YUV (Y-U-V ...)
+
+
+ rgb 5-6-5 bits per component
+
+
+ reverse rgb 5-6-5 bits per component
+
+
+ rgb 5-5-5 bits per component
+
+
+ reverse rgb 5-5-5 bits per component
+
+
+ packed 10-bit 4:2:2 YUV (U0-Y0-V0-Y1 U2-Y2-V2-Y3 U4 ...)
+
+
+ planar 4:4:2:0 AYUV
+
+
+ 8-bit paletted RGB
+
+
+ planar 4:1:0 YUV
+
+
+ planar 4:1:0 YUV (like YUV9 but UV planes swapped)
+
+
+ packed 4:1:1 YUV (Cb-Y0-Y1-Cr-Y2-Y3 ...)
+
+
+ rgb with alpha channel first, 16 bits per channel
+
+
+ packed 4:4:4 YUV with alpha channel, 16 bits per channel (A0-Y0-U0-V0 ...)
+
+
+ packed 4:4:4 RGB, 10 bits per channel
+
+
+ planar 4:2:0 YUV, 10 bits per channel
+
+
+ planar 4:2:0 YUV, 10 bits per channel
+
+
+ planar 4:2:2 YUV, 10 bits per channel
+
+
+ planar 4:2:2 YUV, 10 bits per channel
+
+
+ planar 4:4:4 YUV, 10 bits per channel (Since: 1.2)
+
+
+ planar 4:4:4 YUV, 10 bits per channel (Since: 1.2)
+
+
+ planar 4:4:4 RGB, 8 bits per channel (Since: 1.2)
+
+
+ planar 4:4:4 RGB, 10 bits per channel (Since: 1.2)
+
+
+ planar 4:4:4 RGB, 10 bits per channel (Since: 1.2)
+
+
+ planar 4:2:2 YUV with interleaved UV plane (Since: 1.2)
+
+
+ planar 4:4:4 YUV with interleaved UV plane (Since: 1.2)
+
+
+ NV12 with 64x32 tiling in zigzag pattern (Since: 1.4)
+
+
+ planar 4:4:2:0 YUV, 10 bits per channel (Since: 1.6)
+
+
+ planar 4:4:2:0 YUV, 10 bits per channel (Since: 1.6)
+
+
+ planar 4:4:2:2 YUV, 10 bits per channel (Since: 1.6)
+
+
+ planar 4:4:2:2 YUV, 10 bits per channel (Since: 1.6)
+
+
+ planar 4:4:4:4 YUV, 10 bits per channel (Since: 1.6)
+
+
+ planar 4:4:4:4 YUV, 10 bits per channel (Since: 1.6)
+
+
+ planar 4:2:2 YUV with interleaved VU plane (Since: 1.6)
+
+
+ planar 4:2:0 YUV with interleaved UV plane, 10 bits per channel (Since: 1.10)
+
+
+ planar 4:2:0 YUV with interleaved UV plane, 10 bits per channel (Since: 1.10)
+
+
+ packed 4:4:4 YUV (U-Y-V ...) (Since 1.10)
+
+
+ packed 4:2:2 YUV (V0-Y0-U0-Y1 V2-Y2-U2-Y3 V4 ...)
+
+
+ planar 4:4:4:4 ARGB, 8 bits per channel (Since: 1.12)
+
+
+ planar 4:4:4:4 ARGB, 10 bits per channel (Since: 1.12)
+
+
+ planar 4:4:4:4 ARGB, 10 bits per channel (Since: 1.12)
+
+
+ planar 4:4:4 RGB, 12 bits per channel (Since: 1.12)
+
+
+ planar 4:4:4 RGB, 12 bits per channel (Since: 1.12)
+
+
+ planar 4:4:4:4 ARGB, 12 bits per channel (Since: 1.12)
+
+
+ planar 4:4:4:4 ARGB, 12 bits per channel (Since: 1.12)
+
+
+ planar 4:2:0 YUV, 12 bits per channel (Since: 1.12)
+
+
+ planar 4:2:0 YUV, 12 bits per channel (Since: 1.12)
+
+
+ planar 4:2:2 YUV, 12 bits per channel (Since: 1.12)
+
+
+ planar 4:2:2 YUV, 12 bits per channel (Since: 1.12)
+
+
+ planar 4:4:4 YUV, 12 bits per channel (Since: 1.12)
+
+
+ planar 4:4:4 YUV, 12 bits per channel (Since: 1.12)
+
+
+ 10-bit grayscale, packed into 32bit words (2 bits padding) (Since: 1.14)
+
+
+ 10-bit variant of @GST_VIDEO_FORMAT_NV12, packed into 32bit words (MSB 2 bits padding) (Since: 1.14)
+
+
+ 10-bit variant of @GST_VIDEO_FORMAT_NV16, packed into 32bit words (MSB 2 bits padding) (Since: 1.14)
+
+
+ Converts a FOURCC value into the corresponding #GstVideoFormat.
+If the FOURCC cannot be represented by #GstVideoFormat,
+#GST_VIDEO_FORMAT_UNKNOWN is returned.
+
+ the #GstVideoFormat describing the FOURCC value
+
+
+
+
+ a FOURCC value representing raw YUV video
+
+
+
+
+
+ Find the #GstVideoFormat for the given parameters.
+
+ a #GstVideoFormat or GST_VIDEO_FORMAT_UNKNOWN when the parameters to
+not specify a known format.
+
+
+
+
+ the amount of bits used for a pixel
+
+
+
+ the amount of bits used to store a pixel. This value is bigger than
+ @depth
+
+
+
+ the endianness of the masks, #G_LITTLE_ENDIAN or #G_BIG_ENDIAN
+
+
+
+ the red mask
+
+
+
+ the green mask
+
+
+
+ the blue mask
+
+
+
+ the alpha mask, or 0 if no alpha mask
+
+
+
+
+
+ Convert the @format string to its #GstVideoFormat.
+
+ the #GstVideoFormat for @format or GST_VIDEO_FORMAT_UNKNOWN when the
+string is not a known format.
+
+
+
+
+ a format string
+
+
+
+
+
+ Get the #GstVideoFormatInfo for @format
+
+ The #GstVideoFormatInfo for @format.
+
+
+
+
+ a #GstVideoFormat
+
+
+
+
+
+ Get the default palette of @format. This the palette used in the pack
+function for paletted formats.
+
+ the default palette of @format or %NULL when
+@format does not have a palette.
+
+
+
+
+ a #GstVideoFormat
+
+
+
+ size of the palette in bytes
+
+
+
+
+
+ Converts a #GstVideoFormat value into the corresponding FOURCC. Only
+a few YUV formats have corresponding FOURCC values. If @format has
+no corresponding FOURCC value, 0 is returned.
+
+ the FOURCC corresponding to @format
+
+
+
+
+ a #GstVideoFormat video format
+
+
+
+
+
+ Returns a string containing a descriptive name for
+the #GstVideoFormat if there is one, or NULL otherwise.
+
+ the name corresponding to @format
+
+
+
+
+ a #GstVideoFormat video format
+
+
+
+
+
+
+ The different video flags that a format info can have.
+
+ The video format is YUV, components are numbered
+ 0=Y, 1=U, 2=V.
+
+
+ The video format is RGB, components are numbered
+ 0=R, 1=G, 2=B.
+
+
+ The video is gray, there is one gray component
+ with index 0.
+
+
+ The video format has an alpha components with
+ the number 3.
+
+
+ The video format has data stored in little
+ endianness.
+
+
+ The video format has a palette. The palette
+ is stored in the second plane and indexes are stored in the first plane.
+
+
+ The video format has a complex layout that
+ can't be described with the usual information in the #GstVideoFormatInfo.
+
+
+ This format can be used in a
+ #GstVideoFormatUnpack and #GstVideoFormatPack function.
+
+
+ The format is tiled, there is tiling information
+ in the last plane.
+
+
+
+ Information for a video format.
+
+ #GstVideoFormat
+
+
+
+ string representation of the format
+
+
+
+ use readable description of the format
+
+
+
+ #GstVideoFormatFlags
+
+
+
+ The number of bits used to pack data items. This can be less than 8
+ when multiple pixels are stored in a byte. for values > 8 multiple bytes
+ should be read according to the endianness flag before applying the shift
+ and mask.
+
+
+
+ the number of components in the video format.
+
+
+
+ the number of bits to shift away to get the component data
+
+
+
+
+
+ the depth in bits for each component
+
+
+
+
+
+ the pixel stride of each component. This is the amount of
+ bytes to the pixel immediately to the right. When bits < 8, the stride is
+ expressed in bits. For 24-bit RGB, this would be 3 bytes, for example,
+ while it would be 4 bytes for RGBx or ARGB.
+
+
+
+
+
+ the number of planes for this format. The number of planes can be
+ less than the amount of components when multiple components are packed into
+ one plane.
+
+
+
+ the plane number where a component can be found
+
+
+
+
+
+ the offset in the plane where the first pixel of the components
+ can be found.
+
+
+
+
+
+ subsampling factor of the width for the component. Use
+ GST_VIDEO_SUB_SCALE to scale a width.
+
+
+
+
+
+ subsampling factor of the height for the component. Use
+ GST_VIDEO_SUB_SCALE to scale a height.
+
+
+
+
+
+ the format of the unpacked pixels. This format must have the
+ #GST_VIDEO_FORMAT_FLAG_UNPACK flag set.
+
+
+
+ an unpack function for this format
+
+
+
+ the amount of lines that will be packed
+
+
+
+ an pack function for this format
+
+
+
+ The tiling mode
+
+
+
+ The width of a tile, in bytes, represented as a shift
+
+
+
+ The height of a tile, in bytes, represented as a shift
+
+
+
+
+
+
+
+
+
+ Packs @width pixels from @src to the given planes and strides in the
+format @info. The pixels from source have each component interleaved
+and will be packed into the planes in @data.
+
+This function operates on pack_lines lines, meaning that @src should
+contain at least pack_lines lines with a stride of @sstride and @y
+should be a multiple of pack_lines.
+
+Subsampled formats will use the horizontally and vertically cosited
+component from the source. Subsampling should be performed before
+packing.
+
+Because this function does not have a x coordinate, it is not possible to
+pack pixels starting from an unaligned position. For tiled images this
+means that packing should start from a tile coordinate. For subsampled
+formats this means that a complete pixel needs to be packed.
+
+
+
+
+
+ a #GstVideoFormatInfo
+
+
+
+ flags to control the packing
+
+
+
+ a source array
+
+
+
+ the source array stride
+
+
+
+ pointers to the destination data planes
+
+
+
+ strides of the destination planes
+
+
+
+ the chroma siting of the target when subsampled (not used)
+
+
+
+ the y position in the image to pack to
+
+
+
+ the amount of pixels to pack.
+
+
+
+
+
+ Unpacks @width pixels from the given planes and strides containing data of
+format @info. The pixels will be unpacked into @dest with each component
+interleaved as per @info's unpack_format, which will usually be one of
+#GST_VIDEO_FORMAT_ARGB, #GST_VIDEO_FORMAT_AYUV, #GST_VIDEO_FORMAT_ARGB64 or
+#GST_VIDEO_FORMAT_AYUV64 depending on the format to unpack.
+@dest should at least be big enough to hold @width * bytes_per_pixel bytes
+where bytes_per_pixel relates to the unpack format and will usually be
+either 4 or 8 depending on the unpack format. bytes_per_pixel will be
+the same as the pixel stride for plane 0 for the above formats.
+
+For subsampled formats, the components will be duplicated in the destination
+array. Reconstruction of the missing components can be performed in a
+separate step after unpacking.
+
+
+
+
+
+ a #GstVideoFormatInfo
+
+
+
+ flags to control the unpacking
+
+
+
+ a destination array
+
+
+
+ pointers to the data planes
+
+
+
+ strides of the planes
+
+
+
+ the x position in the image to start from
+
+
+
+ the y position in the image to start from
+
+
+
+ the amount of pixels to unpack.
+
+
+
+
+
+ A video frame obtained from gst_video_frame_map()
+
+ the #GstVideoInfo
+
+
+
+ #GstVideoFrameFlags for the frame
+
+
+
+ the mapped buffer
+
+
+
+ pointer to metadata if any
+
+
+
+ id of the mapped frame. the id can for example be used to
+ indentify the frame in case of multiview video.
+
+
+
+ pointers to the plane data
+
+
+
+
+
+ mappings of the planes
+
+
+
+
+
+
+
+
+
+
+ Copy the contents from @src to @dest.
+
+ TRUE if the contents could be copied.
+
+
+
+
+ a #GstVideoFrame
+
+
+
+ a #GstVideoFrame
+
+
+
+
+
+ Copy the plane with index @plane from @src to @dest.
+
+ TRUE if the contents could be copied.
+
+
+
+
+ a #GstVideoFrame
+
+
+
+ a #GstVideoFrame
+
+
+
+ a plane
+
+
+
+
+
+ Use @info and @buffer to fill in the values of @frame. @frame is usually
+allocated on the stack, and you will pass the address to the #GstVideoFrame
+structure allocated on the stack; gst_video_frame_map() will then fill in
+the structures with the various video-specific information you need to access
+the pixels of the video buffer. You can then use accessor macros such as
+GST_VIDEO_FRAME_COMP_DATA(), GST_VIDEO_FRAME_PLANE_DATA(),
+GST_VIDEO_FRAME_COMP_STRIDE(), GST_VIDEO_FRAME_PLANE_STRIDE() etc.
+to get to the pixels.
+
+|[<!-- language="C" -->
+ GstVideoFrame vframe;
+ ...
+ // set RGB pixels to black one at a time
+ if (gst_video_frame_map (&vframe, video_info, video_buffer, GST_MAP_WRITE)) {
+ guint8 *pixels = GST_VIDEO_FRAME_PLANE_DATA (vframe, 0);
+ guint stride = GST_VIDEO_FRAME_PLANE_STRIDE (vframe, 0);
+ guint pixel_stride = GST_VIDEO_FRAME_COMP_PSTRIDE (vframe, 0);
+
+ for (h = 0; h < height; ++h) {
+ for (w = 0; w < width; ++w) {
+ guint8 *pixel = pixels + h * stride + w * pixel_stride;
+
+ memset (pixel, 0, pixel_stride);
+ }
+ }
+
+ gst_video_frame_unmap (&vframe);
+ }
+ ...
+]|
+
+All video planes of @buffer will be mapped and the pointers will be set in
+@frame->data.
+
+The purpose of this function is to make it easy for you to get to the video
+pixels in a generic way, without you having to worry too much about details
+such as whether the video data is allocated in one contiguous memory chunk
+or multiple memory chunks (e.g. one for each plane); or if custom strides
+and custom plane offsets are used or not (as signalled by GstVideoMeta on
+each buffer). This function will just fill the #GstVideoFrame structure
+with the right values and if you use the accessor macros everything will
+just work and you can access the data easily. It also maps the underlying
+memory chunks for you.
+
+ %TRUE on success.
+
+
+
+
+ pointer to #GstVideoFrame
+
+
+
+ a #GstVideoInfo
+
+
+
+ the buffer to map
+
+
+
+ #GstMapFlags
+
+
+
+
+
+ Use @info and @buffer to fill in the values of @frame with the video frame
+information of frame @id.
+
+When @id is -1, the default frame is mapped. When @id != -1, this function
+will return %FALSE when there is no GstVideoMeta with that id.
+
+All video planes of @buffer will be mapped and the pointers will be set in
+@frame->data.
+
+ %TRUE on success.
+
+
+
+
+ pointer to #GstVideoFrame
+
+
+
+ a #GstVideoInfo
+
+
+
+ the buffer to map
+
+
+
+ the frame id to map
+
+
+
+ #GstMapFlags
+
+
+
+
+
+ Unmap the memory previously mapped with gst_video_frame_map.
+
+
+
+
+
+ a #GstVideoFrame
+
+
+
+
+
+
+ Extra video frame flags
+
+ no flags
+
+
+ The video frame is interlaced. In mixed
+ interlace-mode, this flag specifies if the frame is interlaced or
+ progressive.
+
+
+ The video frame has the top field first
+
+
+ The video frame has the repeat flag
+
+
+ The video frame has one field
+
+
+ The video contains one or
+ more non-mono views
+
+
+ The video frame is the first
+ in a set of corresponding views provided as sequential frames.
+
+
+
+ Additional mapping flags for gst_video_frame_map().
+
+ Don't take another reference of the buffer and store it in
+ the GstVideoFrame. This makes sure that the buffer stays
+ writable while the frame is mapped, but requires that the
+ buffer reference stays valid until the frame is unmapped again.
+
+
+ Offset to define more flags
+
+
+
+ The orientation of the GL texture.
+
+ Top line first in memory, left row first
+
+
+ Bottom line first in memory, left row first
+
+
+ Top line first in memory, right row first
+
+
+ Bottom line first in memory, right row first
+
+
+
+ The GL texture type.
+
+ Luminance texture, GL_LUMINANCE
+
+
+ Luminance-alpha texture, GL_LUMINANCE_ALPHA
+
+
+ RGB 565 texture, GL_RGB
+
+
+ RGB texture, GL_RGB
+
+
+ RGBA texture, GL_RGBA
+
+
+ R texture, GL_RED_EXT
+
+
+ RG texture, GL_RG_EXT
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Extra buffer metadata for uploading a buffer to an OpenGL texture
+ID. The caller of gst_video_gl_texture_upload_meta_upload() must
+have OpenGL set up and call this from a thread where it is valid
+to upload something to an OpenGL texture.
+
+ parent #GstMeta
+
+
+
+ Orientation of the textures
+
+
+
+ Number of textures that are generated
+
+
+
+ Type of each texture
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Uploads the buffer which owns the meta to a specific texture ID.
+
+ %TRUE if uploading succeeded, %FALSE otherwise.
+
+
+
+
+ a #GstVideoGLTextureUploadMeta
+
+
+
+ the texture IDs to upload to
+
+
+
+
+
+
+
+
+
+
+
+
+ disable gamma handling
+
+
+ convert between input and output gamma
+Different gamma conversion modes
+
+
+
+ Information describing image properties. This information can be filled
+in from GstCaps with gst_video_info_from_caps(). The information is also used
+to store the specific video info when mapping a video frame with
+gst_video_frame_map().
+
+Use the provided macros to access the info in this structure.
+
+ the format info of the video
+
+
+
+ the interlace mode
+
+
+
+ additional video flags
+
+
+
+ the width of the video
+
+
+
+ the height of the video
+
+
+
+ the default size of one frame
+
+
+
+ the number of views for multiview video
+
+
+
+ a #GstVideoChromaSite.
+
+
+
+ the colorimetry info
+
+
+
+ the pixel-aspect-ratio numerator
+
+
+
+ the pixel-aspect-ratio demnominator
+
+
+
+ the framerate numerator
+
+
+
+ the framerate demnominator
+
+
+
+ offsets of the planes
+
+
+
+
+
+ strides of the planes
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Allocate a new #GstVideoInfo that is also initialized with
+gst_video_info_init().
+
+ a new #GstVideoInfo. free with gst_video_info_free().
+
+
+
+
+ Adjust the offset and stride fields in @info so that the padding and
+stride alignment in @align is respected.
+
+Extra padding will be added to the right side when stride alignment padding
+is required and @align will be updated with the new padding values.
+
+ %FALSE if alignment could not be applied, e.g. because the
+ size of a frame can't be represented as a 32 bit integer (Since: 1.12)
+
+
+
+
+ a #GstVideoInfo
+
+
+
+ alignment parameters
+
+
+
+
+
+ Converts among various #GstFormat types. This function handles
+GST_FORMAT_BYTES, GST_FORMAT_TIME, and GST_FORMAT_DEFAULT. For
+raw video, GST_FORMAT_DEFAULT corresponds to video frames. This
+function can be used to handle pad queries of the type GST_QUERY_CONVERT.
+
+ TRUE if the conversion was successful.
+
+
+
+
+ a #GstVideoInfo
+
+
+
+ #GstFormat of the @src_value
+
+
+
+ value to convert
+
+
+
+ #GstFormat of the @dest_value
+
+
+
+ pointer to destination value
+
+
+
+
+
+ Copy a GstVideoInfo structure.
+
+ a new #GstVideoInfo. free with gst_video_info_free.
+
+
+
+
+ a #GstVideoInfo
+
+
+
+
+
+ Free a GstVideoInfo structure previously allocated with gst_video_info_new()
+or gst_video_info_copy().
+
+
+
+
+
+ a #GstVideoInfo
+
+
+
+
+
+ Parse @caps and update @info.
+
+ TRUE if @caps could be parsed
+
+
+
+
+ a #GstVideoInfo
+
+
+
+ a #GstCaps
+
+
+
+
+
+ Initialize @info with default values.
+
+
+
+
+
+ a #GstVideoInfo
+
+
+
+
+
+ Compares two #GstVideoInfo and returns whether they are equal or not
+
+ %TRUE if @info and @other are equal, else %FALSE.
+
+
+
+
+ a #GstVideoInfo
+
+
+
+ a #GstVideoInfo
+
+
+
+
+
+ Set the default info for a video frame of @format and @width and @height.
+
+Note: This initializes @info first, no values are preserved. This function
+does not set the offsets correctly for interlaced vertically
+subsampled formats.
+
+ %FALSE if the returned video info is invalid, e.g. because the
+ size of a frame can't be represented as a 32 bit integer (Since: 1.12)
+
+
+
+
+ a #GstVideoInfo
+
+
+
+ the format
+
+
+
+ a width
+
+
+
+ a height
+
+
+
+
+
+ Convert the values of @info into a #GstCaps.
+
+ a new #GstCaps containing the info of @info.
+
+
+
+
+ a #GstVideoInfo
+
+
+
+
+
+
+ The possible values of the #GstVideoInterlaceMode describing the interlace
+mode of the stream.
+
+ all frames are progressive
+
+
+ 2 fields are interleaved in one video
+ frame. Extra buffer flags describe the field order.
+
+
+ frames contains both interlaced and
+ progressive video, the buffer flags describe the frame and fields.
+
+
+ 2 fields are stored in one buffer, use the
+ frame ID to get access to the required field. For multiview (the
+ 'views' property > 1) the fields of view N can be found at frame ID
+ (N * 2) and (N * 2) + 1.
+ Each field has only half the amount of lines as noted in the
+ height property. This mode requires multiple GstVideoMeta metadata
+ to describe the fields.
+
+
+ Convert @mode to a #GstVideoInterlaceMode
+
+ the #GstVideoInterlaceMode of @mode or
+ #GST_VIDEO_INTERLACE_MODE_PROGRESSIVE when @mode is not a valid
+ string representation for a #GstVideoInterlaceMode.
+
+
+
+
+ a mode
+
+
+
+
+
+ Convert @mode to its string representation.
+
+ @mode as a string or NULL if @mode in invalid.
+
+
+
+
+ a #GstVideoInterlaceMode
+
+
+
+
+
+
+ Different color matrix conversion modes
+
+ do conversion between color matrices
+
+
+ use the input color matrix to convert
+ to and from R'G'B
+
+
+ use the output color matrix to convert
+ to and from R'G'B
+
+
+ disable color matrix conversion.
+
+
+
+ Extra buffer metadata describing image properties
+
+ parent #GstMeta
+
+
+
+ the buffer this metadata belongs to
+
+
+
+ additional video flags
+
+
+
+ the video format
+
+
+
+ identifier of the frame
+
+
+
+ the video width
+
+
+
+ the video height
+
+
+
+ the number of planes in the image
+
+
+
+ array of offsets for the planes. This field might not always be
+ valid, it is used by the default implementation of @map.
+
+
+
+
+
+ array of strides for the planes. This field might not always be
+ valid, it is used by the default implementation of @map.
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Map the video plane with index @plane in @meta and return a pointer to the
+first byte of the plane and the stride of the plane.
+
+ TRUE if the map operation was successful.
+
+
+
+
+ a #GstVideoMeta
+
+
+
+ a plane
+
+
+
+ a #GstMapInfo
+
+
+
+ the data of @plane
+
+
+
+ the stride of @plane
+
+
+
+ @GstMapFlags
+
+
+
+
+
+ Unmap a previously mapped plane with gst_video_meta_map().
+
+ TRUE if the memory was successfully unmapped.
+
+
+
+
+ a #GstVideoMeta
+
+
+
+ a plane
+
+
+
+ a #GstMapInfo
+
+
+
+
+
+
+
+
+
+
+
+ Extra data passed to a video transform #GstMetaTransformFunction such as:
+"gst-video-scale".
+
+ the input #GstVideoInfo
+
+
+
+ the output #GstVideoInfo
+
+
+
+ Get the #GQuark for the "gst-video-scale" metadata transform operation.
+
+ a #GQuark
+
+
+
+
+
+ GstVideoMultiviewFlags are used to indicate extra properties of a
+stereo/multiview stream beyond the frame layout and buffer mapping
+that is conveyed in the #GstMultiviewMode.
+
+ No flags
+
+
+ For stereo streams, the
+ normal arrangement of left and right views is reversed.
+
+
+ The left view is vertically
+ mirrored.
+
+
+ The left view is horizontally
+ mirrored.
+
+
+ The right view is
+ vertically mirrored.
+
+
+ The right view is
+ horizontally mirrored.
+
+
+ For frame-packed
+ multiview modes, indicates that the individual
+ views have been encoded with half the true width or height
+ and should be scaled back up for display. This flag
+ is used for overriding input layout interpretation
+ by adjusting pixel-aspect-ratio.
+ For side-by-side, column interleaved or checkerboard packings, the
+ pixel width will be doubled. For row interleaved and top-bottom
+ encodings, pixel height will be doubled.
+
+
+ The video stream contains both
+ mono and multiview portions, signalled on each buffer by the
+ absence or presence of the @GST_VIDEO_BUFFER_FLAG_MULTIPLE_VIEW
+ buffer flag.
+
+
+
+
+
+ #GstVideoMultiviewFramePacking represents the subset of #GstVideoMultiviewMode
+values that can be applied to any video frame without needing extra metadata.
+It can be used by elements that provide a property to override the
+multiview interpretation of a video stream when the video doesn't contain
+any markers.
+
+This enum is used (for example) on playbin, to re-interpret a played
+video stream as a stereoscopic video. The individual enum values are
+equivalent to and have the same value as the matching #GstVideoMultiviewMode.
+
+ A special value indicating
+no frame packing info.
+
+
+ All frames are monoscopic.
+
+
+ All frames represent a left-eye view.
+
+
+ All frames represent a right-eye view.
+
+
+ Left and right eye views are
+provided in the left and right half of the frame respectively.
+
+
+ Left and right eye
+views are provided in the left and right half of the frame, but
+have been sampled using quincunx method, with half-pixel offset
+between the 2 views.
+
+
+ Alternating vertical
+columns of pixels represent the left and right eye view respectively.
+
+
+ Alternating horizontal
+rows of pixels represent the left and right eye view respectively.
+
+
+ The top half of the frame
+contains the left eye, and the bottom half the right eye.
+
+
+ Pixels are arranged with
+alternating pixels representing left and right eye views in a
+checkerboard fashion.
+
+
+
+ All possible stereoscopic 3D and multiview representations.
+In conjunction with #GstVideoMultiviewFlags, describes how
+multiview content is being transported in the stream.
+
+ A special value indicating
+no multiview information. Used in GstVideoInfo and other places to
+indicate that no specific multiview handling has been requested or
+provided. This value is never carried on caps.
+
+
+ All frames are monoscopic.
+
+
+ All frames represent a left-eye view.
+
+
+ All frames represent a right-eye view.
+
+
+ Left and right eye views are
+provided in the left and right half of the frame respectively.
+
+
+ Left and right eye
+views are provided in the left and right half of the frame, but
+have been sampled using quincunx method, with half-pixel offset
+between the 2 views.
+
+
+ Alternating vertical
+columns of pixels represent the left and right eye view respectively.
+
+
+ Alternating horizontal
+rows of pixels represent the left and right eye view respectively.
+
+
+ The top half of the frame
+contains the left eye, and the bottom half the right eye.
+
+
+ Pixels are arranged with
+alternating pixels representing left and right eye views in a
+checkerboard fashion.
+
+
+ Left and right eye views
+are provided in separate frames alternately.
+
+
+ Multiple
+independent views are provided in separate frames in sequence.
+This method only applies to raw video buffers at the moment.
+Specific view identification is via the #GstVideoMultiviewMeta
+and #GstVideoMeta(s) on raw video buffers.
+
+
+ Multiple views are
+provided as separate #GstMemory framebuffers attached to each
+#GstBuffer, described by the #GstVideoMultiviewMeta
+and #GstVideoMeta(s)
+
+
+
+ The #GstVideoMultiviewMode value
+
+Given a string from a caps multiview-mode field,
+output the corresponding #GstVideoMultiviewMode
+or #GST_VIDEO_MULTIVIEW_MODE_NONE
+
+
+
+
+ multiview-mode field string from caps
+
+
+
+
+
+
+ The caps string representation of the mode, or NULL if invalid.
+
+Given a #GstVideoMultiviewMode returns the multiview-mode caps string
+for insertion into a caps structure
+
+
+
+
+ A #GstVideoMultiviewMode value
+
+
+
+
+
+
+ The interface allows unified access to control flipping and autocenter
+operation of video-sources or operators.
+
+ Get the horizontal centering offset from the given object.
+
+ %TRUE in case the element supports centering
+
+
+
+
+ #GstVideoOrientation interface of a #GstElement
+
+
+
+ return location for the result
+
+
+
+
+
+ Get the horizontal flipping state (%TRUE for flipped) from the given object.
+
+ %TRUE in case the element supports flipping
+
+
+
+
+ #GstVideoOrientation interface of a #GstElement
+
+
+
+ return location for the result
+
+
+
+
+
+ Get the vertical centering offset from the given object.
+
+ %TRUE in case the element supports centering
+
+
+
+
+ #GstVideoOrientation interface of a #GstElement
+
+
+
+ return location for the result
+
+
+
+
+
+ Get the vertical flipping state (%TRUE for flipped) from the given object.
+
+ %TRUE in case the element supports flipping
+
+
+
+
+ #GstVideoOrientation interface of a #GstElement
+
+
+
+ return location for the result
+
+
+
+
+
+ Set the horizontal centering offset for the given object.
+
+ %TRUE in case the element supports centering
+
+
+
+
+ #GstVideoOrientation interface of a #GstElement
+
+
+
+ centering offset
+
+
+
+
+
+ Set the horizontal flipping state (%TRUE for flipped) for the given object.
+
+ %TRUE in case the element supports flipping
+
+
+
+
+ #GstVideoOrientation interface of a #GstElement
+
+
+
+ use flipping
+
+
+
+
+
+ Set the vertical centering offset for the given object.
+
+ %TRUE in case the element supports centering
+
+
+
+
+ #GstVideoOrientation interface of a #GstElement
+
+
+
+ centering offset
+
+
+
+
+
+ Set the vertical flipping state (%TRUE for flipped) for the given object.
+
+ %TRUE in case the element supports flipping
+
+
+
+
+ #GstVideoOrientation interface of a #GstElement
+
+
+
+ use flipping
+
+
+
+
+
+ Get the horizontal centering offset from the given object.
+
+ %TRUE in case the element supports centering
+
+
+
+
+ #GstVideoOrientation interface of a #GstElement
+
+
+
+ return location for the result
+
+
+
+
+
+ Get the horizontal flipping state (%TRUE for flipped) from the given object.
+
+ %TRUE in case the element supports flipping
+
+
+
+
+ #GstVideoOrientation interface of a #GstElement
+
+
+
+ return location for the result
+
+
+
+
+
+ Get the vertical centering offset from the given object.
+
+ %TRUE in case the element supports centering
+
+
+
+
+ #GstVideoOrientation interface of a #GstElement
+
+
+
+ return location for the result
+
+
+
+
+
+ Get the vertical flipping state (%TRUE for flipped) from the given object.
+
+ %TRUE in case the element supports flipping
+
+
+
+
+ #GstVideoOrientation interface of a #GstElement
+
+
+
+ return location for the result
+
+
+
+
+
+ Set the horizontal centering offset for the given object.
+
+ %TRUE in case the element supports centering
+
+
+
+
+ #GstVideoOrientation interface of a #GstElement
+
+
+
+ centering offset
+
+
+
+
+
+ Set the horizontal flipping state (%TRUE for flipped) for the given object.
+
+ %TRUE in case the element supports flipping
+
+
+
+
+ #GstVideoOrientation interface of a #GstElement
+
+
+
+ use flipping
+
+
+
+
+
+ Set the vertical centering offset for the given object.
+
+ %TRUE in case the element supports centering
+
+
+
+
+ #GstVideoOrientation interface of a #GstElement
+
+
+
+ centering offset
+
+
+
+
+
+ Set the vertical flipping state (%TRUE for flipped) for the given object.
+
+ %TRUE in case the element supports flipping
+
+
+
+
+ #GstVideoOrientation interface of a #GstElement
+
+
+
+ use flipping
+
+
+
+
+
+
+ #GstVideoOrientationInterface interface.
+
+ parent interface type.
+
+
+
+
+
+ %TRUE in case the element supports flipping
+
+
+
+
+ #GstVideoOrientation interface of a #GstElement
+
+
+
+ return location for the result
+
+
+
+
+
+
+
+
+ %TRUE in case the element supports flipping
+
+
+
+
+ #GstVideoOrientation interface of a #GstElement
+
+
+
+ return location for the result
+
+
+
+
+
+
+
+
+ %TRUE in case the element supports centering
+
+
+
+
+ #GstVideoOrientation interface of a #GstElement
+
+
+
+ return location for the result
+
+
+
+
+
+
+
+
+ %TRUE in case the element supports centering
+
+
+
+
+ #GstVideoOrientation interface of a #GstElement
+
+
+
+ return location for the result
+
+
+
+
+
+
+
+
+ %TRUE in case the element supports flipping
+
+
+
+
+ #GstVideoOrientation interface of a #GstElement
+
+
+
+ use flipping
+
+
+
+
+
+
+
+
+ %TRUE in case the element supports flipping
+
+
+
+
+ #GstVideoOrientation interface of a #GstElement
+
+
+
+ use flipping
+
+
+
+
+
+
+
+
+ %TRUE in case the element supports centering
+
+
+
+
+ #GstVideoOrientation interface of a #GstElement
+
+
+
+ centering offset
+
+
+
+
+
+
+
+
+ %TRUE in case the element supports centering
+
+
+
+
+ #GstVideoOrientation interface of a #GstElement
+
+
+
+ centering offset
+
+
+
+
+
+
+
+ The different video orientation methods.
+
+ Identity (no rotation)
+
+
+ Rotate clockwise 90 degrees
+
+
+ Rotate 180 degrees
+
+
+ Rotate counter-clockwise 90 degrees
+
+
+ Flip horizontally
+
+
+ Flip vertically
+
+
+ Flip across upper left/lower right diagonal
+
+
+ Flip across upper right/lower left diagonal
+
+
+ Select flip method based on image-orientation tag
+
+
+ Current status depends on plugin internal setup
+
+
+
+ The #GstVideoOverlay interface is used for 2 main purposes :
+
+* To get a grab on the Window where the video sink element is going to render.
+ This is achieved by either being informed about the Window identifier that
+ the video sink element generated, or by forcing the video sink element to use
+ a specific Window identifier for rendering.
+* To force a redrawing of the latest video frame the video sink element
+ displayed on the Window. Indeed if the #GstPipeline is in #GST_STATE_PAUSED
+ state, moving the Window around will damage its content. Application
+ developers will want to handle the Expose events themselves and force the
+ video sink element to refresh the Window's content.
+
+Using the Window created by the video sink is probably the simplest scenario,
+in some cases, though, it might not be flexible enough for application
+developers if they need to catch events such as mouse moves and button
+clicks.
+
+Setting a specific Window identifier on the video sink element is the most
+flexible solution but it has some issues. Indeed the application needs to set
+its Window identifier at the right time to avoid internal Window creation
+from the video sink element. To solve this issue a #GstMessage is posted on
+the bus to inform the application that it should set the Window identifier
+immediately. Here is an example on how to do that correctly:
+|[
+static GstBusSyncReply
+create_window (GstBus * bus, GstMessage * message, GstPipeline * pipeline)
+{
+ // ignore anything but 'prepare-window-handle' element messages
+ if (!gst_is_video_overlay_prepare_window_handle_message (message))
+ return GST_BUS_PASS;
+
+ win = XCreateSimpleWindow (disp, root, 0, 0, 320, 240, 0, 0, 0);
+
+ XSetWindowBackgroundPixmap (disp, win, None);
+
+ XMapRaised (disp, win);
+
+ XSync (disp, FALSE);
+
+ gst_video_overlay_set_window_handle (GST_VIDEO_OVERLAY (GST_MESSAGE_SRC (message)),
+ win);
+
+ gst_message_unref (message);
+
+ return GST_BUS_DROP;
+}
+...
+int
+main (int argc, char **argv)
+{
+...
+ bus = gst_pipeline_get_bus (GST_PIPELINE (pipeline));
+ gst_bus_set_sync_handler (bus, (GstBusSyncHandler) create_window, pipeline,
+ NULL);
+...
+}
+]|
+
+## Two basic usage scenarios
+
+There are two basic usage scenarios: in the simplest case, the application
+uses #playbin or #plasink or knows exactly what particular element is used
+for video output, which is usually the case when the application creates
+the videosink to use (e.g. #xvimagesink, #ximagesink, etc.) itself; in this
+case, the application can just create the videosink element, create and
+realize the window to render the video on and then
+call gst_video_overlay_set_window_handle() directly with the XID or native
+window handle, before starting up the pipeline.
+As #playbin and #playsink implement the video overlay interface and proxy
+it transparently to the actual video sink even if it is created later, this
+case also applies when using these elements.
+
+In the other and more common case, the application does not know in advance
+what GStreamer video sink element will be used for video output. This is
+usually the case when an element such as #autovideosink is used.
+In this case, the video sink element itself is created
+asynchronously from a GStreamer streaming thread some time after the
+pipeline has been started up. When that happens, however, the video sink
+will need to know right then whether to render onto an already existing
+application window or whether to create its own window. This is when it
+posts a prepare-window-handle message, and that is also why this message needs
+to be handled in a sync bus handler which will be called from the streaming
+thread directly (because the video sink will need an answer right then).
+
+As response to the prepare-window-handle element message in the bus sync
+handler, the application may use gst_video_overlay_set_window_handle() to tell
+the video sink to render onto an existing window surface. At this point the
+application should already have obtained the window handle / XID, so it
+just needs to set it. It is generally not advisable to call any GUI toolkit
+functions or window system functions from the streaming thread in which the
+prepare-window-handle message is handled, because most GUI toolkits and
+windowing systems are not thread-safe at all and a lot of care would be
+required to co-ordinate the toolkit and window system calls of the
+different threads (Gtk+ users please note: prior to Gtk+ 2.18
+GDK_WINDOW_XID() was just a simple structure access, so generally fine to do
+within the bus sync handler; this macro was changed to a function call in
+Gtk+ 2.18 and later, which is likely to cause problems when called from a
+sync handler; see below for a better approach without GDK_WINDOW_XID()
+used in the callback).
+
+## GstVideoOverlay and Gtk+
+
+|[
+#include <gst/video/videooverlay.h>
+#include <gtk/gtk.h>
+#ifdef GDK_WINDOWING_X11
+#include <gdk/gdkx.h> // for GDK_WINDOW_XID
+#endif
+#ifdef GDK_WINDOWING_WIN32
+#include <gdk/gdkwin32.h> // for GDK_WINDOW_HWND
+#endif
+...
+static guintptr video_window_handle = 0;
+...
+static GstBusSyncReply
+bus_sync_handler (GstBus * bus, GstMessage * message, gpointer user_data)
+{
+ // ignore anything but 'prepare-window-handle' element messages
+ if (!gst_is_video_overlay_prepare_window_handle_message (message))
+ return GST_BUS_PASS;
+
+ if (video_window_handle != 0) {
+ GstVideoOverlay *overlay;
+
+ // GST_MESSAGE_SRC (message) will be the video sink element
+ overlay = GST_VIDEO_OVERLAY (GST_MESSAGE_SRC (message));
+ gst_video_overlay_set_window_handle (overlay, video_window_handle);
+ } else {
+ g_warning ("Should have obtained video_window_handle by now!");
+ }
+
+ gst_message_unref (message);
+ return GST_BUS_DROP;
+}
+...
+static void
+video_widget_realize_cb (GtkWidget * widget, gpointer data)
+{
+#if GTK_CHECK_VERSION(2,18,0)
+ // Tell Gtk+/Gdk to create a native window for this widget instead of
+ // drawing onto the parent widget.
+ // This is here just for pedagogical purposes, GDK_WINDOW_XID will call
+ // it as well in newer Gtk versions
+ if (!gdk_window_ensure_native (widget->window))
+ g_error ("Couldn't create native window needed for GstVideoOverlay!");
+#endif
+
+#ifdef GDK_WINDOWING_X11
+ {
+ gulong xid = GDK_WINDOW_XID (gtk_widget_get_window (video_window));
+ video_window_handle = xid;
+ }
+#endif
+#ifdef GDK_WINDOWING_WIN32
+ {
+ HWND wnd = GDK_WINDOW_HWND (gtk_widget_get_window (video_window));
+ video_window_handle = (guintptr) wnd;
+ }
+#endif
+}
+...
+int
+main (int argc, char **argv)
+{
+ GtkWidget *video_window;
+ GtkWidget *app_window;
+ ...
+ app_window = gtk_window_new (GTK_WINDOW_TOPLEVEL);
+ ...
+ video_window = gtk_drawing_area_new ();
+ g_signal_connect (video_window, "realize",
+ G_CALLBACK (video_widget_realize_cb), NULL);
+ gtk_widget_set_double_buffered (video_window, FALSE);
+ ...
+ // usually the video_window will not be directly embedded into the
+ // application window like this, but there will be many other widgets
+ // and the video window will be embedded in one of them instead
+ gtk_container_add (GTK_CONTAINER (ap_window), video_window);
+ ...
+ // show the GUI
+ gtk_widget_show_all (app_window);
+
+ // realize window now so that the video window gets created and we can
+ // obtain its XID/HWND before the pipeline is started up and the videosink
+ // asks for the XID/HWND of the window to render onto
+ gtk_widget_realize (video_window);
+
+ // we should have the XID/HWND now
+ g_assert (video_window_handle != 0);
+ ...
+ // set up sync handler for setting the xid once the pipeline is started
+ bus = gst_pipeline_get_bus (GST_PIPELINE (pipeline));
+ gst_bus_set_sync_handler (bus, (GstBusSyncHandler) bus_sync_handler, NULL,
+ NULL);
+ gst_object_unref (bus);
+ ...
+ gst_element_set_state (pipeline, GST_STATE_PLAYING);
+ ...
+}
+]|
+
+## GstVideoOverlay and Qt
+
+|[
+#include <glib.h>
+#include <gst/gst.h>
+#include <gst/video/videooverlay.h>
+
+#include <QApplication>
+#include <QTimer>
+#include <QWidget>
+
+int main(int argc, char *argv[])
+{
+ if (!g_thread_supported ())
+ g_thread_init (NULL);
+
+ gst_init (&argc, &argv);
+ QApplication app(argc, argv);
+ app.connect(&app, SIGNAL(lastWindowClosed()), &app, SLOT(quit ()));
+
+ // prepare the pipeline
+
+ GstElement *pipeline = gst_pipeline_new ("xvoverlay");
+ GstElement *src = gst_element_factory_make ("videotestsrc", NULL);
+ GstElement *sink = gst_element_factory_make ("xvimagesink", NULL);
+ gst_bin_add_many (GST_BIN (pipeline), src, sink, NULL);
+ gst_element_link (src, sink);
+
+ // prepare the ui
+
+ QWidget window;
+ window.resize(320, 240);
+ window.show();
+
+ WId xwinid = window.winId();
+ gst_video_overlay_set_window_handle (GST_VIDEO_OVERLAY (sink), xwinid);
+
+ // run the pipeline
+
+ GstStateChangeReturn sret = gst_element_set_state (pipeline,
+ GST_STATE_PLAYING);
+ if (sret == GST_STATE_CHANGE_FAILURE) {
+ gst_element_set_state (pipeline, GST_STATE_NULL);
+ gst_object_unref (pipeline);
+ // Exit application
+ QTimer::singleShot(0, QApplication::activeWindow(), SLOT(quit()));
+ }
+
+ int ret = app.exec();
+
+ window.hide();
+ gst_element_set_state (pipeline, GST_STATE_NULL);
+ gst_object_unref (pipeline);
+
+ return ret;
+}
+]|
+
+ This helper shall be used by classes implementing the #GstVideoOverlay
+interface that want the render rectangle to be controllable using
+properties. This helper will install "render-rectangle" property into the
+class.
+
+Since 1.14
+
+
+
+
+
+ The class on which the properties will be installed
+
+
+
+ The first free property ID to use
+
+
+
+
+
+ This helper shall be used by classes implementing the #GstVideoOverlay
+interface that want the render rectangle to be controllable using
+properties. This helper will parse and set the render rectangle calling
+gst_video_overlay_set_render_rectangle().
+
+ %TRUE if the @property_id matches the GstVideoOverlay property
+
+Since 1.14
+
+
+
+
+ The instance on which the property is set
+
+
+
+ The highest property ID.
+
+
+
+ The property ID
+
+
+
+ The #GValue to be set
+
+
+
+
+
+ Tell an overlay that it has been exposed. This will redraw the current frame
+in the drawable even if the pipeline is PAUSED.
+
+
+
+
+
+ a #GstVideoOverlay to expose.
+
+
+
+
+
+ Tell an overlay that it should handle events from the window system. These
+events are forwarded upstream as navigation events. In some window system,
+events are not propagated in the window hierarchy if a client is listening
+for them. This method allows you to disable events handling completely
+from the #GstVideoOverlay.
+
+
+
+
+
+ a #GstVideoOverlay to expose.
+
+
+
+ a #gboolean indicating if events should be handled or not.
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ This will call the video overlay's set_window_handle method. You
+should use this method to tell to an overlay to display video output to a
+specific window (e.g. an XWindow on X11). Passing 0 as the @handle will
+tell the overlay to stop using that window and create an internal one.
+
+
+
+
+
+ a #GstVideoOverlay to set the window on.
+
+
+
+ a handle referencing the window.
+
+
+
+
+
+ Tell an overlay that it has been exposed. This will redraw the current frame
+in the drawable even if the pipeline is PAUSED.
+
+
+
+
+
+ a #GstVideoOverlay to expose.
+
+
+
+
+
+ This will post a "have-window-handle" element message on the bus.
+
+This function should only be used by video overlay plugin developers.
+
+
+
+
+
+ a #GstVideoOverlay which got a window
+
+
+
+ a platform-specific handle referencing the window
+
+
+
+
+
+ Tell an overlay that it should handle events from the window system. These
+events are forwarded upstream as navigation events. In some window system,
+events are not propagated in the window hierarchy if a client is listening
+for them. This method allows you to disable events handling completely
+from the #GstVideoOverlay.
+
+
+
+
+
+ a #GstVideoOverlay to expose.
+
+
+
+ a #gboolean indicating if events should be handled or not.
+
+
+
+
+
+ This will post a "prepare-window-handle" element message on the bus
+to give applications an opportunity to call
+gst_video_overlay_set_window_handle() before a plugin creates its own
+window.
+
+This function should only be used by video overlay plugin developers.
+
+
+
+
+
+ a #GstVideoOverlay which does not yet have an Window handle set
+
+
+
+
+
+ Configure a subregion as a video target within the window set by
+gst_video_overlay_set_window_handle(). If this is not used or not supported
+the video will fill the area of the window set as the overlay to 100%.
+By specifying the rectangle, the video can be overlayed to a specific region
+of that window only. After setting the new rectangle one should call
+gst_video_overlay_expose() to force a redraw. To unset the region pass -1 for
+the @width and @height parameters.
+
+This method is needed for non fullscreen video overlay in UI toolkits that
+do not support subwindows.
+
+ %FALSE if not supported by the sink.
+
+
+
+
+ a #GstVideoOverlay
+
+
+
+ the horizontal offset of the render area inside the window
+
+
+
+ the vertical offset of the render area inside the window
+
+
+
+ the width of the render area inside the window
+
+
+
+ the height of the render area inside the window
+
+
+
+
+
+ This will call the video overlay's set_window_handle method. You
+should use this method to tell to an overlay to display video output to a
+specific window (e.g. an XWindow on X11). Passing 0 as the @handle will
+tell the overlay to stop using that window and create an internal one.
+
+
+
+
+
+ a #GstVideoOverlay to set the window on.
+
+
+
+ a handle referencing the window.
+
+
+
+
+
+
+ Functions to create and handle overlay compositions on video buffers.
+
+An overlay composition describes one or more overlay rectangles to be
+blended on top of a video buffer.
+
+This API serves two main purposes:
+
+* it can be used to attach overlay information (subtitles or logos)
+ to non-raw video buffers such as GL/VAAPI/VDPAU surfaces. The actual
+ blending of the overlay can then be done by e.g. the video sink that
+ processes these non-raw buffers.
+
+* it can also be used to blend overlay rectangles on top of raw video
+ buffers, thus consolidating blending functionality for raw video in
+ one place.
+
+Together, this allows existing overlay elements to easily handle raw
+and non-raw video as input in without major changes (once the overlays
+have been put into a #GstOverlayComposition object anyway) - for raw
+video the overlay can just use the blending function to blend the data
+on top of the video, and for surface buffers it can just attach them to
+the buffer and let the sink render the overlays.
+
+ Creates a new video overlay composition object to hold one or more
+overlay rectangles.
+
+ a new #GstVideoOverlayComposition. Unref with
+ gst_video_overlay_composition_unref() when no longer needed.
+
+
+
+
+ a #GstVideoOverlayRectangle to add to the
+ composition
+
+
+
+
+
+ Adds an overlay rectangle to an existing overlay composition object. This
+must be done right after creating the overlay composition.
+
+
+
+
+
+ a #GstVideoOverlayComposition
+
+
+
+ a #GstVideoOverlayRectangle to add to the
+ composition
+
+
+
+
+
+ Blends the overlay rectangles in @comp on top of the raw video data
+contained in @video_buf. The data in @video_buf must be writable and
+mapped appropriately.
+
+Since @video_buf data is read and will be modified, it ought be
+mapped with flag GST_MAP_READWRITE.
+
+
+
+
+
+ a #GstVideoOverlayComposition
+
+
+
+ a #GstVideoFrame containing raw video data in a
+ supported format. It should be mapped using GST_MAP_READWRITE
+
+
+
+
+
+ Makes a copy of @comp and all contained rectangles, so that it is possible
+to modify the composition and contained rectangles (e.g. add additional
+rectangles or change the render co-ordinates or render dimension). The
+actual overlay pixel data buffers contained in the rectangles are not
+copied.
+
+ a new #GstVideoOverlayComposition equivalent
+ to @comp.
+
+
+
+
+ a #GstVideoOverlayComposition to copy
+
+
+
+
+
+ Returns the @n-th #GstVideoOverlayRectangle contained in @comp.
+
+ the @n-th rectangle, or NULL if @n is out of
+ bounds. Will not return a new reference, the caller will need to
+ obtain her own reference using gst_video_overlay_rectangle_ref()
+ if needed.
+
+
+
+
+ a #GstVideoOverlayComposition
+
+
+
+ number of the rectangle to get
+
+
+
+
+
+ Returns the sequence number of this composition. Sequence numbers are
+monotonically increasing and unique for overlay compositions and rectangles
+(meaning there will never be a rectangle with the same sequence number as
+a composition).
+
+ the sequence number of @comp
+
+
+
+
+ a #GstVideoOverlayComposition
+
+
+
+
+
+ Takes ownership of @comp and returns a version of @comp that is writable
+(i.e. can be modified). Will either return @comp right away, or create a
+new writable copy of @comp and unref @comp itself. All the contained
+rectangles will also be copied, but the actual overlay pixel data buffers
+contained in the rectangles are not copied.
+
+ a writable #GstVideoOverlayComposition
+ equivalent to @comp.
+
+
+
+
+ a #GstVideoOverlayComposition to copy
+
+
+
+
+
+ Returns the number of #GstVideoOverlayRectangle<!-- -->s contained in @comp.
+
+ the number of rectangles
+
+
+
+
+ a #GstVideoOverlayComposition
+
+
+
+
+
+
+ Extra buffer metadata describing image overlay data.
+
+ parent #GstMeta
+
+
+
+ the attached #GstVideoOverlayComposition
+
+
+
+
+
+
+
+
+
+ Overlay format flags.
+
+ no flags
+
+
+ RGB are premultiplied by A/255.
+
+
+ a global-alpha value != 1 is set.
+
+
+
+ #GstVideoOverlay interface
+
+ parent interface type.
+
+
+
+
+
+
+
+
+
+ a #GstVideoOverlay to expose.
+
+
+
+
+
+
+
+
+
+
+
+
+ a #GstVideoOverlay to expose.
+
+
+
+ a #gboolean indicating if events should be handled or not.
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ a #GstVideoOverlay to set the window on.
+
+
+
+ a handle referencing the window.
+
+
+
+
+
+
+
+
+
+ An opaque video overlay rectangle object. A rectangle contains a single
+overlay rectangle which can be added to a composition.
+
+ Creates a new video overlay rectangle with ARGB or AYUV pixel data.
+The layout in case of ARGB of the components in memory is B-G-R-A
+on little-endian platforms
+(corresponding to #GST_VIDEO_FORMAT_BGRA) and A-R-G-B on big-endian
+platforms (corresponding to #GST_VIDEO_FORMAT_ARGB). In other words,
+pixels are treated as 32-bit words and the lowest 8 bits then contain
+the blue component value and the highest 8 bits contain the alpha
+component value. Unless specified in the flags, the RGB values are
+non-premultiplied. This is the format that is used by most hardware,
+and also many rendering libraries such as Cairo, for example.
+The pixel data buffer must have #GstVideoMeta set.
+
+ a new #GstVideoOverlayRectangle. Unref with
+ gst_video_overlay_rectangle_unref() when no longer needed.
+
+
+
+
+ a #GstBuffer pointing to the pixel memory
+
+
+
+ the X co-ordinate on the video where the top-left corner of this
+ overlay rectangle should be rendered to
+
+
+
+ the Y co-ordinate on the video where the top-left corner of this
+ overlay rectangle should be rendered to
+
+
+
+ the render width of this rectangle on the video
+
+
+
+ the render height of this rectangle on the video
+
+
+
+ flags
+
+
+
+
+
+ Makes a copy of @rectangle, so that it is possible to modify it
+(e.g. to change the render co-ordinates or render dimension). The
+actual overlay pixel data buffers contained in the rectangle are not
+copied.
+
+ a new #GstVideoOverlayRectangle equivalent
+ to @rectangle.
+
+
+
+
+ a #GstVideoOverlayRectangle to copy
+
+
+
+
+
+ Retrieves the flags associated with a #GstVideoOverlayRectangle.
+This is useful if the caller can handle both premultiplied alpha and
+non premultiplied alpha, for example. By knowing whether the rectangle
+uses premultiplied or not, it can request the pixel data in the format
+it is stored in, to avoid unnecessary conversion.
+
+ the #GstVideoOverlayFormatFlags associated with the rectangle.
+
+
+
+
+ a #GstVideoOverlayRectangle
+
+
+
+
+
+ Retrieves the global-alpha value associated with a #GstVideoOverlayRectangle.
+
+ the global-alpha value associated with the rectangle.
+
+
+
+
+ a #GstVideoOverlayRectangle
+
+
+
+
+
+
+ a #GstBuffer holding the ARGB pixel data with
+ width and height of the render dimensions as per
+ gst_video_overlay_rectangle_get_render_rectangle(). This function does
+ not return a reference, the caller should obtain a reference of her own
+ with gst_buffer_ref() if needed.
+
+
+
+
+ a #GstVideoOverlayRectangle
+
+
+
+ flags
+ If a global_alpha value != 1 is set for the rectangle, the caller
+ should set the #GST_VIDEO_OVERLAY_FORMAT_FLAG_GLOBAL_ALPHA flag
+ if he wants to apply global-alpha himself. If the flag is not set
+ global_alpha is applied internally before returning the pixel-data.
+
+
+
+
+
+
+ a #GstBuffer holding the AYUV pixel data with
+ width and height of the render dimensions as per
+ gst_video_overlay_rectangle_get_render_rectangle(). This function does
+ not return a reference, the caller should obtain a reference of her own
+ with gst_buffer_ref() if needed.
+
+
+
+
+ a #GstVideoOverlayRectangle
+
+
+
+ flags
+ If a global_alpha value != 1 is set for the rectangle, the caller
+ should set the #GST_VIDEO_OVERLAY_FORMAT_FLAG_GLOBAL_ALPHA flag
+ if he wants to apply global-alpha himself. If the flag is not set
+ global_alpha is applied internally before returning the pixel-data.
+
+
+
+
+
+
+ a #GstBuffer holding the pixel data with
+ format as originally provided and specified in video meta with
+ width and height of the render dimensions as per
+ gst_video_overlay_rectangle_get_render_rectangle(). This function does
+ not return a reference, the caller should obtain a reference of her own
+ with gst_buffer_ref() if needed.
+
+
+
+
+ a #GstVideoOverlayRectangle
+
+
+
+ flags
+ If a global_alpha value != 1 is set for the rectangle, the caller
+ should set the #GST_VIDEO_OVERLAY_FORMAT_FLAG_GLOBAL_ALPHA flag
+ if he wants to apply global-alpha himself. If the flag is not set
+ global_alpha is applied internally before returning the pixel-data.
+
+
+
+
+
+ Retrieves the pixel data as it is. This is useful if the caller can
+do the scaling itself when handling the overlaying. The rectangle will
+need to be scaled to the render dimensions, which can be retrieved using
+gst_video_overlay_rectangle_get_render_rectangle().
+
+ a #GstBuffer holding the ARGB pixel data with
+ #GstVideoMeta set. This function does not return a reference, the caller
+ should obtain a reference of her own with gst_buffer_ref() if needed.
+
+
+
+
+ a #GstVideoOverlayRectangle
+
+
+
+ flags.
+ If a global_alpha value != 1 is set for the rectangle, the caller
+ should set the #GST_VIDEO_OVERLAY_FORMAT_FLAG_GLOBAL_ALPHA flag
+ if he wants to apply global-alpha himself. If the flag is not set
+ global_alpha is applied internally before returning the pixel-data.
+
+
+
+
+
+ Retrieves the pixel data as it is. This is useful if the caller can
+do the scaling itself when handling the overlaying. The rectangle will
+need to be scaled to the render dimensions, which can be retrieved using
+gst_video_overlay_rectangle_get_render_rectangle().
+
+ a #GstBuffer holding the AYUV pixel data with
+ #GstVideoMeta set. This function does not return a reference, the caller
+ should obtain a reference of her own with gst_buffer_ref() if needed.
+
+
+
+
+ a #GstVideoOverlayRectangle
+
+
+
+ flags.
+ If a global_alpha value != 1 is set for the rectangle, the caller
+ should set the #GST_VIDEO_OVERLAY_FORMAT_FLAG_GLOBAL_ALPHA flag
+ if he wants to apply global-alpha himself. If the flag is not set
+ global_alpha is applied internally before returning the pixel-data.
+
+
+
+
+
+ Retrieves the pixel data as it is. This is useful if the caller can
+do the scaling itself when handling the overlaying. The rectangle will
+need to be scaled to the render dimensions, which can be retrieved using
+gst_video_overlay_rectangle_get_render_rectangle().
+
+ a #GstBuffer holding the pixel data with
+ #GstVideoMeta set. This function does not return a reference, the caller
+ should obtain a reference of her own with gst_buffer_ref() if needed.
+
+
+
+
+ a #GstVideoOverlayRectangle
+
+
+
+ flags.
+ If a global_alpha value != 1 is set for the rectangle, the caller
+ should set the #GST_VIDEO_OVERLAY_FORMAT_FLAG_GLOBAL_ALPHA flag
+ if he wants to apply global-alpha himself. If the flag is not set
+ global_alpha is applied internally before returning the pixel-data.
+
+
+
+
+
+ Retrieves the render position and render dimension of the overlay
+rectangle on the video.
+
+ TRUE if valid render dimensions were retrieved.
+
+
+
+
+ a #GstVideoOverlayRectangle
+
+
+
+ address where to store the X render offset
+
+
+
+ address where to store the Y render offset
+
+
+
+ address where to store the render width
+
+
+
+ address where to store the render height
+
+
+
+
+
+ Returns the sequence number of this rectangle. Sequence numbers are
+monotonically increasing and unique for overlay compositions and rectangles
+(meaning there will never be a rectangle with the same sequence number as
+a composition).
+
+Using the sequence number of a rectangle as an indicator for changed
+pixel-data of a rectangle is dangereous. Some API calls, like e.g.
+gst_video_overlay_rectangle_set_global_alpha(), automatically update
+the per rectangle sequence number, which is misleading for renderers/
+consumers, that handle global-alpha themselves. For them the
+pixel-data returned by gst_video_overlay_rectangle_get_pixels_*()
+wont be different for different global-alpha values. In this case a
+renderer could also use the GstBuffer pointers as a hint for changed
+pixel-data.
+
+ the sequence number of @rectangle
+
+
+
+
+ a #GstVideoOverlayRectangle
+
+
+
+
+
+ Sets the global alpha value associated with a #GstVideoOverlayRectangle. Per-
+pixel alpha values are multiplied with this value. Valid
+values: 0 <= global_alpha <= 1; 1 to deactivate.
+
+@rectangle must be writable, meaning its refcount must be 1. You can
+make the rectangles inside a #GstVideoOverlayComposition writable using
+gst_video_overlay_composition_make_writable() or
+gst_video_overlay_composition_copy().
+
+
+
+
+
+ a #GstVideoOverlayRectangle
+
+
+
+ Global alpha value (0 to 1.0)
+
+
+
+
+
+ Sets the render position and dimensions of the rectangle on the video.
+This function is mainly for elements that modify the size of the video
+in some way (e.g. through scaling or cropping) and need to adjust the
+details of any overlays to match the operation that changed the size.
+
+@rectangle must be writable, meaning its refcount must be 1. You can
+make the rectangles inside a #GstVideoOverlayComposition writable using
+gst_video_overlay_composition_make_writable() or
+gst_video_overlay_composition_copy().
+
+
+
+
+
+ a #GstVideoOverlayRectangle
+
+
+
+ render X position of rectangle on video
+
+
+
+ render Y position of rectangle on video
+
+
+
+ render width of rectangle
+
+
+
+ render height of rectangle
+
+
+
+
+
+
+ The different flags that can be used when packing and unpacking.
+
+ No flag
+
+
+ When the source has a smaller depth
+ than the target format, set the least significant bits of the target
+ to 0. This is likely sightly faster but less accurate. When this flag
+ is not specified, the most significant bits of the source are duplicated
+ in the least significant bits of the destination.
+
+
+ The source is interlaced. The unpacked
+ format will be interlaced as well with each line containing
+ information from alternating fields. (Since 1.2)
+
+
+
+ Different primaries conversion modes
+
+ disable conversion between primaries
+
+
+ do conversion between primaries only
+ when it can be merged with color matrix conversion.
+
+
+ fast conversion between primaries
+
+
+
+ Helper structure representing a rectangular area.
+
+ X coordinate of rectangle's top-left point
+
+
+
+ Y coordinate of rectangle's top-left point
+
+
+
+ width of the rectangle
+
+
+
+ height of the rectangle
+
+
+
+
+ Extra buffer metadata describing an image region of interest
+
+ parent #GstMeta
+
+
+
+ GQuark describing the semantic of the Roi (f.i. a face, a pedestrian)
+
+
+
+ identifier of this particular ROI
+
+
+
+ identifier of its parent ROI, used f.i. for ROI hierarchisation.
+
+
+
+ x component of upper-left corner
+
+
+
+ y component of upper-left corner
+
+
+
+ bounding box width
+
+
+
+ bounding box height
+
+
+
+ list of #GstStructure containing element-specific params for downstream, see gst_video_region_of_interest_meta_add_params(). (Since: 1.14)
+
+
+
+
+
+ Attach element-specific parameters to @meta meant to be used by downstream
+elements which may handle this ROI.
+The name of @s is used to identify the element these parameters are meant for.
+
+This is typically used to tell encoders how they should encode this specific region.
+For example, a structure named "roi/x264enc" could be used to give the
+QP offsets this encoder should use when encoding the region described in @meta.
+Multiple parameters can be defined for the same meta so different encoders
+can be supported by cross platform applications).
+
+
+
+
+
+ a #GstVideoRegionOfInterestMeta
+
+
+
+ a #GstStructure
+
+
+
+
+
+ Retrieve the parameter for @meta having @name as structure name,
+or %NULL if there is none.
+
+See also: gst_video_region_of_interest_meta_add_param()
+
+ a #GstStructure
+
+
+
+
+ a #GstVideoRegionOfInterestMeta
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ #GstVideoResampler is a structure which holds the information
+required to perform various kinds of resampling filtering.
+
+ the input size
+
+
+
+ the output size
+
+
+
+ the maximum number of taps
+
+
+
+ the number of phases
+
+
+
+ array with the source offset for each output element
+
+
+
+ array with the phase to use for each output element
+
+
+
+ array with new number of taps for each phase
+
+
+
+ the taps for all phases
+
+
+
+
+
+
+
+
+ Clear a previously initialized #GstVideoResampler @resampler.
+
+
+
+
+
+ a #GstVideoResampler
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Different resampler flags.
+
+ no flags
+
+
+ when no taps are given, half the
+ number of calculated taps. This can be used when making scalers
+ for the different fields of an interlaced picture. Since 1.10
+
+
+
+ Different subsampling and upsampling methods
+
+ Duplicates the samples when
+ upsampling and drops when downsampling
+
+
+ Uses linear interpolation to reconstruct
+ missing samples and averaging to downsample
+
+
+ Uses cubic interpolation
+
+
+ Uses sinc interpolation
+
+
+ Uses lanczos interpolation
+
+
+
+ #GstVideoScaler is a utility object for rescaling and resampling
+video frames using various interpolation / sampling methods.
+
+ Scale a rectangle of pixels in @src with @src_stride to @dest with
+@dest_stride using the horizontal scaler @hscaler and the vertical
+scaler @vscale.
+
+One or both of @hscale and @vscale can be NULL to only perform scaling in
+one dimension or do a copy without scaling.
+
+@x and @y are the coordinates in the destination image to process.
+
+
+
+
+
+ a horzontal #GstVideoScaler
+
+
+
+ a vertical #GstVideoScaler
+
+
+
+ a #GstVideoFormat for @srcs and @dest
+
+
+
+ source pixels
+
+
+
+ source pixels stride
+
+
+
+ destination pixels
+
+
+
+ destination pixels stride
+
+
+
+ the horizontal destination offset
+
+
+
+ the vertical destination offset
+
+
+
+ the number of output pixels to scale
+
+
+
+ the number of output lines to scale
+
+
+
+
+
+ Combine a scaler for Y and UV into one scaler for the packed @format.
+
+ a new horizontal videoscaler for @format.
+
+
+
+
+ a scaler for the Y component
+
+
+
+ a scaler for the U and V components
+
+
+
+ the input video format
+
+
+
+ the output video format
+
+
+
+
+
+ Free a previously allocated #GstVideoScaler @scale.
+
+
+
+
+
+ a #GstVideoScaler
+
+
+
+
+
+ For a given pixel at @out_offset, get the first required input pixel at
+@in_offset and the @n_taps filter coefficients.
+
+Note that for interlaced content, @in_offset needs to be incremented with
+2 to get the next input line.
+
+ an array of @n_tap gdouble values with filter coefficients.
+
+
+
+
+ a #GstVideoScaler
+
+
+
+ an output offset
+
+
+
+ result input offset
+
+
+
+ result n_taps
+
+
+
+
+
+ Get the maximum number of taps for @scale.
+
+ the maximum number of taps
+
+
+
+
+ a #GstVideoScaler
+
+
+
+
+
+ Horizontally scale the pixels in @src to @dest, starting from @dest_offset
+for @width samples.
+
+
+
+
+
+ a #GstVideoScaler
+
+
+
+ a #GstVideoFormat for @src and @dest
+
+
+
+ source pixels
+
+
+
+ destination pixels
+
+
+
+ the horizontal destination offset
+
+
+
+ the number of pixels to scale
+
+
+
+
+
+ Vertically combine @width pixels in the lines in @src_lines to @dest.
+@dest is the location of the target line at @dest_offset and
+@srcs are the input lines for @dest_offset, as obtained with
+gst_video_scaler_get_info().
+
+
+
+
+
+ a #GstVideoScaler
+
+
+
+ a #GstVideoFormat for @srcs and @dest
+
+
+
+ source pixels lines
+
+
+
+ destination pixels
+
+
+
+ the vertical destination offset
+
+
+
+ the number of pixels to scale
+
+
+
+
+
+ Make a new @method video scaler. @in_size source lines/pixels will
+be scaled to @out_size destination lines/pixels.
+
+@n_taps specifies the amount of pixels to use from the source for one output
+pixel. If n_taps is 0, this function chooses a good value automatically based
+on the @method and @in_size/@out_size.
+
+ a #GstVideoResample
+
+
+
+
+ a #GstVideoResamplerMethod
+
+
+
+ #GstVideoScalerFlags
+
+
+
+ number of taps to use
+
+
+
+ number of source elements
+
+
+
+ number of destination elements
+
+
+
+ extra options
+
+
+
+
+
+
+ Different scale flags.
+
+ no flags
+
+
+ Set up a scaler for interlaced content
+
+
+
+ Provides useful functions and a base class for video sinks.
+
+GstVideoSink will configure the default base sink to drop frames that
+arrive later than 20ms as this is considered the default threshold for
+observing out-of-sync frames.
+
+ Takes @src rectangle and position it at the center of @dst rectangle with or
+without @scaling. It handles clipping if the @src rectangle is bigger than
+the @dst one and @scaling is set to FALSE.
+
+
+
+
+
+ the #GstVideoRectangle describing the source area
+
+
+
+ the #GstVideoRectangle describing the destination area
+
+
+
+ a pointer to a #GstVideoRectangle which will receive the result area
+
+
+
+ a #gboolean indicating if scaling should be applied or not
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Whether to show video frames during preroll. If set to %FALSE, video
+frames will only be rendered in PLAYING state.
+
+
+
+
+
+
+ video width (derived class needs to set this)
+
+
+
+ video height (derived class needs to set this)
+
+
+
+
+
+
+
+
+
+
+
+
+ The video sink class structure. Derived classes should override the
+@show_frame virtual function.
+
+ the parent class structure
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Enum value describing the available tiling modes.
+
+ Unknown or unset tile mode
+
+
+ Every four adjacent blocks - two
+ horizontally and two vertically are grouped together and are located
+ in memory in Z or flipped Z order. In case of odd rows, the last row
+ of blocks is arranged in linear order.
+
+
+
+ Enum value describing the most common tiling types.
+
+ Tiles are indexed. Use
+ gst_video_tile_get_index () to retrieve the tile at the requested
+ coordinates.
+
+
+
+ @field_count must be 0 for progressive video and 1 or 2 for interlaced.
+
+A representation of a SMPTE time code.
+
+@hours must be positive and less than 24. Will wrap around otherwise.
+@minutes and @seconds must be positive and less than 60.
+@frames must be less than or equal to @config.fps_n / @config.fps_d
+These values are *NOT* automatically normalized.
+
+ the corresponding #GstVideoTimeCodeConfig
+
+
+
+ the hours field of #GstVideoTimeCode
+
+
+
+ the minutes field of #GstVideoTimeCode
+
+
+
+ the seconds field of #GstVideoTimeCode
+
+
+
+ the frames field of #GstVideoTimeCode
+
+
+
+ Interlaced video field count
+
+
+
+ @field_count is 0 for progressive, 1 or 2 for interlaced.
+@latest_daiy_jam reference is stolen from caller.
+
+ a new #GstVideoTimeCode with the given values.
+The values are not checked for being in a valid range. To see if your
+timecode actually has valid content, use #gst_video_time_code_is_valid.
+
+
+
+
+ Numerator of the frame rate
+
+
+
+ Denominator of the frame rate
+
+
+
+ The latest daily jam of the #GstVideoTimeCode
+
+
+
+ #GstVideoTimeCodeFlags
+
+
+
+ the hours field of #GstVideoTimeCode
+
+
+
+ the minutes field of #GstVideoTimeCode
+
+
+
+ the seconds field of #GstVideoTimeCode
+
+
+
+ the frames field of #GstVideoTimeCode
+
+
+
+ Interlaced video field count
+
+
+
+
+
+
+ a new empty #GstVideoTimeCode
+
+
+
+
+ The resulting config->latest_daily_jam is set to
+midnight, and timecode is set to the given time.
+
+ the #GVideoTimeCode representation of @dt.
+
+
+
+
+ Numerator of the frame rate
+
+
+
+ Denominator of the frame rate
+
+
+
+ #GDateTime to convert
+
+
+
+ #GstVideoTimeCodeFlags
+
+
+
+ Interlaced video field count
+
+
+
+
+
+
+ a new #GstVideoTimeCode from the given string
+
+
+
+
+ The string that represents the #GstVideoTimeCode
+
+
+
+
+
+ Adds or subtracts @frames amount of frames to @tc. tc needs to
+contain valid data, as verified by #gst_video_time_code_is_valid.
+
+
+
+
+
+ a valid #GstVideoTimeCode
+
+
+
+ How many frames to add or subtract
+
+
+
+
+
+ This makes a component-wise addition of @tc_inter to @tc. For example,
+adding ("01:02:03:04", "00:01:00:00") will return "01:03:03:04".
+When it comes to drop-frame timecodes,
+adding ("00:00:00;00", "00:01:00:00") will return "00:01:00;02"
+because of drop-frame oddities. However,
+adding ("00:09:00;02", "00:01:00:00") will return "00:10:00;00"
+because this time we can have an exact minute.
+
+ A new #GstVideoTimeCode with @tc_inter added.
+
+
+
+
+ The #GstVideoTimeCode where the diff should be added. This
+must contain valid timecode values.
+
+
+
+ The #GstVideoTimeCodeInterval to add to @tc.
+The interval must contain valid values, except that for drop-frame
+timecode, it may also contain timecodes which would normally
+be dropped. These are then corrected to the next reasonable timecode.
+
+
+
+
+
+ Initializes @tc with empty/zero/NULL values.
+
+
+
+
+
+ a #GstVideoTimeCode
+
+
+
+
+
+ Compares @tc1 and @tc2 . If both have latest daily jam information, it is
+taken into account. Otherwise, it is assumed that the daily jam of both
+@tc1 and @tc2 was at the same time. Both time codes must be valid.
+
+ 1 if @tc1 is after @tc2, -1 if @tc1 is before @tc2, 0 otherwise.
+
+
+
+
+ a #GstVideoTimeCode
+
+
+
+ another #GstVideoTimeCode
+
+
+
+
+
+
+ a new #GstVideoTimeCode with the same values as @tc .
+
+
+
+
+ a #GstVideoTimeCode
+
+
+
+
+
+
+ how many frames have passed since the daily jam of @tc .
+
+
+
+
+ a valid #GstVideoTimeCode
+
+
+
+
+
+ Frees @tc .
+
+
+
+
+
+ a #GstVideoTimeCode
+
+
+
+
+
+ Adds one frame to @tc .
+
+
+
+
+
+ a valid #GstVideoTimeCode
+
+
+
+
+
+ @field_count is 0 for progressive, 1 or 2 for interlaced.
+@latest_daiy_jam reference is stolen from caller.
+
+Initializes @tc with the given values.
+The values are not checked for being in a valid range. To see if your
+timecode actually has valid content, use #gst_video_time_code_is_valid.
+
+
+
+
+
+ a #GstVideoTimeCode
+
+
+
+ Numerator of the frame rate
+
+
+
+ Denominator of the frame rate
+
+
+
+ The latest daily jam of the #GstVideoTimeCode
+
+
+
+ #GstVideoTimeCodeFlags
+
+
+
+ the hours field of #GstVideoTimeCode
+
+
+
+ the minutes field of #GstVideoTimeCode
+
+
+
+ the seconds field of #GstVideoTimeCode
+
+
+
+ the frames field of #GstVideoTimeCode
+
+
+
+ Interlaced video field count
+
+
+
+
+
+ The resulting config->latest_daily_jam is set to
+midnight, and timecode is set to the given time.
+
+
+
+
+
+ a #GstVideoTimeCode
+
+
+
+ Numerator of the frame rate
+
+
+
+ Denominator of the frame rate
+
+
+
+ #GDateTime to convert
+
+
+
+ #GstVideoTimeCodeFlags
+
+
+
+ Interlaced video field count
+
+
+
+
+
+
+ whether @tc is a valid timecode (supported frame rate,
+hours/minutes/seconds/frames not overflowing)
+
+
+
+
+ #GstVideoTimeCode to check
+
+
+
+
+
+
+ how many nsec have passed since the daily jam of @tc .
+
+
+
+
+ a valid #GstVideoTimeCode
+
+
+
+
+
+ The @tc.config->latest_daily_jam is required to be non-NULL.
+
+ the #GDateTime representation of @tc.
+
+
+
+
+ A valid #GstVideoTimeCode to convert
+
+
+
+
+
+
+ the SMPTE ST 2059-1:2015 string representation of @tc. That will
+take the form hh:mm:ss:ff . The last separator (between seconds and frames)
+may vary:
+
+';' for drop-frame, non-interlaced content and for drop-frame interlaced
+field 2
+',' for drop-frame interlaced field 1
+':' for non-drop-frame, non-interlaced content and for non-drop-frame
+interlaced field 2
+'.' for non-drop-frame interlaced field 1
+
+
+
+
+ #GstVideoTimeCode to convert
+
+
+
+
+
+
+ Supported frame rates: 30000/1001, 60000/1001 (both with and without drop
+frame), and integer frame rates e.g. 25/1, 30/1, 50/1, 60/1.
+
+The configuration of the time code.
+
+ Numerator of the frame rate
+
+
+
+ Denominator of the frame rate
+
+
+
+ the corresponding #GstVideoTimeCodeFlags
+
+
+
+ The latest daily jam information, if present, or NULL
+
+
+
+
+ Flags related to the time code information.
+For drop frame, only 30000/1001 and 60000/1001 frame rates are supported.
+
+ No flags
+
+
+ Whether we have drop frame rate
+
+
+ Whether we have interlaced video
+
+
+
+ A representation of a difference between two #GstVideoTimeCode instances.
+Will not necessarily correspond to a real timecode (e.g. 00:00:10;00)
+
+ the hours field of #GstVideoTimeCodeInterval
+
+
+
+ the minutes field of #GstVideoTimeCodeInterval
+
+
+
+ the seconds field of #GstVideoTimeCodeInterval
+
+
+
+ the frames field of #GstVideoTimeCodeInterval
+
+
+
+
+ a new #GstVideoTimeCodeInterval with the given values.
+
+
+
+
+ the hours field of #GstVideoTimeCodeInterval
+
+
+
+ the minutes field of #GstVideoTimeCodeInterval
+
+
+
+ the seconds field of #GstVideoTimeCodeInterval
+
+
+
+ the frames field of #GstVideoTimeCodeInterval
+
+
+
+
+
+ @tc_inter_str must only have ":" as separators.
+
+ a new #GstVideoTimeCodeInterval from the given string
+
+
+
+
+ The string that represents the #GstVideoTimeCodeInterval
+
+
+
+
+
+ Initializes @tc with empty/zero/NULL values.
+
+
+
+
+
+ a #GstVideoTimeCodeInterval
+
+
+
+
+
+
+ a new #GstVideoTimeCodeInterval with the same values as @tc .
+
+
+
+
+ a #GstVideoTimeCodeInterval
+
+
+
+
+
+ Frees @tc .
+
+
+
+
+
+ a #GstVideoTimeCodeInterval
+
+
+
+
+
+ Initializes @tc with the given values.
+
+
+
+
+
+ a #GstVideoTimeCodeInterval
+
+
+
+ the hours field of #GstVideoTimeCodeInterval
+
+
+
+ the minutes field of #GstVideoTimeCodeInterval
+
+
+
+ the seconds field of #GstVideoTimeCodeInterval
+
+
+
+ the frames field of #GstVideoTimeCodeInterval
+
+
+
+
+
+
+ Extra buffer metadata describing the GstVideoTimeCode of the frame.
+
+Each frame is assumed to have its own timecode, i.e. they are not
+automatically incremented/interpolated.
+
+ parent #GstMeta
+
+
+
+ the GstVideoTimeCode to attach
+
+
+
+
+
+
+
+
+
+ The video transfer function defines the formula for converting between
+non-linear RGB (R'G'B') and linear RGB
+
+ unknown transfer function
+
+
+ linear RGB, gamma 1.0 curve
+
+
+ Gamma 1.8 curve
+
+
+ Gamma 2.0 curve
+
+
+ Gamma 2.2 curve
+
+
+ Gamma 2.2 curve with a linear segment in the lower
+ range
+
+
+ Gamma 2.2 curve with a linear segment in the
+ lower range
+
+
+ Gamma 2.4 curve with a linear segment in the lower
+ range
+
+
+ Gamma 2.8 curve
+
+
+ Logarithmic transfer characteristic
+ 100:1 range
+
+
+ Logarithmic transfer characteristic
+ 316.22777:1 range
+
+
+ Gamma 2.2 curve with a linear segment in the lower
+ range. Used for BT.2020 with 12 bits per
+ component. Since: 1.6
+
+
+ Gamma 2.19921875. Since: 1.8
+
+
+
+ Attaches GstVideoAffineTransformationMeta metadata to @buffer with
+the given parameters.
+
+ the #GstVideoAffineTransformationMeta on @buffer.
+
+
+
+
+ a #GstBuffer
+
+
+
+
+
+ Attaches GstVideoGLTextureUploadMeta metadata to @buffer with the given
+parameters.
+
+ the #GstVideoGLTextureUploadMeta on @buffer.
+
+
+
+
+ a #GstBuffer
+
+
+
+ the #GstVideoGLTextureOrientation
+
+
+
+ the number of textures
+
+
+
+ array of #GstVideoGLTextureType
+
+
+
+ the function to upload the buffer to a specific texture ID
+
+
+
+ user data for the implementor of @upload
+
+
+
+ function to copy @user_data
+
+
+
+ function to free @user_data
+
+
+
+
+
+ Attaches GstVideoMeta metadata to @buffer with the given parameters and the
+default offsets and strides for @format and @width x @height.
+
+This function calculates the default offsets and strides and then calls
+gst_buffer_add_video_meta_full() with them.
+
+ the #GstVideoMeta on @buffer.
+
+
+
+
+ a #GstBuffer
+
+
+
+ #GstVideoFrameFlags
+
+
+
+ a #GstVideoFormat
+
+
+
+ the width
+
+
+
+ the height
+
+
+
+
+
+ Attaches GstVideoMeta metadata to @buffer with the given parameters.
+
+ the #GstVideoMeta on @buffer.
+
+
+
+
+ a #GstBuffer
+
+
+
+ #GstVideoFrameFlags
+
+
+
+ a #GstVideoFormat
+
+
+
+ the width
+
+
+
+ the height
+
+
+
+ number of planes
+
+
+
+ offset of each plane
+
+
+
+ stride of each plane
+
+
+
+
+
+ Sets an overlay composition on a buffer. The buffer will obtain its own
+reference to the composition, meaning this function does not take ownership
+of @comp.
+
+ a #GstVideoOverlayCompositionMeta
+
+
+
+
+ a #GstBuffer
+
+
+
+ a #GstVideoOverlayComposition
+
+
+
+
+
+ Attaches #GstVideoRegionOfInterestMeta metadata to @buffer with the given
+parameters.
+
+ the #GstVideoRegionOfInterestMeta on @buffer.
+
+
+
+
+ a #GstBuffer
+
+
+
+ Type of the region of interest (e.g. "face")
+
+
+
+ X position
+
+
+
+ Y position
+
+
+
+ width
+
+
+
+ height
+
+
+
+
+
+ Attaches #GstVideoRegionOfInterestMeta metadata to @buffer with the given
+parameters.
+
+ the #GstVideoRegionOfInterestMeta on @buffer.
+
+
+
+
+ a #GstBuffer
+
+
+
+ Type of the region of interest (e.g. "face")
+
+
+
+ X position
+
+
+
+ Y position
+
+
+
+ width
+
+
+
+ height
+
+
+
+
+
+ Attaches #GstVideoTimeCodeMeta metadata to @buffer with the given
+parameters.
+
+ the #GstVideoTimeCodeMeta on @buffer.
+
+
+
+
+ a #GstBuffer
+
+
+
+ a #GstVideoTimeCode
+
+
+
+
+
+ Attaches #GstVideoTimeCodeMeta metadata to @buffer with the given
+parameters.
+
+ the #GstVideoTimeCodeMeta on @buffer.
+
+
+
+
+ a #GstBuffer
+
+
+
+ framerate numerator
+
+
+
+ framerate denominator
+
+
+
+ a #GDateTime for the latest daily jam
+
+
+
+ a #GstVideoTimeCodeFlags
+
+
+
+ hours since the daily jam
+
+
+
+ minutes since the daily jam
+
+
+
+ seconds since the daily jam
+
+
+
+ frames since the daily jam
+
+
+
+ fields since the daily jam
+
+
+
+
+
+ Find the #GstVideoMeta on @buffer with the lowest @id.
+
+Buffers can contain multiple #GstVideoMeta metadata items when dealing with
+multiview buffers.
+
+ the #GstVideoMeta with lowest id (usually 0) or %NULL when there
+is no such metadata on @buffer.
+
+
+
+
+ a #GstBuffer
+
+
+
+
+
+ Find the #GstVideoMeta on @buffer with the given @id.
+
+Buffers can contain multiple #GstVideoMeta metadata items when dealing with
+multiview buffers.
+
+ the #GstVideoMeta with @id or %NULL when there is no such metadata
+on @buffer.
+
+
+
+
+ a #GstBuffer
+
+
+
+ a metadata id
+
+
+
+
+
+ Find the #GstVideoRegionOfInterestMeta on @buffer with the given @id.
+
+Buffers can contain multiple #GstVideoRegionOfInterestMeta metadata items if
+multiple regions of interests are marked on a frame.
+
+ the #GstVideoRegionOfInterestMeta with @id or %NULL when there is
+no such metadata on @buffer.
+
+
+
+
+ a #GstBuffer
+
+
+
+ a metadata id
+
+
+
+
+
+ Get the video alignment from the bufferpool configuration @config in
+in @align
+
+ %TRUE if @config could be parsed correctly.
+
+
+
+
+ a #GstStructure
+
+
+
+ a #GstVideoAlignment
+
+
+
+
+
+ Set the video alignment in @align to the bufferpool configuration
+@config
+
+
+
+
+
+ a #GstStructure
+
+
+
+ a #GstVideoAlignment
+
+
+
+
+
+ Convenience function to check if the given message is a
+"prepare-window-handle" message from a #GstVideoOverlay.
+
+ whether @msg is a "prepare-window-handle" message
+
+
+
+
+ a #GstMessage
+
+
+
+
+
+ Inspect a #GstEvent and return the #GstNavigationEventType of the event, or
+#GST_NAVIGATION_EVENT_INVALID if the event is not a #GstNavigation event.
+
+
+
+
+
+ A #GstEvent to inspect.
+
+
+
+
+
+ Inspect a #GstNavigation command event and retrieve the enum value of the
+associated command.
+
+ TRUE if the navigation command could be extracted, otherwise FALSE.
+
+
+
+
+ A #GstEvent to inspect.
+
+
+
+ Pointer to GstNavigationCommand to receive the
+ type of the navigation event.
+
+
+
+
+
+
+
+
+
+
+ A #GstEvent to inspect.
+
+
+
+ A pointer to a location to receive
+ the string identifying the key press. The returned string is owned by the
+ event, and valid only until the event is unreffed.
+
+
+
+
+
+ Retrieve the details of either a #GstNavigation mouse button press event or
+a mouse button release event. Determine which type the event is using
+gst_navigation_event_get_type() to retrieve the #GstNavigationEventType.
+
+ TRUE if the button number and both coordinates could be extracted,
+ otherwise FALSE.
+
+
+
+
+ A #GstEvent to inspect.
+
+
+
+ Pointer to a gint that will receive the button
+ number associated with the event.
+
+
+
+ Pointer to a gdouble to receive the x coordinate of the
+ mouse button event.
+
+
+
+ Pointer to a gdouble to receive the y coordinate of the
+ mouse button event.
+
+
+
+
+
+ Inspect a #GstNavigation mouse movement event and extract the coordinates
+of the event.
+
+ TRUE if both coordinates could be extracted, otherwise FALSE.
+
+
+
+
+ A #GstEvent to inspect.
+
+
+
+ Pointer to a gdouble to receive the x coordinate of the
+ mouse movement.
+
+
+
+ Pointer to a gdouble to receive the y coordinate of the
+ mouse movement.
+
+
+
+
+
+ Check a bus message to see if it is a #GstNavigation event, and return
+the #GstNavigationMessageType identifying the type of the message if so.
+
+ The type of the #GstMessage, or
+#GST_NAVIGATION_MESSAGE_INVALID if the message is not a #GstNavigation
+notification.
+
+
+
+
+ A #GstMessage to inspect.
+
+
+
+
+
+ Creates a new #GstNavigation message with type
+#GST_NAVIGATION_MESSAGE_ANGLES_CHANGED for notifying an application
+that the current angle, or current number of angles available in a
+multiangle video has changed.
+
+ The new #GstMessage.
+
+
+
+
+ A #GstObject to set as source of the new message.
+
+
+
+ The currently selected angle.
+
+
+
+ The number of viewing angles now available.
+
+
+
+
+
+ Creates a new #GstNavigation message with type
+#GST_NAVIGATION_MESSAGE_COMMANDS_CHANGED
+
+ The new #GstMessage.
+
+
+
+
+ A #GstObject to set as source of the new message.
+
+
+
+
+
+ Creates a new #GstNavigation message with type
+#GST_NAVIGATION_MESSAGE_EVENT.
+
+ The new #GstMessage.
+
+
+
+
+ A #GstObject to set as source of the new message.
+
+
+
+ A navigation #GstEvent
+
+
+
+
+
+ Creates a new #GstNavigation message with type
+#GST_NAVIGATION_MESSAGE_MOUSE_OVER.
+
+ The new #GstMessage.
+
+
+
+
+ A #GstObject to set as source of the new message.
+
+
+
+ %TRUE if the mouse has entered a clickable area of the display.
+%FALSE if it over a non-clickable area.
+
+
+
+
+
+ Parse a #GstNavigation message of type GST_NAVIGATION_MESSAGE_ANGLES_CHANGED
+and extract the @cur_angle and @n_angles parameters.
+
+ %TRUE if the message could be successfully parsed. %FALSE if not.
+
+
+
+
+ A #GstMessage to inspect.
+
+
+
+ A pointer to a #guint to receive the new
+ current angle number, or NULL
+
+
+
+ A pointer to a #guint to receive the new angle
+ count, or NULL.
+
+
+
+
+
+ Parse a #GstNavigation message of type #GST_NAVIGATION_MESSAGE_EVENT
+and extract contained #GstEvent. The caller must unref the @event when done
+with it.
+
+ %TRUE if the message could be successfully parsed. %FALSE if not.
+
+
+
+
+ A #GstMessage to inspect.
+
+
+
+ a pointer to a #GstEvent to receive
+ the contained navigation event.
+
+
+
+
+
+ Parse a #GstNavigation message of type #GST_NAVIGATION_MESSAGE_MOUSE_OVER
+and extract the active/inactive flag. If the mouse over event is marked
+active, it indicates that the mouse is over a clickable area.
+
+ %TRUE if the message could be successfully parsed. %FALSE if not.
+
+
+
+
+ A #GstMessage to inspect.
+
+
+
+ A pointer to a gboolean to receive the
+ active/inactive state, or NULL.
+
+
+
+
+
+ Inspect a #GstQuery and return the #GstNavigationQueryType associated with
+it if it is a #GstNavigation query.
+
+ The #GstNavigationQueryType of the query, or
+#GST_NAVIGATION_QUERY_INVALID
+
+
+
+
+ The query to inspect
+
+
+
+
+
+ Create a new #GstNavigation angles query. When executed, it will
+query the pipeline for the set of currently available angles, which may be
+greater than one in a multiangle video.
+
+ The new query.
+
+
+
+
+ Create a new #GstNavigation commands query. When executed, it will
+query the pipeline for the set of currently available commands.
+
+ The new query.
+
+
+
+
+ Parse the current angle number in the #GstNavigation angles @query into the
+#guint pointed to by the @cur_angle variable, and the number of available
+angles into the #guint pointed to by the @n_angles variable.
+
+ %TRUE if the query could be successfully parsed. %FALSE if not.
+
+
+
+
+ a #GstQuery
+
+
+
+ Pointer to a #guint into which to store the
+ currently selected angle value from the query, or NULL
+
+
+
+ Pointer to a #guint into which to store the
+ number of angles value from the query, or NULL
+
+
+
+
+
+ Parse the number of commands in the #GstNavigation commands @query.
+
+ %TRUE if the query could be successfully parsed. %FALSE if not.
+
+
+
+
+ a #GstQuery
+
+
+
+ the number of commands in this query.
+
+
+
+
+
+ Parse the #GstNavigation command query and retrieve the @nth command from
+it into @cmd. If the list contains less elements than @nth, @cmd will be
+set to #GST_NAVIGATION_COMMAND_INVALID.
+
+ %TRUE if the query could be successfully parsed. %FALSE if not.
+
+
+
+
+ a #GstQuery
+
+
+
+ the nth command to retrieve.
+
+
+
+ a pointer to store the nth command into.
+
+
+
+
+
+ Set the #GstNavigation angles query result field in @query.
+
+
+
+
+
+ a #GstQuery
+
+
+
+ the current viewing angle to set.
+
+
+
+ the number of viewing angles to set.
+
+
+
+
+
+ Set the #GstNavigation command query result fields in @query. The number
+of commands passed must be equal to @n_commands.
+
+
+
+
+
+ a #GstQuery
+
+
+
+ the number of commands to set.
+
+
+
+ An array containing @n_cmds
+ @GstNavigationCommand values.
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Lets you blend the @src image into the @dest image
+
+
+
+
+
+ The #GstVideoFrame where to blend @src in
+
+
+
+ the #GstVideoFrame that we want to blend into
+
+
+
+ The x offset in pixel where the @src image should be blended
+
+
+
+ the y offset in pixel where the @src image should be blended
+
+
+
+ the global_alpha each per-pixel alpha value is multiplied
+ with
+
+
+
+
+
+ Scales a buffer containing RGBA (or AYUV) video. This is an internal
+helper function which is used to scale subtitle overlays, and may be
+deprecated in the near future. Use #GstVideoScaler to scale video buffers
+instead.
+
+
+
+
+
+ the #GstVideoInfo describing the video data in @src_buffer
+
+
+
+ the source buffer containing video pixels to scale
+
+
+
+ the height in pixels to scale the video data in @src_buffer to
+
+
+
+ the width in pixels to scale the video data in @src_buffer to
+
+
+
+ pointer to a #GstVideoInfo structure that will be filled in
+ with the details for @dest_buffer
+
+
+
+ a pointer to a #GstBuffer variable, which will be
+ set to a newly-allocated buffer containing the scaled pixels.
+
+
+
+
+
+ Given the Pixel Aspect Ratio and size of an input video frame, and the
+pixel aspect ratio of the intended display device, calculates the actual
+display ratio the video will be rendered with.
+
+ A boolean indicating success and a calculated Display Ratio in the
+dar_n and dar_d parameters.
+The return value is FALSE in the case of integer overflow or other error.
+
+
+
+
+ Numerator of the calculated display_ratio
+
+
+
+ Denominator of the calculated display_ratio
+
+
+
+ Width of the video frame in pixels
+
+
+
+ Height of the video frame in pixels
+
+
+
+ Numerator of the pixel aspect ratio of the input video.
+
+
+
+ Denominator of the pixel aspect ratio of the input video.
+
+
+
+ Numerator of the pixel aspect ratio of the display device
+
+
+
+ Denominator of the pixel aspect ratio of the display device
+
+
+
+
+
+ Convert @s to a #GstVideoChromaSite
+
+ a #GstVideoChromaSite or %GST_VIDEO_CHROMA_SITE_UNKNOWN when @s does
+not contain a valid chroma description.
+
+
+
+
+ a chromasite string
+
+
+
+
+
+ Perform resampling of @width chroma pixels in @lines.
+
+
+
+
+
+ a #GstVideoChromaResample
+
+
+
+ pixel lines
+
+
+
+ the number of pixels on one line
+
+
+
+
+
+ Create a new resampler object for the given parameters. When @h_factor or
+@v_factor is > 0, upsampling will be used, otherwise subsampling is
+performed.
+
+ a new #GstVideoChromaResample that should be freed with
+ gst_video_chroma_resample_free() after usage.
+
+
+
+
+ a #GstVideoChromaMethod
+
+
+
+ a #GstVideoChromaSite
+
+
+
+ #GstVideoChromaFlags
+
+
+
+ the #GstVideoFormat
+
+
+
+ horizontal resampling factor
+
+
+
+ vertical resampling factor
+
+
+
+
+
+ Converts @site to its string representation.
+
+ a string describing @site.
+
+
+
+
+ a #GstVideoChromaSite
+
+
+
+
+
+ Get the coefficients used to convert between Y'PbPr and R'G'B' using @matrix.
+
+When:
+
+|[
+ 0.0 <= [Y',R',G',B'] <= 1.0)
+ (-0.5 <= [Pb,Pr] <= 0.5)
+]|
+
+the general conversion is given by:
+
+|[
+ Y' = Kr*R' + (1-Kr-Kb)*G' + Kb*B'
+ Pb = (B'-Y')/(2*(1-Kb))
+ Pr = (R'-Y')/(2*(1-Kr))
+]|
+
+and the other way around:
+
+|[
+ R' = Y' + Cr*2*(1-Kr)
+ G' = Y' - Cb*2*(1-Kb)*Kb/(1-Kr-Kb) - Cr*2*(1-Kr)*Kr/(1-Kr-Kb)
+ B' = Y' + Cb*2*(1-Kb)
+]|
+
+ TRUE if @matrix was a YUV color format and @Kr and @Kb contain valid
+ values.
+
+
+
+
+ a #GstVideoColorMatrix
+
+
+
+ result red channel coefficient
+
+
+
+ result blue channel coefficient
+
+
+
+
+
+ Get information about the chromaticity coordinates of @primaries.
+
+ a #GstVideoColorPrimariesInfo for @primaries.
+
+
+
+
+ a #GstVideoColorPrimaries
+
+
+
+
+
+ Compute the offset and scale values for each component of @info. For each
+component, (c[i] - offset[i]) / scale[i] will scale the component c[i] to the
+range [0.0 .. 1.0].
+
+The reverse operation (c[i] * scale[i]) + offset[i] can be used to convert
+the component values in range [0.0 .. 1.0] back to their representation in
+@info and @range.
+
+
+
+
+
+ a #GstVideoColorRange
+
+
+
+ a #GstVideoFormatInfo
+
+
+
+ output offsets
+
+
+
+
+
+ output scale
+
+
+
+
+
+
+
+ Convert @val to its gamma decoded value. This is the inverse operation of
+@gst_video_color_transfer_encode().
+
+For a non-linear value L' in the range [0..1], conversion to the linear
+L is in general performed with a power function like:
+
+|[
+ L = L' ^ gamma
+]|
+
+Depending on @func, different formulas might be applied. Some formulas
+encode a linear segment in the lower range.
+
+ the gamme decoded value of @val
+
+
+
+
+ a #GstVideoTransferFunction
+
+
+
+ a value
+
+
+
+
+
+ Convert @val to its gamma encoded value.
+
+For a linear value L in the range [0..1], conversion to the non-linear
+(gamma encoded) L' is in general performed with a power function like:
+
+|[
+ L' = L ^ (1 / gamma)
+]|
+
+Depending on @func, different formulas might be applied. Some formulas
+encode a linear segment in the lower range.
+
+ the gamme encoded value of @val
+
+
+
+
+ a #GstVideoTransferFunction
+
+
+
+ a value
+
+
+
+
+
+ Converts a raw video buffer into the specified output caps.
+
+The output caps can be any raw video formats or any image formats (jpeg, png, ...).
+
+The width, height and pixel-aspect-ratio can also be specified in the output caps.
+
+ The converted #GstSample, or %NULL if an error happened (in which case @err
+will point to the #GError).
+
+
+
+
+ a #GstSample
+
+
+
+ the #GstCaps to convert to
+
+
+
+ the maximum amount of time allowed for the processing.
+
+
+
+
+
+ Converts a raw video buffer into the specified output caps.
+
+The output caps can be any raw video formats or any image formats (jpeg, png, ...).
+
+The width, height and pixel-aspect-ratio can also be specified in the output caps.
+
+@callback will be called after conversion, when an error occured or if conversion didn't
+finish after @timeout. @callback will always be called from the thread default
+%GMainContext, see g_main_context_get_thread_default(). If GLib before 2.22 is used,
+this will always be the global default main context.
+
+@destroy_notify will be called after the callback was called and @user_data is not needed
+anymore.
+
+
+
+
+
+ a #GstSample
+
+
+
+ the #GstCaps to convert to
+
+
+
+ the maximum amount of time allowed for the processing.
+
+
+
+ %GstVideoConvertSampleCallback that will be called after conversion.
+
+
+
+ extra data that will be passed to the @callback
+
+
+
+ %GDestroyNotify to be called after @user_data is not needed anymore
+
+
+
+
+
+ Create a new converter object to convert between @in_info and @out_info
+with @config.
+
+ a #GstVideoConverter or %NULL if conversion is not possible.
+
+
+
+
+ a #GstVideoInfo
+
+
+
+ a #GstVideoInfo
+
+
+
+ a #GstStructure with configuration options
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Make a new dither object for dithering lines of @format using the
+algorithm described by @method.
+
+Each component will be quantized to a multiple of @quantizer. Better
+performance is achived when @quantizer is a power of 2.
+
+@width is the width of the lines that this ditherer will handle.
+
+ a new #GstVideoDither
+
+
+
+
+ a #GstVideoDitherMethod
+
+
+
+ a #GstVideoDitherFlags
+
+
+
+ a #GstVideoFormat
+
+
+
+ quantizer
+
+
+
+ the width of the lines
+
+
+
+
+
+ Checks if an event is a force key unit event. Returns true for both upstream
+and downstream force key unit events.
+
+ %TRUE if the event is a valid force key unit event
+
+
+
+
+ A #GstEvent to check
+
+
+
+
+
+ Creates a new downstream force key unit event. A downstream force key unit
+event can be sent down the pipeline to request downstream elements to produce
+a key unit. A downstream force key unit event must also be sent when handling
+an upstream force key unit event to notify downstream that the latter has been
+handled.
+
+To parse an event created by gst_video_event_new_downstream_force_key_unit() use
+gst_video_event_parse_downstream_force_key_unit().
+
+ The new GstEvent
+
+
+
+
+ the timestamp of the buffer that starts a new key unit
+
+
+
+ the stream_time of the buffer that starts a new key unit
+
+
+
+ the running_time of the buffer that starts a new key unit
+
+
+
+ %TRUE to produce headers when starting a new key unit
+
+
+
+ integer that can be used to number key units
+
+
+
+
+
+ Creates a new Still Frame event. If @in_still is %TRUE, then the event
+represents the start of a still frame sequence. If it is %FALSE, then
+the event ends a still frame sequence.
+
+To parse an event created by gst_video_event_new_still_frame() use
+gst_video_event_parse_still_frame().
+
+ The new GstEvent
+
+
+
+
+ boolean value for the still-frame state of the event.
+
+
+
+
+
+ Creates a new upstream force key unit event. An upstream force key unit event
+can be sent to request upstream elements to produce a key unit.
+
+@running_time can be set to request a new key unit at a specific
+running_time. If set to GST_CLOCK_TIME_NONE, upstream elements will produce a
+new key unit as soon as possible.
+
+To parse an event created by gst_video_event_new_downstream_force_key_unit() use
+gst_video_event_parse_downstream_force_key_unit().
+
+ The new GstEvent
+
+
+
+
+ the running_time at which a new key unit should be produced
+
+
+
+ %TRUE to produce headers when starting a new key unit
+
+
+
+ integer that can be used to number key units
+
+
+
+
+
+ Get timestamp, stream-time, running-time, all-headers and count in the force
+key unit event. See gst_video_event_new_downstream_force_key_unit() for a
+full description of the downstream force key unit event.
+
+@running_time will be adjusted for any pad offsets of pads it was passing through.
+
+ %TRUE if the event is a valid downstream force key unit event.
+
+
+
+
+ A #GstEvent to parse
+
+
+
+ A pointer to the timestamp in the event
+
+
+
+ A pointer to the stream-time in the event
+
+
+
+ A pointer to the running-time in the event
+
+
+
+ A pointer to the all_headers flag in the event
+
+
+
+ A pointer to the count field of the event
+
+
+
+
+
+ Parse a #GstEvent, identify if it is a Still Frame event, and
+return the still-frame state from the event if it is.
+If the event represents the start of a still frame, the in_still
+variable will be set to TRUE, otherwise FALSE. It is OK to pass NULL for the
+in_still variable order to just check whether the event is a valid still-frame
+event.
+
+Create a still frame event using gst_video_event_new_still_frame()
+
+ %TRUE if the event is a valid still-frame event. %FALSE if not
+
+
+
+
+ A #GstEvent to parse
+
+
+
+
+ A boolean to receive the still-frame status from the event, or NULL
+
+
+
+
+
+ Get running-time, all-headers and count in the force key unit event. See
+gst_video_event_new_upstream_force_key_unit() for a full description of the
+upstream force key unit event.
+
+Create an upstream force key unit event using gst_video_event_new_upstream_force_key_unit()
+
+@running_time will be adjusted for any pad offsets of pads it was passing through.
+
+ %TRUE if the event is a valid upstream force-key-unit event. %FALSE if not
+
+
+
+
+ A #GstEvent to parse
+
+
+
+ A pointer to the running_time in the event
+
+
+
+ A pointer to the all_headers flag in the event
+
+
+
+ A pointer to the count field in the event
+
+
+
+
+
+ Convert @order to a #GstVideoFieldOrder
+
+ the #GstVideoFieldOrder of @order or
+ #GST_VIDEO_FIELD_ORDER_UNKNOWN when @order is not a valid
+ string representation for a #GstVideoFieldOrder.
+
+
+
+
+ a field order
+
+
+
+
+
+ Convert @order to its string representation.
+
+ @order as a string or NULL if @order in invalid.
+
+
+
+
+ a #GstVideoFieldOrder
+
+
+
+
+
+ Converts a FOURCC value into the corresponding #GstVideoFormat.
+If the FOURCC cannot be represented by #GstVideoFormat,
+#GST_VIDEO_FORMAT_UNKNOWN is returned.
+
+ the #GstVideoFormat describing the FOURCC value
+
+
+
+
+ a FOURCC value representing raw YUV video
+
+
+
+
+
+ Find the #GstVideoFormat for the given parameters.
+
+ a #GstVideoFormat or GST_VIDEO_FORMAT_UNKNOWN when the parameters to
+not specify a known format.
+
+
+
+
+ the amount of bits used for a pixel
+
+
+
+ the amount of bits used to store a pixel. This value is bigger than
+ @depth
+
+
+
+ the endianness of the masks, #G_LITTLE_ENDIAN or #G_BIG_ENDIAN
+
+
+
+ the red mask
+
+
+
+ the green mask
+
+
+
+ the blue mask
+
+
+
+ the alpha mask, or 0 if no alpha mask
+
+
+
+
+
+ Convert the @format string to its #GstVideoFormat.
+
+ the #GstVideoFormat for @format or GST_VIDEO_FORMAT_UNKNOWN when the
+string is not a known format.
+
+
+
+
+ a format string
+
+
+
+
+
+ Get the #GstVideoFormatInfo for @format
+
+ The #GstVideoFormatInfo for @format.
+
+
+
+
+ a #GstVideoFormat
+
+
+
+
+
+ Get the default palette of @format. This the palette used in the pack
+function for paletted formats.
+
+ the default palette of @format or %NULL when
+@format does not have a palette.
+
+
+
+
+ a #GstVideoFormat
+
+
+
+ size of the palette in bytes
+
+
+
+
+
+ Converts a #GstVideoFormat value into the corresponding FOURCC. Only
+a few YUV formats have corresponding FOURCC values. If @format has
+no corresponding FOURCC value, 0 is returned.
+
+ the FOURCC corresponding to @format
+
+
+
+
+ a #GstVideoFormat video format
+
+
+
+
+
+ Returns a string containing a descriptive name for
+the #GstVideoFormat if there is one, or NULL otherwise.
+
+ the name corresponding to @format
+
+
+
+
+ a #GstVideoFormat video format
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Given the nominal duration of one video frame,
+this function will check some standard framerates for
+a close match (within 0.1%) and return one if possible,
+
+It will calculate an arbitrary framerate if no close
+match was found, and return %FALSE.
+
+It returns %FALSE if a duration of 0 is passed.
+
+ %TRUE if a close "standard" framerate was
+recognised, and %FALSE otherwise.
+
+
+
+
+ Nominal duration of one frame
+
+
+
+ Numerator of the calculated framerate
+
+
+
+ Denominator of the calculated framerate
+
+
+
+
+
+ Convert @mode to a #GstVideoInterlaceMode
+
+ the #GstVideoInterlaceMode of @mode or
+ #GST_VIDEO_INTERLACE_MODE_PROGRESSIVE when @mode is not a valid
+ string representation for a #GstVideoInterlaceMode.
+
+
+
+
+ a mode
+
+
+
+
+
+ Convert @mode to its string representation.
+
+ @mode as a string or NULL if @mode in invalid.
+
+
+
+
+ a #GstVideoInterlaceMode
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Get the #GQuark for the "gst-video-scale" metadata transform operation.
+
+ a #GQuark
+
+
+
+
+
+ A const #GValue containing a list of stereo video modes
+
+Utility function that returns a #GValue with a GstList of packed stereo
+video modes with double the height of a single view for use in
+caps negotiations. Currently this is top-bottom and row-interleaved.
+
+
+
+
+
+ A const #GValue containing a list of stereo video modes
+
+Utility function that returns a #GValue with a GstList of packed
+stereo video modes that have double the width/height of a single
+view for use in caps negotiation. Currently this is just
+'checkerboard' layout.
+
+
+
+
+
+ A const #GValue containing a list of stereo video modes
+
+Utility function that returns a #GValue with a GstList of packed stereo
+video modes with double the width of a single view for use in
+caps negotiations. Currently this is side-by-side, side-by-side-quincunx
+and column-interleaved.
+
+
+
+
+
+ A const #GValue containing a list of mono video modes
+
+Utility function that returns a #GValue with a GstList of mono video
+modes (mono/left/right) for use in caps negotiations.
+
+
+
+
+
+ A const #GValue containing a list of 'unpacked' stereo video modes
+
+Utility function that returns a #GValue with a GstList of unpacked
+stereo video modes (separated/frame-by-frame/frame-by-frame-multiview)
+for use in caps negotiations.
+
+
+
+
+
+ A boolean indicating whether the
+ #GST_VIDEO_MULTIVIEW_FLAG_HALF_ASPECT flag should be set.
+
+Utility function that heuristically guess whether a
+frame-packed stereoscopic video contains half width/height
+encoded views, or full-frame views by looking at the
+overall display aspect ratio.
+
+
+
+
+ A #GstVideoMultiviewMode
+
+
+
+ Video frame width in pixels
+
+
+
+ Video frame height in pixels
+
+
+
+ Numerator of the video pixel-aspect-ratio
+
+
+
+ Denominator of the video pixel-aspect-ratio
+
+
+
+
+
+
+ The #GstVideoMultiviewMode value
+
+Given a string from a caps multiview-mode field,
+output the corresponding #GstVideoMultiviewMode
+or #GST_VIDEO_MULTIVIEW_MODE_NONE
+
+
+
+
+ multiview-mode field string from caps
+
+
+
+
+
+
+ The caps string representation of the mode, or NULL if invalid.
+
+Given a #GstVideoMultiviewMode returns the multiview-mode caps string
+for insertion into a caps structure
+
+
+
+
+ A #GstVideoMultiviewMode value
+
+
+
+
+
+ Utility function that transforms the width/height/PAR
+and multiview mode and flags of a #GstVideoInfo into
+the requested mode.
+
+
+
+
+
+ A #GstVideoInfo structure to operate on
+
+
+
+ A #GstVideoMultiviewMode value
+
+
+
+ A set of #GstVideoMultiviewFlags
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ This helper shall be used by classes implementing the #GstVideoOverlay
+interface that want the render rectangle to be controllable using
+properties. This helper will install "render-rectangle" property into the
+class.
+
+Since 1.14
+
+
+
+
+
+ The class on which the properties will be installed
+
+
+
+ The first free property ID to use
+
+
+
+
+
+ This helper shall be used by classes implementing the #GstVideoOverlay
+interface that want the render rectangle to be controllable using
+properties. This helper will parse and set the render rectangle calling
+gst_video_overlay_set_render_rectangle().
+
+ %TRUE if the @property_id matches the GstVideoOverlay property
+
+Since 1.14
+
+
+
+
+ The instance on which the property is set
+
+
+
+ The highest property ID.
+
+
+
+ The property ID
+
+
+
+ The #GValue to be set
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+ Make a new @method video scaler. @in_size source lines/pixels will
+be scaled to @out_size destination lines/pixels.
+
+@n_taps specifies the amount of pixels to use from the source for one output
+pixel. If n_taps is 0, this function chooses a good value automatically based
+on the @method and @in_size/@out_size.
+
+ a #GstVideoResample
+
+
+
+
+ a #GstVideoResamplerMethod
+
+
+
+ #GstVideoScalerFlags
+
+
+
+ number of taps to use
+
+
+
+ number of source elements
+
+
+
+ number of destination elements
+
+
+
+ extra options
+
+
+
+
+
+ Get the tile index of the tile at coordinates @x and @y in the tiled
+image of @x_tiles by @y_tiles.
+
+Use this method when @mode is of type %GST_VIDEO_TILE_MODE_INDEXED.
+
+ the index of the tile at @x and @y in the tiled image of
+ @x_tiles by @y_tiles.
+
+
+
+
+ a #GstVideoTileMode
+
+
+
+ x coordinate
+
+
+
+ y coordinate
+
+
+
+ number of horizintal tiles
+
+
+
+ number of vertical tiles
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+
+