From 6923d5008c4f3420537b9c8e4eed997668128602 Mon Sep 17 00:00:00 2001 From: Edward Hervey Date: Fri, 28 Jun 2024 14:22:19 +0200 Subject: [PATCH] girs: Update for fedora 40 gobject-introspection Part-of: --- girs/GES-1.0.gir | 904 ++++++++------ girs/Gst-1.0.gir | 1941 ++++++++++++++++++++---------- girs/GstAllocators-1.0.gir | 8 +- girs/GstAnalytics-1.0.gir | 8 +- girs/GstApp-1.0.gir | 177 +-- girs/GstAudio-1.0.gir | 865 +++++++++---- girs/GstBadAudio-1.0.gir | 226 +++- girs/GstBadBaseCameraBin-1.0.gir | 25 +- girs/GstBase-1.0.gir | 699 +++++++++-- girs/GstCheck-1.0.gir | 14 +- girs/GstCodecs-1.0.gir | 60 +- girs/GstController-1.0.gir | 42 +- girs/GstCuda-1.0.gir | 32 +- girs/GstGL-1.0.gir | 405 +++++-- girs/GstInsertBin-1.0.gir | 2 +- girs/GstMse-1.0.gir | 118 +- girs/GstNet-1.0.gir | 32 +- girs/GstPbutils-1.0.gir | 108 +- girs/GstPlay-1.0.gir | 140 +-- girs/GstPlayer-1.0.gir | 110 +- girs/GstRtp-1.0.gir | 297 +++-- girs/GstRtsp-1.0.gir | 457 +++---- girs/GstRtspServer-1.0.gir | 468 ++++--- girs/GstSdp-1.0.gir | 4 +- girs/GstTag-1.0.gir | 112 +- girs/GstTranscoder-1.0.gir | 54 +- girs/GstVa-1.0.gir | 18 +- girs/GstValidate-1.0.gir | 196 +-- girs/GstVideo-1.0.gir | 1319 ++++++++++++-------- girs/GstVulkan-1.0.gir | 242 ++-- girs/GstWebRTC-1.0.gir | 258 ++-- 31 files changed, 6122 insertions(+), 3219 deletions(-) diff --git a/girs/GES-1.0.gir b/girs/GES-1.0.gir index af8abefa19..3ae0166e01 100644 --- a/girs/GES-1.0.gir +++ b/girs/GES-1.0.gir @@ -142,7 +142,7 @@ reloading - + Returns an asset with the given properties. If such an asset already exists in the cache (it has been previously created in GES), then a reference to the existing asset is returned. Otherwise, a newly created @@ -192,7 +192,7 @@ asset, or %NULL if an error occurred. - + Requests an asset with the given properties asynchronously (see ges_asset_request()). When the asset has been initialized or fetched from the cache, the given callback function will be called. The @@ -312,6 +312,7 @@ error occurred. + Deprecated: 1.18: This vmethod is no longer called. @@ -326,6 +327,10 @@ error occurred. + A method called by a #GESProject when an asset has +failed to load. @error is the error given by +ges_asset_request_finish (). Returns: %TRUE if a new id for @self was +passed to @proposed_new_id. @@ -343,6 +348,13 @@ error occurred. + A method to be called when an asset is being requested +asynchronously. This will be after the properties of the asset have +been set, so it is tasked with (re)loading the 'state' of the asset. +The return value should indicated whether the loading is complete, is +carrying on asynchronously, or an error occurred. The default +implementation will simply return that loading is already complete (the +asset is already in a usable state after the properties have been set). @@ -385,7 +397,7 @@ error occurred. - + Gets the #GESAsset:extractable-type of the asset. @@ -399,7 +411,7 @@ error occurred. - + Gets the #GESAsset:id of the asset. @@ -413,7 +425,7 @@ error occurred. - + Gets the default #GESAsset:proxy of the asset. @@ -427,7 +439,7 @@ error occurred. - + Gets the #GESAsset:proxy-target of the asset. Note that the proxy target may have loaded with an error, so you should @@ -464,7 +476,7 @@ that @asset has. - + Sets the #GESAsset:proxy for the asset. If @proxy is among the existing proxies of the asset (see @@ -519,11 +531,11 @@ list. - + The #GESExtractable object type that can be extracted from the asset. - + The ID of the asset. This should be unique amongst all assets with the same #GESAsset:extractable-type. Depending on the associated #GESExtractable implementation, this id may convey some information @@ -535,7 +547,7 @@ of the extractable type to see whether they differ from the default behaviour. - + The default proxy for this asset, or %NULL if it has no proxy. A proxy will act as a substitute for the original asset when the original is requested (see ges_asset_request()). @@ -544,7 +556,7 @@ Setting this property will not usually remove the existing proxy, but will replace it as the default (see ges_asset_set_proxy()). - + The asset that this asset is a proxy for, or %NULL if it is not a proxy for another asset. @@ -577,6 +589,13 @@ asset it is now the proxy of/no longer the proxy of. + A method to be called when an asset is being requested +asynchronously. This will be after the properties of the asset have +been set, so it is tasked with (re)loading the 'state' of the asset. +The return value should indicated whether the loading is complete, is +carrying on asynchronously, or an error occurred. The default +implementation will simply return that loading is already complete (the +asset is already in a usable state after the properties have been set). @@ -590,6 +609,11 @@ asset it is now the proxy of/no longer the proxy of. + A method that returns a new object of the asset's +#GESAsset:extractable-type, or %NULL if an error occurs. The default +implementation will fetch the properties of the #GESExtractable from +its get_parameters_from_id() class method and set them on a new +#GESAsset:extractable-type #GObject, which is returned. @@ -622,6 +646,7 @@ error occurred. + Deprecated: 1.18: This vmethod is no longer called. @@ -638,6 +663,10 @@ error occurred. + A method called by a #GESProject when an asset has +failed to load. @error is the error given by +ges_asset_request_finish (). Returns: %TRUE if a new id for @self was +passed to @proposed_new_id. @@ -676,7 +705,7 @@ asynchronously error - + @@ -726,7 +755,7 @@ You can use the following children properties through the - + @@ -821,7 +850,7 @@ track. - + @@ -887,7 +916,7 @@ modify these fields, or add additional ones. - + @@ -927,7 +956,7 @@ be created by clips. - + @@ -937,7 +966,7 @@ be created by clips. - + The location of the file/resource to use. @@ -967,7 +996,7 @@ be created by clips. - + @@ -1183,10 +1212,10 @@ as core children, nor as additional effects. - + - + @@ -1251,7 +1280,7 @@ property name/value pairs - + @@ -1302,7 +1331,7 @@ property name/value pairs - + @@ -1530,6 +1559,12 @@ do_time_effect_change (GESClip * clip) + Method to create the core #GESTrackElement of a clip +of this class. If a clip of this class may create several track elements per +track type, this should be left as %NULL, and +GESClipClass::create_track_elements should be used instead. Otherwise, you +should implement this class method and leave +GESClipClass::create_track_elements as the default implementation The #GESTrackElement created @@ -1549,6 +1584,10 @@ given @type or an error occurred. + Method to create the (multiple) core +#GESTrackElement-s of a clip of this class. If +GESClipClass::create_track_element is implemented, this should be kept as the +default implementation A list of @@ -1754,7 +1793,7 @@ match any type - + Gets the #GESClip:duration-limit of the clip. @@ -1829,7 +1868,7 @@ be performed. - + Gets the #GESClip:layer of the clip. @@ -1844,7 +1883,7 @@ be performed. - + Gets the #GESClip:supported-formats of the clip. @@ -2070,7 +2109,7 @@ would not be able to adapt itself once the effect is removed. - + Sets the #GESClip:supported-formats of the clip. This should normally only be called by subclasses, which should be responsible for updating its value, rather than the user. @@ -2219,7 +2258,7 @@ the start and end of the clip - + The maximum #GESTimelineElement:duration that can be *currently* set for the clip, taking into account the #GESTimelineElement:in-point, #GESTimelineElement:max-duration, #GESTrackElement:active, and @@ -2236,7 +2275,7 @@ variables, its #GESTimelineElement:duration will be set to the new limit. - + The layer this clip lies in. If you want to connect to this property's #GObject::notify signal, @@ -2244,7 +2283,7 @@ you should connect to it with g_signal_connect_after() since the signal emission may be stopped internally. - + The #GESTrackType-s that the clip supports, which it can create #GESTrackElement-s for. Note that this can be a combination of #GESTrackType flags to indicate support for several @@ -2338,7 +2377,7 @@ natural frame rate. - + Gets track types for which objects extracted from @self can create #GESTrackElement @@ -2353,7 +2392,7 @@ a layer - + Sets track types for which objects extracted from @self can create #GESTrackElement @@ -2370,7 +2409,7 @@ a layer - + The formats supported by the asset. @@ -2420,7 +2459,7 @@ a layer - + @@ -2459,7 +2498,7 @@ default implementation - + @@ -2511,7 +2550,7 @@ Result: (transfer full): A help string. - + @@ -2555,6 +2594,7 @@ The #GESContainer-s to group + Virtual method to add a child @@ -2569,6 +2609,7 @@ The #GESContainer-s to group + Virtual method that is called right after a #GESTimelineElement is added @@ -2583,6 +2624,7 @@ The #GESContainer-s to group + Virtual method that is called right after a #GESTimelineElement is removed @@ -2638,6 +2680,7 @@ be moved to. -1 means no move + Virtual method to remove a child @@ -2824,7 +2867,7 @@ new #GESContainer-s created from the splitting of @container. - + The span of the container's children's #GESTimelineElement:priority values, which is the number of integers that lie between (inclusive) the minimum and maximum priorities found amongst the container's @@ -2892,6 +2935,7 @@ may be stopped internally. + Virtual method that is called right after a #GESTimelineElement is added @@ -2908,6 +2952,7 @@ may be stopped internally. + Virtual method that is called right after a #GESTimelineElement is removed @@ -2924,6 +2969,7 @@ may be stopped internally. + Virtual method to add a child @@ -2940,6 +2986,7 @@ may be stopped internally. + Virtual method to remove a child @@ -2956,6 +3003,8 @@ may be stopped internally. + Virtual method to ungroup a container into a list of +containers @@ -2978,6 +3027,8 @@ new #GESContainer-s created from the splitting of @container. + Virtual method to group a list of containers together under a +single container @@ -2993,6 +3044,7 @@ new #GESContainer-s created from the splitting of @container. + Deprecated @@ -3042,7 +3094,7 @@ be moved to. -1 means no move - + @@ -3136,7 +3188,7 @@ if no track elements are created or an error occurred. - + The timeout to use for the discoverer @@ -3149,7 +3201,7 @@ if no track elements are created or an error occurred. - + Whether to use the cache or not @@ -3162,7 +3214,7 @@ if no track elements are created or an error occurred. - + Sets the timeout to use for the discoverer @@ -3179,7 +3231,7 @@ if no track elements are created or an error occurred. - + Sets whether to use the cache or not @@ -3196,11 +3248,11 @@ if no track elements are created or an error occurred. - + The timeout (in milliseconds) for the #GstDiscoverer operations - + @@ -3254,7 +3306,7 @@ discovery. - + @@ -3266,23 +3318,23 @@ discovery. The edges of an object contain in a #GESTimeline or #GESTrack - + Represents the start of an object. - + Represents the start of an object. - + Represents the end of an object. - + Represents the end of an object. - + Represent the fact we are not working with any edge of an object. - + Represent the fact we are not working with any edge of an object. @@ -3397,7 +3449,7 @@ by the same amount such that the snapped edges will touch. You can also find more explanation about the behaviour of those modes at: [trim, ripple and roll](http://pitivi.org/manual/trimming.html) and [clip management](http://pitivi.org/manual/usingclips.html). - + The element is edited the normal way (default). If acting on the element as a whole (#GES_EDGE_NONE), this will MOVE the element by MOVING its toplevel. When acting on the start of the @@ -3408,7 +3460,7 @@ and [clip management](http://pitivi.org/manual/usingclips.html). of the element (#GES_EDGE_END), this will END-TRIM the element, leaving its toplevel unchanged. - + The element is edited the normal way (default). If acting on the element as a whole (#GES_EDGE_NONE), this will MOVE the element by MOVING its toplevel. When acting on the start of the @@ -3419,7 +3471,7 @@ and [clip management](http://pitivi.org/manual/usingclips.html). of the element (#GES_EDGE_END), this will END-TRIM the element, leaving its toplevel unchanged. - + The element is edited in ripple mode: moving itself as well as later elements, keeping their relative times. This edits the element the same as #GES_EDIT_MODE_NORMAL. In addition, if @@ -3433,7 +3485,7 @@ and [clip management](http://pitivi.org/manual/usingclips.html). edited element. These additional elements will also be shifted by the same shift in layers as the edited element. - + The element is edited in ripple mode: moving itself as well as later elements, keeping their relative times. This edits the element the same as #GES_EDIT_MODE_NORMAL. In addition, if @@ -3447,7 +3499,7 @@ and [clip management](http://pitivi.org/manual/usingclips.html). edited element. These additional elements will also be shifted by the same shift in layers as the edited element. - + The element is edited in roll mode: swapping its content for its neighbour's, or vis versa, in the timeline output. This edits the element the same as #GES_EDIT_MODE_TRIM. In addition, @@ -3465,7 +3517,7 @@ and [clip management](http://pitivi.org/manual/usingclips.html). whole (#GES_EDGE_NONE) is not defined. The element can not shift layers under this mode. - + The element is edited in roll mode: swapping its content for its neighbour's, or vis versa, in the timeline output. This edits the element the same as #GES_EDIT_MODE_TRIM. In addition, @@ -3483,19 +3535,19 @@ and [clip management](http://pitivi.org/manual/usingclips.html). whole (#GES_EDGE_NONE) is not defined. The element can not shift layers under this mode. - + The element is edited in trim mode. When acting on the start of the element, this will START-TRIM it. When acting on the end of the element, this will END-TRIM it. Acting on the element as a whole (#GES_EDGE_NONE) is not defined. - + The element is edited in trim mode. When acting on the start of the element, this will START-TRIM it. When acting on the end of the element, this will END-TRIM it. Acting on the element as a whole (#GES_EDGE_NONE) is not defined. - + The element is edited in slide mode (not yet implemented): moving the element replacing or consuming content on each end. When acting on the element as a whole, this will MOVE the @@ -3508,7 +3560,7 @@ and [clip management](http://pitivi.org/manual/usingclips.html). (#GES_EDGE_START and #GES_EDGE_END) is not defined. The element can not shift layers under this mode. - + The element is edited in slide mode (not yet implemented): moving the element replacing or consuming content on each end. When acting on the element as a whole, this will MOVE the @@ -3581,7 +3633,7 @@ wrong. - + The description of the effect bin with a gst-launch-style pipeline description. @@ -3630,7 +3682,7 @@ to what track type the effect should be used in. - + @@ -3729,14 +3781,14 @@ The asset ID of an effect clip is in the form: - + 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. @@ -3766,10 +3818,10 @@ Example: "videobalance saturation=1.5 hue=+0.5" - + - + @@ -3848,6 +3900,11 @@ that is compatible with @self's current state. + This method is called after the #GESAsset of an object is +set. If your class supports the asset of an object changing, then you +can use this method to change the parameters of the object to match the +new asset #GESAsset:id. If setting the asset should be able to fail, +you should implement @set_asset_full instead. @@ -3862,6 +3919,9 @@ that is compatible with @self's current state. + Like @set_asset, but also allows you to return %FALSE +to indicate a failure to change the object in response to a change in +its asset. @@ -3995,6 +4055,11 @@ the new asset on the object. + This method is called after the #GESAsset of an object is +set. If your class supports the asset of an object changing, then you +can use this method to change the parameters of the object to match the +new asset #GESAsset:id. If setting the asset should be able to fail, +you should implement @set_asset_full instead. @@ -4011,6 +4076,9 @@ the new asset on the object. + Like @set_asset, but also allows you to return %FALSE +to indicate a failure to change the object in response to a change in +its asset. @@ -4027,6 +4095,11 @@ the new asset on the object. + The method to call to get the object +properties corresponding to a given asset #GESAsset:id. The default +implementation will simply return no parameters. The default #GESAsset +will call this to set the returned properties on the extracted object, +but other subclasses may ignore this method. @@ -4043,6 +4116,14 @@ the new asset on the object. + The method to fetch the #GESAsset:id of some associated asset. +Note that it may be the case that the object does not have its asset +set, or even that an asset with such an #GESAsset:id does not exist in +the GES cache. Instead, this should return the #GESAsset:id that is +_compatible_ with the current state of the object. The default +implementation simply returns the currently set asset ID, or the type name +of the object, which is what is used as the #GESAsset:id by default, +if no asset is set. @@ -4059,6 +4140,15 @@ that is compatible with @self's current state. + The method to call to get the actual +#GESAsset:extractable-type an asset should have set, given the +requested #GESAsset:id. The default implementation simply returns the +same type as given. You can overwrite this if it is more appropriate +to extract the object from a subclass, depending on the requested +#GESAsset:id. Note that when an asset is requested, this method will be +called before the other class methods. In particular, this means that +the @check_id and @get_parameters_from_id class methods of the returned +type will be used (instead of our own). @@ -4075,6 +4165,8 @@ that is compatible with @self's current state. + The method to set metadata on an asset. This is called +on initiation of the asset, but before it begins to load its state. @@ -4184,6 +4276,7 @@ the asset for the #GESFormatter that has the highest @rank + Whether the URI can be loaded @@ -4441,7 +4534,7 @@ else FALSE. - + @@ -4560,7 +4653,7 @@ ges_container_group() instead, which can return a different - + An overwrite of the #GESTimelineElement:duration property. For a #GESGroup, this is the difference between the earliest #GESTimelineElement:start time and the latest end time (given by @@ -4568,23 +4661,23 @@ ges_container_group() instead, which can return a different its children. - + An overwrite of the #GESTimelineElement:in-point property. This has no meaning for a group and should not be set. - + An overwrite of the #GESTimelineElement:max-duration property. This has no meaning for a group and should not be set. - + An overwrite of the #GESTimelineElement:priority property. Setting #GESTimelineElement priorities is deprecated as all priority management is now done by GES itself. - + An overwrite of the #GESTimelineElement:start property. For a #GESGroup, this is the earliest #GESTimelineElement:start time amongst its children. @@ -4613,7 +4706,7 @@ amongst its children. - + @@ -4626,7 +4719,7 @@ files, do not set the in-point property. - + The location of the file/resource to use. @@ -4656,7 +4749,7 @@ files, do not set the in-point property. - + @@ -4690,6 +4783,7 @@ layers will simply play in addition. + method to get the objects contained in the layer @@ -4871,7 +4965,7 @@ ges_layer_set_active_for_tracks(). - + Gets the #GESLayer:auto-transition of the layer. @@ -4943,7 +5037,7 @@ be the end time of the final clip). - + Get the priority of the layer. When inside a timeline, this is its index in the timeline. See ges_timeline_move_layer(). @@ -5044,7 +5138,7 @@ in the @layer's timeline - + Sets #GESLayer:auto-transition for the layer. Use ges_timeline_set_auto_transition() if you want all layers within a #GESTimeline to have #GESLayer:auto-transition set to %TRUE. Use this @@ -5067,7 +5161,7 @@ the layer - + Sets the layer to the given priority. See #GESLayer:priority. use #ges_timeline_move_layer instead. This deprecation means that you will not need to handle layer priorities at all yourself, GES @@ -5101,7 +5195,7 @@ will make sure there is never 'gaps' between layer priorities. - + Whether to automatically create a #GESTransitionClip whenever two #GESSource-s that both belong to a #GESClip in the layer overlap. See #GESTimeline for what counts as an overlap. @@ -5111,7 +5205,7 @@ When a layer is added to a #GESTimeline, if this property is left as will be set to %TRUE as well. - + The priority of the layer in the #GESTimeline. 0 is the highest priority. Conceptually, a timeline is a stack of layers, and the priority of the layer represents its position in the stack. Two @@ -5198,6 +5292,7 @@ efficient way of providing the list of contained #GESClip-s. + method to get the objects contained in the layer @@ -5250,7 +5345,7 @@ efficient way of providing the list of contained #GESClip-s. - + @@ -5312,7 +5407,7 @@ assets (string). A timed #GESMetaContainer object. - + Current position (in nanoseconds) of the #GESMarker @@ -5324,10 +5419,10 @@ assets (string). - + Marker does not serve any special purpose. - + Marker can be a snapping target. @@ -5425,7 +5520,7 @@ marker by 1. - + Flags indicating how markers on the list should be treated. @@ -6594,13 +6689,13 @@ the latter case, @value will be %NULL. - + The metadata is readable - + The metadata is writable - + The metadata is readable and writable @@ -6651,7 +6746,7 @@ information. - + 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>. @@ -6690,7 +6785,7 @@ multifile://20:50@/home/you/sequence/\%04d.png - + @@ -6749,10 +6844,10 @@ multifile://20:50@/home/you/sequence/\%04d.png - + - + @@ -6789,7 +6884,7 @@ not considered overlays. - + @@ -6825,7 +6920,7 @@ ges_pipeline_set_render_settings(). - + Gets the #GESPipeline:mode of the pipeline. @@ -6991,7 +7086,7 @@ native size - + Sets the #GESPipeline:mode of the pipeline. Note that the pipeline will be set to #GST_STATE_NULL during this call to @@ -7045,7 +7140,7 @@ result to - + Takes the given timeline and sets it as the #GESPipeline:timeline for the pipeline. @@ -7079,14 +7174,14 @@ immediately before the #GESPipeline:audio-sink. This exposes the #playsink:audio-sink property of the internal #playsink. - + The pipeline's mode. In preview mode (for audio or video, or both) the pipeline can display the timeline's content to an end user. In rendering mode the pipeline can encode the timeline's content and save it to a file. - + The timeline used by this pipeline, whose content it will play and render, or %NULL if the pipeline does not yet have a timeline. @@ -7131,24 +7226,24 @@ immediately before the #GESPipeline:video-sink. This exposes the The various modes a #GESPipeline can be configured to. - + Output the #GESPipeline:timeline's audio to the soundcard - + Output the #GESPipeline:timeline's video to the screen - + Output both the #GESPipeline:timeline's audio and video to the soundcard and screen (default) - + Render the #GESPipeline:timeline with forced decoding (the underlying #encodebin has its #encodebin:avoid-reencoding property set to %FALSE) - + Render the #GESPipeline:timeline, avoiding decoding/reencoding (the underlying #encodebin has its #encodebin:avoid-reencoding property set to %TRUE). @@ -7156,7 +7251,7 @@ avoiding decoding/reencoding (the underlying #encodebin has its > is enabled. - + @@ -7195,7 +7290,7 @@ is really not in good shape and is deprecated. - + @@ -7534,7 +7629,7 @@ MT safe. - + Retrieve the uri that is currently set on @project @@ -7671,7 +7766,7 @@ as defined in #ges_find_formatter_for_uri - + @@ -7947,7 +8042,7 @@ only to find out what the new location is. - + @@ -8106,10 +8201,10 @@ or %NULL if there was an error. - + - + @@ -8286,7 +8381,7 @@ or %NULL if there was an error. - + Get the volume of the test audio signal applied on @self. @@ -8300,7 +8395,7 @@ or %NULL if there was an error. - + Get the #GESVideoTestPattern which is applied on @self. @@ -8345,7 +8440,7 @@ or %NULL if there was an error. - + Sets whether the audio track of this clip is muted or not. @@ -8362,7 +8457,7 @@ or %NULL if there was an error. - + Sets the volume of the test audio signal. @@ -8379,7 +8474,7 @@ or %NULL if there was an error. - + Sets which video pattern to display on @self. @@ -8396,19 +8491,19 @@ or %NULL if there was an error. - + 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. @@ -8435,24 +8530,24 @@ or %NULL if there was an error. - + Horizontal alignment of the text. - + align text left - + align text center - + align text right - + align text on xpos position - + @@ -8728,7 +8823,7 @@ of the text render by @self. - + Get the color used by @source. @@ -8742,7 +8837,7 @@ of the text render by @self. - + Get the pango font description used by @self. @@ -8756,7 +8851,7 @@ of the text render by @self. - + Get the horizontal aligment used by @self. @@ -8770,7 +8865,7 @@ of the text render by @self. - + Get the text currently set on @self. @@ -8784,7 +8879,7 @@ of the text render by @self. - + Get the vertical aligment used by @self. @@ -8798,7 +8893,7 @@ of the text render by @self. - + Get the horizontal position used by @source. @@ -8812,7 +8907,7 @@ of the text render by @self. - + Get the vertical position used by @source. @@ -8826,7 +8921,7 @@ of the text render by @self. - + Sets the color of the text. @@ -8843,7 +8938,7 @@ of the text render by @self. - + Sets the pango font description of the text @@ -8877,7 +8972,7 @@ of the text render by @self. - + Sets the text this clip will render. @@ -8912,7 +9007,7 @@ made. - + Sets the horizontal position of the text. @@ -8929,7 +9024,7 @@ made. - + Sets the vertical position of the text. @@ -8946,31 +9041,31 @@ made. - + 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 @@ -8997,30 +9092,30 @@ made. - + - + 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 - + @@ -9425,7 +9520,7 @@ actually taken into account. - + Gets #GESTimeline:auto-transition for the timeline. @@ -9439,7 +9534,7 @@ actually taken into account. - + Get the current #GESTimeline:duration of the timeline @@ -9603,7 +9698,7 @@ or %NULL if there is an error. - + Gets the #GESTimeline:snapping-distance for the timeline. @@ -9818,7 +9913,7 @@ loaded from, before defaulting to the formatter with highest rank. - + Sets #GESTimeline:auto-transition for the timeline. This will also set the corresponding #GESLayer:auto-transition for all of the timeline's layers to the same value. See ges_layer_set_auto_transition() if you @@ -9839,7 +9934,7 @@ to @timeline's layers - + Sets #GESTimeline:snapping-distance for the timeline. This new value will only effect future snappings and will not be used to snap the current element positions within the timeline. @@ -9873,20 +9968,20 @@ actually taken into account. - + Whether to automatically create a transition whenever two #GESSource-s overlap in a track of the timeline. See #GESLayer:auto-transition if you want this to only happen in some layers. - + The current duration (in nanoseconds) of the timeline. A timeline 'starts' at time 0, so this is the maximum end time of all of its #GESTimelineElement-s. - + The distance (in nanoseconds) at which a #GESTimelineElement being moved within the timeline should snap one of its #GESSource-s with another #GESSource-s edge. See #GESEditMode for which edges can @@ -10370,6 +10465,8 @@ ges_timeline_element_list_children_properties(). + Prepare @copy for pasting as a copy of @self. At least by +copying the children properties of @self into @copy. @@ -10449,6 +10546,11 @@ of #GESTrack it can exist in, or will create #GESTrackElement-s for. + List the children properties that have been +registered for the element. The default implementation is able to fetch +all of these, so should be sufficient. If you overwrite this, you +should still call the default implementation to get the full list, and +then edit its content. @@ -10504,6 +10606,9 @@ specification of the child property + Paste @self, which is the @copy prepared by @deep_copy, into +the timeline at the given @paste_position, with @ref_element as a +reference, which is the @self that was passed to @deep_copy. @@ -11072,7 +11177,7 @@ optionally by more name/return location pairs, followed by %NULL - + Gets the #GESTimelineElement:duration for the element. @@ -11118,7 +11223,7 @@ layer. - + Gets the #GESTimelineElement:max-duration for the element. @@ -11132,7 +11237,7 @@ layer. - + Gets the #GESTimelineElement:name for the element. @@ -11178,7 +11283,7 @@ not the case. - + Gets the #GESTimelineElement:parent for the element. @@ -11193,7 +11298,7 @@ not the case. - + Gets the #GESTimelineElement:priority for the element. @@ -11207,7 +11312,7 @@ not the case. - + Gets the #GESTimelineElement:start for the element. @@ -11221,7 +11326,7 @@ not the case. - + Gets the #GESTimelineElement:timeline for the element. @@ -11600,7 +11705,7 @@ name/value pairs, followed by %NULL - + Sets #GESTimelineElement:duration for the element. Whilst the element is part of a #GESTimeline, this is the same as @@ -11647,7 +11752,7 @@ this method will fail. - + Sets #GESTimelineElement:max-duration for the element. If the new maximum duration is below the current #GESTimelineElement:in-point of the element, this method will fail. @@ -11667,7 +11772,7 @@ the element, this method will fail. - + Sets the #GESTimelineElement:name for the element. If %NULL is given for @name, then the library will instead generate a new name based on the type name of the element, such as the name "uriclip3" for a @@ -11701,7 +11806,7 @@ guaranteed. - + Sets the #GESTimelineElement:parent for the element. This is used internally and you should normally not call this. A @@ -11731,7 +11836,7 @@ If @parent is not %NULL, you must ensure it already has a - + Sets the priority of the element within the containing layer. All priority management is done by GES itself now. To set #GESEffect priorities #ges_clip_set_top_effect_index should @@ -11752,7 +11857,7 @@ be used. - + Sets #GESTimelineElement:start for the element. If the element has a parent, this will also move its siblings with the same shift. @@ -11778,7 +11883,7 @@ would place the timeline in an unsupported configuration. - + Sets the #GESTimelineElement:timeline of the element. This is used internally and you should normally not call this. A @@ -11829,7 +11934,7 @@ See ges_timeline_element_edit() with #GES_EDIT_MODE_TRIM and - + The duration that the element is in effect for in the timeline (a time difference in nanoseconds using the time coordinates of the timeline). For example, for a source element, this would determine @@ -11838,7 +11943,7 @@ operation element, this would determine for how long its effect should be applied to any source content. - + The initial offset to use internally when outputting content (in nanoseconds, but in the time coordinates of the internal content). @@ -11854,7 +11959,7 @@ For elements that have no internal content, this should be kept as 0. - + The full duration of internal content that is available (a time difference in nanoseconds using the time coordinates of the internal content). @@ -11871,24 +11976,24 @@ For elements that have no internal content, or whose content is indefinite, this should be kept as #GST_CLOCK_TIME_NONE. - + The name of the element. This should be unique within its timeline. - + The parent container of the element. - + The priority of the element. Priority management is now done by GES itself. - + Whether the element should be serialized. - + The starting position of the element in the timeline (in nanoseconds and in the time coordinates of the timeline). For example, for a source element, this would determine the time at which it should @@ -11897,7 +12002,7 @@ would determine the time at which it should start applying its effect to any source content. - + The timeline that the element lies within. @@ -12019,6 +12124,8 @@ and emitting the notify signal. + Method called just before the #GESTimelineElement:parent +is set. @@ -12038,6 +12145,25 @@ and emitting the notify signal. + Method called just before the #GESTimelineElement:start is +set. This method should check whether the #GESTimelineElement:start can +be changed to the new value and to otherwise prepare the element in +response to what the new value will be. A return of %FALSE means that +the property should not be set. A return of %TRUE means that the +property should be set to the value given to the setter and a notify +emitted. A return of -1 means that the property should not be set but +the setter should still return %TRUE (normally because the method +already handled setting the value, potentially to a snapped value, and +emitted the notify signal). +#GESTimelineElement:duration is set. This method should check +whether the #GESTimelineElement:duration can be changed to the new +value and to otherwise prepare the element in response to what the new +value will be. A return of %FALSE means that the property should not be +set. A return of %TRUE means that the property should be set to the +value given to the setter and a notify emitted. A return of -1 means +that the property should not be set but the setter should still return +%TRUE (normally because the method already handled setting the value, +potentially to a snapped value, and emitted the notify signal). @@ -12057,6 +12183,12 @@ and emitting the notify signal. + Method called just before the +#GESTimelineElement:in-point is set to a new value. This method should +not set the #GESTimelineElement:in-point itself, but should check +whether it can be changed to the new value and to otherwise prepare the +element in response to what the new value will be. A return of %FALSE +means that the property should not be set. @@ -12095,6 +12227,12 @@ and emitting the notify signal. + Method called just before the +#GESTimelineElement:max-duration is set. This method should +not set the #GESTimelineElement:max-duration itself, but should check +whether it can be changed to the new value and to otherwise prepare the +element in response to what the new value will be. A return of %FALSE +means that the property should not be set. @@ -12114,6 +12252,8 @@ and emitting the notify signal. + Method called just before the +#GESTimelineElement:priority is set. @@ -12133,6 +12273,8 @@ and emitting the notify signal. + Set this method to overwrite a redirect to +ges_timeline_element_edit() in ges_timeline_element_ripple(). @@ -12153,6 +12295,8 @@ failure. + Set this method to overwrite a redirect to +ges_timeline_element_edit() in ges_timeline_element_ripple_end(). @@ -12192,6 +12336,8 @@ failure. + Set this method to overwrite a redirect to +ges_timeline_element_edit() in ges_timeline_element_roll_end(). @@ -12211,6 +12357,8 @@ failure. + Set this method to overwrite a redirect to +ges_timeline_element_edit() in ges_timeline_element_trim(). @@ -12230,6 +12378,8 @@ failure. + Prepare @copy for pasting as a copy of @self. At least by +copying the children properties of @self into @copy. @@ -12246,6 +12396,9 @@ failure. + Paste @self, which is the @copy prepared by @deep_copy, into +the timeline at the given @paste_position, with @ref_element as a +reference, which is the @self that was passed to @deep_copy. @@ -12265,6 +12418,11 @@ failure. + List the children properties that have been +registered for the element. The default implementation is able to fetch +all of these, so should be sufficient. If you overwrite this, you +should still call the default implementation to get the full list, and +then edit its content. @@ -12281,6 +12439,11 @@ failure. + Find @child, and its registered child property @pspec, +corresponding to the child property specified by @prop_name. The +default implementation will search for the first child that matches. If +you overwrite this, you will likely still want to call the default +vmethod, which has access to the registered parameter specifications. @@ -12311,6 +12474,7 @@ specification of the child property + Return a the track types for the element. @@ -12417,10 +12581,10 @@ not the case. - + - + @@ -12454,7 +12618,7 @@ See #GESTitleSource for more information about exposed properties - + Get the pango font description used by @self. use #ges_timeline_element_get_children_properties instead. See #GESTitleSource for more information about exposed properties @@ -12470,7 +12634,7 @@ See #GESTitleSource for more information about exposed properties - + Get the horizontal aligment used by @self. use #ges_timeline_element_get_children_properties instead. See #GESTitleSource for more information about exposed properties @@ -12486,7 +12650,7 @@ See #GESTitleSource for more information about exposed properties - + Get the text currently set on @self. use #ges_timeline_element_get_children_properties instead. See #GESTitleSource for more information about exposed properties @@ -12518,7 +12682,7 @@ See #GESTitleSource for more information about exposed properties - + Get the vertical aligment used by @self. use #ges_timeline_element_get_children_properties instead. See #GESTitleSource for more information about exposed properties @@ -12534,7 +12698,7 @@ See #GESTitleSource for more information about exposed properties - + Get the horizontal position used by @self. use #ges_timeline_element_get_children_properties instead. See #GESTitleSource for more information about exposed properties @@ -12550,7 +12714,7 @@ See #GESTitleSource for more information about exposed properties - + Get the vertical position used by @self. use #ges_timeline_element_get_children_property instead @@ -12565,7 +12729,7 @@ See #GESTitleSource for more information about exposed properties - + Sets the background of the text. use #ges_timeline_element_set_children_properties instead. See #GESTitleSource for more information about exposed properties @@ -12584,7 +12748,7 @@ See #GESTitleSource for more information about exposed properties - + Sets the color of the text. use #ges_timeline_element_set_children_properties instead. See #GESTitleSource for more information about exposed properties @@ -12603,7 +12767,7 @@ See #GESTitleSource for more information about exposed properties - + Sets the pango font description of the text. use #ges_timeline_element_set_children_properties instead. See #GESTitleSource for more information about exposed properties @@ -12622,7 +12786,7 @@ See #GESTitleSource for more information about exposed properties - + Sets the horizontal aligment of the text. use #ges_timeline_element_set_children_properties instead. See #GESTitleSource for more information about exposed properties @@ -12641,7 +12805,7 @@ See #GESTitleSource for more information about exposed properties - + Sets the text this clip will render. use #ges_timeline_element_set_children_properties instead. See #GESTitleSource for more information about exposed properties @@ -12661,7 +12825,7 @@ made. - + Sets the vertical aligment of the text. use #ges_timeline_element_set_children_properties instead. See #GESTitleSource for more information about exposed properties @@ -12680,7 +12844,7 @@ See #GESTitleSource for more information about exposed properties - + Sets the horizontal position of the text. use #ges_timeline_element_set_children_properties instead. See #GESTitleSource for more information about exposed properties @@ -12699,7 +12863,7 @@ See #GESTitleSource for more information about exposed properties - + Sets the vertical position of the text. use #ges_timeline_element_set_children_properties instead. See #GESTitleSource for more information about exposed properties @@ -12718,56 +12882,56 @@ See #GESTitleSource for more information about exposed properties - + The background of the text use #ges_timeline_element_set_children_properties or #ges_timeline_element_get_children_properties instead. See #GESTitleSource for more information about exposed properties - + The color of the text use #ges_timeline_element_set_children_properties or #ges_timeline_element_get_children_properties instead. See #GESTitleSource for more information about exposed properties - + Pango font description string use #ges_timeline_element_set_children_properties or #ges_timeline_element_get_children_properties instead. See #GESTitleSource for more information about exposed properties - + Horizontal alignment of the text use #ges_timeline_element_set_children_properties or #ges_timeline_element_get_children_properties instead. See #GESTitleSource for more information about exposed properties - + The text to diplay use #ges_timeline_element_set_children_properties or #ges_timeline_element_get_children_properties instead. See #GESTitleSource for more information about exposed properties - + Vertical alignent of the text use #ges_timeline_element_set_children_properties or #ges_timeline_element_get_children_properties instead. See #GESTitleSource for more information about exposed properties - + The horizontal position of the text use #ges_timeline_element_set_children_properties or #ges_timeline_element_get_children_properties instead. See #GESTitleSource for more information about exposed properties - + The vertical position of the text use #ges_timeline_element_set_children_properties or #ges_timeline_element_get_children_properties instead. @@ -12797,7 +12961,7 @@ See #GESTitleSource for more information about exposed properties - + @@ -13087,7 +13251,7 @@ made. - + @@ -13228,7 +13392,7 @@ needed to be committed. - + Get the #GESTrack:caps of the track. @@ -13261,7 +13425,7 @@ all the #GESTrackElement-s in @track. - + Gets the #GESTrack:mixing of the track. @@ -13275,7 +13439,7 @@ all the #GESTrackElement-s in @track. - + Gets the #GESTrack:restriction-caps of the track. @@ -13367,7 +13531,7 @@ function already set appropriately. - + Sets the #GESTrack:mixing for the track. @@ -13384,7 +13548,7 @@ function already set appropriately. - + Sets the #GESTrack:restriction-caps for the track. > **NOTE**: Restriction caps are **not** taken into account when @@ -13457,7 +13621,7 @@ width=400". - + The capabilities used to choose the output of the #GESTrack's elements. Internally, this is used to select output streams when several may be available, by determining whether its #GstPad is @@ -13477,17 +13641,17 @@ this. Default value: #GST_CAPS_ANY. - + Current duration of the track Default value: O - + The #nlecomposition:id of the underlying #nlecomposition. - + Whether the track should support the mixing of #GESLayer data, such as composing the video data of each layer (when part of the video data is transparent, the next layer will become visible) or adding @@ -13495,7 +13659,7 @@ together the audio data. As such, for audio and video tracks, you'll likely want to keep this set to %TRUE. - + The capabilities that specifies the final output format of the #GESTrack. For example, for a video track, it would specify the height, width, framerate and other properties of the stream. @@ -13507,7 +13671,7 @@ You may change this property after the track has been added to a Default value: #GST_CAPS_ANY. - + The track type of the track. This controls the type of #GESTrackElement-s that can be added to the track. This should match with the track's #GESTrack:caps. @@ -13868,7 +14032,7 @@ for @trackelement. - + Gets #GESTrackElement:auto-clamp-control-sources. @@ -14056,7 +14220,7 @@ nleobject that @object wraps. - + Get the #GESTrackElement:track for the element. @@ -14071,7 +14235,7 @@ or %NULL if it does not belong to a track. - + Gets the #GESTrackElement:track-type for the element. @@ -14224,7 +14388,7 @@ binding from - + Sets #GESTrackElement:active for the element. @@ -14242,7 +14406,7 @@ binding from - + Sets #GESTrackElement:auto-clamp-control-sources. If set to %TRUE, this will immediately clamp all the control sources. @@ -14398,7 +14562,7 @@ gst_direct_control_binding_new_absolute() instead. - + Sets #GESTrackElement:has-internal-source for the element. If this is set to %FALSE, this method will also set the #GESTimelineElement:in-point of the element to 0 and its @@ -14421,7 +14585,7 @@ set to %FALSE, this method will also set the - + Sets the #GESTrackElement:track-type for the element. @@ -14438,13 +14602,13 @@ set to %FALSE, this method will also set the - + Whether the effect of the element should be applied in its #GESTrackElement:track. If set to %FALSE, it will not be used in the output of the track. - + Whether the control sources on the element (see ges_track_element_set_control_source()) will be automatically updated whenever the #GESTimelineElement:in-point or out-point of the @@ -14456,7 +14620,7 @@ per control source. Default value: %TRUE - + This property is used to determine whether the 'internal time' properties of the element have any meaning. In particular, unless this is set to %TRUE, the #GESTimelineElement:in-point and @@ -14493,12 +14657,12 @@ content and manipulate the timing of their data streams (time effects). - + The track that this element belongs to, or %NULL if it does not belong to a track. - + The track type of the element, which determines the type of track the element can be added to (see #GESTrack:track-type). This should correspond to the type of data that the element can produce or @@ -14597,7 +14761,7 @@ property of the track element. - + Get the GESAssetTrackType the #GESTrackElement extracted from @self should get into @@ -14612,7 +14776,7 @@ should get into - + Set the #GESTrackType the #GESTrackElement extracted from @self should get into @@ -14630,7 +14794,7 @@ should get into - + @@ -14679,7 +14843,7 @@ should get into - + @@ -14821,10 +14985,10 @@ usage. - + - + @@ -14834,19 +14998,19 @@ 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 @@ -14935,7 +15099,7 @@ or %NULL if something went wrong - + a #GESVideoStandardTransitionType representing the wipe to use @@ -14967,10 +15131,10 @@ to apply. - + - + @@ -14999,7 +15163,7 @@ the URI points to a file of some type. - + Get the location of the resource. @@ -15041,7 +15205,7 @@ the URI points to a file of some type. - + Sets whether the clip is a still image or not. @@ -15058,7 +15222,7 @@ the URI points to a file of some type. - + Sets whether the audio track of this clip is muted or not. @@ -15075,19 +15239,19 @@ the URI points to a file of some type. - + 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. @@ -15122,7 +15286,7 @@ before create_track_elements is called. - + Creates a #GESUriClipAsset for @uri Example of request of a GESUriClipAsset: @@ -15174,7 +15338,7 @@ ges_uri_clip_asset_new (uri, (GAsyncReadyCallback) filesource_asset_loaded_cb, u - + Creates a #GESUriClipAsset for @uri synchonously. You should avoid to use it in application, and rather create #GESUriClipAsset asynchronously @@ -15191,7 +15355,7 @@ You can also use multi file uris for #GESMultiFileSource. - + Gets duration of the file represented by @self @@ -15267,11 +15431,11 @@ are different as those can be extended 'infinitely'. - + The duration (in nanoseconds) of the media file - + The duration (in nanoseconds) of the media file @@ -15344,7 +15508,7 @@ are different as those can be extended 'infinitely'. - + @@ -15358,10 +15522,10 @@ are different as those can be extended 'infinitely'. - + - + @@ -15451,7 +15615,7 @@ contains one frame) - + @@ -15521,6 +15685,10 @@ account. + method to return the GstElement to put in the source topbin. +Other elements will be queued, like a videoscale. +In the case of a VideoUriSource, for example, the subclass will return a decodebin, +and we will append a videoscale. @@ -15601,285 +15769,285 @@ account. - + - + 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 - + Similar to crossfade, but fade in the front video without fading out the background one 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 @@ -15944,7 +16112,7 @@ the background one - + @@ -16008,7 +16176,7 @@ modify these fields, or add additional ones. - + @@ -16021,7 +16189,7 @@ modify these fields, or add additional ones. - + Get the border property of @self, this value represents the border width of the transition. Use ges_timeline_element_get_child_property instead. @@ -16038,7 +16206,7 @@ the border width of the transition. - + Get the transition type used by @trans. @@ -16068,7 +16236,7 @@ the direction of the transition. - + 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 @@ -16110,7 +16278,7 @@ for later use. - + Sets the transition being used to @type. @@ -16128,16 +16296,16 @@ for later use. - + This value represents the border width of the transition. - + This value represents the direction of the transition. Use ges_timeline_element_[sg]et_child_property instead. - + @@ -16164,7 +16332,7 @@ for later use. - + @@ -16174,7 +16342,7 @@ for later use. - + The location of the file/resource to use. @@ -16204,7 +16372,7 @@ for later use. - + @@ -16233,7 +16401,7 @@ for later use. - + diff --git a/girs/Gst-1.0.gir b/girs/Gst-1.0.gir index 2c2f64bf3b..0def2f8a85 100644 --- a/girs/Gst-1.0.gir +++ b/girs/Gst-1.0.gir @@ -398,6 +398,7 @@ use an alignment of 7. + implementation that acquires memory @@ -421,6 +422,7 @@ use an alignment of 7. + implementation that releases memory @@ -446,26 +448,26 @@ use an alignment of 7. Flags for allocators. - + The allocator has a custom alloc function. Only elements designed to work with this allocator should be using it, other elements should ignore it from allocation propositions. This implies %GST_ALLOCATOR_FLAG_NO_COPY. - + When copying a #GstMemory allocated with this allocator, the copy will instead be allocated using the default allocator. Use this when allocating a new memory is an heavy opperation that should only be done with a #GstBufferPool for example. - + 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. @@ -1551,13 +1553,13 @@ of @bin. See also gst_element_sync_state_with_parent(). - + 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. @@ -1874,19 +1876,19 @@ this message should chain up to the parent class implementation so the 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 - + 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. - + The last enum in the series of flags for bins. Derived classes can use this as first value in a list of flags. - + @@ -3500,91 +3502,91 @@ Either @nbuf or the #GstBuffer pointed to by @obuf may be %NULL. 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 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 the end of a frame boundary, for audio this is the start of a talkspurt. for RTP packets this matches the marker flag in the RTP packet header. - + 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. - + This buffer is important and should not be dropped. This can be used to mark important buffers, e.g. to flag RTP packets @@ -3592,7 +3594,7 @@ 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. - + additional media specific flags can be added starting from this flag. @@ -3627,7 +3629,7 @@ When @meta is set to %NULL, the item will be removed from the buffer. - + Buffer lists are an object containing a list of buffers. Buffer lists are created with gst_buffer_list_new() and filled with data @@ -4658,21 +4660,21 @@ allocating buffers. 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. @@ -4917,21 +4919,21 @@ successful. - + 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 @@ -5580,7 +5582,7 @@ matching message was posted on the bus. - + Enables async message delivery support for bus watches, gst_bus_pop() and similar API. Without this only the synchronous message handlers are called. @@ -5683,10 +5685,10 @@ gst_bus_enable_sync_message_emission() before. The standard flags that a bus may have. - + The bus is currently dropping all messages - + offset to define more flags @@ -5720,7 +5722,7 @@ Note that this function is used as a #GSourceFunc which means that returning - + @@ -5752,13 +5754,13 @@ message should not be unreffed by the sync 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 @@ -5780,6 +5782,7 @@ array cannot grow. + Reallocate @data. @@ -5946,6 +5949,219 @@ evaluates to @def_return. + + Output an debugging message in the given category. + +There is no need to finish the end of the message string with a newline +character, a newline character will be added automatically. + + + + category to use + + + printf-style message to output + + + + + Output an debugging message belonging to the given object in the given category. + +There is no need to finish the end of the message string with a newline +character, a newline character will be added automatically. + + + + category to use + + + the #GObject the message belongs to + + + printf-style message to output + + + + + Output an error message in the given category. + +There is no need to finish the end of the message string with a newline +character, a newline character will be added automatically. + + + + category to use + + + printf-style message to output + + + + + Output an error message belonging to the given object in the given category. + +There is no need to finish the end of the message string with a newline +character, a newline character will be added automatically. + + + + category to use + + + the #GObject the message belongs to + + + printf-style message to output + + + + + Output an fixme message in the given category. + +There is no need to finish the end of the message string with a newline +character, a newline character will be added automatically. + + + + category to use + + + printf-style message to output + + + + + Output a fixme message belonging to the given object in the given category. + +There is no need to finish the end of the message string with a newline +character, a newline character will be added automatically. + + + + category to use + + + the #GObject the message belongs to + + + printf-style message to output + + + + + Output an informational message in the given category. + +There is no need to finish the end of the message string with a newline +character, a newline character will be added automatically. + + + + category to use + + + printf-style message to output + + + + + Output an informational message belonging to the given object in the given +category. + +There is no need to finish the end of the message string with a newline +character, a newline character will be added automatically. + + + + category to use + + + the #GObject the message belongs to + + + printf-style message to output + + + + + Outputs a debugging message. This is the most general macro for outputting +debugging messages. You will probably want to use one of the ones described +below. + +There is no need to finish the end of the debug message with a newline +character, a newline character will be added automatically. + + + + category to use + + + the severity of the message + + + the #GObject the message belongs to or %NULL if none + + + A printf-style message to output + + + + + Outputs a debugging message. This is the most general macro for outputting +debugging messages. You will probably want to use one of the ones described +below. + +There is no need to finish the end of the debug message with a newline +character, a newline character will be added automatically. + + + + category to use + + + the severity of the message + + + the identifier of the object this message + relates to, or %NULL if none + + + A printf-style message to output + + + + + Output an logging message in the given category. + +There is no need to finish the end of the message string with a newline +character, a newline character will be added automatically. + + + + category to use + + + printf-style message to output + + + + + Output an logging message belonging to the given object in the given category. + +There is no need to finish the end of the message string with a newline +character, a newline character will be added automatically. + + + + category to use + + + the #GObject the message belongs to + + + printf-style message to output + + + Output a hexdump of @data in the given category. @@ -6016,6 +6232,73 @@ character, a newline character will be added automatically. + + Output a tracing message in the given category. + +There is no need to finish the end of the message string with a newline +character, a newline character will be added automatically. + + + + category to use + + + printf-style message to output + + + + + Output a tracing message belonging to the given object in the given +category. + +There is no need to finish the end of the message string with a newline +character, a newline character will be added automatically. + + + + category to use + + + the #GObject the message belongs to + + + printf-style message to output + + + + + Output an warning message in the given category. + +There is no need to finish the end of the message string with a newline +character, a newline character will be added automatically. + + + + category to use + + + printf-style message to output + + + + + Output a warning message belonging to the given object in the given category. + +There is no need to finish the end of the message string with a newline +character, a newline character will be added automatically. + + + + category to use + + + the #GObject the message belongs to + + + printf-style message to output + + + Check whether a GStreamer version equal to or greater than major.minor.micro is present. @@ -7559,7 +7842,7 @@ of @new_caps. - + #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. @@ -8178,7 +8461,7 @@ removed from the caps if %FALSE is returned. Extra flags for a caps. - + Caps has no specific content, but can contain anything. @@ -8229,10 +8512,10 @@ caps and intersect them with this mode. 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. @@ -8695,6 +8978,7 @@ usage. For plain #GObject @target is the same as @object. + Fetch a child object by name @@ -8715,6 +8999,7 @@ usage. For plain #GObject @target is the same as @object. + Fetch a child object by index @@ -8735,6 +9020,7 @@ usage. For plain #GObject @target is the same as @object. + Get the number of children in @parent @@ -8750,6 +9036,7 @@ usage. For plain #GObject @target is the same as @object. + Called when @child is added to @parent @@ -8772,6 +9059,7 @@ usage. For plain #GObject @target is the same as @object. + Called when @child is removed from @parent @@ -9413,7 +9701,7 @@ given invalid input. - + Gets the amount of time that master and slave clocks are sampled. @@ -9637,7 +9925,7 @@ is set on the clock, and is intended to be called by subclasses only. - + Sets the amount of time, in nanoseconds, to sample master and slave clocks @@ -9767,13 +10055,13 @@ is not set on the clock, or if the clock is already synced. - + - + - + @@ -10011,88 +10299,88 @@ not be extended or allocated using a custom allocator. 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 - + 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) - + time since Epoch, but using International Atomic Time as reference (Since: 1.18) - + #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. @@ -10585,7 +10873,7 @@ property, %FALSE otherwise - + @@ -10770,7 +11058,7 @@ or %NULL if the property isn't controlled. - + @@ -10928,53 +11216,53 @@ undefined contain NANs. 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. @@ -11071,6 +11359,18 @@ the metadata on @transbuf. + + Output a debugging message in the default category. + +There is no need to finish the end of the message string with a newline +character, a newline character will be added automatically. + + + + printf-style message to output + + + @@ -11244,6 +11544,37 @@ freed by the caller. + + Output a debugging message for the given identifier in the default category. + +There is no need to finish the end of the message string with a newline +character, a newline character will be added automatically. + + + + An identifier of the message provider + + + printf-style message to output + + + + + Output a debugging message belonging to the given object in the default +category. + +There is no need to finish the end of the message string with a newline +character, a newline character will be added automatically. + + + + the #GObject the message belongs to + + + printf-style message to output + + + Evaluates to 2 strings, that describe the pad. Often used in debugging statements. @@ -11526,7 +11857,7 @@ Used to generate `gst_dynamic_type_register_*(GstPlugin* plugin)`. - + Struct to store date, time and timezone information altogether. #GstDateTime is refcounted and immutable. @@ -12212,69 +12543,69 @@ debugging message. 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. @@ -12292,27 +12623,27 @@ pointer to a void pointer 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 @@ -12320,37 +12651,37 @@ and GST_DEBUG_BIN_TO_DOT_FILE_WITH_TS(). 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 @@ -12358,23 +12689,23 @@ message is, the greater the probability that the debugging system outputs it. - + 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. @@ -12392,7 +12723,7 @@ message is, the greater the probability that the debugging system outputs it. - + Gets the string representation of a #GstDebugMessage. This function is used @@ -12499,7 +12830,7 @@ create a unique name. - + Getter for the #GstCaps that this device supports. @@ -12514,7 +12845,7 @@ gst_caps_unref() when done. - + 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. @@ -12530,7 +12861,7 @@ classes of the #GstDeviceProvider that produced this device. - + Gets the user-friendly name of the device. @@ -12544,7 +12875,7 @@ classes of the #GstDeviceProvider that produced this device. - + Gets the extra properties of a device. @@ -12623,16 +12954,16 @@ device in the PLAYING state. - + - + - + - + @@ -12661,6 +12992,8 @@ device in the PLAYING state. + Creates the fully configured element to access this device. + Subclasses need to override this and return a new element. @@ -12682,6 +13015,9 @@ create a unique name. + This only needs to be implemented by subclasses if the + element can be reconfigured to use a different device. See the documentation + for gst_device_reconfigure_element(). @@ -12945,7 +13281,7 @@ will be emitted on the bus when the list of devices changes. - + @@ -12974,10 +13310,10 @@ will be emitted on the bus when the list of devices changes. - + - + @@ -13020,6 +13356,9 @@ from all relevant providers. + Returns a list of devices that are currently available. + This should never block. The devices should not have a parent and should + be floating. @@ -13385,6 +13724,9 @@ all devices again. + Returns a list of devices that are currently available. + This should never block. The devices should not have a parent and should + be floating. @@ -13400,6 +13742,8 @@ all devices again. + Starts monitoring for new devices. Only subclasses that can know + that devices have been added or remove need to implement this method. @@ -13415,6 +13759,8 @@ all devices again. + Stops monitoring for new devices. Only subclasses that implement + the start() method need to implement this method. @@ -13750,11 +14096,11 @@ metadata. Free with g_strfreev() when no longer needed. - + The opaque #GstDeviceProviderFactoryClass data structure. - + @@ -13793,7 +14139,7 @@ plugin_init (GstPlugin * plugin) - + @@ -14457,6 +14803,48 @@ application will be informed. + + Output an error message in the default category. + +There is no need to finish the end of the message string with a newline +character, a newline character will be added automatically. + + + + printf-style message to output + + + + + Output an error message for the given identifier in the default category. + +There is no need to finish the end of the message string with a newline +character, a newline character will be added automatically. + + + + An identifier of the message provider + + + printf-style message to output + + + + + Output an error message belonging to the given object in the default category. + +There is no need to finish the end of the message string with a newline +character, a newline character will be added automatically. + + + + the #GObject the message belongs to + + + printf-style message to output + + + Builds a string using errno describing the previously failed system call. To be used as the debug argument in #GST_ELEMENT_ERROR. @@ -14924,6 +15312,7 @@ MT safe. + called when a request pad is to be released @@ -15092,6 +15481,7 @@ MT safe. + called immediately after a new state was set. @@ -17202,6 +17592,7 @@ functionality. + called when a new pad is requested @@ -17232,6 +17623,7 @@ request. Can be %NULL. + called when a request pad is to be released @@ -17248,6 +17640,7 @@ request. Can be %NULL. + get the state of the element @@ -17283,6 +17676,7 @@ MT safe. + set a new state on the element @@ -17304,6 +17698,7 @@ MT safe. + called by @set_state to perform an incremental state change @@ -17323,6 +17718,7 @@ MT safe. + called immediately after a new state was set. @@ -17345,6 +17741,7 @@ MT safe. + set a #GstBus on the element @@ -17363,6 +17760,7 @@ MT safe. + gets the #GstClock provided by the element @@ -17381,6 +17779,7 @@ MT safe. + set the #GstClock on the element @@ -17404,6 +17803,7 @@ MT safe. + send a #GstEvent to the element @@ -17424,6 +17824,7 @@ as flushing seeks and steps) will emit %GST_MESSAGE_ASYNC_DONE. + perform a #GstQuery on the element @@ -17445,6 +17846,8 @@ MT safe. + called when a message is posted on the element. Chain up to + the parent class' handler to have it posted on the bus. @@ -17467,6 +17870,7 @@ MT safe. + set a #GstContext on the element @@ -18292,30 +18696,30 @@ make a copy of the protocol string array if you need to. - + 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 @@ -20104,128 +20508,128 @@ 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. - + Notify downstream that a playback rate override should be applied as soon as possible. (Since: 1.18) - + 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) - + Sent by the pipeline to notify elements that handle the instant-rate-change event about the running-time when the rate multiplier should be applied (or was applied). (Since: 1.18) - + 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. @@ -20291,24 +20695,66 @@ A lower value represents a higher-priority event. #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. + + Output a fixme message in the default category. + +There is no need to finish the end of the message string with a newline +character, a newline character will be added automatically. + + + + printf-style message to output + + + + + Output a fixme message for the given identifier in the default category. + +There is no need to finish the end of the message string with a newline +character, a newline character will be added automatically. + + + + An identifier of the message provider + + + printf-style message to output + + + + + Output a fixme message belonging to the given object in the default category. + +There is no need to finish the end of the message string with a newline +character, a newline character will be added automatically. + + + + the #GObject the message belongs to + + + printf-style message to output + + + A mask value with all bits set, for use as a GstFlagSet mask where all flag bits must match @@ -20371,77 +20817,77 @@ when serializing, for easier debugging. 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 using GST_ELEMENT_ERROR() 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) @@ -20598,7 +21044,7 @@ Can be used as a default value in variables used to store group_id. The different flags that can be set on #GST_EVENT_GAP events. See gst_event_set_gap_flags() for details. - + The #GST_EVENT_GAP signals missing data, for example because of packet loss. @@ -20838,9 +21284,53 @@ is unlinked and links to the new target are established. if @newtarget is - + + + Output an informational message in the default category. + +There is no need to finish the end of the message string with a newline +character, a newline character will be added automatically. + + + + printf-style message to output + + + + + Output an informational message for the given identifier the default +category. + +There is no need to finish the end of the message string with a newline +character, a newline character will be added automatically. + + + + An identifier of the message provider + + + printf-style message to output + + + + + Output an informational message belonging to the given object in the default +category. + +There is no need to finish the end of the message string with a newline +character, a newline character will be added automatically. + + + + the #GObject the message belongs to + + + printf-style message to output + + + @@ -22330,13 +22820,13 @@ The function will be called with the iterator lock held. The result of a #GstIteratorItemFunction. - + Skip this item - + Return item - + Stop after this item. @@ -22387,16 +22877,16 @@ The function will be called with the iterator lock held. The result of gst_iterator_next(). - + No more items in the iterator - + An item was retrieved - + Datastructure changed while iterating - + An error happened @@ -22430,30 +22920,72 @@ The function will be called with the iterator lock held. + + Output a logging message in the default category. + +There is no need to finish the end of the message string with a newline +character, a newline character will be added automatically. + + + + printf-style message to output + + + + + Output a logging message for the given identifier in the default category. + +There is no need to finish the end of the message string with a newline +character, a newline character will be added automatically. + + + + An identifier of the message provider + + + printf-style message to output + + + + + Output a logging message belonging to the given object in the default category. + +There is no need to finish the end of the message string with a newline +character, a newline character will be added automatically. + + + + the #GObject the message belongs to + + + printf-style message to output + + + 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. @@ -22465,16 +22997,16 @@ deciding where to go while developing code. 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 @@ -23006,13 +23538,13 @@ gst_mini_object_lock() and gst_mini_object_unlock(). Flags used when mapping memory - + map for read access - + map for write access - + first flag that can be used for custom purposes @@ -23485,32 +24017,32 @@ If @size is set to -1, all bytes starting at @offset are copied. 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. (DEPRECATED: do not use in new code, instead you should create a custom GstAllocator for memory pooling instead of relying on the GstBuffer they were originally attached to.) - + 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 @@ -26344,32 +26876,32 @@ of @new_message. 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. It is possible to specify a redirection url to the error messages by setting a `redirect-location` field into the error message, application or high level bins might use the information as required. - + 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 @@ -26377,152 +26909,152 @@ 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) - + Message indicating a #GstDevice was changed a #GstDeviceProvider (Since: 1.16) - + Message sent by elements to request the running time from the pipeline when an instant rate change should be applied (which may be in the past when the answer arrives). (Since: 1.18) - + mask for all of the above messages. @@ -26920,19 +27452,19 @@ when a pooled buffer is returned. 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. @@ -27670,20 +28202,20 @@ and the memory associated with the object is freed. 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. @@ -28073,6 +28605,7 @@ Either @newobj and the value pointed to by @oldobj may be %NULL. + default signal handler @@ -28220,7 +28753,7 @@ curve or apply a control curve sample by sample. - + 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() @@ -28242,7 +28775,7 @@ MT safe. This function grabs and releases @object's LOCK. - + Returns the parent of @object. This function increases the refcount of the parent object so you should gst_object_unref() it after usage. @@ -28533,7 +29066,7 @@ The control-rate should not change if the element is in %GST_STATE_PAUSED or - + 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 @@ -28558,7 +29091,7 @@ MT safe. This function grabs and releases @object's LOCK. - + 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()). @@ -28653,10 +29186,10 @@ this might deadlock the dispose function. - + - + 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 @@ -28728,6 +29261,7 @@ the elements contained in that bin. + default signal handler @@ -28754,12 +29288,12 @@ the elements contained in that bin. 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) - + Flag that's set when the object has been constructed. This can be used by API such as base class setters to differentiate between the case where they're called from a subclass's instance init function (and where the @@ -28767,7 +29301,7 @@ object isn't fully constructed yet, and so one shouldn't do anything but set values in the instance structure), and the case where the object is constructed. - + subclasses can add additional flags starting from this flag @@ -30389,7 +30923,7 @@ incremented ref-count or %NULL when pad has no caps. Unref after usage. - + Gets the direction of the pad. The direction of the pad is decided at construction time so this function does not take the LOCK. @@ -30435,7 +30969,7 @@ No locking is performed in this function. - + Get the offset applied to the running time of @pad. @pad has to be a source pad. @@ -31813,7 +32347,7 @@ of the peer sink pad, if present. - + Set the offset that will be applied to the running time of @pad. Upon next buffer, every sticky events (notably segment) will be pushed again with their running time adjusted. For that reason this is only reliable on @@ -32025,10 +32559,10 @@ be renegotiated to something else. - + - + The offset that will be applied to the running time of the pad. @@ -32369,13 +32903,13 @@ post an error on the bus and return an appropriate #GstFlowReturn value. The direction of a pad. - + direction is unknown. - + the pad is a source pad. - + the pad is a sink pad. @@ -32434,63 +32968,63 @@ return it. 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 @@ -32610,28 +33144,28 @@ specified, expensive but safe @GST_PAD_LINK_CHECK_CAPS are performed. > 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()). @@ -32662,25 +33196,25 @@ specified, expensive but safe @GST_PAD_LINK_CHECK_CAPS are performed. 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 @@ -32688,13 +33222,13 @@ specified, expensive but safe @GST_PAD_LINK_CHECK_CAPS are performed. 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 @@ -32714,18 +33248,18 @@ pad operates in push or pull 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(). - + @@ -32848,31 +33382,31 @@ The callback is allowed to modify the data pointer in @info. 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, passing the data. For blocking probes this will cause data flow to unblock, unless there are also other blocking probes installed. - + pass the data item in the block probe and block on the next item. Note, that if there are multiple pad probes installed and any probe returns PASS, the data will be passed. - + 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 @@ -32887,73 +33421,73 @@ The callback is allowed to modify the data pointer in @info. 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 @@ -33166,7 +33700,7 @@ and with the given arguments. - + Gets the capabilities of the pad template. @@ -33233,11 +33767,11 @@ expose "stable" caps to the reader. - + The capabilities of the pad described by the pad template. - + The direction of the pad described by the pad template. @@ -33245,11 +33779,11 @@ expose "stable" caps to the reader. 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. @@ -33329,7 +33863,7 @@ expose "stable" caps to the reader. Flags for the padtemplate - + first flag that can be used by subclasses. @@ -33435,7 +33969,7 @@ for re-use. - + Opaque structure. @@ -33500,28 +34034,28 @@ of %GST_PARSE_ERROR_NO_SUCH_ELEMENT was returned. 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. @@ -33534,19 +34068,19 @@ of %GST_PARSE_ERROR_NO_SUCH_ELEMENT was returned. 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) @@ -33640,7 +34174,7 @@ MT safe. - + Check if @pipeline will automatically flush messages when going to the NULL state. @@ -33711,7 +34245,7 @@ MT safe. - + Get the configured delay (see gst_pipeline_set_delay()). @@ -33727,7 +34261,7 @@ MT safe. - + Gets the latency that should be configured on the pipeline. See gst_pipeline_set_latency(). @@ -33775,7 +34309,7 @@ MT safe. - + 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. @@ -33828,7 +34362,7 @@ MT safe. - + 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 @@ -33854,7 +34388,7 @@ MT safe. - + 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 @@ -33901,19 +34435,19 @@ MT safe. - + 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(). @@ -33959,14 +34493,14 @@ see gst_pipeline_set_delay() for more information on this option. Pipeline flags - + this pipeline works with a fixed clock - + offset to define more flags - + @@ -34563,38 +35097,38 @@ The cache is flushed every time the registry is rebuilt. - + Ignore enum members when generating the plugins cache. This is useful if the members of the enum are generated dynamically, in order not to expose incorrect documentation to the end user. - + 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. @@ -34664,13 +35198,13 @@ BSD, MIT/X11, Proprietary, unknown. 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 @@ -34883,7 +35417,7 @@ the most appropriate feature. - + @@ -34927,10 +35461,10 @@ to get a list of plugins that match certain criteria. The plugin loading state - + Temporarily loaded plugins - + The plugin won't be scanned (again) @@ -34973,7 +35507,7 @@ register each #GstPluginFeature. - + 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 @@ -35862,6 +36396,7 @@ data @tag names can be something like e.g. "comment". Supplying %NULL for the + virtual method to get list of presets @@ -35880,6 +36415,7 @@ data @tag names can be something like e.g. "comment". Supplying %NULL for the + virtual methods to get properties that are persistent @@ -35898,6 +36434,7 @@ data @tag names can be something like e.g. "comment". Supplying %NULL for the + virtual methods to load a preset into properties @@ -35917,6 +36454,7 @@ data @tag names can be something like e.g. "comment". Supplying %NULL for the + virtual methods to save properties into a preset @@ -35936,6 +36474,7 @@ data @tag names can be something like e.g. "comment". Supplying %NULL for the + virtual methods to rename a preset @@ -35959,6 +36498,7 @@ data @tag names can be something like e.g. "comment". Supplying %NULL for the + virtual methods to remove a preset @@ -35978,6 +36518,7 @@ data @tag names can be something like e.g. "comment". Supplying %NULL for the + virtual methods to set textual meta data to a preset @@ -36005,6 +36546,7 @@ data @tag names can be something like e.g. "comment". Supplying %NULL for the + virtual methods to get textual meta data from a preset @@ -36041,19 +36583,19 @@ or no value for the given @tag 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. @@ -36278,18 +36820,18 @@ immediately with the current result. 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. @@ -36447,24 +36989,24 @@ be %NULL. Unref target pad after usage. - + 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. @@ -38523,68 +39065,68 @@ Either @new_query or the #GstQuery pointed to by @old_query may be %NULL. 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) - + the bitrate query (since 1.16) - + Query stream selection capability. @@ -38634,13 +39176,13 @@ Either @new_query or the #GstQuery pointed to by @old_query may be %NULL. #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. @@ -38907,16 +39449,16 @@ 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 @@ -39480,64 +40022,64 @@ replacing a previously-added one by the same name) - + 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. @@ -39962,7 +40504,7 @@ strings. - + A #GstSample is a small object containing data, a type, timing and extra arbitrary information. @@ -40208,25 +40750,25 @@ sample will be freed. 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. @@ -40282,67 +40824,67 @@ 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) - + When doing fast forward or fast reverse playback, request that elements only decode keyframes and forward predicted frames and skip all other content (for example B-Frames), for formats that have keyframes and forward predicted frames. (Since: 1.18) - + Signals that a rate change should be applied immediately. Only valid if start/stop position are GST_CLOCK_TIME_NONE, the playback direction does not change @@ -40352,13 +40894,13 @@ elements that handle them. 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 @@ -41072,45 +41614,45 @@ to get the real negative 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) - + Decode only keyframes or forward predicted frames, where possible (Since: 1.18) - + Do not decode any audio, where possible (Since: 1.6) - + No special flags specified. - + Serialize using the old format for nested structures. - + Serialization fails if a value cannot be serialized instead of using placeholder "NULL" value (e.g. pointers, objects). @@ -41189,16 +41731,16 @@ Setting @max_threads to 0 effectively freezes the pool. - + - + Try to retrieve the minimum information available, which may be none on some platforms (Since: 1.18) - + Try to retrieve as much information as possible, including source information when getting the stack trace @@ -41207,21 +41749,21 @@ Setting @max_threads to 0 effectively freezes the pool. 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. @@ -41230,14 +41772,14 @@ gst_element_set_state() and checked using gst_element_get_state(). These are the different state changes an element goes through. %GST_STATE_NULL &rArr; %GST_STATE_PLAYING is called an upwards state change and %GST_STATE_PLAYING &rArr; %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. @@ -41249,7 +41791,7 @@ and %GST_STATE_PLAYING &rArr; %GST_STATE_NULL a downwards state change. - + state change from PAUSED to PLAYING. * Most elements ignore this state change. * The pipeline selects a #GstClock and distributes this to all the children @@ -41266,7 +41808,7 @@ and %GST_STATE_PLAYING &rArr; %GST_STATE_NULL a downwards state change. - + state change from PLAYING to PAUSED. * Most elements ignore this state change. * The pipeline calculates the running_time based on the last selected @@ -41281,7 +41823,7 @@ and %GST_STATE_PLAYING &rArr; %GST_STATE_NULL a downwards state change. - + state change from PAUSED to READY. * Sinks unblock any waits in the preroll. * Elements unblock any waits on devices @@ -41291,25 +41833,25 @@ and %GST_STATE_PLAYING &rArr; %GST_STATE_NULL a downwards state change. - + 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) @@ -41331,16 +41873,16 @@ they should make sure to go back to real 'PAUSED' state (prerolling for example) 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. @@ -41487,7 +42029,7 @@ a new one will be automatically generated - + Retrieve the caps for @stream, if any @@ -41501,7 +42043,7 @@ a new one will be automatically generated - + Retrieve the current stream flags for @stream @@ -41515,7 +42057,7 @@ a new one will be automatically generated - + Returns the stream ID of @stream. @@ -41530,7 +42072,7 @@ during the lifetime of @stream. - + Retrieve the stream type for @stream @@ -41544,7 +42086,7 @@ during the lifetime of @stream. - + Retrieve the tags for @stream, if any @@ -41558,7 +42100,7 @@ during the lifetime of @stream. - + Set the caps for the #GstStream @@ -41575,7 +42117,7 @@ during the lifetime of @stream. - + Set the @flags for the @stream. @@ -41592,7 +42134,7 @@ during the lifetime of @stream. - + Set the stream type of @stream @@ -41609,7 +42151,7 @@ during the lifetime of @stream. - + Set the tags for the #GstStream @@ -41626,23 +42168,23 @@ during the lifetime of @stream. - + 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. @@ -41708,6 +42250,7 @@ Applications can activate streams from a collection by using the + default signal handler for the stream-notify signal @@ -41776,7 +42319,7 @@ The caller should not modify the returned #GstStream - + Returns the upstream id of the @collection. @@ -41790,7 +42333,7 @@ The caller should not modify the returned #GstStream - + stream-id @@ -41834,6 +42377,7 @@ streams within the collection. + default signal handler for the stream-notify signal @@ -41858,62 +42402,62 @@ streams within the collection. - + 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. @@ -41924,20 +42468,20 @@ can't be decrypted because no suitable key is available. - + 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 @@ -41945,31 +42489,31 @@ can't be decrypted because no suitable key is available. 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 @@ -41980,19 +42524,19 @@ 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. @@ -44555,10 +45099,10 @@ It is a programming error if both @newstr and the value pointed to by The type of a %GST_MESSAGE_STRUCTURE_CHANGE. - + Pad linking is starting or done. - + Pad unlinking is starting or done. @@ -44751,7 +45295,7 @@ MT safe. - + @@ -44777,7 +45321,7 @@ MT safe. - + @@ -45608,6 +46152,18 @@ returned by gst_toc_entry_set_loop() to indicate infinite looping. + + Output a tracing message in the default category. + +There is no need to finish the end of the message string with a newline +character, a newline character will be added automatically. + + + + printf-style message to output + + + @@ -45692,6 +46248,36 @@ returned by gst_toc_entry_set_loop() to indicate infinite looping. + + Output a tracing message for the given identifier in the default category. + +There is no need to finish the end of the message string with a newline +character, a newline character will be added automatically. + + + + An identifier of the message provider + + + printf-style message to output + + + + + Output a tracing message belonging to the given object in the default category. + +There is no need to finish the end of the message string with a newline +character, a newline character will be added automatically. + + + + the #GObject the message belongs to + + + printf-style message to output + + + @@ -45798,19 +46384,19 @@ Used to generate `gst_type_find_register_*(GstPlugin* plugin)`. Extra tag flags used when registering tags. - + undefined flag - + tag is meta data - + tag is encoded - + tag is decoded - + number of tag flags @@ -47147,38 +47733,38 @@ In the table below this is shown for the cases that a tag exists in the list | PREPEND | B, A | A | B | ø | | KEEP | A | A | B | ø | | KEEP_ALL | A | A | ø | ø | - + 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 @@ -48100,6 +48686,7 @@ instead. + prepare the threadpool @@ -48114,6 +48701,7 @@ instead. + make sure all threads are stopped @@ -48128,6 +48716,7 @@ instead. + start a new thread @@ -48155,6 +48744,7 @@ instead. + join a thread @@ -48209,18 +48799,18 @@ instead. - + The different states a task can be in - + the task is started and running - + the task is stopped - + the task is paused @@ -48257,7 +48847,7 @@ instead. - + #GstToc functions are used to create/free #GstToc and #GstTocEntry structures. Also they are used to convert #GstToc into #GstStructure and vice versa. @@ -48448,7 +49038,7 @@ a track listing from different sources). - + Create new #GstTocEntry structure. @@ -48735,25 +49325,25 @@ values, %FALSE otherwise. 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) @@ -48776,26 +49366,26 @@ There are two types of TOC entries: alternatives or parts in a sequence. 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 @@ -48907,7 +49497,7 @@ contextual data, which they must not modify. - + @@ -48971,10 +49561,10 @@ the factory is not loaded. - + - + @@ -49044,21 +49634,21 @@ the category "GST_TRACER". - + 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. @@ -49069,16 +49659,16 @@ 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 @@ -49087,6 +49677,7 @@ events may contain values for different #GstPads. stream. + Method to peek data. @@ -49106,6 +49697,7 @@ stream. + Method to suggest #GstCaps with a given probability. @@ -49129,6 +49721,7 @@ stream. + Returns the length of current data. @@ -49466,7 +50059,7 @@ e.g. assume a certain media type based on the file extension. - + @@ -49489,22 +50082,22 @@ e.g. assume a certain media type based on the file extension. 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. @@ -49517,17 +50110,17 @@ in doing a reliable typefind. 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 @@ -49662,6 +50255,7 @@ Returns #GST_URI_UNKNOWN if the @handler isn't implemented correctly. + Method to tell whether the element handles source or sink URI. @@ -49675,6 +50269,7 @@ Returns #GST_URI_UNKNOWN if the @handler isn't implemented correctly. + Method to return the list of protocols handled by the element. @@ -49690,6 +50285,7 @@ Returns #GST_URI_UNKNOWN if the @handler isn't implemented correctly. + Method to return the URI currently handled by the element. @@ -49708,6 +50304,7 @@ Returns #GST_URI_UNKNOWN if the @handler isn't implemented correctly. + Method to set a new URI. @@ -49729,13 +50326,13 @@ Returns #GST_URI_UNKNOWN if the @handler isn't implemented correctly. The different types of URI direction. - + The URI direction is unknown - + The URI is a consumer. - + The URI is a producer. @@ -49786,7 +50383,7 @@ Returns #GST_URI_UNKNOWN if the @handler isn't implemented correctly. - + A #GstUri object can be used to parse and split a URI string into its constituent parts. Two #GstUri objects can be joined to make a new #GstUri using the algorithm described in RFC3986. @@ -51378,7 +51975,7 @@ Free-function: g_free - + #GstVecDeque is an object that provides standard double-ended queue (deque) functionality based on an array instead of linked lists. This reduces the overhead caused by memory management by a large factor. @@ -51835,6 +52432,48 @@ of size @struct_size, with an initial queue size of @initial_size. + + Output a warning message in the default category. + +There is no need to finish the end of the message string with a newline +character, a newline character will be added automatically. + + + + printf-style message to output + + + + + Output a warning message for the given identifier in the default category. + +There is no need to finish the end of the message string with a newline +character, a newline character will be added automatically. + + + + An identifier of the message provider + + + printf-style message to output + + + + + Output a warning message belonging to the given object in the default category. + +There is no need to finish the end of the message string with a newline +character, a newline character will be added automatically. + + + + the #GObject the message belongs to + + + printf-style message to output + + + Store a 16 bit unsigned integer value in big endian format into the memory buffer. diff --git a/girs/GstAllocators-1.0.gir b/girs/GstAllocators-1.0.gir index 6828ff3b93..60e8ece52a 100644 --- a/girs/GstAllocators-1.0.gir +++ b/girs/GstAllocators-1.0.gir @@ -161,10 +161,10 @@ number and the height is scaled according to the sub-sampling. - + - + @@ -405,6 +405,8 @@ The memory is only mmapped on gst_buffer_map() request. + Implementations shall return the physicall memory address + that is backing the provided memory, or 0 if none. @@ -426,6 +428,8 @@ The memory is only mmapped on gst_buffer_map() request. + Implementations shall return the physicall memory address + that is backing the provided memory, or 0 if none. diff --git a/girs/GstAnalytics-1.0.gir b/girs/GstAnalytics-1.0.gir index f074af230a..0ad873c7c6 100644 --- a/girs/GstAnalytics-1.0.gir +++ b/girs/GstAnalytics-1.0.gir @@ -200,6 +200,9 @@ have a static lifetime (never be freed). + A pointer to a function that will be called +when the containing meta is transform to potentially copy the data +into a new Mtd into the new meta. @@ -225,6 +228,9 @@ have a static lifetime (never be freed). + A pointer to a function that will be called when the +containing meta is cleared to potetially do cleanup (ex. _unref or release) +resources it was using. @@ -354,7 +360,7 @@ identified by @id is stored. Only use for criteria. - + An opaque #GstMeta that can be used to hold various types of results from analysis processes. diff --git a/girs/GstApp-1.0.gir b/girs/GstApp-1.0.gir index 8fbc44a4f5..336da1aca9 100644 --- a/girs/GstApp-1.0.gir +++ b/girs/GstApp-1.0.gir @@ -52,13 +52,13 @@ and/or use gtk-doc annotations. --> Buffer dropping scheme to avoid the element's internal queue to block when full. - + Not Leaky - + Leaky on upstream (new buffers) - + Leaky on downstream (old buffers) @@ -309,7 +309,7 @@ condition. - + Get the configured caps on @appsink. @@ -323,7 +323,7 @@ condition. - + Check if @appsink will drop old buffers when the maximum amount of queued data is reached (meaning max buffers, time or bytes limit, whichever is hit first). @@ -339,7 +339,7 @@ filled. - + Check if appsink will emit the "new-preroll" and "new-sample" signals. @@ -354,7 +354,7 @@ signals. - + Get the maximum amount of buffers that can be queued in @appsink. @@ -368,7 +368,7 @@ signals. - + Get the maximum total size, in bytes, that can be queued in @appsink. @@ -382,7 +382,7 @@ signals. - + Get the maximum total duration that can be queued in @appsink. @@ -396,7 +396,7 @@ signals. - + Check if @appsink will wait for all buffers to be consumed when an EOS is received. @@ -573,7 +573,7 @@ way. - + 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, @@ -593,7 +593,7 @@ you must check the caps on the samples to get the actual used caps. - + Instruct @appsink to drop old buffers when the maximum amount of queued data is reached, that is, when any configured limit is hit (max-buffers, max-time or max-bytes). @@ -611,7 +611,7 @@ data is reached, that is, when any configured limit is hit (max-buffers, max-tim - + 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. @@ -630,7 +630,7 @@ the application prefers to operate in pull mode. - + 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, unless 'drop' is set, in which @@ -650,7 +650,7 @@ case new buffers will be discarded. - + Set the maximum total size 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, unless 'drop' is set, in which @@ -670,7 +670,7 @@ case new buffers will be discarded. - + Set the maximum total duration 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, unless 'drop' is set, in which @@ -690,7 +690,7 @@ case new buffers will be discarded. - + Instruct @appsink to wait for all buffers to be consumed when an EOS is received. @@ -811,34 +811,34 @@ condition. - + - + - + - + - + - + Maximum amount of buffers in the queue (0 = unlimited). - + Maximum amount of bytes in the queue (0 = unlimited) - + Maximum total duration of data in the queue (0 = unlimited) - + Wait for all buffers to be processed after receiving an EOS. In cases where it is uncertain if an @appsink will have a consumer for its buffers @@ -1075,6 +1075,8 @@ for the EOS condition. gst_app_sink_set_callbacks(). + Called when the end-of-stream has been reached. This callback + is called from the streaming thread. @@ -1091,6 +1093,11 @@ gst_app_sink_set_callbacks(). + Called when a new preroll sample is available. + This callback is called from the streaming thread. + The new preroll sample can be retrieved with + gst_app_sink_pull_preroll() either from this callback + or from any other thread. @@ -1107,6 +1114,11 @@ gst_app_sink_set_callbacks(). + Called when a new sample is available. + This callback is called from the streaming thread. + The new sample can be retrieved with + gst_app_sink_pull_sample() either from this callback + or from any other thread. @@ -1123,6 +1135,14 @@ gst_app_sink_set_callbacks(). + Called when a new event is available. + This callback is called from the streaming thread. + The new event can be retrieved with + gst_app_sink_pull_event() either from this callback + or from any other thread. + The callback should return %TRUE if the event has been handled, + %FALSE otherwise. + Since: 1.20 @@ -1139,6 +1159,12 @@ gst_app_sink_set_callbacks(). + Called when the propose_allocation query is available. + This callback is called from the streaming thread. + The allocation query can be retrieved with + gst_app_sink_propose_allocation() either from this callback + or from any other thread. + Since: 1.24 @@ -1305,7 +1331,7 @@ Call gst_mini_object_unref() after usage. - + @@ -1526,7 +1552,7 @@ element is the last buffer of the stream. - + Get the configured caps on @appsrc. @@ -1540,7 +1566,7 @@ element is the last buffer of the stream. - + Get the number of currently queued buffers inside @appsrc. @@ -1554,7 +1580,7 @@ element is the last buffer of the stream. - + Get the number of currently queued bytes inside @appsrc. @@ -1568,7 +1594,7 @@ element is the last buffer of the stream. - + Get the amount of currently queued time inside @appsrc. @@ -1582,7 +1608,7 @@ element is the last buffer of the stream. - + Get the duration of the stream in nanoseconds. A value of GST_CLOCK_TIME_NONE means that the duration is not known. @@ -1597,7 +1623,7 @@ not known. - + Check if appsrc will emit the "new-preroll" and "new-buffer" signals. @@ -1633,7 +1659,7 @@ signals. - + Returns the currently set #GstAppLeakyType. See gst_app_src_set_leaky_type() for more details. @@ -1648,7 +1674,7 @@ for more details. - + Get the maximum amount of buffers that can be queued in @appsrc. @@ -1662,7 +1688,7 @@ for more details. - + Get the maximum amount of bytes that can be queued in @appsrc. @@ -1676,7 +1702,7 @@ for more details. - + Get the maximum amount of time that can be queued in @appsrc. @@ -1690,7 +1716,7 @@ for more details. - + Get the size of the stream in bytes. A value of -1 means that the size is not known. @@ -1705,7 +1731,7 @@ not known. - + Get the stream type. Control the stream type of @appsrc with gst_app_src_set_stream_type(). @@ -1833,7 +1859,7 @@ way. - + 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 @@ -1853,7 +1879,7 @@ buffers must match the caps or left NULL. - + Set the duration of the stream in nanoseconds. A value of GST_CLOCK_TIME_NONE means that the duration is not known. @@ -1871,7 +1897,7 @@ not known. - + 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. @@ -1912,7 +1938,7 @@ default latency calculations for pseudo-live sources will be used. - + When set to any other value than GST_APP_LEAKY_TYPE_NONE then the appsrc will drop any buffers that are pushed into it once its internal queue is full. The selected type defines whether to drop the oldest or new @@ -1932,7 +1958,7 @@ buffers. - + Set the maximum amount of buffers that can be queued in @appsrc. After the maximum amount of buffers are queued, @appsrc will emit the "enough-data" signal. @@ -1951,7 +1977,7 @@ After the maximum amount of buffers are queued, @appsrc will emit the - + 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. @@ -1970,7 +1996,7 @@ After the maximum amount of bytes are queued, @appsrc will emit the - + Set the maximum amount of time that can be queued in @appsrc. After the maximum amount of time are queued, @appsrc will emit the "enough-data" signal. @@ -1989,7 +2015,7 @@ After the maximum amount of time are queued, @appsrc will emit the - + Set the size of the stream in bytes. A value of -1 means that the size is not known. @@ -2007,7 +2033,7 @@ not known. - + Set the stream type on @appsrc. For seekable streams, the "seek" signal must be connected to. @@ -2027,46 +2053,46 @@ A stream_type stream - + When max-bytes are queued and after the enough-data signal has been emitted, block any further push-buffer calls until the amount of queued bytes drops below the max-bytes limit. - + The GstCaps that will negotiated downstream and will be put on outgoing buffers. - + The number of currently queued buffers inside appsrc. - + The number of currently queued bytes inside appsrc. - + The amount of currently queued time inside appsrc. - + The total duration in nanoseconds of the data stream. If the total duration is known, it is recommended to configure it with this property. - + Make appsrc emit the "need-data", "enough-data" and "seek-data" signals. This option is by default enabled for backwards compatibility reasons but can disabled when needed because signal emission is expensive. - + The format to use for segment events. When the source is producing timestamped buffers this property should be set to GST_FORMAT_TIME. - + When enabled, appsrc will check GstSegment in GstSample which was pushed via gst_app_src_push_sample() or "push-sample" signal action. If a GstSegment is changed, corresponding segment event will be followed @@ -2077,55 +2103,55 @@ GstAppSrc::format should be time. However, possibly #GstAppSrc can support other formats. - + Instruct the source to behave like a live source. This includes that it will only push out buffers in the PLAYING state. - + When set to any other value than GST_APP_LEAKY_TYPE_NONE then the appsrc will drop any buffers that are pushed into it once its internal queue is full. The selected type defines whether to drop the oldest or new buffers. - + The maximum amount of buffers that can be queued internally. After the maximum amount of buffers are queued, appsrc will emit the "enough-data" signal. - + The maximum amount of bytes that can be queued internally. After the maximum amount of bytes are queued, appsrc will emit the "enough-data" signal. - + - + The maximum amount of time that can be queued internally. After the maximum amount of time are queued, appsrc will emit the "enough-data" signal. - + The minimum latency of the source. A value of -1 will use the default latency calculations of #GstBaseSrc. - + Make appsrc emit the "need-data" signal when the amount of bytes in the queue drops below this percentage of max-bytes. - + The total size in bytes of the data stream. If the total size is known, it is recommended to configure it with this property. - + The type of stream that this source is producing. For seekable streams the application should connect to the seek-data signal. @@ -2259,6 +2285,9 @@ This callback is only called for seekable stream types. gst_app_src_set_callbacks(). + Called when the appsrc needs more data. A buffer or EOS should be + pushed to appsrc from this thread or another thread. @length is just a hint + and when it is set to -1, any number of bytes can be pushed into @appsrc. @@ -2278,6 +2307,9 @@ gst_app_src_set_callbacks(). + Called when appsrc has enough data. It is recommended that the + application stops calling push-buffer until the need_data callback is + emitted again to avoid excessive buffer queueing. @@ -2294,6 +2326,9 @@ gst_app_src_set_callbacks(). + Called when a seek should be performed to the offset. + The next push-buffer should produce buffers from the new @offset. + This callback is only called for seekable stream types. @@ -2454,20 +2489,20 @@ extracted - + 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 index 0fbffd648c..8f262cbfe7 100644 --- a/girs/GstAudio-1.0.gir +++ b/girs/GstAudio-1.0.gir @@ -1247,6 +1247,10 @@ additional information in the info #GstStructure of the returned sample: - "size" G_TYPE_UINT size of the input buffer in samples + Aggregates one input buffer to the output + buffer. The in_offset and out_offset are in "frames", which is + the size of a sample times the number of channels. Returns TRUE if + any non-silence was added to the buffer @@ -1276,6 +1280,7 @@ additional information in the info #GstStructure of the returned sample: + Create a new output buffer contains num_frames frames. @@ -1306,20 +1311,20 @@ additional information in the info #GstStructure of the returned sample: - + - + - + Causes the element to aggregate on a timeout even when no live source is connected to its sinks. See #GstAggregator:min-upstream-latency for a companion property: in the vast majority of cases where you plan to plug in live sources with a non-zero latency, you should set it to a non-zero value. - + Don't wait for inactive pads when live. An inactive pad is a pad that hasn't yet received a buffer, but that has been waited on at least once. @@ -1330,10 +1335,10 @@ data flow, for example the user may decide to connect it later, but wants to configure it already. - + - + Output block size in nanoseconds, expressed as a fraction. @@ -1359,6 +1364,7 @@ but wants to configure it already. + Create a new output buffer contains num_frames frames. @@ -1375,6 +1381,10 @@ but wants to configure it already. + Aggregates one input buffer to the output + buffer. The in_offset and out_offset are in "frames", which is + the size of a sample times the number of channels. Returns TRUE if + any non-silence was added to the buffer @@ -1442,13 +1452,14 @@ See #GstAudioAggregator for more details. - + The default implementation of GstPad used with #GstAudioAggregator + Convert a buffer from one format to another. @@ -1469,6 +1480,8 @@ See #GstAudioAggregator for more details. + Called when either the input or output + formats have changed. @@ -1479,7 +1492,7 @@ See #GstAudioAggregator for more details. - + Emit QoS messages when dropping buffers. @@ -1505,6 +1518,7 @@ See #GstAudioAggregator for more details. + Convert a buffer from one format to another. @@ -1527,6 +1541,8 @@ See #GstAudioAggregator for more details. + Called when either the input or output + formats have changed. @@ -1545,10 +1561,10 @@ See #GstAudioAggregator for more details. - + - + @@ -1573,6 +1589,10 @@ the returned buffer (see gst_object_set_parent()). + payload data in a format suitable to write to the sink. If no + payloading is required, returns a reffed copy of the original + buffer, else returns the payloaded buffer with all other metadata + copied. @@ -1602,7 +1622,7 @@ the returned buffer (see gst_object_set_parent()). - + Get the current alignment threshold, in nanoseconds, used by @sink. @@ -1616,7 +1636,7 @@ the returned buffer (see gst_object_set_parent()). - + Get the current discont wait, in nanoseconds, used by @sink. @@ -1630,7 +1650,7 @@ the returned buffer (see gst_object_set_parent()). - + Get the current drift tolerance, in microseconds, used by @sink. @@ -1644,7 +1664,7 @@ the returned buffer (see gst_object_set_parent()). - + Queries whether @sink will provide a clock or not. See also gst_audio_base_sink_set_provide_clock. @@ -1659,7 +1679,7 @@ gst_audio_base_sink_set_provide_clock. - + Get the current slave method used by @sink. @@ -1690,7 +1710,7 @@ for the custom slave method. - + Controls the sink's alignment threshold. @@ -1739,7 +1759,7 @@ method were used. - + Controls how long the sink will wait before creating a discontinuity. @@ -1756,7 +1776,7 @@ method were used. - + Controls the sink's drift tolerance. @@ -1773,7 +1793,7 @@ method were used. - + 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 @@ -1793,7 +1813,7 @@ NULL. - + Controls how clock slaving will be performed in @sink. @@ -1810,32 +1830,32 @@ NULL. - + - + - + - + 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. - + - + - + @@ -1877,6 +1897,7 @@ functionality. + create and return a #GstAudioRingBuffer to write to. @@ -1892,6 +1913,10 @@ functionality. + payload data in a format suitable to write to the sink. If no + payloading is required, returns a reffed copy of the original + buffer, else returns the payloaded buffer with all other metadata + copied. @@ -1971,42 +1996,42 @@ discontinuity happens. 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) @@ -2047,7 +2072,7 @@ returned buffer (see gst_object_set_parent()). - + Queries whether @src will provide a clock or not. See also gst_audio_base_src_set_provide_clock. @@ -2062,7 +2087,7 @@ gst_audio_base_src_set_provide_clock. - + Get the current slave method used by @src. @@ -2076,7 +2101,7 @@ gst_audio_base_src_set_provide_clock. - + 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. @@ -2095,7 +2120,7 @@ of @src. If @provide is %FALSE, gst_element_provide_clock() will return NULL. - + Controls how clock slaving will be performed in @src. @@ -2112,24 +2137,24 @@ of @src. If @provide is %FALSE, gst_element_provide_clock() will return NULL. - + Actual configured size of audio buffer in microseconds. - + Actual configured audio latency in microseconds. - + - + - + - + @@ -2168,6 +2193,7 @@ functionality. + create and return a #GstAudioRingBuffer to read from. @@ -2188,24 +2214,24 @@ functionality. - + 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. @@ -2481,6 +2507,7 @@ The track entries in the TOC will be sorted by track number. + closing the device @@ -2492,6 +2519,7 @@ The track entries in the TOC will be sorted by track number. + opening the device @@ -2506,6 +2534,7 @@ The track entries in the TOC will be sorted by track number. + reading a sector @@ -2540,13 +2569,13 @@ copy of the structure (and take ownership of the taglist if there is one). - + - + - + @@ -2577,6 +2606,7 @@ copy of the structure (and take ownership of the taglist if there is one). + opening the device @@ -2593,6 +2623,7 @@ copy of the structure (and take ownership of the taglist if there is one). + closing the device @@ -2606,6 +2637,7 @@ copy of the structure (and take ownership of the taglist if there is one). + reading a sector @@ -2630,14 +2662,14 @@ copy of the structure (and take ownership of the taglist if there is one). 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 - + @@ -2680,7 +2712,7 @@ on the pipeline's #GstBus instead. - + Free memory allocated by @mix. @@ -2820,19 +2852,19 @@ Perform channel mixing on @in_data and write the result to @out_data. 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 @@ -2861,100 +2893,100 @@ 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 implicitly 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) @@ -3155,7 +3187,7 @@ be used. - + This object is used to convert audio samples from one format to another. The object can perform conversion of: @@ -3461,14 +3493,14 @@ option and values. 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(). @@ -3565,6 +3597,9 @@ Things that subclass need to take care of: data for indicated duration. + Optional. + Called when the element changes to GST_STATE_NULL. + Allows closing external resources. @@ -3576,6 +3611,12 @@ Things that subclass need to take care of: + Optional. + Setup the allocation parameters for allocating output + buffers. The passed in query contains the result of the + downstream allocation query. + Subclasses should chain up to the parent implementation to + invoke the default handler. @@ -3590,6 +3631,11 @@ Things that subclass need to take care of: + Optional. + Instructs subclass to clear any codec caches and discard + any pending samples and not yet returned decoded data. + @hard indicates whether a FLUSH is being processed, + or otherwise a DISCONT (or conceptually similar). @@ -3604,6 +3650,11 @@ Things that subclass need to take care of: + Optional. + Allows for a custom sink getcaps implementation. + If not implemented, + default returns gst_audio_decoder_proxy_getcaps + applied to sink template caps. @@ -3618,6 +3669,11 @@ Things that subclass need to take care of: + Provides input data (or NULL to clear any remaining data) + to subclass. Input data ref management is performed by + base class, subclass should not care or intervene, + and input data is only valid until next call to base class, + most notably a call to gst_audio_decoder_finish_frame(). @@ -3648,6 +3704,9 @@ negotiate fails. + Optional. + Called when the element changes to GST_STATE_READY. + Allows opening external resources. @@ -3679,6 +3738,10 @@ negotiate fails. + Optional. + Called just prior to pushing (encoded data) buffer downstream. + Subclass has full discretionary access to buffer, + and a not OK flow return will abort downstream pushing. @@ -3693,6 +3756,10 @@ negotiate fails. + Optional. + Propose buffer allocation parameters for upstream elements. + Subclasses should chain up to the parent implementation to + invoke the default handler. @@ -3707,6 +3774,7 @@ negotiate fails. + Notifies subclass of incoming data format (caps). @@ -3721,6 +3789,9 @@ negotiate fails. + Optional. + Event handler on the sink pad. Subclasses should chain up to + the parent implementation to invoke the default handler. @@ -3735,6 +3806,11 @@ negotiate fails. + Optional. + Query handler on the sink pad. This function should + return TRUE if the query could be performed. Subclasses + should chain up to the parent implementation to invoke the + default handler. Since: 1.6 @@ -3749,6 +3825,9 @@ negotiate fails. + Optional. + Event handler on the src pad. Subclasses should chain up to + the parent implementation to invoke the default handler. @@ -3763,6 +3842,11 @@ negotiate fails. + Optional. + Query handler on the source pad. This function should + return TRUE if the query could be performed. Subclasses + should chain up to the parent implementation to invoke the + default handler. Since: 1.6 @@ -3777,6 +3861,9 @@ negotiate fails. + Optional. + Called when the element starts processing. + Allows opening external resources. @@ -3788,6 +3875,9 @@ negotiate fails. + Optional. + Called when the element stops processing. + Allows closing external resources. @@ -3799,6 +3889,11 @@ negotiate fails. + Optional. Transform the metadata on the input buffer to the + output buffer. By default this method copies all meta without + tags and meta with only the "audio" tag. subclasses can + implement this method and return %TRUE if the metadata is to be + copied. Since: 1.6 @@ -4001,7 +4096,7 @@ latency. - + currently configured decoder tolerated error count. @@ -4014,7 +4109,7 @@ latency. - + Queries decoder's latency aggregation. @@ -4067,7 +4162,7 @@ MT safe. - + Queries decoder packet loss concealment handling. @@ -4096,7 +4191,7 @@ MT safe. - + Queries current audio jitter tolerance threshold. @@ -4260,7 +4355,7 @@ so the pipeline can reconfigure its global 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 @@ -4280,7 +4375,7 @@ GST_AUDIO_DECODER_MAX_ERRORS. - + Sets decoder minimum aggregation latency. MT safe. @@ -4362,7 +4457,7 @@ caps features. - + Enable or disable decoder packet loss concealment, provided subclass and codec are capable and allow handling plc. @@ -4399,7 +4494,7 @@ MT safe. - + Configures decoder audio jitter tolerance threshold. MT safe. @@ -4440,18 +4535,18 @@ handler with %GST_PAD_SET_ACCEPT_INTERSECT and - + Maximum number of tolerated consecutive decode errors. See gst_audio_decoder_set_max_errors() for more details. - + - + - + @@ -4491,6 +4586,9 @@ overridden. + Optional. + Called when the element starts processing. + Allows opening external resources. @@ -4504,6 +4602,9 @@ overridden. + Optional. + Called when the element stops processing. + Allows closing external resources. @@ -4517,6 +4618,7 @@ overridden. + Notifies subclass of incoming data format (caps). @@ -4533,6 +4635,11 @@ overridden. + Optional. + Allows chopping incoming data into manageable units (frames) + for subsequent decoding. This division is at subclass + discretion and may or may not correspond to 1 (or more) + frames as defined by audio format. @@ -4555,6 +4662,11 @@ overridden. + Provides input data (or NULL to clear any remaining data) + to subclass. Input data ref management is performed by + base class, subclass should not care or intervene, + and input data is only valid until next call to base class, + most notably a call to gst_audio_decoder_finish_frame(). @@ -4571,6 +4683,11 @@ overridden. + Optional. + Instructs subclass to clear any codec caches and discard + any pending samples and not yet returned decoded data. + @hard indicates whether a FLUSH is being processed, + or otherwise a DISCONT (or conceptually similar). @@ -4587,6 +4704,10 @@ overridden. + Optional. + Called just prior to pushing (encoded data) buffer downstream. + Subclass has full discretionary access to buffer, + and a not OK flow return will abort downstream pushing. @@ -4603,6 +4724,9 @@ overridden. + Optional. + Event handler on the sink pad. Subclasses should chain up to + the parent implementation to invoke the default handler. @@ -4619,6 +4743,9 @@ overridden. + Optional. + Event handler on the src pad. Subclasses should chain up to + the parent implementation to invoke the default handler. @@ -4635,6 +4762,9 @@ overridden. + Optional. + Called when the element changes to GST_STATE_READY. + Allows opening external resources. @@ -4648,6 +4778,9 @@ overridden. + Optional. + Called when the element changes to GST_STATE_NULL. + Allows closing external resources. @@ -4661,6 +4794,10 @@ overridden. + Optional. + Negotiate with downstream and configure buffer pools, etc. + Subclasses should chain up to the parent implementation to + invoke the default handler. @@ -4676,6 +4813,12 @@ overridden. + Optional. + Setup the allocation parameters for allocating output + buffers. The passed in query contains the result of the + downstream allocation query. + Subclasses should chain up to the parent implementation to + invoke the default handler. @@ -4692,6 +4835,10 @@ overridden. + Optional. + Propose buffer allocation parameters for upstream elements. + Subclasses should chain up to the parent implementation to + invoke the default handler. @@ -4708,6 +4855,11 @@ overridden. + Optional. + Query handler on the sink pad. This function should + return TRUE if the query could be performed. Subclasses + should chain up to the parent implementation to invoke the + default handler. Since: 1.6 @@ -4724,6 +4876,11 @@ overridden. + Optional. + Query handler on the source pad. This function should + return TRUE if the query could be performed. Subclasses + should chain up to the parent implementation to invoke the + default handler. Since: 1.6 @@ -4740,6 +4897,11 @@ overridden. + Optional. + Allows for a custom sink getcaps implementation. + If not implemented, + default returns gst_audio_decoder_proxy_getcaps + applied to sink template caps. @@ -4756,6 +4918,11 @@ overridden. + Optional. Transform the metadata on the input buffer to the + output buffer. By default this method copies all meta without + tags and meta with only the "audio" tag. subclasses can + implement this method and return %TRUE if the metadata is to be + copied. Since: 1.6 @@ -4783,21 +4950,21 @@ overridden. - + Set of available dithering methods. - + No dithering - + Rectangular dithering - + Triangular dithering (default) - + High frequency triangular dithering @@ -4938,6 +5105,9 @@ Things that subclass need to take care of: + Optional. + Called when the element changes to GST_STATE_NULL. + Allows closing external resources. @@ -4949,6 +5119,12 @@ Things that subclass need to take care of: + Optional. + Setup the allocation parameters for allocating output + buffers. The passed in query contains the result of the + downstream allocation query. + Subclasses should chain up to the parent implementation to + invoke the default handler. @@ -4963,6 +5139,9 @@ Things that subclass need to take care of: + Optional. + Instructs subclass to clear any codec caches and discard + any pending samples and not yet returned encoded data. @@ -4974,6 +5153,11 @@ Things that subclass need to take care of: + Optional. + Allows for a custom sink getcaps implementation (e.g. + for multichannel input specification). If not implemented, + default returns gst_audio_encoder_proxy_getcaps + applied to sink template caps. @@ -4988,6 +5172,12 @@ Things that subclass need to take care of: + Provides input samples (or NULL to clear any remaining data) + according to directions as configured by the subclass + using the API. Input data ref management is performed + by base class, subclass should not care or intervene, + and input data is only valid until next call to base class, + most notably a call to gst_audio_encoder_finish_frame(). @@ -5018,6 +5208,9 @@ negotiate fails. + Optional. + Called when the element changes to GST_STATE_READY. + Allows opening external resources. @@ -5029,6 +5222,10 @@ negotiate fails. + Optional. + Called just prior to pushing (encoded data) buffer downstream. + Subclass has full discretionary access to buffer, + and a not OK flow return will abort downstream pushing. @@ -5043,6 +5240,10 @@ negotiate fails. + Optional. + Propose buffer allocation parameters for upstream elements. + Subclasses should chain up to the parent implementation to + invoke the default handler. @@ -5057,6 +5258,8 @@ negotiate fails. + Notifies subclass of incoming data format. + GstAudioInfo contains the format according to provided caps. @@ -5071,6 +5274,9 @@ negotiate fails. + Optional. + Event handler on the sink pad. Subclasses should chain up to + the parent implementation to invoke the default handler. @@ -5085,6 +5291,11 @@ negotiate fails. + Optional. + Query handler on the sink pad. This function should + return TRUE if the query could be performed. Subclasses + should chain up to the parent implementation to invoke the + default handler. Since: 1.6 @@ -5099,6 +5310,9 @@ negotiate fails. + Optional. + Event handler on the src pad. Subclasses should chain up to + the parent implementation to invoke the default handler. @@ -5113,6 +5327,11 @@ negotiate fails. + Optional. + Query handler on the source pad. This function should + return TRUE if the query could be performed. Subclasses + should chain up to the parent implementation to invoke the + default handler. Since: 1.6 @@ -5127,6 +5346,9 @@ negotiate fails. + Optional. + Called when the element starts processing. + Allows opening external resources. @@ -5138,6 +5360,9 @@ negotiate fails. + Optional. + Called when the element stops processing. + Allows closing external resources. @@ -5149,6 +5374,11 @@ negotiate fails. + Optional. Transform the metadata on the input buffer to the + output buffer. By default this method copies all meta without + tags and meta with only the "audio" tag. subclasses can + implement this method and return %TRUE if the metadata is to be + copied. Since: 1.6 @@ -5328,7 +5558,7 @@ MT safe. - + @@ -5374,7 +5604,7 @@ latency. - + Queries if the encoder will handle granule marking. @@ -5390,7 +5620,7 @@ MT safe. - + Queries encoder perfect timestamp behaviour. @@ -5406,7 +5636,7 @@ MT safe. - + Queries current audio jitter tolerance threshold. @@ -5624,7 +5854,7 @@ MT safe. - + @@ -5738,7 +5968,7 @@ MT safe. - + Enable or disable encoder perfect output timestamp preference. MT safe. @@ -5757,7 +5987,7 @@ MT safe. - + Configures encoder audio jitter tolerance threshold. MT safe. @@ -5776,16 +6006,16 @@ MT safe. - + - + - + - + @@ -5824,6 +6054,9 @@ needed. At minimum @set_format and @handle_frame needs to be overridden. + Optional. + Called when the element starts processing. + Allows opening external resources. @@ -5837,6 +6070,9 @@ needed. At minimum @set_format and @handle_frame needs to be overridden. + Optional. + Called when the element stops processing. + Allows closing external resources. @@ -5850,6 +6086,8 @@ needed. At minimum @set_format and @handle_frame needs to be overridden. + Notifies subclass of incoming data format. + GstAudioInfo contains the format according to provided caps. @@ -5866,6 +6104,12 @@ needed. At minimum @set_format and @handle_frame needs to be overridden. + Provides input samples (or NULL to clear any remaining data) + according to directions as configured by the subclass + using the API. Input data ref management is performed + by base class, subclass should not care or intervene, + and input data is only valid until next call to base class, + most notably a call to gst_audio_encoder_finish_frame(). @@ -5882,6 +6126,9 @@ needed. At minimum @set_format and @handle_frame needs to be overridden. + Optional. + Instructs subclass to clear any codec caches and discard + any pending samples and not yet returned encoded data. @@ -5895,6 +6142,10 @@ needed. At minimum @set_format and @handle_frame needs to be overridden. + Optional. + Called just prior to pushing (encoded data) buffer downstream. + Subclass has full discretionary access to buffer, + and a not OK flow return will abort downstream pushing. @@ -5911,6 +6162,9 @@ needed. At minimum @set_format and @handle_frame needs to be overridden. + Optional. + Event handler on the sink pad. Subclasses should chain up to + the parent implementation to invoke the default handler. @@ -5927,6 +6181,9 @@ needed. At minimum @set_format and @handle_frame needs to be overridden. + Optional. + Event handler on the src pad. Subclasses should chain up to + the parent implementation to invoke the default handler. @@ -5943,6 +6200,11 @@ needed. At minimum @set_format and @handle_frame needs to be overridden. + Optional. + Allows for a custom sink getcaps implementation (e.g. + for multichannel input specification). If not implemented, + default returns gst_audio_encoder_proxy_getcaps + applied to sink template caps. @@ -5959,6 +6221,9 @@ needed. At minimum @set_format and @handle_frame needs to be overridden. + Optional. + Called when the element changes to GST_STATE_READY. + Allows opening external resources. @@ -5972,6 +6237,9 @@ needed. At minimum @set_format and @handle_frame needs to be overridden. + Optional. + Called when the element changes to GST_STATE_NULL. + Allows closing external resources. @@ -5985,6 +6253,10 @@ needed. At minimum @set_format and @handle_frame needs to be overridden. + Optional. + Negotiate with downstream and configure buffer pools, etc. + Subclasses should chain up to the parent implementation to + invoke the default handler. @@ -6000,6 +6272,12 @@ needed. At minimum @set_format and @handle_frame needs to be overridden. + Optional. + Setup the allocation parameters for allocating output + buffers. The passed in query contains the result of the + downstream allocation query. + Subclasses should chain up to the parent implementation to + invoke the default handler. @@ -6016,6 +6294,10 @@ needed. At minimum @set_format and @handle_frame needs to be overridden. + Optional. + Propose buffer allocation parameters for upstream elements. + Subclasses should chain up to the parent implementation to + invoke the default handler. @@ -6032,6 +6314,11 @@ needed. At minimum @set_format and @handle_frame needs to be overridden. + Optional. Transform the metadata on the input buffer to the + output buffer. By default this method copies all meta without + tags and meta with only the "audio" tag. subclasses can + implement this method and return %TRUE if the metadata is to be + copied. Since: 1.6 @@ -6054,6 +6341,11 @@ needed. At minimum @set_format and @handle_frame needs to be overridden. + Optional. + Query handler on the sink pad. This function should + return TRUE if the query could be performed. Subclasses + should chain up to the parent implementation to invoke the + default handler. Since: 1.6 @@ -6070,6 +6362,11 @@ needed. At minimum @set_format and @handle_frame needs to be overridden. + Optional. + Query handler on the source pad. This function should + return TRUE if the query could be performed. Subclasses + should chain up to the parent implementation to invoke the + default handler. Since: 1.6 @@ -6091,7 +6388,7 @@ needed. At minimum @set_format and @handle_frame needs to be overridden. - + @@ -6110,6 +6407,7 @@ Derived classes should override the #GstAudioFilterClass.setup() and virtual functions in their class_init function. + virtual function called whenever the format changes @@ -6145,6 +6443,7 @@ function. + virtual function called whenever the format changes @@ -6188,152 +6487,152 @@ This function is usually used from within a GObject class_init function. 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 @@ -6437,19 +6736,19 @@ versions were printing a critical warning and returned %NULL. 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 @@ -6831,10 +7130,10 @@ Note: This initializes @info first, no values are preserved. Layout of the audio samples for the different channels. - + interleaved audio - + non-interleaved audio @@ -6904,28 +7203,28 @@ meta as well as extracting it. 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 slightly faster but less accurate. When this flag @@ -6933,7 +7232,7 @@ meta as well as extracting it. in the least significant bits of the destination. - + Free a #GstAudioQuantize. @@ -7039,14 +7338,14 @@ the @dither and @ns parameters. 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. @@ -7278,50 +7577,50 @@ for @quality in @options. The different filter interpolation methods. - + no interpolation - + linear interpolation of the filter coefficients. - + cubic interpolation of the filter coefficients. 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. @@ -7329,21 +7628,21 @@ for @quality in @options. 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 @@ -7608,6 +7907,7 @@ MT safe. + resume processing of samples after pause @@ -8459,6 +8759,7 @@ called to fill the memory at @data with @len bytes of samples. + open the device, don't set any params or allocate anything @@ -8476,6 +8777,7 @@ MT safe. + allocate the resources for the ringbuffer using the given spec @@ -8497,6 +8799,7 @@ MT safe. + free resources of the ringbuffer @@ -8514,6 +8817,7 @@ MT safe. + close the device @@ -8531,6 +8835,7 @@ MT safe. + start processing of samples @@ -8548,6 +8853,7 @@ MT safe. + pause processing of samples @@ -8565,6 +8871,7 @@ MT safe. + resume processing of samples after pause @@ -8578,6 +8885,7 @@ MT safe. + stop processing of samples @@ -8595,6 +8903,7 @@ MT safe. + get number of frames queued in device @@ -8612,6 +8921,8 @@ MT safe. + activate the thread that starts pulling and monitoring the +consumed segments in the device. @@ -8632,6 +8943,7 @@ FALSE on error. + write samples into the ringbuffer @@ -8671,6 +8983,10 @@ with a flush or stop. + Optional. + Clear the entire ringbuffer. + Subclasses should chain up to the parent implementation to + invoke the default handler. @@ -8692,56 +9008,56 @@ with a flush or stop. 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) - + samples in DSD format (Since: 1.24) - + @@ -8801,16 +9117,16 @@ only the rate, channels, position, and bpf fields in @info are populated. 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) @@ -8840,6 +9156,7 @@ together with #GstAudioBaseSink using a default implementation of a #GstAudioRingBuffer that uses threads. + Close the device. @@ -8851,6 +9168,8 @@ together with #GstAudioBaseSink using a default implementation of a + Return how many frames are still in the device. Participates in + computing the time for audio clocks and drives the synchronisation. @@ -8862,6 +9181,8 @@ together with #GstAudioBaseSink using a default implementation of a + Open the device. No configuration needs to be done at this point. + This function is also used to check if the device is available. @@ -8873,6 +9194,9 @@ together with #GstAudioBaseSink using a default implementation of a + Pause the device and unblock write as fast as possible. + For retro compatibility, the audio sink will fallback + to calling reset if this vmethod is not provided. Since: 1.18 @@ -8884,6 +9208,7 @@ together with #GstAudioBaseSink using a default implementation of a + Prepare the device to operate with the specified parameters. @@ -8898,6 +9223,9 @@ together with #GstAudioBaseSink using a default implementation of a + Returns as quickly as possible from a write and flush any pending + samples from the device. + This vmethod is deprecated. Please provide pause and stop instead. @@ -8909,6 +9237,7 @@ together with #GstAudioBaseSink using a default implementation of a + Resume the device. Since: 1.18 @@ -8920,6 +9249,10 @@ together with #GstAudioBaseSink using a default implementation of a + Stop the device and unblock write as fast as possible. + Pending samples are flushed from the device. + For retro compatibility, the audio sink will fallback + to calling reset if this vmethod is not provided. Since: 1.18 @@ -8931,6 +9264,7 @@ together with #GstAudioBaseSink using a default implementation of a + Undo operations done in prepare. @@ -8981,6 +9315,8 @@ together with #GstAudioBaseSink using a default implementation of a + Open the device. No configuration needs to be done at this point. + This function is also used to check if the device is available. @@ -8994,6 +9330,7 @@ together with #GstAudioBaseSink using a default implementation of a + Prepare the device to operate with the specified parameters. @@ -9010,6 +9347,7 @@ together with #GstAudioBaseSink using a default implementation of a + Undo operations done in prepare. @@ -9023,6 +9361,7 @@ together with #GstAudioBaseSink using a default implementation of a + Close the device. @@ -9036,6 +9375,10 @@ together with #GstAudioBaseSink using a default implementation of a + Write data to the device. + This vmethod is allowed to block until all the data is written. + If such is the case then it is expected that pause, stop and + reset will unblock the write when called. @@ -9058,6 +9401,8 @@ together with #GstAudioBaseSink using a default implementation of a + Return how many frames are still in the device. Participates in + computing the time for audio clocks and drives the synchronisation. @@ -9071,6 +9416,9 @@ together with #GstAudioBaseSink using a default implementation of a + Returns as quickly as possible from a write and flush any pending + samples from the device. + This vmethod is deprecated. Please provide pause and stop instead. @@ -9084,6 +9432,9 @@ together with #GstAudioBaseSink using a default implementation of a + Pause the device and unblock write as fast as possible. + For retro compatibility, the audio sink will fallback + to calling reset if this vmethod is not provided. Since: 1.18 @@ -9097,6 +9448,7 @@ together with #GstAudioBaseSink using a default implementation of a + Resume the device. Since: 1.18 @@ -9110,6 +9462,10 @@ together with #GstAudioBaseSink using a default implementation of a + Stop the device and unblock write as fast as possible. + Pending samples are flushed from the device. + For retro compatibility, the audio sink will fallback + to calling reset if this vmethod is not provided. Since: 1.18 @@ -9160,6 +9516,7 @@ together with #GstAudioBaseSrc using a default implementation of a #GstAudioRingBuffer that uses threads. + close the device @@ -9171,6 +9528,7 @@ together with #GstAudioBaseSrc using a default implementation of a + the number of frames queued in the device @@ -9182,6 +9540,7 @@ together with #GstAudioBaseSrc using a default implementation of a + open the device with the specified caps @@ -9193,6 +9552,7 @@ together with #GstAudioBaseSrc using a default implementation of a + configure device with format @@ -9232,6 +9592,7 @@ together with #GstAudioBaseSrc using a default implementation of a + unblock a read to the device and reset. @@ -9243,6 +9604,7 @@ together with #GstAudioBaseSrc using a default implementation of a + undo the configuration @@ -9274,6 +9636,7 @@ functionality. + open the device with the specified caps @@ -9287,6 +9650,7 @@ functionality. + configure device with format @@ -9303,6 +9667,7 @@ functionality. + undo the configuration @@ -9316,6 +9681,7 @@ functionality. + close the device @@ -9329,6 +9695,7 @@ functionality. + read samples from the audio device @@ -9355,6 +9722,7 @@ functionality. + the number of frames queued in the device @@ -9368,6 +9736,7 @@ functionality. + unblock a read to the device and reset. @@ -9386,7 +9755,7 @@ functionality. - + #GstAudioStreamAlign provides a helper object that helps tracking audio stream alignment and discontinuities, and detects discontinuities if possible. @@ -9778,31 +10147,31 @@ in a more readable fashion. Enum value describing how DSD bits are grouped. - + unknown / invalid DSD format - + 8 DSD bits in 1 byte - + 16 DSD bits in 2 bytes, little endian order - + 16 DSD bits in 2 bytes, big endian order - + 32 DSD bits in 4 bytes, little endian order - + 32 DSD bits in 4 bytes, big endian order - + number of valid DSD formats - + 16 DSD bits in 2 bytes, native endianness - + 32 DSD bits in 4 bytes, native endianness @@ -10400,7 +10769,7 @@ The volume property is defined to be a linear volume factor. - + Returns %TRUE if the stream is muted @@ -10413,7 +10782,7 @@ The volume property is defined to be a linear volume factor. - + The current stream volume as linear factor @@ -10430,7 +10799,7 @@ The volume property is defined to be a linear volume factor. - + @@ -10446,7 +10815,7 @@ The volume property is defined to be a linear volume factor. - + @@ -10466,10 +10835,10 @@ The volume property is defined to be a linear volume factor. - + - + diff --git a/girs/GstBadAudio-1.0.gir b/girs/GstBadAudio-1.0.gir index 9ba916248b..9d9b5490ab 100644 --- a/girs/GstBadAudio-1.0.gir +++ b/girs/GstBadAudio-1.0.gir @@ -251,6 +251,12 @@ is defined this way (this is all done by the base class automatically): subsong duration regardless of the output mode. + Optional. + Sets up the allocation parameters for allocating output + buffers. The passed in query contains the result of the + downstream allocation query. + Subclasses should chain up to the parent implementation to + invoke the default handler. @@ -265,6 +271,11 @@ is defined this way (this is all done by the base class automatically): + Always required. + Allocates an output buffer, fills it with decoded audio samples, and must be passed on to + *buffer . The number of decoded samples must be passed on to *num_samples. + If decoding finishes or the decoding is no longer possible (for example, due to an + unrecoverable error), this function returns FALSE, otherwise TRUE. @@ -282,6 +293,12 @@ is defined this way (this is all done by the base class automatically): + Optional. + Returns the current subsong. + If the current subsong mode is not GST_NONSTREAM_AUDIO_SUBSONG_MODE_SINGLE, this + function's return value is undefined. + If this function is implemented by the subclass, + @get_num_subsongs should be implemented as well. @@ -293,6 +310,13 @@ is defined this way (this is all done by the base class automatically): + Optional. + Returns a tag list containing the main song tags, or NULL if there are + no such tags. Returned tags will be unref'd. Use this vfunc instead of + manually pushing a tag event downstream to avoid edge cases where not yet + pushed sticky tag events get overwritten before they are pushed (can for + example happen with decodebin if tags are pushed downstream before the + decodebin pads are linked). @@ -304,6 +328,8 @@ is defined this way (this is all done by the base class automatically): + Optional. + Returns the number of loops for playback. @@ -315,6 +341,16 @@ is defined this way (this is all done by the base class automatically): + Optional. + Returns the number of subsongs available. + The return values 0 and 1 have a similar, but distinct, meaning. + If this function returns 0, then this decoder does not support subsongs at all. + @get_current_subsong must then also always return 0. In other words, this function + either never returns 0, or never returns anything else than 0. + A return value of 1 means that the media contains either only one or no subsongs + (the entire song is then considered to be one single subsong). 1 also means that only + this very media has no or just one subsong, and the decoder itself can + support multiple subsongs. @@ -326,6 +362,8 @@ is defined this way (this is all done by the base class automatically): + Optional. + Returns the duration of a subsong. Returns GST_CLOCK_TIME_NONE if duration is unknown. @@ -340,6 +378,9 @@ is defined this way (this is all done by the base class automatically): + Optional. + Returns tags for a subsong, or NULL if there are no tags. + Returned tags will be unref'd. @@ -354,6 +395,10 @@ is defined this way (this is all done by the base class automatically): + Always required. + Returns a bitmask containing the output modes the subclass supports. + The mask is formed by a bitwise OR combination of integers, which can be calculated + this way: 1 << GST_NONSTREAM_AUDIO_OUTPUT_MODE_<mode> , where mode is either STEADY or LOOPING @@ -365,6 +410,14 @@ is defined this way (this is all done by the base class automatically): + Required if loads_from_sinkpad is set to TRUE (the default value). + Loads the media from the given buffer. The entire media is supplied at once, + so after this call, loading should be finished. This function + can also make use of a suggested initial subsong & subsong mode and initial + playback position (but isn't required to). In case it chooses a different starting + position, the function must pass this position to *initial_position. + The subclass does not have to unref the input buffer; the base class does that + already. @@ -394,6 +447,10 @@ is defined this way (this is all done by the base class automatically): + Required if loads_from_sinkpad is set to FALSE. + Loads the media in a way defined by the custom sink. Data is not supplied; + the derived class has to handle this on its own. Otherwise, this function is + identical to @load_from_buffer. @@ -431,6 +488,10 @@ is defined this way (this is all done by the base class automatically): + Optional. + Proposes buffer allocation parameters for upstream elements. + Subclasses should chain up to the parent implementation to + invoke the default handler. @@ -445,6 +506,14 @@ is defined this way (this is all done by the base class automatically): + Optional. + Called when a seek event is received by the parent class. + new_position is a pointer to a GstClockTime integer which + contains a position relative to the current subsong. + Minimum is 0, maximum is the subsong length. + After this function finishes, new_position is set to the + actual new position (which may differ from the request + position, depending on the decoder). @@ -459,6 +528,15 @@ is defined this way (this is all done by the base class automatically): + Optional. + Sets the current subsong. This function is allowed to switch to a different + subsong than the required one, and can optionally make use of the suggested initial + position. In case it chooses a different starting position, the function must pass + this position to *initial_position. + This function switches the subsong mode to GST_NONSTREAM_AUDIO_SUBSONG_MODE_SINGLE + automatically. + If this function is implemented by the subclass, @get_current_subsong and + @get_num_subsongs should be implemented as well. @@ -476,6 +554,19 @@ is defined this way (this is all done by the base class automatically): + Optional. + Sets the number of loops for playback. If this is called during playback, + the subclass must set any internal loop counters to zero. A loop value of -1 + means infinite looping; 0 means no looping; and when the num_loops is greater than 0, + playback should loop exactly num_loops times. If this function is implemented, + @get_num_loops should be implemented as well. The function can ignore the given values + and choose another; however, @get_num_loops should return this other value afterwards. + It is up to the subclass to define where the loop starts and ends. It can mean that only + a subset at the end or in the middle of a song is repeated, for example. + If the current subsong mode is GST_NONSTREAM_AUDIO_SUBSONG_MODE_SINGLE, then the subsong + is repeated this many times. If it is GST_NONSTREAM_AUDIO_SUBSONG_MODE_ALL, then all + subsongs are repeated this many times. With GST_NONSTREAM_AUDIO_SUBSONG_MODE_DECODER_DEFAULT, + the behavior is decoder specific. @@ -490,6 +581,11 @@ is defined this way (this is all done by the base class automatically): + Optional. + Sets the output mode the subclass has to use. Unlike with most other functions, the subclass + cannot choose a different mode; it must use the requested one. + If the output mode is set to LOOPING, @gst_nonstream_audio_decoder_handle_loop + must be called after playback moved back to the start of a loop. @@ -507,6 +603,13 @@ is defined this way (this is all done by the base class automatically): + Optional. + Sets the current subsong mode. Since this might influence the current playback position, + this function must set the initial_position integer argument to a defined value. + If the playback position is not affected at all, it must be set to GST_CLOCK_TIME_NONE. + If the subsong is restarted after the mode switch, it is recommended to set the value + to the position in the playback right after the switch (or 0 if the subsongs are always + reset back to the beginning). @@ -524,6 +627,11 @@ is defined this way (this is all done by the base class automatically): + Optional. + Called when a position query is received by the parent class. + The position that this function returns must be relative to + the current subsong. Thus, the minimum is 0, and the maximum + is the subsong length. @@ -697,16 +805,16 @@ gst_nonstream_audio_decoder_set_output_format(). - + - + - + - + @@ -804,6 +912,14 @@ loads_from_sinkpad is TRUE. + Optional. + Called when a seek event is received by the parent class. + new_position is a pointer to a GstClockTime integer which + contains a position relative to the current subsong. + Minimum is 0, maximum is the subsong length. + After this function finishes, new_position is set to the + actual new position (which may differ from the request + position, depending on the decoder). @@ -820,6 +936,11 @@ loads_from_sinkpad is TRUE. + Optional. + Called when a position query is received by the parent class. + The position that this function returns must be relative to + the current subsong. Thus, the minimum is 0, and the maximum + is the subsong length. @@ -833,6 +954,14 @@ loads_from_sinkpad is TRUE. + Required if loads_from_sinkpad is set to TRUE (the default value). + Loads the media from the given buffer. The entire media is supplied at once, + so after this call, loading should be finished. This function + can also make use of a suggested initial subsong & subsong mode and initial + playback position (but isn't required to). In case it chooses a different starting + position, the function must pass this position to *initial_position. + The subclass does not have to unref the input buffer; the base class does that + already. @@ -864,6 +993,10 @@ loads_from_sinkpad is TRUE. + Required if loads_from_sinkpad is set to FALSE. + Loads the media in a way defined by the custom sink. Data is not supplied; + the derived class has to handle this on its own. Otherwise, this function is + identical to @load_from_buffer. @@ -892,6 +1025,13 @@ loads_from_sinkpad is TRUE. + Optional. + Returns a tag list containing the main song tags, or NULL if there are + no such tags. Returned tags will be unref'd. Use this vfunc instead of + manually pushing a tag event downstream to avoid edge cases where not yet + pushed sticky tag events get overwritten before they are pushed (can for + example happen with decodebin if tags are pushed downstream before the + decodebin pads are linked). @@ -905,6 +1045,15 @@ loads_from_sinkpad is TRUE. + Optional. + Sets the current subsong. This function is allowed to switch to a different + subsong than the required one, and can optionally make use of the suggested initial + position. In case it chooses a different starting position, the function must pass + this position to *initial_position. + This function switches the subsong mode to GST_NONSTREAM_AUDIO_SUBSONG_MODE_SINGLE + automatically. + If this function is implemented by the subclass, @get_current_subsong and + @get_num_subsongs should be implemented as well. @@ -924,6 +1073,12 @@ loads_from_sinkpad is TRUE. + Optional. + Returns the current subsong. + If the current subsong mode is not GST_NONSTREAM_AUDIO_SUBSONG_MODE_SINGLE, this + function's return value is undefined. + If this function is implemented by the subclass, + @get_num_subsongs should be implemented as well. @@ -937,6 +1092,16 @@ loads_from_sinkpad is TRUE. + Optional. + Returns the number of subsongs available. + The return values 0 and 1 have a similar, but distinct, meaning. + If this function returns 0, then this decoder does not support subsongs at all. + @get_current_subsong must then also always return 0. In other words, this function + either never returns 0, or never returns anything else than 0. + A return value of 1 means that the media contains either only one or no subsongs + (the entire song is then considered to be one single subsong). 1 also means that only + this very media has no or just one subsong, and the decoder itself can + support multiple subsongs. @@ -950,6 +1115,8 @@ loads_from_sinkpad is TRUE. + Optional. + Returns the duration of a subsong. Returns GST_CLOCK_TIME_NONE if duration is unknown. @@ -966,6 +1133,9 @@ loads_from_sinkpad is TRUE. + Optional. + Returns tags for a subsong, or NULL if there are no tags. + Returned tags will be unref'd. @@ -982,6 +1152,13 @@ loads_from_sinkpad is TRUE. + Optional. + Sets the current subsong mode. Since this might influence the current playback position, + this function must set the initial_position integer argument to a defined value. + If the playback position is not affected at all, it must be set to GST_CLOCK_TIME_NONE. + If the subsong is restarted after the mode switch, it is recommended to set the value + to the position in the playback right after the switch (or 0 if the subsongs are always + reset back to the beginning). @@ -1001,6 +1178,19 @@ loads_from_sinkpad is TRUE. + Optional. + Sets the number of loops for playback. If this is called during playback, + the subclass must set any internal loop counters to zero. A loop value of -1 + means infinite looping; 0 means no looping; and when the num_loops is greater than 0, + playback should loop exactly num_loops times. If this function is implemented, + @get_num_loops should be implemented as well. The function can ignore the given values + and choose another; however, @get_num_loops should return this other value afterwards. + It is up to the subclass to define where the loop starts and ends. It can mean that only + a subset at the end or in the middle of a song is repeated, for example. + If the current subsong mode is GST_NONSTREAM_AUDIO_SUBSONG_MODE_SINGLE, then the subsong + is repeated this many times. If it is GST_NONSTREAM_AUDIO_SUBSONG_MODE_ALL, then all + subsongs are repeated this many times. With GST_NONSTREAM_AUDIO_SUBSONG_MODE_DECODER_DEFAULT, + the behavior is decoder specific. @@ -1017,6 +1207,8 @@ loads_from_sinkpad is TRUE. + Optional. + Returns the number of loops for playback. @@ -1030,6 +1222,10 @@ loads_from_sinkpad is TRUE. + Always required. + Returns a bitmask containing the output modes the subclass supports. + The mask is formed by a bitwise OR combination of integers, which can be calculated + this way: 1 << GST_NONSTREAM_AUDIO_OUTPUT_MODE_<mode> , where mode is either STEADY or LOOPING @@ -1043,6 +1239,11 @@ loads_from_sinkpad is TRUE. + Optional. + Sets the output mode the subclass has to use. Unlike with most other functions, the subclass + cannot choose a different mode; it must use the requested one. + If the output mode is set to LOOPING, @gst_nonstream_audio_decoder_handle_loop + must be called after playback moved back to the start of a loop. @@ -1062,6 +1263,11 @@ loads_from_sinkpad is TRUE. + Always required. + Allocates an output buffer, fills it with decoded audio samples, and must be passed on to + *buffer . The number of decoded samples must be passed on to *num_samples. + If decoding finishes or the decoding is no longer possible (for example, due to an + unrecoverable error), this function returns FALSE, otherwise TRUE. @@ -1094,6 +1300,12 @@ loads_from_sinkpad is TRUE. + Optional. + Sets up the allocation parameters for allocating output + buffers. The passed in query contains the result of the + downstream allocation query. + Subclasses should chain up to the parent implementation to + invoke the default handler. @@ -1110,6 +1322,10 @@ loads_from_sinkpad is TRUE. + Optional. + Proposes buffer allocation parameters for upstream elements. + Subclasses should chain up to the parent implementation to + invoke the default handler. @@ -1472,7 +1688,7 @@ Free-function: gst_buffer_unref - + diff --git a/girs/GstBadBaseCameraBin-1.0.gir b/girs/GstBadBaseCameraBin-1.0.gir index 9ea665eb1e..10a488af84 100644 --- a/girs/GstBadBaseCameraBin-1.0.gir +++ b/girs/GstBadBaseCameraBin-1.0.gir @@ -54,6 +54,7 @@ and/or use gtk-doc annotations. --> + construct pipeline @@ -96,6 +97,7 @@ and/or use gtk-doc annotations. --> + set the zoom @@ -110,6 +112,7 @@ and/or use gtk-doc annotations. --> + configure pipeline for the chosen settings @@ -167,7 +170,7 @@ and/or use gtk-doc annotations. --> - + Set the chosen #GstCameraBinMode capture mode. @@ -214,16 +217,16 @@ and/or use gtk-doc annotations. --> - + - + - + - + When %TRUE, preview images should be posted to the bus when captures are made @@ -234,7 +237,7 @@ captures are made - + When TRUE new capture can be prepared. If FALSE capturing is ongoing and starting a new capture immediately is not possible. @@ -244,7 +247,7 @@ function, please schedule a new thread to do it. If you're using glib's mainloop you can use g_idle_add() for example. - + @@ -308,6 +311,7 @@ mainloop you can use g_idle_add() for example. + construct pipeline @@ -321,6 +325,7 @@ mainloop you can use g_idle_add() for example. + configure pipeline for the chosen settings @@ -334,6 +339,7 @@ mainloop you can use g_idle_add() for example. + set the zoom @@ -350,6 +356,7 @@ mainloop you can use g_idle_add() for example. + set the mode @@ -416,9 +423,9 @@ mainloop you can use g_idle_add() for example. - + - + diff --git a/girs/GstBase-1.0.gir b/girs/GstBase-1.0.gir index b93a292b13..5bf37c7d33 100644 --- a/girs/GstBase-1.0.gir +++ b/girs/GstBase-1.0.gir @@ -962,7 +962,7 @@ buffer in the list before freeing the list after usage. - + @@ -1029,6 +1029,13 @@ Control is given to the subclass when all pads have data. This class used to live in gst-plugins-bad and was moved to core. + Mandatory. + Called when buffers are queued on all sinkpads. Classes + should iterate the GstElement->sinkpads and peek or steal + buffers from the #GstAggregatorPads. If the subclass returns + GST_FLOW_EOS, sending of the eos event will be taken care + of. Once / if a buffer has been constructed from the + aggregated buffers, the subclass should call _finish_buffer. @@ -1043,6 +1050,14 @@ This class used to live in gst-plugins-bad and was moved to core. + Optional. + Called when a buffer is received on a sink pad, the task of + clipping it and translating it to the current segment falls + on the subclass. The function should use the segment of data + and the negotiated media type on the pad to perform + clipping of input buffer. This function takes ownership of + buf and should output a buffer or return NULL in + if the buffer should be dropped. @@ -1060,6 +1075,10 @@ This class used to live in gst-plugins-bad and was moved to core. + Optional. + Called when a new pad needs to be created. Allows subclass that + don't have a single sink pad template to provide a pad based + on the provided information. @@ -1080,6 +1099,11 @@ This class used to live in gst-plugins-bad and was moved to core. + Optional. + Allows the subclass to influence the allocation choices. + Setup the allocation parameters for allocating output + buffers. The passed in query contains the result of the + downstream allocation query. @@ -1132,6 +1156,10 @@ sent before pushing the buffer. + Optional. + Fixate and return the src pad caps provided. The function takes + ownership of @caps and returns a fixated version of + @caps. @caps is not guaranteed to be writable. @@ -1146,6 +1174,10 @@ sent before pushing the buffer. + Optional. + Called after a successful flushing seek, once all the flush + stops have been received. Flush pad-specific data in + #GstAggregatorPad->flush. @@ -1157,6 +1189,12 @@ sent before pushing the buffer. + Optional. + Called when the element needs to know the running time of the next + rendered buffer for live pipelines. This causes deadline + based aggregation to occur. Defaults to returning + GST_CLOCK_TIME_NONE causing the element to wait for buffers + on all sink pads before aggregating. @@ -1184,6 +1222,8 @@ if #GstAggregatorClass::negotiate fails. + Optional. + Notifies subclasses what caps format has been negotiated @@ -1220,6 +1260,8 @@ control aggregating parameters for a given set of input samples. + Optional. + Allows the subclass to handle the allocation query from upstream. @@ -1240,6 +1282,9 @@ control aggregating parameters for a given set of input samples. + Optional. + Called when an event is received on a sink pad, the subclass + should always chain up. @@ -1257,6 +1302,9 @@ control aggregating parameters for a given set of input samples. + Optional. + Called when an event is received on a sink pad before queueing up + serialized events. The subclass should always chain up (Since: 1.18). @@ -1274,6 +1322,9 @@ control aggregating parameters for a given set of input samples. + Optional. + Called when a query is received on a sink pad, the subclass + should always chain up. @@ -1291,6 +1342,9 @@ control aggregating parameters for a given set of input samples. + Optional. + Called when a query is received on a sink pad before queueing up + serialized queries. The subclass should always chain up (Since: 1.18). @@ -1308,6 +1362,9 @@ control aggregating parameters for a given set of input samples. + Optional. + Called when the src pad is activated, it will start/stop its + pad task right after that call. @@ -1325,6 +1382,9 @@ control aggregating parameters for a given set of input samples. + Optional. + Called when an event is received on the src pad, the subclass + should always chain up. @@ -1339,6 +1399,9 @@ control aggregating parameters for a given set of input samples. + Optional. + Called when a query is received on the src pad, the subclass + should always chain up. @@ -1353,6 +1416,10 @@ control aggregating parameters for a given set of input samples. + Optional. + Called when the element goes from READY to PAUSED. + The subclass should get ready to process + aggregated buffers. @@ -1364,6 +1431,9 @@ control aggregating parameters for a given set of input samples. + Optional. + Called when the element goes from PAUSED to READY. + The subclass should free all resources and reset its state. @@ -1495,7 +1565,7 @@ by @trans; free it after use it - + 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. @@ -1648,7 +1718,7 @@ sure upstream has had a fair chance to start up. - + 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 if the values changed. @@ -1727,14 +1797,14 @@ if it is used at all. - + Enables the emission of signals such as #GstAggregator::samples-selected - + - + Force minimum upstream latency (in nanoseconds). When sources with a higher latency are expected to be plugged in dynamically after the aggregator has started playing, this allows overriding the minimum @@ -1742,10 +1812,10 @@ latency reported by the initial source(s). This is only taken into account when larger than the actually reported minimum latency. - + - + @@ -1809,6 +1879,10 @@ _finish_buffer from inside that function. + Optional. + Called after a successful flushing seek, once all the flush + stops have been received. Flush pad-specific data in + #GstAggregatorPad->flush. @@ -1822,6 +1896,14 @@ _finish_buffer from inside that function. + Optional. + Called when a buffer is received on a sink pad, the task of + clipping it and translating it to the current segment falls + on the subclass. The function should use the segment of data + and the negotiated media type on the pad to perform + clipping of input buffer. This function takes ownership of + buf and should output a buffer or return NULL in + if the buffer should be dropped. @@ -1841,6 +1923,13 @@ _finish_buffer from inside that function. + Optional. + Called when a subclass calls gst_aggregator_finish_buffer() + from their aggregate function to push out a buffer. + Subclasses can override this to modify or decorate buffers + before they get pushed out. This function takes ownership + of the buffer passed. Subclasses that override this method + should always chain up to the parent class virtual method. @@ -1859,6 +1948,9 @@ _finish_buffer from inside that function. + Optional. + Called when an event is received on a sink pad, the subclass + should always chain up. @@ -1878,6 +1970,9 @@ _finish_buffer from inside that function. + Optional. + Called when a query is received on a sink pad, the subclass + should always chain up. @@ -1897,6 +1992,9 @@ _finish_buffer from inside that function. + Optional. + Called when an event is received on the src pad, the subclass + should always chain up. @@ -1913,6 +2011,9 @@ _finish_buffer from inside that function. + Optional. + Called when a query is received on the src pad, the subclass + should always chain up. @@ -1929,6 +2030,9 @@ _finish_buffer from inside that function. + Optional. + Called when the src pad is activated, it will start/stop its + pad task right after that call. @@ -1948,6 +2052,13 @@ _finish_buffer from inside that function. + Mandatory. + Called when buffers are queued on all sinkpads. Classes + should iterate the GstElement->sinkpads and peek or steal + buffers from the #GstAggregatorPads. If the subclass returns + GST_FLOW_EOS, sending of the eos event will be taken care + of. Once / if a buffer has been constructed from the + aggregated buffers, the subclass should call _finish_buffer. @@ -1964,6 +2075,9 @@ _finish_buffer from inside that function. + Optional. + Called when the element goes from PAUSED to READY. + The subclass should free all resources and reset its state. @@ -1977,6 +2091,10 @@ _finish_buffer from inside that function. + Optional. + Called when the element goes from READY to PAUSED. + The subclass should get ready to process + aggregated buffers. @@ -1990,6 +2108,12 @@ _finish_buffer from inside that function. + Optional. + Called when the element needs to know the running time of the next + rendered buffer for live pipelines. This causes deadline + based aggregation to occur. Defaults to returning + GST_CLOCK_TIME_NONE causing the element to wait for buffers + on all sink pads before aggregating. @@ -2003,6 +2127,10 @@ _finish_buffer from inside that function. + Optional. + Called when a new pad needs to be created. Allows subclass that + don't have a single sink pad template to provide a pad based + on the provided information. @@ -2025,6 +2153,12 @@ _finish_buffer from inside that function. + Lets subclasses update the #GstCaps representing + the src pad caps before usage. The result should end up + in @ret. Return %GST_AGGREGATOR_FLOW_NEED_DATA to indicate that the + element needs more information (caps, a buffer, etc) to + choose the correct caps. Should return ANY caps if the + stream has not caps at all. @@ -2044,6 +2178,10 @@ _finish_buffer from inside that function. + Optional. + Fixate and return the src pad caps provided. The function takes + ownership of @caps and returns a fixated version of + @caps. @caps is not guaranteed to be writable. @@ -2060,6 +2198,8 @@ _finish_buffer from inside that function. + Optional. + Notifies subclasses what caps format has been negotiated @@ -2076,6 +2216,11 @@ _finish_buffer from inside that function. + Optional. + Allows the subclass to influence the allocation choices. + Setup the allocation parameters for allocating output + buffers. The passed in query contains the result of the + downstream allocation query. @@ -2092,6 +2237,8 @@ _finish_buffer from inside that function. + Optional. + Allows the subclass to handle the allocation query from upstream. @@ -2114,6 +2261,8 @@ _finish_buffer from inside that function. + Optional. + Negotiate the caps with the peer (Since: 1.18). @@ -2129,6 +2278,9 @@ _finish_buffer from inside that function. + Optional. + Called when an event is received on a sink pad before queueing up + serialized events. The subclass should always chain up (Since: 1.18). @@ -2148,6 +2300,9 @@ _finish_buffer from inside that function. + Optional. + Called when a query is received on a sink pad before queueing up + serialized queries. The subclass should always chain up (Since: 1.18). @@ -2216,6 +2371,10 @@ _finish_buffer from inside that function. This class used to live in gst-plugins-bad and was moved to core. + Optional + Called when the pad has received a flush stop, this is the place + to flush any information specific to the pad, it allows for individual + pads to be flushed while others might not be. @@ -2230,6 +2389,9 @@ This class used to live in gst-plugins-bad and was moved to core. + Optional + Called before input buffers are queued in the pad, return %TRUE + if the buffer should be skipped. @@ -2334,7 +2496,7 @@ usage. - + Enables the emission of signals such as #GstAggregatorPad::buffer-consumed @@ -2370,6 +2532,10 @@ usage. + Optional + Called when the pad has received a flush stop, this is the place + to flush any information specific to the pad, it allows for individual + pads to be flushed while others might not be. @@ -2386,6 +2552,9 @@ usage. + Optional + Called before input buffers are queued in the pad, return %TRUE + if the buffer should be skipped. @@ -2410,21 +2579,21 @@ usage. - + - + - + Start at running time 0. - + Start at the running time of the first buffer that is received. - + Start at the running time selected by the `start-time` property. @@ -2935,6 +3104,8 @@ Things that subclass need to take care of: frame intervals. + Optional. + Convert between formats. @@ -2958,6 +3129,10 @@ Things that subclass need to take care of: + Optional. + Called until it doesn't return GST_FLOW_OK anymore for + the first buffers. Can be used by the subclass to detect + the stream format. @@ -2972,6 +3147,8 @@ Things that subclass need to take care of: + Optional. + Allows the subclass to do its own sink get caps if needed. @@ -3013,6 +3190,12 @@ if desired. + Optional. + Called just prior to pushing a frame (after any pending + events have been sent) to give subclass a chance to perform + additional actions at this time (e.g. tag sending) or to + decide whether this buffer should be dropped or not + (e.g. custom segment clipping). @@ -3027,6 +3210,8 @@ if desired. + Optional. + Allows the subclass to be notified of the actual caps set. @@ -3041,6 +3226,10 @@ if desired. + Optional. + Event handler on the sink pad. This function should chain + up to the parent implementation to let the default handler + run. @@ -3055,6 +3244,10 @@ if desired. + Optional. + Query handler on the sink pad. This function should chain + up to the parent implementation to let the default handler + run (Since: 1.2) @@ -3069,6 +3262,9 @@ if desired. + Optional. + Event handler on the source pad. Should chain up to the + parent to let the default handler run. @@ -3083,6 +3279,9 @@ if desired. + Optional. + Query handler on the source pad. Should chain up to the + parent to let the default handler run (Since: 1.2) @@ -3097,6 +3296,9 @@ if desired. + Optional. + Called when the element starts processing. + Allows opening external resources. @@ -3108,6 +3310,9 @@ if desired. + Optional. + Called when the element stops processing. + Allows closing external resources. @@ -3536,7 +3741,7 @@ into the frame data that the picture starts. - + 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 @@ -3579,6 +3784,9 @@ needed. At minimum @handle_frame needs to be overridden. + Optional. + Called when the element starts processing. + Allows opening external resources. @@ -3592,6 +3800,9 @@ needed. At minimum @handle_frame needs to be overridden. + Optional. + Called when the element stops processing. + Allows closing external resources. @@ -3605,6 +3816,8 @@ needed. At minimum @handle_frame needs to be overridden. + Optional. + Allows the subclass to be notified of the actual caps set. @@ -3621,6 +3834,16 @@ needed. At minimum @handle_frame needs to be overridden. + Parses the input data into valid frames as defined by subclass + which should be passed to gst_base_parse_finish_frame(). + The frame's input buffer is guaranteed writable, + whereas the input frame ownership is held by caller + (so subclass should make a copy if it needs to hang on). + Input buffer (data) is provided by baseclass with as much + metadata set as possible by baseclass according to upstream + information and/or subclass settings, + though subclass may still set buffer timestamp and duration + if desired. @@ -3640,6 +3863,12 @@ needed. At minimum @handle_frame needs to be overridden. + Optional. + Called just prior to pushing a frame (after any pending + events have been sent) to give subclass a chance to perform + additional actions at this time (e.g. tag sending) or to + decide whether this buffer should be dropped or not + (e.g. custom segment clipping). @@ -3656,6 +3885,8 @@ needed. At minimum @handle_frame needs to be overridden. + Optional. + Convert between formats. @@ -3681,6 +3912,10 @@ needed. At minimum @handle_frame needs to be overridden. + Optional. + Event handler on the sink pad. This function should chain + up to the parent implementation to let the default handler + run. @@ -3697,6 +3932,9 @@ needed. At minimum @handle_frame needs to be overridden. + Optional. + Event handler on the source pad. Should chain up to the + parent to let the default handler run. @@ -3713,6 +3951,8 @@ needed. At minimum @handle_frame needs to be overridden. + Optional. + Allows the subclass to do its own sink get caps if needed. @@ -3729,6 +3969,10 @@ needed. At minimum @handle_frame needs to be overridden. + Optional. + Called until it doesn't return GST_FLOW_OK anymore for + the first buffers. Can be used by the subclass to detect + the stream format. @@ -3745,6 +3989,10 @@ needed. At minimum @handle_frame needs to be overridden. + Optional. + Query handler on the sink pad. This function should chain + up to the parent implementation to let the default handler + run (Since: 1.2) @@ -3761,6 +4009,9 @@ needed. At minimum @handle_frame needs to be overridden. + Optional. + Query handler on the source pad. Should chain up to the + parent to let the default handler run (Since: 1.2) @@ -3937,7 +4188,7 @@ allocated on the stack. when the first non-queued frame is finished - + @@ -4056,6 +4307,11 @@ perform an ASYNC state change. This feature is mostly usable when dealing with non-synchronized streams or sparse streams. + Subclasses should override this when they can provide an + alternate method of spawning a thread to drive the pipeline in pull mode. + Should start or stop the pulling thread, depending on the value of the + "active" argument. Called after actually activating the sink pad in pull + mode. The default implementation starts a task on the sink pad. @@ -4070,6 +4326,7 @@ with non-synchronized streams or sparse streams. + Override this to handle events arriving on the sink pad @@ -4084,6 +4341,8 @@ with non-synchronized streams or sparse streams. + Only useful in pull mode. Implement if you have + ideas about what should be the default values for the caps you support. @@ -4136,6 +4395,8 @@ with non-synchronized streams or sparse streams. + Called to prepare the buffer for @render and @preroll. This + function is called before synchronisation is performed. @@ -4150,6 +4411,8 @@ with non-synchronized streams or sparse streams. + Called to prepare the buffer list for @render_list. This + function is called before synchronisation is performed. @@ -4164,6 +4427,7 @@ with non-synchronized streams or sparse streams. + Called to present the preroll buffer if desired. @@ -4178,6 +4442,7 @@ with non-synchronized streams or sparse streams. + configure the allocation query @@ -4192,6 +4457,7 @@ with non-synchronized streams or sparse streams. + perform a #GstQuery on the element. @@ -4206,6 +4472,8 @@ with non-synchronized streams or sparse streams. + Called when a buffer should be presented or output, at the + correct moment if the #GstBaseSink has been set to sync to the clock. @@ -4220,6 +4488,8 @@ with non-synchronized streams or sparse streams. + Same as @render but used with buffer lists instead of + buffers. @@ -4234,6 +4504,7 @@ with non-synchronized streams or sparse streams. + Notify subclass of changed caps @@ -4248,6 +4519,7 @@ with non-synchronized streams or sparse streams. + Start processing. Ideal for opening resources in the subclass @@ -4259,6 +4531,7 @@ with non-synchronized streams or sparse streams. + Stop processing. Subclasses should use this to close resources. @@ -4270,6 +4543,8 @@ with non-synchronized streams or sparse streams. + Unlock any pending access to the resource. Subclasses should + unblock any blocked function ASAP and call gst_base_sink_wait_preroll() @@ -4281,6 +4556,11 @@ with non-synchronized streams or sparse streams. + Clear the previous unlock request. Subclasses should clear + any state they set during #GstBaseSinkClass::unlock, and be ready to + continue where they left off after gst_base_sink_wait_preroll(), + gst_base_sink_wait() or gst_wait_sink_wait_clock() return or + #GstBaseSinkClass::render is called again. @@ -4292,6 +4572,9 @@ with non-synchronized streams or sparse streams. + Override this to implement custom logic to wait for the event + time (for events like EOS and GAP). Subclasses should always first + chain up to the default implementation. @@ -4329,7 +4612,7 @@ continue. Any other return value should be returned from the render vmethod. - + Get the number of bytes that the sink will pull when it is operating in pull mode. @@ -4360,7 +4643,7 @@ current segment. - + 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. @@ -4395,7 +4678,7 @@ Free-function: gst_sample_unref - + Get the maximum amount of bits per second that the sink will render. @@ -4409,7 +4692,7 @@ Free-function: gst_sample_unref - + Gets the max lateness value. See gst_base_sink_set_max_lateness() for more details. @@ -4426,7 +4709,7 @@ unlimited time. - + Get the processing deadline of @sink. see gst_base_sink_set_processing_deadline() for more information about the processing deadline. @@ -4442,7 +4725,7 @@ the processing deadline. - + Get the render delay of @sink. see gst_base_sink_set_render_delay() for more information about the render delay. @@ -4457,7 +4740,7 @@ information about the render delay. - + Return various #GstBaseSink statistics. This function returns a #GstStructure with name `application/x-gst-base-sink-stats` with the following fields: @@ -4476,7 +4759,7 @@ with name `application/x-gst-base-sink-stats` with the following fields: - + Checks if @sink is currently configured to synchronize against the clock. @@ -4491,7 +4774,7 @@ clock. - + Get the time that will be inserted between frames to control the maximum buffers per second. @@ -4506,7 +4789,7 @@ maximum buffers per second. - + Get the synchronisation offset of @sink. @@ -4625,7 +4908,7 @@ against the clock or when it is dealing with sparse streams. - + Set the number of bytes that the sink will pull when it is operating in pull mode. @@ -4678,7 +4961,7 @@ property. - + Set the maximum amount of bits per second that the sink will render. @@ -4695,7 +4978,7 @@ property. - + 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 @@ -4715,7 +4998,7 @@ an unlimited time. - + Maximum amount of time (in nanoseconds) that the pipeline can take for processing the buffer. This is added to the latency of live pipelines. @@ -4753,7 +5036,7 @@ This function is usually called by subclasses. - + 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 @@ -4778,7 +5061,7 @@ This function is usually called by subclasses. - + 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 @@ -4799,7 +5082,7 @@ contents. - + 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. @@ -4818,7 +5101,7 @@ will render. - + 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 @@ -4938,55 +5221,55 @@ continue. Any other return value should be returned from the render vmethod. - + 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. - + - + Maximum amount of time (in nanoseconds) that the pipeline can take for processing the buffer. This is added to the latency of live pipelines. - + - + 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. - + Various #GstBaseSink statistics. This property returns a #GstStructure with name `application/x-gst-base-sink-stats` with the following fields: @@ -4995,16 +5278,16 @@ with name `application/x-gst-base-sink-stats` with the following fields: - "rendered" G_TYPE_UINT64 Number of rendered frames - + - + 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. @@ -5086,6 +5369,7 @@ output/present buffers. + Called to get sink pad caps from the subclass @@ -5102,6 +5386,7 @@ output/present buffers. + Notify subclass of changed caps @@ -5118,6 +5403,8 @@ output/present buffers. + Only useful in pull mode. Implement if you have + ideas about what should be the default values for the caps you support. @@ -5134,6 +5421,11 @@ output/present buffers. + Subclasses should override this when they can provide an + alternate method of spawning a thread to drive the pipeline in pull mode. + Should start or stop the pulling thread, depending on the value of the + "active" argument. Called after actually activating the sink pad in pull + mode. The default implementation starts a task on the sink pad. @@ -5150,6 +5442,8 @@ output/present buffers. + Called to get the start and end times for synchronising + the passed buffer to the clock @@ -5174,6 +5468,7 @@ output/present buffers. + configure the allocation query @@ -5190,6 +5485,7 @@ output/present buffers. + Start processing. Ideal for opening resources in the subclass @@ -5203,6 +5499,7 @@ output/present buffers. + Stop processing. Subclasses should use this to close resources. @@ -5216,6 +5513,8 @@ output/present buffers. + Unlock any pending access to the resource. Subclasses should + unblock any blocked function ASAP and call gst_base_sink_wait_preroll() @@ -5229,6 +5528,11 @@ output/present buffers. + Clear the previous unlock request. Subclasses should clear + any state they set during #GstBaseSinkClass::unlock, and be ready to + continue where they left off after gst_base_sink_wait_preroll(), + gst_base_sink_wait() or gst_wait_sink_wait_clock() return or + #GstBaseSinkClass::render is called again. @@ -5242,6 +5546,7 @@ output/present buffers. + perform a #GstQuery on the element. @@ -5258,6 +5563,7 @@ output/present buffers. + Override this to handle events arriving on the sink pad @@ -5274,6 +5580,9 @@ output/present buffers. + Override this to implement custom logic to wait for the event + time (for events like EOS and GAP). Subclasses should always first + chain up to the default implementation. @@ -5290,6 +5599,8 @@ output/present buffers. + Called to prepare the buffer for @render and @preroll. This + function is called before synchronisation is performed. @@ -5306,6 +5617,8 @@ output/present buffers. + Called to prepare the buffer list for @render_list. This + function is called before synchronisation is performed. @@ -5322,6 +5635,7 @@ output/present buffers. + Called to present the preroll buffer if desired. @@ -5338,6 +5652,8 @@ output/present buffers. + Called when a buffer should be presented or output, at the + correct moment if the #GstBaseSink has been set to sync to the clock. @@ -5354,6 +5670,8 @@ output/present buffers. + Same as @render but used with buffer lists instead of + buffers. @@ -5375,7 +5693,7 @@ output/present buffers. - + @@ -5537,6 +5855,7 @@ implementation will call alloc if no allocated @buf is provided and then call fi + configure the allocation query @@ -5551,6 +5870,7 @@ implementation will call alloc if no allocated @buf is provided and then call fi + Perform seeking on the resource to the indicated segment. @@ -5565,6 +5885,7 @@ implementation will call alloc if no allocated @buf is provided and then call fi + Override this to implement custom event handling. @@ -5579,6 +5900,8 @@ implementation will call alloc if no allocated @buf is provided and then call fi + Ask the subclass to fill the buffer with data for offset and size. The + passed buffer is guaranteed to hold the requested amount of bytes. @@ -5669,6 +5992,7 @@ out. The base class will sync on the clock using these times. + Check if the source can seek @@ -5700,6 +6024,12 @@ buffer is allocated. + Prepare the #GstSegment that will be passed to the + #GstBaseSrcClass::do_seek vmethod for executing a seek + request. Sub-classes should override this if they support seeking in + formats other than the configured native format. By default, it tries to + convert the seek arguments to the configured native format and prepare a + segment in that format. @@ -5717,6 +6047,7 @@ buffer is allocated. + Handle a requested query. @@ -5749,6 +6080,10 @@ buffer is allocated. + Start processing. Subclasses should open resources and prepare + to produce data. Implementation should call gst_base_src_start_complete() + when the operation completes, either from the current thread or any other + thread that finishes the start operation asynchronously. @@ -5760,6 +6095,7 @@ buffer is allocated. + Stop processing. Subclasses should use this to close resources. @@ -5771,6 +6107,12 @@ buffer is allocated. + Unlock any pending access to the resource. Subclasses should unblock + any blocked function ASAP. In particular, any `create()` function in + progress should be unblocked and should return GST_FLOW_FLUSHING. Any + future #GstBaseSrcClass::create function call should also return + GST_FLOW_FLUSHING until the #GstBaseSrcClass::unlock_stop function has + been called. @@ -5782,6 +6124,9 @@ buffer is allocated. + Clear the previous unlock request. Subclasses should clear any + state they set during #GstBaseSrcClass::unlock, such as clearing command + queues. @@ -5817,7 +6162,7 @@ used - + Get the number of bytes that @src will push out with each buffer. @@ -5845,7 +6190,7 @@ by the src; unref it after usage. - + Query if @src timestamps outgoing buffers based on the current running_time. @@ -6047,7 +6392,7 @@ blocking operation should be unblocked with the unlock vmethod. - + 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 @@ -6073,7 +6418,7 @@ returns %GST_FLOW_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. @@ -6109,7 +6454,7 @@ returns %GST_FLOW_EOS. - + Configure @src to automatically timestamp outgoing buffers based on the current running_time of the pipeline. This property is mostly useful for live sources. @@ -6282,20 +6627,20 @@ continue. Any other return value should be returned from the create vmethod. - + See gst_base_src_set_automatic_eos() - + - + - + - + @@ -6368,6 +6713,7 @@ buffers. + Called to get the caps to report @@ -6384,6 +6730,7 @@ buffers. + Negotiated the caps with the peer. @@ -6399,6 +6746,8 @@ buffers. + Called during negotiation if caps need fixating. Implement instead of + setting a fixate function on the source pad. @@ -6416,6 +6765,7 @@ buffers. + Notify subclass of changed output caps @@ -6435,6 +6785,7 @@ buffers. + configure the allocation query @@ -6451,6 +6802,10 @@ buffers. + Start processing. Subclasses should open resources and prepare + to produce data. Implementation should call gst_base_src_start_complete() + when the operation completes, either from the current thread or any other + thread that finishes the start operation asynchronously. @@ -6464,6 +6819,7 @@ buffers. + Stop processing. Subclasses should use this to close resources. @@ -6477,6 +6833,9 @@ buffers. + Given a buffer, return the start and stop time when it + should be pushed out. The base class will sync on the clock using + these times. @@ -6499,6 +6858,8 @@ buffers. + Return the total size of the resource, in the format set by + gst_base_src_set_format(). @@ -6516,6 +6877,7 @@ buffers. + Check if the source can seek @@ -6529,6 +6891,12 @@ buffers. + Prepare the #GstSegment that will be passed to the + #GstBaseSrcClass::do_seek vmethod for executing a seek + request. Sub-classes should override this if they support seeking in + formats other than the configured native format. By default, it tries to + convert the seek arguments to the configured native format and prepare a + segment in that format. @@ -6548,6 +6916,7 @@ buffers. + Perform seeking on the resource to the indicated segment. @@ -6564,6 +6933,12 @@ buffers. + Unlock any pending access to the resource. Subclasses should unblock + any blocked function ASAP. In particular, any `create()` function in + progress should be unblocked and should return GST_FLOW_FLUSHING. Any + future #GstBaseSrcClass::create function call should also return + GST_FLOW_FLUSHING until the #GstBaseSrcClass::unlock_stop function has + been called. @@ -6577,6 +6952,9 @@ buffers. + Clear the previous unlock request. Subclasses should clear any + state they set during #GstBaseSrcClass::unlock, such as clearing command + queues. @@ -6590,6 +6968,7 @@ buffers. + Handle a requested query. @@ -6606,6 +6985,7 @@ buffers. + Override this to implement custom event handling. @@ -6622,6 +7002,13 @@ buffers. + Ask the subclass to create a buffer with offset and size. When the + subclass returns GST_FLOW_OK, it MUST return a buffer of the requested size + unless fewer bytes are available because an EOS condition is near. No + buffer should be returned when the return value is different from + GST_FLOW_OK. A return value of GST_FLOW_EOS signifies that the end of + stream is reached. The default implementation will call + #GstBaseSrcClass::alloc and then call #GstBaseSrcClass::fill. @@ -6644,6 +7031,8 @@ buffers. + Ask the subclass to allocate a buffer with for offset and size. The + default implementation will create a new buffer from the negotiated allocator. @@ -6666,6 +7055,8 @@ buffers. + Ask the subclass to fill the buffer with data for offset and size. The + passed buffer is guaranteed to hold the requested amount of bytes. @@ -6706,7 +7097,7 @@ buffers. offset to define more flags - + @@ -6816,6 +7207,10 @@ It provides for: * Implied %FALSE if ONLY transform function is implemented. + Optional. + Subclasses can override this method to check if @caps can be + handled by the element. The default implementation might not be + the most optimal way to check this in all cases. @@ -6833,6 +7228,10 @@ It provides for: + Optional. + This method is called right before the base class will + start processing. Dynamic properties or other delayed + configuration could be performed in this method. @@ -6847,6 +7246,10 @@ It provides for: + Optional. + Copy the metadata from the input buffer to the output buffer. + The default implementation will copy the flags, timestamps and + offsets of the buffer. @@ -6864,6 +7267,14 @@ It provides for: + Setup the allocation parameters for allocating output + buffers. The passed in query contains the result of the + downstream allocation query. This function is only called + when not operating in passthrough mode. The default + implementation will remove all memory dependent metadata. + If there is a @filter_meta method implementation, it will + be called for all metadata API in the downstream query, + otherwise the metadata API is removed. @@ -6878,6 +7289,9 @@ It provides for: + Return %TRUE if the metadata API should be proposed in the + upstream allocation query. The default implementation is %NULL + and will cause all metadata to be removed. @@ -6966,6 +7380,14 @@ It provides for: + Propose buffer allocation parameters for upstream elements. + This function must be implemented if the element reads or + writes the buffer content. The query that was passed to + the decide_allocation is passed in this method (or %NULL + when the element is in passthrough mode). The default + implementation will pass the query downstream when in + passthrough mode and will copy all the filtered metadata + API in non-passthrough mode. @@ -6983,6 +7405,10 @@ It provides for: + Optional. + Handle a requested query. Subclasses that implement this + must chain up to the parent if they didn't handle the + query @@ -7000,6 +7426,7 @@ It provides for: + Allows the subclass to be notified of the actual caps set. @@ -7045,6 +7472,9 @@ It provides for: + Optional. + Called when the element starts processing. + Allows opening external resources. @@ -7056,6 +7486,9 @@ It provides for: + Optional. + Called when the element stops processing. + Allows closing external resources. @@ -7067,6 +7500,13 @@ It provides for: + Function which accepts a new input buffer and pre-processes it. + The default implementation performs caps (re)negotiation, then + QoS if needed, and places the input buffer into the @queued_buf + member variable. If the buffer is dropped due to QoS, it returns + GST_BASE_TRANSFORM_FLOW_DROPPED. If this input buffer is not + contiguous with any previous input buffer, then @is_discont + is set to %TRUE. (Since: 1.6) @@ -7084,6 +7524,10 @@ It provides for: + Required if the element does not operate in-place. + Transforms one incoming buffer to one outgoing buffer. + The function is allowed to change size/timestamp/duration + of the outgoing buffer. @@ -7101,6 +7545,9 @@ It provides for: + Optional. Given the pad in this direction and the given + caps, what caps are allowed on the other pad in this + element ? @@ -7121,6 +7568,8 @@ It provides for: + Required if the element operates in-place. + Transform the incoming buffer in-place. @@ -7135,6 +7584,10 @@ It provides for: + Optional. Transform the metadata on the input buffer to the + output buffer. By default this method copies all meta without + tags. Subclasses can implement this method and return %TRUE if + the metadata is to be copied. @@ -7494,7 +7947,7 @@ downstream - + @@ -7549,6 +8002,9 @@ same type and quantity) it should provide @transform_ip. + Optional. Given the pad in this direction and the given + caps, what caps are allowed on the other pad in this + element ? @@ -7571,6 +8027,10 @@ same type and quantity) it should provide @transform_ip. + Optional. Given the pad in this direction and the given + caps, fixate the caps on the other pad. The function takes + ownership of @othercaps and returns a fixated version of + @othercaps. @othercaps is not guaranteed to be writable. @@ -7593,6 +8053,10 @@ same type and quantity) it should provide @transform_ip. + Optional. + Subclasses can override this method to check if @caps can be + handled by the element. The default implementation might not be + the most optimal way to check this in all cases. @@ -7612,6 +8076,7 @@ same type and quantity) it should provide @transform_ip. + Allows the subclass to be notified of the actual caps set. @@ -7631,6 +8096,10 @@ same type and quantity) it should provide @transform_ip. + Optional. + Handle a requested query. Subclasses that implement this + must chain up to the parent if they didn't handle the + query @@ -7650,6 +8119,14 @@ same type and quantity) it should provide @transform_ip. + Setup the allocation parameters for allocating output + buffers. The passed in query contains the result of the + downstream allocation query. This function is only called + when not operating in passthrough mode. The default + implementation will remove all memory dependent metadata. + If there is a @filter_meta method implementation, it will + be called for all metadata API in the downstream query, + otherwise the metadata API is removed. @@ -7666,6 +8143,9 @@ same type and quantity) it should provide @transform_ip. + Return %TRUE if the metadata API should be proposed in the + upstream allocation query. The default implementation is %NULL + and will cause all metadata to be removed. @@ -7688,6 +8168,14 @@ same type and quantity) it should provide @transform_ip. + Propose buffer allocation parameters for upstream elements. + This function must be implemented if the element reads or + writes the buffer content. The query that was passed to + the decide_allocation is passed in this method (or %NULL + when the element is in passthrough mode). The default + implementation will pass the query downstream when in + passthrough mode and will copy all the filtered metadata + API in non-passthrough mode. @@ -7707,6 +8195,11 @@ same type and quantity) it should provide @transform_ip. + Optional. Given the size of a buffer in the given direction + with the given caps, calculate the size in bytes of a buffer + on the other pad with the given other caps. + The default implementation uses get_unit_size and keeps + the number of units the same. @@ -7735,6 +8228,8 @@ same type and quantity) it should provide @transform_ip. + Required if the transform is not in-place. + Get the size in bytes of one unit for the given caps. @@ -7754,6 +8249,9 @@ same type and quantity) it should provide @transform_ip. + Optional. + Called when the element starts processing. + Allows opening external resources. @@ -7767,6 +8265,9 @@ same type and quantity) it should provide @transform_ip. + Optional. + Called when the element stops processing. + Allows closing external resources. @@ -7780,6 +8281,9 @@ same type and quantity) it should provide @transform_ip. + Optional. + Event handler on the sink pad. The default implementation + handles the event and forwards it downstream. @@ -7796,6 +8300,9 @@ same type and quantity) it should provide @transform_ip. + Optional. + Event handler on the source pad. The default implementation + handles the event and forwards it upstream. @@ -7812,6 +8319,15 @@ same type and quantity) it should provide @transform_ip. + Optional. + Subclasses can override this to do their own + allocation of output buffers. Elements that only do + analysis can return a subbuffer or even just + return a reference to the input buffer (if in + passthrough mode). The default implementation will + use the negotiated allocator or bufferpool and + transform_size to allocate an output buffer or it + will return the input buffer in passthrough mode. @@ -7831,6 +8347,10 @@ same type and quantity) it should provide @transform_ip. + Optional. + Copy the metadata from the input buffer to the output buffer. + The default implementation will copy the flags, timestamps and + offsets of the buffer. @@ -7850,6 +8370,10 @@ same type and quantity) it should provide @transform_ip. + Optional. Transform the metadata on the input buffer to the + output buffer. By default this method copies all meta without + tags. Subclasses can implement this method and return %TRUE if + the metadata is to be copied. @@ -7872,6 +8396,10 @@ same type and quantity) it should provide @transform_ip. + Optional. + This method is called right before the base class will + start processing. Dynamic properties or other delayed + configuration could be performed in this method. @@ -7888,6 +8416,10 @@ same type and quantity) it should provide @transform_ip. + Required if the element does not operate in-place. + Transforms one incoming buffer to one outgoing buffer. + The function is allowed to change size/timestamp/duration + of the outgoing buffer. @@ -7907,6 +8439,8 @@ same type and quantity) it should provide @transform_ip. + Required if the element operates in-place. + Transform the incoming buffer in-place. @@ -7923,6 +8457,13 @@ same type and quantity) it should provide @transform_ip. + Function which accepts a new input buffer and pre-processes it. + The default implementation performs caps (re)negotiation, then + QoS if needed, and places the input buffer into the @queued_buf + member variable. If the buffer is dropped due to QoS, it returns + GST_BASE_TRANSFORM_FLOW_DROPPED. If this input buffer is not + contiguous with any previous input buffer, then @is_discont + is set to %TRUE. (Since: 1.6) @@ -7942,6 +8483,15 @@ same type and quantity) it should provide @transform_ip. + Called after each new input buffer is submitted repeatedly + until it either generates an error or fails to generate an output + buffer. The default implementation takes the contents of the + @queued_buf variable, generates an output buffer if needed + by calling the class @prepare_output_buffer, and then + calls either @transform or @transform_ip. Elements that don't + do 1-to-1 transformations of input to output buffers can either + return GST_BASE_TRANSFORM_FLOW_DROPPED or simply not generate + an output buffer until they are ready to do so. (Since: 1.6) @@ -7963,7 +8513,7 @@ same type and quantity) it should provide @transform_ip. - + @@ -11207,7 +11757,7 @@ memory and resources allocated for it. - + @@ -12071,7 +12621,7 @@ gst_collect_pads_set_flushing nor gst_collect_pads_clear from this function. - + @@ -12408,13 +12958,13 @@ MT Safe. - + - + - + @@ -12577,7 +13127,7 @@ This function should also drop the reference to @object the owner of the - + @@ -12596,7 +13146,7 @@ This function should also drop the reference to @object the owner of the - + 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. @@ -13003,6 +13553,7 @@ no allocated @buf is provided and then call fill. + Ask the subclass to fill the buffer with data. @@ -13035,6 +13586,10 @@ buffers. + Ask the subclass to create a buffer. The subclass decides which + size this buffer should be. Other then that, refer to + #GstBaseSrc<!-- -->.create() for more details. If this method is + not implemented, @alloc followed by @fill will be called. @@ -13051,6 +13606,9 @@ buffers. + Ask the subclass to allocate a buffer. The subclass decides which + size this buffer should be. The default implementation will create + a new buffer from the negotiated allocator. @@ -13067,6 +13625,7 @@ buffers. + Ask the subclass to fill the buffer with data. @@ -13088,7 +13647,7 @@ buffers. - + #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. @@ -13569,7 +14128,7 @@ of size @struct_size, with an initial queue size of @initial_size. - + The opaque #GstTypeFindData structure. diff --git a/girs/GstCheck-1.0.gir b/girs/GstCheck-1.0.gir index f729651ce1..a562e9cacf 100644 --- a/girs/GstCheck-1.0.gir +++ b/girs/GstCheck-1.0.gir @@ -31,7 +31,7 @@ and/or use gtk-doc annotations. --> - + Opaque structure containing data about a log filter function. @@ -2124,10 +2124,10 @@ MT safe. - + - + Opaque handle representing a GstHarness stress testing thread. @@ -2145,7 +2145,7 @@ MT safe. - + Opaque consistency checker handle. @@ -2674,10 +2674,10 @@ with information about the pending clock notification - + - + When a #GstTestClock is constructed it will have a certain start time set. If the clock was created using gst_test_clock_new_with_start_time() then this property contains the value of the @start_time argument. If @@ -2700,7 +2700,7 @@ this property contains the value 0. - + diff --git a/girs/GstCodecs-1.0.gir b/girs/GstCodecs-1.0.gir index cc58e6407b..3ff6a2b674 100644 --- a/girs/GstCodecs-1.0.gir +++ b/girs/GstCodecs-1.0.gir @@ -384,7 +384,7 @@ decoding process for the #GstAV1Picture - + @@ -862,7 +862,7 @@ the @system_frame_number - + The compliance controls the behavior of the decoder to handle some subtle cases and contexts, such as the low-latency DPB bumping or mapping the baseline profile as the constrained-baseline profile, @@ -1085,17 +1085,17 @@ etc. - + The decoder behavior is automatically choosen. - + The decoder behavior strictly conforms to the SPEC. All the decoder behaviors conform to the SPEC, not including any nonstandard behavior which is not mentioned in the SPEC. - + The decoder behavior normally conforms to the SPEC. Most behaviors conform to the SPEC but including some nonstandard features which are widely used or @@ -1109,7 +1109,7 @@ etc. have problems when a real baseline stream comes with FMO or ASO. - + The decoder behavior flexibly conforms to the SPEC. It uses the nonstandard features more aggressively in order to get better performance(for @@ -1119,10 +1119,10 @@ etc. disorder when reference frames POC decrease in decoder order. - + - + Store the @picture @@ -2217,10 +2217,10 @@ the @system_frame_number - + - + Store the @picture and perform increase pic_latency_cnt as defined in @@ -3215,10 +3215,10 @@ decoding process for the #GstMpeg2Picture - + - + Store the @picture @@ -3616,6 +3616,9 @@ outputted + Optional. + Called per one #GstVp8Picture to notify subclass to finish + decoding process for the #GstVp8Picture @@ -3649,6 +3652,10 @@ preferred by subclass or not. + Optional. + Called whenever new #GstVp8Picture is created. + Subclass can set implementation specific user data + on the #GstVp8Picture via gst_vp8_picture_set_user_data @@ -3666,6 +3673,7 @@ preferred by subclass or not. + Notifies subclass of SPS update @@ -3683,6 +3691,11 @@ preferred by subclass or not. + Called with a #GstVp8Picture which is required to be outputted. + Subclass can retrieve parent #GstVideoCodecFrame by using + gst_video_decoder_get_frame() with system_frame_number + and the #GstVideoCodecFrame must be consumed by subclass via + gst_video_decoder_{finish,drop,release}_frame(). @@ -3700,6 +3713,9 @@ preferred by subclass or not. + Optional. + Called per one #GstVp8Picture to notify subclass to prepare + decoding process for the #GstVp8Picture @@ -3743,6 +3759,7 @@ preferred by subclass or not. + Notifies subclass of SPS update @@ -3762,6 +3779,10 @@ preferred by subclass or not. + Optional. + Called whenever new #GstVp8Picture is created. + Subclass can set implementation specific user data + on the #GstVp8Picture via gst_vp8_picture_set_user_data @@ -3781,6 +3802,9 @@ preferred by subclass or not. + Optional. + Called per one #GstVp8Picture to notify subclass to prepare + decoding process for the #GstVp8Picture @@ -3816,6 +3840,9 @@ preferred by subclass or not. + Optional. + Called per one #GstVp8Picture to notify subclass to finish + decoding process for the #GstVp8Picture @@ -3832,6 +3859,11 @@ preferred by subclass or not. + Called with a #GstVp8Picture which is required to be outputted. + Subclass can retrieve parent #GstVideoCodecFrame by using + gst_video_decoder_get_frame() with system_frame_number + and the #GstVideoCodecFrame must be consumed by subclass via + gst_video_decoder_{finish,drop,release}_frame(). @@ -3875,7 +3907,7 @@ preferred by subclass or not. - + @@ -4294,7 +4326,7 @@ profile. - + diff --git a/girs/GstController-1.0.gir b/girs/GstController-1.0.gir index 0c3b8f52e3..5b2af56ea2 100644 --- a/girs/GstController-1.0.gir +++ b/girs/GstController-1.0.gir @@ -269,7 +269,7 @@ target property range without any transformations. - + @@ -503,7 +503,7 @@ All functions are MT-safe. - + @@ -529,22 +529,22 @@ All functions are MT-safe. - + 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) @@ -568,21 +568,21 @@ All functions are MT-safe. - + 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. @@ -590,7 +590,7 @@ To get a n nanosecond shift to the left use "(GST_SECOND / frequency) - n". - + Specifies the waveform that should be used for this #GstLFOControlSource. @@ -620,24 +620,24 @@ To get a n nanosecond shift to the left use - + The various waveform modes available. - + sine waveform - + square waveform - + saw waveform - + reverse saw waveform - + triangle waveform @@ -1007,7 +1007,7 @@ time. - + @@ -1028,7 +1028,7 @@ All functions are MT-safe. - + @@ -1054,7 +1054,7 @@ All functions are MT-safe. - + diff --git a/girs/GstCuda-1.0.gir b/girs/GstCuda-1.0.gir index bf2924b8c9..eda536c5a3 100644 --- a/girs/GstCuda-1.0.gir +++ b/girs/GstCuda-1.0.gir @@ -352,7 +352,7 @@ This method is conceptually identical to gst_buffer_pool_set_active method. - + @@ -386,7 +386,7 @@ This method is conceptually identical to gst_buffer_pool_set_active method. - + @@ -506,24 +506,24 @@ so all CUDA functions that operate on the current context are affected. - + - + External resource interop API support - + OS handle supportability in virtual memory management - + - + - + Virtual memory management supportability @@ -540,7 +540,7 @@ so all CUDA functions that operate on the current context are affected. - + @@ -872,12 +872,12 @@ CUDA stream is in use CUDA memory allocation method - + - + Memory allocated via cuMemAlloc or cuMemAllocPitch - + Memory allocated via cuMemCreate and cuMemMap @@ -978,10 +978,10 @@ failure - + - + @@ -1128,7 +1128,7 @@ inactive. - + @@ -1206,7 +1206,7 @@ failure - + diff --git a/girs/GstGL-1.0.gir b/girs/GstGL-1.0.gir index b55654f4ad..4a40d42fbf 100644 --- a/girs/GstGL-1.0.gir +++ b/girs/GstGL-1.0.gir @@ -40,23 +40,23 @@ synchronization metadata on buffers from the pool. - + no API - + Desktop OpenGL up to and including 3.1. The compatibility profile when the OpenGL version is >= 3.2 - + Desktop OpenGL >= 3.2 core profile - + OpenGL ES 1.x - + OpenGL ES 2.x and 3.x - + Any OpenGL API @@ -530,6 +530,9 @@ context. It also provided some wrappers around #GstBaseTransform's context is available and current in the calling thread. + called in the GL thread when caps are set on @filter. + Note: this will also be called when changing OpenGL contexts + where #GstBaseTransform::set_caps may not. @@ -547,6 +550,7 @@ context is available and current in the calling thread. + called in the GL thread to setup the element GL state. @@ -558,6 +562,7 @@ context is available and current in the calling thread. + called in the GL thread to setup the element GL state. @@ -636,6 +641,7 @@ context is available and current in the calling thread. + called in the GL thread to setup the element GL state. @@ -649,6 +655,7 @@ context is available and current in the calling thread. + called in the GL thread to setup the element GL state. @@ -662,6 +669,9 @@ context is available and current in the calling thread. + called in the GL thread when caps are set on @filter. + Note: this will also be called when changing OpenGL contexts + where #GstBaseTransform::set_caps may not. @@ -686,7 +696,7 @@ context is available and current in the calling thread. - + @@ -850,6 +860,7 @@ multiple times. This must be called before any other GstGLBaseMemory operation. Opaque #GstGLBaseMemoryAllocator struct + a #GstGLBaseMemoryAllocatorAllocFunction a newly allocated #GstGLBaseMemory from @allocator and @params @@ -1022,14 +1033,14 @@ function to allocate and OpenGL resources needed for your application - + generic failure - + the implementation is too old and doesn't implement enough features - + a resource could not be found @@ -1040,11 +1051,11 @@ function to allocate and OpenGL resources needed for your application - + the texture needs downloading to the data pointer - + the data pointer needs uploading to the texture @@ -1184,7 +1195,7 @@ is available and is not available within this element. - + @@ -1194,6 +1205,7 @@ context. It also provided some wrappers around #GstBaseSrc's `start()` and current in the calling thread. + called in the GL thread to fill the current video texture. @@ -1208,6 +1220,7 @@ current in the calling thread. + called in the GL thread to setup the element GL state. @@ -1219,6 +1232,7 @@ current in the calling thread. + called in the GL thread to setup the element GL state. @@ -1229,7 +1243,7 @@ current in the calling thread. - + @@ -1275,6 +1289,7 @@ current in the calling thread. + called in the GL thread to setup the element GL state. @@ -1288,6 +1303,7 @@ current in the calling thread. + called in the GL thread to setup the element GL state. @@ -1301,6 +1317,7 @@ current in the calling thread. + called in the GL thread to fill the current video texture. @@ -1322,7 +1339,7 @@ current in the calling thread. - + @@ -1497,7 +1514,7 @@ gst_buffer_pool_set_config() will cause this function to return a new - + @@ -1720,17 +1737,17 @@ yuv to rgb - + - + none - + slow - + non-conformant @@ -1749,16 +1766,16 @@ yuv to rgb - + none - + window - + pbuffer - + pixmap @@ -1980,6 +1997,7 @@ case. + choose a format for the framebuffer @@ -1991,6 +2009,7 @@ case. + create the OpenGL context @@ -2008,6 +2027,7 @@ case. + destroy the OpenGL context @@ -2748,6 +2768,7 @@ MT-safe + get the backing platform specific OpenGL context @@ -2763,6 +2784,7 @@ MT-safe + get the available OpenGL api's that this context can work with @@ -2793,6 +2815,7 @@ MT-safe + get an function pointer to an OpenGL function @@ -2809,6 +2832,7 @@ MT-safe + call eglMakeCurrent or similar @@ -2828,6 +2852,7 @@ MT-safe + choose a format for the framebuffer @@ -2841,6 +2866,7 @@ MT-safe + create the OpenGL context @@ -2860,6 +2886,7 @@ MT-safe + destroy the OpenGL context @@ -2873,6 +2900,7 @@ MT-safe + swap the default framebuffer's front/back buffers @@ -2970,22 +2998,22 @@ MT-safe OpenGL context errors. - + Failed for an unspecified reason - + The configuration requested is not correct - + The OpenGL API requested is not correct - + The OpenGL libraries are too old - + glXCreateContext (or similar) failed - + A resource is not available @@ -2995,7 +3023,7 @@ MT-safe - + @@ -3421,54 +3449,54 @@ display's object lock held. - + - + no display type - + X11 display - + Wayland display - + Cocoa display - + Win32 display - + Dispmanx display - + EGL display - + Vivante Framebuffer display - + Mesa3D GBM display - + EGLDevice display. - + EAGL display. - + WinRT display. - + Android display. - + Mesa3D surfaceless display using the EGL_PLATFORM_SURFACELESS_MESA extension. - + any display type @@ -3488,6 +3516,10 @@ single input and producing a single output with a #GstGLFramebuffer + perform operations on the input and output buffers. In general, + you should avoid using this method if at all possible. One valid + use-case for using this is keeping previous buffers for future calculations. + Note: If @filter exists, then @filter_texture is not run @@ -3527,6 +3559,7 @@ single input and producing a single output with a #GstGLFramebuffer + perform initialization when the Framebuffer object is created @@ -3538,6 +3571,7 @@ single input and producing a single output with a #GstGLFramebuffer + mirror from #GstBaseTransform @@ -3555,6 +3589,8 @@ single input and producing a single output with a #GstGLFramebuffer + Perform sub-class specific modifications of the + caps to be processed between upload on input and before download for output. @@ -3738,6 +3774,7 @@ See also: gst_gl_filter_render_to_target() + mirror from #GstBaseTransform @@ -3757,6 +3794,10 @@ See also: gst_gl_filter_render_to_target() + perform operations on the input and output buffers. In general, + you should avoid using this method if at all possible. One valid + use-case for using this is keeping previous buffers for future calculations. + Note: If @filter exists, then @filter_texture is not run @@ -3776,6 +3817,8 @@ See also: gst_gl_filter_render_to_target() + given @in_tex, transform it into @out_tex. Not used + if @filter exists @@ -3799,6 +3842,7 @@ See also: gst_gl_filter_render_to_target() + perform initialization when the Framebuffer object is created @@ -3812,6 +3856,8 @@ See also: gst_gl_filter_render_to_target() + Perform sub-class specific modifications of the + caps to be processed between upload on input and before download for output. @@ -3861,68 +3907,68 @@ See also: gst_gl_filter_render_to_target() - + Single component replicated across R, G, and B textures components - + Single component stored in the A texture component - + Combination of #GST_GL_LUMINANCE and #GST_GL_ALPHA - + Single component stored in the R texture component - + Single 8-bit component stored in the R texture component - + Two components stored in the R and G texture components - + Two 8-bit components stored in the R and G texture components - + Three components stored in the R, G, and B texture components - + Three 8-bit components stored in the R, G, and B texture components - + Three components of bit depth 5, 6 and 5 stored in the R, G, and B texture components respectively. - + Three 16-bit components stored in the R, G, and B texture components - + Four components stored in the R, G, B, and A texture components respectively. - + Four 8-bit components stored in the R, G, B, and A texture components respectively. - + Four 16-bit components stored in the R, G, B, and A texture components respectively. - + A single 16-bit component for depth information. - + A 24-bit component for depth information and a 8-bit component for stencil informat. - + - + Single 16-bit component stored in the R texture component - + Two 16-bit components stored in the R and G texture components @@ -4225,10 +4271,10 @@ with. - + - + Structure containing function pointers to OpenGL functions. Each field is named exactly the same as the OpenGL function without the @@ -4933,7 +4979,7 @@ manually. - + @@ -4997,7 +5043,7 @@ manually. - + @@ -5044,26 +5090,26 @@ manually. - + no platform - + the EGL platform used primarily with the X11, wayland and android window systems as well as on embedded Linux - + the GLX platform used primarily with the X11 window system - + the WGL platform used primarily on Windows - + the CGL platform used primarily on OS X - + the EAGL platform used primarily on iOS - + any OpenGL platform @@ -5263,13 +5309,13 @@ GStreamer elements. - + no query - + query the time elapsed - + query the current time @@ -5489,13 +5535,13 @@ multiple times. This must be called before any other GstGLRenderbuffer operatio Compilation stage that caused an error - + Compilation error occurred - + Link error occurred - + General program error occurred @@ -5507,19 +5553,19 @@ multiple times. This must be called before any other GstGLRenderbuffer operatio GLSL profiles - + no profile supported/available - + OpenGL|ES profile - + OpenGL core profile - + OpenGL compatibility profile - + any OpenGL/OpenGL|ES profile @@ -5783,60 +5829,60 @@ multiple times. This must be called before any other GstGLRenderbuffer operatio - + GLSL version list - + no version - + version 100 (only valid for ES) - + version 110 (only valid for compatibility desktop GL) - + version 120 (only valid for compatibility desktop GL) - + version 130 (only valid for compatibility desktop GL) - + version 140 (only valid for compatibility desktop GL) - + version 150 (valid for compatibility/core desktop GL) - + version 300 (only valid for ES) - + version 310 (only valid for ES) - + version 320 (only valid for ES) - + version 330 (valid for compatibility/core desktop GL) - + version 400 (valid for compatibility/core desktop GL) - + version 410 (valid for compatibility/core desktop GL) - + version 420 (valid for compatibility/core desktop GL) - + version 430 (valid for compatibility/core desktop GL) - + version 440 (valid for compatibility/core desktop GL) - + version 450 (valid for compatibility/core desktop GL) @@ -7025,7 +7071,7 @@ Note: must be called in the GL thread and @shader must have been linked. - + @@ -7049,18 +7095,18 @@ Note: must be called in the GL thread and @shader must have been linked. - + Output anaglyph type to generate when downmixing to mono - + Dubois optimised Green-Magenta anaglyph - + Dubois optimised Red-Cyan anaglyph - + Dubois optimised Amber-Blue anaglyph @@ -7081,6 +7127,7 @@ with the CPU or with other OpenGL contexts. + set a sync point in the OpenGL command stream @@ -7097,6 +7144,7 @@ with the CPU or with other OpenGL contexts. + the same as @set_sync but called from @context's thread @@ -7113,6 +7161,7 @@ with the CPU or with other OpenGL contexts. + execute a wait on the previously set sync point into the OpenGL command stream @@ -7129,6 +7178,7 @@ with the CPU or with other OpenGL contexts. + the same as @wait but called from @context's thread @@ -7145,6 +7195,7 @@ with the CPU or with other OpenGL contexts. + wait for the previously set sync point to pass from the CPU @@ -7161,6 +7212,7 @@ with the CPU or with other OpenGL contexts. + the same as @wait_cpu but called from @context's thread @@ -7177,6 +7229,7 @@ with the CPU or with other OpenGL contexts. + copy @data into a new #GstGLSyncMeta @@ -7199,6 +7252,7 @@ with the CPU or with other OpenGL contexts. + free @data @@ -7215,6 +7269,7 @@ with the CPU or with other OpenGL contexts. + free @data in @context's thread @@ -7297,17 +7352,17 @@ gst_gl_value_set_texture_target_from_mask(), gst_gl_value_get_texture_target_mask(), and gst_gl_value_set_texture_target() functions can be used for handling texture targets with #GValue's when e.g. dealing with #GstCaps. - + no texture target - + 2D texture target (`GL_TEXTURE_2D`) - + rectangle texture target (`GL_TEXTURE_RECTANGLE`) - + external oes texture target (`GL_TEXTURE_EXTERNAL_OES`) @@ -7580,23 +7635,23 @@ gst_gl_upload_set_caps() creating a new #GstBuffer in @outbuf_ptr. - + - + No further processing required - + An unspecified error occurred - + The configuration is unsupported. - + This element requires a reconfiguration. - + private return value. @@ -8098,19 +8153,19 @@ setting the caps with gst_gl_view_convert_set_caps(). - + - + - + - + - + @@ -8182,7 +8237,7 @@ setting the caps with gst_gl_view_convert_set_caps(). - + @@ -8203,6 +8258,7 @@ either be a user visible window (onscreen) or hidden (offscreen). + close the connection to the display @@ -8241,6 +8297,7 @@ either be a user visible window (onscreen) or hidden (offscreen). + Gets the current windowing system display connection the windowing system display handle for this @window @@ -8254,6 +8311,9 @@ either be a user visible window (onscreen) or hidden (offscreen). + Gets the current window handle that this #GstGLWindow is + rendering into. This may return a different value to + what is passed into @set_window_handle the window handle we are currently rendering into @@ -8302,6 +8362,7 @@ from the @window. + open the connection to the display @@ -9089,6 +9150,7 @@ direct handle comparision. + Gets the current windowing system display connection @@ -9104,6 +9166,7 @@ direct handle comparision. + Set a window handle to render into @@ -9122,6 +9185,9 @@ direct handle comparision. + Gets the current window handle that this #GstGLWindow is + rendering into. This may return a different value to + what is passed into @set_window_handle @@ -9137,6 +9203,7 @@ direct handle comparision. + redraw the window with the specified dimensions @@ -9151,6 +9218,7 @@ direct handle comparision. + run the mainloop @@ -9165,6 +9233,7 @@ direct handle comparision. + send a quit to the mainloop @@ -9179,6 +9248,7 @@ direct handle comparision. + invoke a function on the window thread. Required to be reentrant. @@ -9201,6 +9271,8 @@ direct handle comparision. + invoke a function on the window thread. @run may or may + not have been called. Required to be reentrant. @@ -9227,6 +9299,7 @@ direct handle comparision. + open the connection to the display @@ -9240,6 +9313,7 @@ direct handle comparision. + close the connection to the display @@ -9253,6 +9327,9 @@ direct handle comparision. + whether to handle 'extra' events from the windowing system. + Basic events like surface moves and resizes are still valid + things to listen for. @@ -9271,6 +9348,8 @@ direct handle comparision. + request that the window change surface size. The + implementation is free to ignore this information. @@ -9293,6 +9372,7 @@ direct handle comparision. + request that the window be shown to the user @@ -9307,6 +9387,7 @@ direct handle comparision. + request a rectangle to render into. See #GstVideoOverlay @@ -9338,6 +9419,7 @@ direct handle comparision. + request a resize to occur when possible @@ -9352,6 +9434,8 @@ direct handle comparision. + Whether the window takes care of glViewport setup. + and the user does not need to deal with viewports @@ -9367,6 +9451,7 @@ direct handle comparision. + Whether the window has output surface or not. (Since: 1.18) @@ -9388,13 +9473,13 @@ direct handle comparision. - + failed for a unspecified reason - + the implementation is too old - + no such resource was found @@ -9404,7 +9489,7 @@ direct handle comparision. - + @@ -9476,6 +9561,30 @@ user-defined purposes. + + Stores a debug message in @ad for later output + + + + the #GstGLAsyncDebug to store the message in + + + the #GstDebugCategory to output the message in + + + the #GstDebugLevel + + + a #GObject to associate with the debug message + + + a printf style format string + + + the list of arguments for @format + + + Stores a debug message in @ad for later output @@ -11201,6 +11310,23 @@ multiple times. This must be called before any other GstGLMemory operation. + + + + + + + + + + + + + + + + + @@ -11261,6 +11387,23 @@ GStreamer elements. + + + + + + + + + + + + + + + + + diff --git a/girs/GstInsertBin-1.0.gir b/girs/GstInsertBin-1.0.gir index 718ba18ca6..74dce73b07 100644 --- a/girs/GstInsertBin-1.0.gir +++ b/girs/GstInsertBin-1.0.gir @@ -409,7 +409,7 @@ operation is requested. - + diff --git a/girs/GstMse-1.0.gir b/girs/GstMse-1.0.gir index 4818df5a86..78211aa927 100644 --- a/girs/GstMse-1.0.gir +++ b/girs/GstMse-1.0.gir @@ -142,7 +142,7 @@ and set an error. - + Gets a #GstSourceBufferList containing all the Source Buffers currently associated with this Media Source that are considered "active." For a Source Buffer to be considered active, either its video track is @@ -163,7 +163,7 @@ well. - + Gets the current duration of @self. [Specification](https://www.w3.org/TR/media-source-2/#dom-mediasource-duration) @@ -197,7 +197,7 @@ the current live seekable range. - + Gets the current playback position of the Media Source. @@ -211,7 +211,7 @@ the current live seekable range. - + Gets the current Ready State of the Media Source. [Specification](https://www.w3.org/TR/media-source-2/#dom-mediasource-readystate) @@ -227,7 +227,7 @@ the current live seekable range. - + Gets a #GstSourceBufferList containing all the Source Buffers currently associated with this Media Source. This object will reflect any future changes to the parent Media Source as well. @@ -268,7 +268,7 @@ changes to the parent Media Source as well. - + Sets the duration of @self. [Specification](https://www.w3.org/TR/media-source-2/#dom-mediasource-duration) @@ -316,30 +316,30 @@ If the ready state is not %GST_MEDIA_SOURCE_READY_STATE_OPEN, or the supplied - + A #GstSourceBufferList of every #GstSourceBuffer in this Media Source that is considered active [Specification](https://www.w3.org/TR/media-source-2/#dom-mediasource-activesourcebuffers) - + The Duration of the Media Source as a #GstClockTime [Specification](https://www.w3.org/TR/media-source-2/#dom-mediasource-duration) - + The position of the player consuming from the Media Source - + The Ready State of the Media Source [Specification](https://www.w3.org/TR/media-source-2/#dom-mediasource-readystate) - + A #GstSourceBufferList of every #GstSourceBuffer in this Media Source [Specification](https://www.w3.org/TR/media-source-2/#dom-mediasource-sourcebuffers) @@ -378,13 +378,13 @@ gst_media_source_end_of_stream(). Reasons for ending a #GstMediaSource using gst_media_source_end_of_stream(). [Specification](https://www.w3.org/TR/media-source-2/#dom-endofstreamerror) - + End the stream successfully - + End the stream due to a networking error - + End the stream due to a decoding error @@ -393,15 +393,15 @@ gst_media_source_end_of_stream(). These values correspond directly to those in the Web IDL specification. [Specification](https://webidl.spec.whatwg.org/#idl-DOMException-error-names) - + - + - + - + - + Any error type that can be reported by the Media Source API. @@ -429,15 +429,15 @@ HTML specification, only representing a single @start and @end time. Describes the possible states of the Media Source. [Specification](https://www.w3.org/TR/media-source-2/#dom-readystate) - + The #GstMediaSource is not connected to any playback element. - + The #GstMediaSource is connected to a playback element and ready to append data to its #GstSourceBuffer (s). - + gst_media_source_end_of_stream() has been called on the current #GstMediaSource @@ -455,7 +455,7 @@ Once added to a Pipeline, this element should be attached to a Media Source using gst_media_source_attach(). - + Gets the duration of @self. [Specification](https://html.spec.whatwg.org/multipage/media.html#dom-media-duration) @@ -471,7 +471,7 @@ using gst_media_source_attach(). - + the number of audio tracks available from this source @@ -484,7 +484,7 @@ using gst_media_source_attach(). - + the number of text tracks available from this source @@ -497,7 +497,7 @@ using gst_media_source_attach(). - + the number of video tracks available from this source @@ -510,7 +510,7 @@ using gst_media_source_attach(). - + Gets the current playback position of @self. [Specification](https://html.spec.whatwg.org/multipage/media.html#current-playback-position) @@ -526,7 +526,7 @@ using gst_media_source_attach(). - + The Ready State of @self, describing to what level it can supply content for the current #GstMseSrc:position. This is a separate concept from #GstMediaSource:ready-state: and corresponds to the HTML Media Element's @@ -545,31 +545,31 @@ Ready State. - + The duration of the stream as a #GstClockTime [Specification](https://html.spec.whatwg.org/multipage/media.html#dom-media-duration) - + The number of audio tracks in the Media Source - + The number of text tracks in the Media Source - + The number of video tracks in the Media Source - + The playback position as a #GstClockTime [Specification](https://html.spec.whatwg.org/multipage/media.html#current-playback-position) - + The Ready State of this element, describing to what level it can supply content for the current #GstMseSrc:position. This is a separate concept from #GstMediaSource:ready-state: and corresponds to the HTML Media @@ -601,23 +601,23 @@ directly to the ready state of a HTML Media Element and is a separate concept from #GstMediaSourceReadyState. [Specification](https://html.spec.whatwg.org/multipage/media.html#ready-states) - + No information is available about the stream - + The duration is available and video dimensions are available if the stream contains video - + The current playback position can be presented but future information is not available - + There is data for the current position and some amount in the future and any text tracks are ready. - + Either there is enough data to play the stream through at the current playback and input rate or the input buffer is full. @@ -707,7 +707,7 @@ to the Source Buffer must be of the supplied @type afterward. - + [Specification](https://www.w3.org/TR/media-source-2/#dom-sourcebuffer-mode) @@ -721,7 +721,7 @@ to the Source Buffer must be of the supplied @type afterward. - + Returns the current append window end time. Any segment processed that starts after this value will be ignored. @@ -738,7 +738,7 @@ after this value will be ignored. - + Returns the current append window start time. Any segment processed that ends earlier than this value will be ignored. @@ -755,7 +755,7 @@ earlier than this value will be ignored. - + Returns a sequence of #GstMediaSourceRange values representing which segments of @self are buffered in memory. @@ -774,7 +774,7 @@ of @self are buffered in memory. - + Returns the current content type of @self. @@ -788,7 +788,7 @@ of @self are buffered in memory. - + [Specification](https://www.w3.org/TR/media-source-2/#dom-sourcebuffer-timestampoffset) @@ -802,7 +802,7 @@ of @self are buffered in memory. - + [Specification](https://www.w3.org/TR/media-source-2/#dom-sourcebuffer-updating) @@ -840,7 +840,7 @@ of @self are buffered in memory. - + Changes the Append Mode of @self. This influences what timestamps will be assigned to media processed by this Source Buffer. In Segment mode, the timestamps in each segment determine the position of each sample after it @@ -908,7 +908,7 @@ ignored. - + Attempt to set the timestamp offset of @self. Any media processed after this value is set will have this value added to its start time. @@ -929,7 +929,7 @@ value is set will have this value added to its start time. - + Affects how timestamps of processed media segments are interpreted. In %GST_SOURCE_BUFFER_APPEND_MODE_SEGMENTS, the start timestamp of a processed media segment is used directly along with @@ -941,21 +941,21 @@ most recently appended segment. [Specification](https://www.w3.org/TR/media-source-2/#dom-sourcebuffer-mode) - + Any segments processed which have a start time greater than this value will be ignored by this Source Buffer. [Specification](https://www.w3.org/TR/media-source-2/#dom-sourcebuffer-appendwindowend) - + Any segments processed which end before this value will be ignored by this Source Buffer. [Specification](https://www.w3.org/TR/media-source-2/#dom-sourcebuffer-appendwindowstart) - + The set of Time Intervals that have been loaded into the current Source Buffer @@ -964,18 +964,18 @@ Buffer - + The MIME content-type of the data stream - + The next media segment appended to the current Source Buffer will have its start timestamp increased by this amount. [Specification](https://www.w3.org/TR/media-source-2/#dom-sourcebuffer-timestampoffset) - + Whether the current source buffer is still asynchronously processing previously issued commands. @@ -1030,9 +1030,9 @@ gst_source_buffer_append_buffer(). [Specification](https://www.w3.org/TR/media-source-2/#dom-appendmode) - + - + @@ -1061,7 +1061,7 @@ It is used by #GstMediaSource to provide direct access to its child informing clients which of the Source Buffers are active through #GstMediaSource:active-source-buffers. - + [Specification](https://www.w3.org/TR/media-source-2/#dom-sourcebufferlist-length) @@ -1096,7 +1096,7 @@ the highest index in the list, it will return `NULL`. - + The number of #GstSourceBuffer<!-- -->s contained by this structure [Specification](https://www.w3.org/TR/media-source-2/#dom-sourcebufferlist-length) diff --git a/girs/GstNet-1.0.gir b/girs/GstNet-1.0.gir index 6e0605afcd..068d5ae9e9 100644 --- a/girs/GstNet-1.0.gir +++ b/girs/GstNet-1.0.gir @@ -186,10 +186,10 @@ clock. - + - + @@ -198,16 +198,16 @@ clock. - + - + - + - + @@ -233,7 +233,7 @@ clock. - + @@ -421,19 +421,19 @@ The #GstNetTimeProvider typically wraps the clock used by a #GstPipeline. - + - + - + - + @@ -459,7 +459,7 @@ The #GstNetTimeProvider typically wraps the clock used by a #GstPipeline. - + @@ -602,16 +602,16 @@ gst_clock_is_synced(). - + - + - + @@ -639,7 +639,7 @@ gst_clock_is_synced(). - + diff --git a/girs/GstPbutils-1.0.gir b/girs/GstPbutils-1.0.gir index 15827df078..d6b8c2f4ac 100644 --- a/girs/GstPbutils-1.0.gir +++ b/girs/GstPbutils-1.0.gir @@ -109,10 +109,10 @@ new frame. - + - + @@ -185,39 +185,39 @@ 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 @@ -487,7 +487,7 @@ pending URIS (if any). - + The duration (in nanoseconds) after which the discovery of an individual URI will timeout. @@ -495,7 +495,7 @@ If the discovery of a URI times out, the %GST_DISCOVERER_TIMEOUT will be set on the result flags. - + @@ -1115,47 +1115,47 @@ 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 @@ -1596,7 +1596,7 @@ NULL. See gst_encoding_profile_get_restriction() for more details. - + @@ -1687,7 +1687,7 @@ the list of contained #GstEncodingProfile. - + @@ -1791,7 +1791,7 @@ later during the encoding. - + The properties that are going to be set on the underlying element @@ -2012,7 +2012,7 @@ during the encoding - + This allows setting the muxing/encoding element properties. **Set properties generically** @@ -2226,7 +2226,7 @@ serialization format". - + A #GstStructure defining the properties to be set to the element the profile represents. @@ -2241,7 +2241,7 @@ element-properties,row-mt=true, end-usage=vbr - + @@ -2582,7 +2582,7 @@ constance framerate. - + @@ -2690,7 +2690,7 @@ constance framerate. - + Opaque context structure for the plugin installation. Use the provided API to set details on it. @@ -2874,49 +2874,49 @@ gst_install_plugins_sync(), and also the result code passed to the 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 occurred 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 occurred 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 @@ -2969,33 +2969,33 @@ Actual releases have 0, GIT versions have 1, prerelease versions have 2-... Flags that are returned by gst_pb_utils_get_caps_description_flags() and describe the format of the caps. - + Caps describe a container format. - + Caps describe an audio format, or a container format that can store audio. - + Caps describe an video format, or a container format that can store video. - + Caps describe an image format, or a container format that can store image. - + Caps describe an subtitle format, or a container format that can store subtitles. - + Container format is a tags container. - + Container format can store any kind of stream type. - + Caps describe a metadata format, or a container format that can store metadata. diff --git a/girs/GstPlay-1.0.gir b/girs/GstPlay-1.0.gir index 0f1dabaf68..2b7bc5451d 100644 --- a/girs/GstPlay-1.0.gir +++ b/girs/GstPlay-1.0.gir @@ -585,7 +585,7 @@ matching #GstPlayVideoInfo. - + Retrieve the current value of audio-video-offset property @@ -635,7 +635,7 @@ gst_structure_free() after usage or gst_play_set_config(). - + A Function to get current audio #GstPlayAudioInfo instance. @@ -651,7 +651,7 @@ The caller should free it with g_object_unref() - + A Function to get current subtitle #GstPlaySubtitleInfo instance. @@ -667,7 +667,7 @@ The caller should free it with g_object_unref() - + A Function to get current video #GstPlayVideoInfo instance. @@ -698,7 +698,7 @@ The caller should free it with g_object_unref() - + Retrieves the duration of the media stream that self represents. @@ -713,7 +713,7 @@ nanoseconds. - + A Function to get the current media info #GstPlayMediaInfo instance. @@ -782,7 +782,7 @@ fill memory. To avoid that, the bus has to be set "flushing". - + %TRUE if the currently-playing stream is muted. @@ -795,7 +795,7 @@ fill memory. To avoid that, the bus has to be set "flushing". - + The internal playbin instance. @@ -810,7 +810,7 @@ The caller should free it with g_object_unref() - + the absolute position time, in nanoseconds, of the @@ -824,7 +824,7 @@ currently-playing stream. - + current playback rate @@ -852,7 +852,7 @@ currently-playing stream. - + Retrieve the current value of subtitle-video-offset property @@ -866,7 +866,7 @@ currently-playing stream. - + Gets the URI of the currently-playing stream. @@ -908,7 +908,7 @@ Currently supported settings are: - + Returns the current volume level, as a percentage between 0 and 1. @@ -1017,7 +1017,7 @@ Sets the audio track @stream_index. - + Sets audio-video-offset property by value of @offset @@ -1118,7 +1118,7 @@ value. - + %TRUE if the currently-playing stream should be muted. @@ -1135,7 +1135,7 @@ value. - + Playback at specified rate @@ -1207,7 +1207,7 @@ rendered. - + Sets subtitle-video-offset property by value of @offset @@ -1224,7 +1224,7 @@ rendered. - + Sets the next URI to play. @@ -1313,7 +1313,7 @@ Sets the video track @stream_index. - + Sets the volume level of the stream as a percentage between 0 and 1. @@ -1344,55 +1344,55 @@ in the stream. - + - + - + - + - + - + - + - + - + - + - + - + - + - + - + - + @@ -1465,24 +1465,24 @@ in the stream. - + - + - + hue or color balance. - + brightness or black level. - + color saturation or chroma gain. - + contrast or luma gain. @@ -1502,7 +1502,7 @@ gain. - + generic error. @@ -1751,7 +1751,7 @@ matching #GstPlayVideoInfo. - + @@ -1760,43 +1760,43 @@ matching #GstPlayVideoInfo. Types of messages that will be posted on the play API bus. See also #gst_play_get_message_bus() - + Source element was initalized for set URI - + Sink position changed - + Duration of stream changed - + State changed, see #GstPlayState - + Pipeline is in buffering state, message contains the percentage value of the decoding buffer - + Sink has received EOS - + Message contains an error - + Message contains an error - + Video sink received format in different dimensions than before - + A media-info property has changed, message contains current #GstPlayMediaInfo - + The volume of the audio ouput has changed - + Audio muting flag has been toggled - + Any pending seeking operation has been completed @@ -2067,7 +2067,7 @@ it on the created adapter object. - + The #GstPlay owning this signal adapter. @@ -2080,7 +2080,7 @@ it on the created adapter object. - + @@ -2224,7 +2224,7 @@ it on the created adapter object. - + @@ -2246,16 +2246,16 @@ it on the created adapter object. - + the play is stopped. - + the play is buffering. - + the play is paused. - + the play is currently playing a stream. @@ -2351,7 +2351,7 @@ of the given @info (ex: "audio", "video", "subtitle") - + @@ -2371,7 +2371,7 @@ of the given @info (ex: "audio", "video", "subtitle") - + @@ -2471,7 +2471,7 @@ of the given @info (ex: "audio", "video", "subtitle") - + @@ -2549,7 +2549,7 @@ for details. - + The currently set, platform specific window @@ -2601,7 +2601,7 @@ do not support subwindows. - + Sets the platform specific window handle into which the video should be rendered @@ -2622,11 +2622,11 @@ should be rendered - + - + diff --git a/girs/GstPlayer-1.0.gir b/girs/GstPlayer-1.0.gir index 92efe7e756..d881796ce7 100644 --- a/girs/GstPlayer-1.0.gir +++ b/girs/GstPlayer-1.0.gir @@ -533,7 +533,7 @@ matching #GstPlayerVideoInfo. - + Retrieve the current value of audio-video-offset property @@ -583,7 +583,7 @@ gst_structure_free() after usage or gst_player_set_config(). - + A Function to get current audio #GstPlayerAudioInfo instance. @@ -599,7 +599,7 @@ The caller should free it with g_object_unref() - + A Function to get current subtitle #GstPlayerSubtitleInfo instance. @@ -615,7 +615,7 @@ The caller should free it with g_object_unref() - + A Function to get current video #GstPlayerVideoInfo instance. @@ -646,7 +646,7 @@ The caller should free it with g_object_unref() - + Retrieves the duration of the media stream that self represents. @@ -661,7 +661,7 @@ nanoseconds. - + A Function to get the current media info #GstPlayerMediaInfo instance. @@ -705,7 +705,7 @@ The caller should free it with g_object_unref() - + %TRUE if the currently-playing stream is muted. @@ -718,7 +718,7 @@ The caller should free it with g_object_unref() - + The internal playbin instance. @@ -733,7 +733,7 @@ The caller should free it with g_object_unref() - + the absolute position time, in nanoseconds, of the @@ -747,7 +747,7 @@ currently-playing stream. - + current playback rate @@ -775,7 +775,7 @@ currently-playing stream. - + Retrieve the current value of subtitle-video-offset property @@ -789,7 +789,7 @@ currently-playing stream. - + Gets the URI of the currently-playing stream. @@ -831,7 +831,7 @@ Currently supported settings are: - + Returns the current volume level, as a percentage between 0 and 1. @@ -940,7 +940,7 @@ Sets the audio track @stream_idex. - + Sets audio-video-offset property by value of @offset @@ -1041,7 +1041,7 @@ value. - + %TRUE if the currently-playing stream should be muted. @@ -1058,7 +1058,7 @@ value. - + Playback at specified rate @@ -1130,7 +1130,7 @@ rendered. - + Sets subtitle-video-offset property by value of @offset @@ -1147,7 +1147,7 @@ rendered. - + Sets the next URI to play. @@ -1236,7 +1236,7 @@ Sets the video track @stream_index. - + Sets the volume level of the stream as a percentage between 0 and 1. This volume is a linear factor. For showing the volume in a GUI it @@ -1271,58 +1271,58 @@ in the stream. - + - + - + - + - + - + - + - + - + - + - + - + - + - + - + - + @@ -1514,24 +1514,24 @@ unknown. - + - + - + hue or color balance. - + brightness or black level. - + color saturation or chroma gain. - + contrast or luma gain. @@ -1551,7 +1551,7 @@ gain. - + generic error. @@ -1596,7 +1596,7 @@ or the thread default one if %NULL is used. See gst_player_new(). - + @@ -1825,7 +1825,7 @@ matching #GstPlayerVideoInfo. - + @@ -1910,16 +1910,16 @@ matching #GstPlayerVideoInfo. - + the player is stopped. - + the player is buffering. - + the player is paused. - + the player is currently playing a stream. @@ -2014,7 +2014,7 @@ of the given @info (ex: "audio", "video", "subtitle") - + @@ -2034,7 +2034,7 @@ of the given @info (ex: "audio", "video", "subtitle") - + @@ -2136,7 +2136,7 @@ unknown. - + @@ -2214,7 +2214,7 @@ for details. - + The currently set, platform specific window @@ -2266,7 +2266,7 @@ do not support subwindows. - + Sets the platform specific window handle into which the video should be rendered @@ -2287,11 +2287,11 @@ should be rendered - + - + diff --git a/girs/GstRtp-1.0.gir b/girs/GstRtp-1.0.gir index c8d35af735..6b1f52933a 100644 --- a/girs/GstRtp-1.0.gir +++ b/girs/GstRtp-1.0.gir @@ -318,47 +318,47 @@ gst_rtcp_buffer_validate_reduced(). 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 @@ -1959,113 +1959,113 @@ is set as zero, @lost_packets or @dup_packets will be zero. 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 - + H.323 callable address - + Application Specific Identifier (RFC6776) - + Reporting Group Identifier (RFC8861) - + RtpStreamId SDES item (RFC8852). - + RepairedRtpStreamId SDES item (RFC8852). - + CLUE CaptId (RFC8849) - + MID SDES item (RFC8843). Different RTCP packet types. - + Invalid type - + Sender report - + Receiver report - + Source description - + Goodbye - + Application defined - + Transport layer feedback. - + Payload-specific feedback. - + Extended report. Types of RTCP Extended Reports, those are defined in RFC 3611 and other RFCs according to the [IANA registry](https://www.iana.org/assignments/rtcp-xr-block-types/rtcp-xr-block-types.xhtml). - + Invalid XR Report Block - + Loss RLE Report Block - + Duplicate RLE Report Block - + Packet Receipt Times Report Block - + Receiver Reference Time Report Block - + Delay since the last Receiver Report - + Statistics Summary Report Block - + VoIP Metrics Report Block @@ -2295,7 +2295,7 @@ audio codec - + @@ -2335,7 +2335,7 @@ audio codec - + @@ -2374,6 +2374,7 @@ Be aware that in case gst_rtp_base_depayload_push_list() is used each buffer will see the same list of RTP header extensions. + custom event handling @@ -2388,6 +2389,7 @@ each buffer will see the same list of RTP header extensions. + signal the depayloader about packet loss @@ -2402,6 +2404,11 @@ each buffer will see the same list of RTP header extensions. + process incoming rtp packets. Subclass must implement either + this method or @process_rtp_packet to process incoming rtp packets. + If the child returns a buffer without a valid timestamp, the timestamp + of the provided buffer will be applied to the result buffer and the + buffer will be pushed. If this function returns %NULL, nothing is pushed. @@ -2416,6 +2423,14 @@ each buffer will see the same list of RTP header extensions. + Same as the process virtual function, but slightly more +efficient, since it is passed the rtp buffer structure that has already +been mapped (with GST_MAP_READ) by the base class and thus does not have +to be mapped again by the subclass. Can be used by the subclass to process +incoming rtp packets. If the subclass returns a buffer without a valid +timestamp, the timestamp of the input buffer will be applied to the result +buffer and the output buffer will be pushed out. If this function returns +%NULL, nothing is pushed out. Since: 1.6. @@ -2430,6 +2445,7 @@ each buffer will see the same list of RTP header extensions. + configure the depayloader @@ -2626,14 +2642,14 @@ the outgoing buffer when it didn't have a timestamp already. - + If enabled, the depayloader will automatically try to enable all the RTP header extensions provided in the sink caps, saving the application the need to handle these extensions manually using the GstRTPBaseDepayload::request-extension: signal. - + A list of already enabled RTP header extensions. This may be useful for finding out which extensions are already enabled (with add-extension signal) and picking a non-conflicting ID for a new extension that needs to be added on top of the existing ones. @@ -2646,14 +2662,14 @@ Dynamic updates of this property can be received by subscribing to its correspon "notify::extensions". - + Max seqnum reorder before the sender is assumed to have restarted. When max-reorder is set to 0 all reordered/duplicate packets are considered coming from a restarted sender. - + Add RTP source information found in RTP header as meta to output buffer. @@ -2748,6 +2764,7 @@ necessary attributes as required by the extension implementation. + configure the depayloader @@ -2764,6 +2781,11 @@ necessary attributes as required by the extension implementation. + process incoming rtp packets. Subclass must implement either + this method or @process_rtp_packet to process incoming rtp packets. + If the child returns a buffer without a valid timestamp, the timestamp + of the provided buffer will be applied to the result buffer and the + buffer will be pushed. If this function returns %NULL, nothing is pushed. @@ -2780,6 +2802,7 @@ necessary attributes as required by the extension implementation. + signal the depayloader about packet loss @@ -2796,6 +2819,7 @@ necessary attributes as required by the extension implementation. + custom event handling @@ -2812,6 +2836,14 @@ necessary attributes as required by the extension implementation. + Same as the process virtual function, but slightly more +efficient, since it is passed the rtp buffer structure that has already +been mapped (with GST_MAP_READ) by the base class and thus does not have +to be mapped again by the subclass. Can be used by the subclass to process +incoming rtp packets. If the subclass returns a buffer without a valid +timestamp, the timestamp of the input buffer will be applied to the result +buffer and the output buffer will be pushed out. If this function returns +%NULL, nothing is pushed out. Since: 1.6. @@ -2833,13 +2865,14 @@ necessary attributes as required by the extension implementation. - + Provides a base class for RTP payloaders + get desired caps @@ -2857,6 +2890,7 @@ necessary attributes as required by the extension implementation. + process data @@ -2871,6 +2905,7 @@ necessary attributes as required by the extension implementation. + custom query handling @@ -2888,6 +2923,7 @@ necessary attributes as required by the extension implementation. + configure the payloader @@ -2902,6 +2938,7 @@ necessary attributes as required by the extension implementation. + custom event handling on the sinkpad @@ -2916,6 +2953,7 @@ necessary attributes as required by the extension implementation. + custom event handling on the srcpad @@ -3152,14 +3190,14 @@ Variable arguments should be in the form field name, field type - + If enabled, the payloader will automatically try to enable all the RTP header extensions provided in the src caps, saving the application the need to handle these extensions manually using the GstRTPBasePayload::request-extension: signal. - + A list of already enabled RTP header extensions. This may be useful for finding out which extensions are already enabled (with add-extension signal) and picking a non-conflicting ID for a new extension that needs to be added on top of the existing ones. @@ -3172,22 +3210,22 @@ Dynamic updates of this property can be received by subscribing to its correspon "notify::extensions". - + - + Minimum duration of the packet data in ns (can't go above MTU) - + - + Make the payloader timestamp packets according to the Rate-Control=no behaviour specified in the ONVIF replay spec. - + 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 @@ -3205,14 +3243,14 @@ 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) - + Make the RTP packets' timestamps be scaled with the segment's rate (corresponding to RTSP speed parameter). Disabling this property means the timestamps will not be affected by the set delivery speed (RTSP speed). @@ -3223,18 +3261,18 @@ video. This is achieved by the client setting RTSP Speed to 2 while the server has this property disabled. - + - + - + Enable writing the CSRC field in allocated RTP header based on RTP source information found in the input buffer's #GstRTPSourceMeta. - + @@ -3254,10 +3292,10 @@ the last processed buffer and current state of the stream being payloaded: * `timestamp-offset` :#G_TYPE_UINT, The current offset added to the timestamp - + - + @@ -3380,6 +3418,7 @@ necessary attributes as required by the extension implementation. + get desired caps @@ -3399,6 +3438,7 @@ necessary attributes as required by the extension implementation. + configure the payloader @@ -3415,6 +3455,7 @@ necessary attributes as required by the extension implementation. + process data @@ -3431,6 +3472,7 @@ necessary attributes as required by the extension implementation. + custom event handling on the sinkpad @@ -3447,6 +3489,7 @@ necessary attributes as required by the extension implementation. + custom event handling on the srcpad @@ -3463,6 +3506,7 @@ necessary attributes as required by the extension implementation. + custom query handling @@ -3487,7 +3531,7 @@ necessary attributes as required by the extension implementation. - + @@ -4553,28 +4597,28 @@ 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 @@ -4618,6 +4662,7 @@ information given in the input @buffer. + retrieve the supported flags the flags supported by this instance of @ext @@ -4664,6 +4709,8 @@ information given in the input @buffer. + set the necessary attributes that may be signaled e.g. with + an SDP. @@ -5143,6 +5190,7 @@ from gst_rtp_header_extension_get_max_size(). + retrieve the supported flags @@ -5158,6 +5206,13 @@ from gst_rtp_header_extension_get_max_size(). + retrieve the maximum size for this extension based on the + information available from input_meta. Implementations should attempt + to provide as accurate information as possible as the returned value + will be used to control the amount of possible data in the payload. + Implementations must return the maximum as the allocated size for + writing the extension will be at least the size of the returned value. + Return the amount of data read or <0 on failure. @@ -5177,6 +5232,8 @@ from gst_rtp_header_extension_get_max_size(). + write into @data the information for this extension. Various + information is provided to help writing extensions in particular cases. @@ -5215,6 +5272,8 @@ from gst_rtp_header_extension_get_max_size(). + read from a rtp payloaded buffer and extract the extension + information, optionally adding some meta onto the output buffer. @@ -5249,6 +5308,8 @@ from gst_rtp_header_extension_get_max_size(). + read any information from sink caps that the header + extension needs for its function. @@ -5268,6 +5329,8 @@ from gst_rtp_header_extension_get_max_size(). + update depayloader non-RTP (depayloaded) caps with + the information parsed from RTP header. @@ -5287,6 +5350,8 @@ from gst_rtp_header_extension_get_max_size(). + set the necessary attributes that may be signaled e.g. with + an SDP. @@ -5306,6 +5371,8 @@ from gst_rtp_header_extension_get_max_size(). + write the necessary caps field/s for the configured + attributes e.g. as signalled with SDP. @@ -5350,34 +5417,34 @@ from gst_rtp_header_extension_get_max_size(). Direction to which to apply the RTP Header Extension - + Neither send nor receive RTP Header Extensions - + Only send RTP Header Extensions @GST_RTP_HEADER_EXTENSION_DIRECTION_RECVONLY: Only receive RTP Header Extensions - + - + Send and receive RTP Header Extensions ext - + RTP header extension direction is inherited from the stream Flags that apply to a RTP Audio/Video header extension. - + The one byte rtp extension header. 1-16 data bytes per extension with a maximum of 14 extension ids in total. - + The two byte rtp extension header. 256 data bytes per extension with a maximum of 255 (or 256 including appbits) extensions in total. @@ -5396,82 +5463,82 @@ 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) @@ -5550,19 +5617,19 @@ types specified with @payload_type. 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) diff --git a/girs/GstRtsp-1.0.gir b/girs/GstRtsp-1.0.gir index 9bfcb58f3b..290f33325d 100644 --- a/girs/GstRtsp-1.0.gir +++ b/girs/GstRtsp-1.0.gir @@ -34,13 +34,13 @@ and/or use gtk-doc annotations. --> Authentication methods, ordered by strength - + no authentication - + basic authentication - + digest authentication @@ -78,7 +78,7 @@ and/or use gtk-doc annotations. --> - + This object manages the RTSP connection to the server. It provides function to receive and send bytes and messages. @@ -1302,10 +1302,10 @@ read from @socket which should be used before starting to read new data. The possible events for the connection. - + connection is readable - + connection is writable @@ -1817,217 +1817,217 @@ read from @socket which should be used before starting to read new data. 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 @@ -2773,46 +2773,46 @@ gst_rtsp_message_init_data() on stack allocated #GstRTSPMessage structures. 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). @@ -2832,40 +2832,40 @@ gst_rtsp_message_init_data() on stack allocated #GstRTSPMessage structures. 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) @@ -2976,202 +2976,202 @@ UTC times will be converted to nanoseconds since 1900. 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 - + RTSP request is successful, but was redirected. - + some unspecified error occurred - + invalid arguments were provided to a function - + an operation was canceled - + no memory was available for the operation - + a host resolve error occurred - + function not implemented - + a system error occurred, errno contains more details - + a parsing error occurred - + windows networking could not start - + windows networking stack has wrong version - + end-of-file was reached - + a network problem occurred, h_errno contains more details - + the host is not an IP host - + a timeout occurred - + 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 - + - + - + - + - + - + - + - + - + - + - + - + RTSP request is temporarily redirected - + RTSP request is permanently redirected - + - + - + - + - + - + - + - + - + - + - + - + - + - + - + - + - + - + - + - + - + - + - + - + - + - + - + - + - + - + - + - + - + - + - + @@ -3234,31 +3234,31 @@ UTC times will be converted to nanoseconds since 1900. 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 @@ -3662,16 +3662,16 @@ with gst_rtsp_url_free(). The supported RTSP versions. - + unknown/invalid version - + version 1.0 - + version 1.1. - + version 2.0. @@ -3689,7 +3689,7 @@ with gst_rtsp_url_free(). - + Opaque RTSP watch object that can be used for asynchronous RTSP operations. @@ -4004,6 +4004,7 @@ communication. Free with gst_rtsp_watch_unref () after usage. Callback functions from a #GstRTSPWatch. + callback when a message was received @@ -4023,6 +4024,7 @@ communication. Free with gst_rtsp_watch_unref () after usage. + callback when a message was sent @@ -4042,6 +4044,7 @@ communication. Free with gst_rtsp_watch_unref () after usage. + callback when the connection is closed @@ -4058,6 +4061,7 @@ communication. Free with gst_rtsp_watch_unref () after usage. + callback when an error occurred @@ -4077,6 +4081,8 @@ communication. Free with gst_rtsp_watch_unref () after usage. + a client started a tunneled connection. The tunnelid of the + connection must be saved. @@ -4093,6 +4099,9 @@ communication. Free with gst_rtsp_watch_unref () after usage. + a client finished a tunneled connection. In this callback + you usually pair the tunnelid of this connection with the saved one using + gst_rtsp_connection_do_tunnel(). @@ -4109,6 +4118,8 @@ communication. Free with gst_rtsp_watch_unref () after usage. + callback when an error occurred with more information than + the @error callback. @@ -4134,6 +4145,7 @@ communication. Free with gst_rtsp_watch_unref () after usage. + callback when the post connection of a tunnel is closed. @@ -4150,6 +4162,9 @@ communication. Free with gst_rtsp_watch_unref () after usage. + callback when an HTTP response to the GET request + is about to be sent for a tunneled connection. The response can be + modified in the callback. Since: 1.4. diff --git a/girs/GstRtspServer-1.0.gir b/girs/GstRtspServer-1.0.gir index 7aa03b02d1..dc42527585 100644 --- a/girs/GstRtspServer-1.0.gir +++ b/girs/GstRtspServer-1.0.gir @@ -545,7 +545,7 @@ after use. - + @@ -632,6 +632,10 @@ g_free() after usage. + check the authentication of a client. The default implementation + checks if the authentication in the header matches one of the basic + authentication tokens. This function should set the authgroup field + in the context. @@ -646,6 +650,10 @@ g_free() after usage. + check if a resource can be accessed. this function should + call authenticate to authenticate the client when needed. The method + should also construct and send an appropriate response message on + error. @@ -1021,6 +1029,10 @@ no one else overrides it. + check the authentication of a client. The default implementation + checks if the authentication in the header matches one of the basic + authentication tokens. This function should set the authgroup field + in the context. @@ -1037,6 +1049,10 @@ no one else overrides it. + check if a resource can be accessed. this function should + call authenticate to authenticate the client when needed. The method + should also construct and send an appropriate response message on + error. @@ -1099,7 +1115,7 @@ no one else overrides it. - + @@ -1137,6 +1153,8 @@ possibility to adjust the error code. + called to give the application the possibility to adjust + the range, seek flags, rate and rate-control. Since 1.18 @@ -1166,6 +1184,9 @@ possibility to adjust the error code. + called to give the implementation the possibility to + adjust the response to a play request, for example if extra headers were + parsed when #GstRTSPClientClass.adjust_play_mode was called. Since 1.18 @@ -1222,6 +1243,9 @@ possibility to adjust the error code. + called when the stream in media needs to be configured. + The default implementation will configure the blocksize on the payloader when + spcified in the request headers. @@ -1242,6 +1266,8 @@ possibility to adjust the error code. + called when the client transport needs to be + configured. @@ -1259,6 +1285,7 @@ possibility to adjust the error code. + called when the SDP needs to be created for media. @@ -1335,6 +1362,7 @@ possibility to adjust the error code. + called to create path from uri. @@ -1377,6 +1405,8 @@ possibility to adjust the error code. + get parameters. This function should also initialize the + RTSP response(ctx->response) via a call to gst_rtsp_message_init_response() @@ -1391,6 +1421,8 @@ possibility to adjust the error code. + set parameters. This function should also initialize the + RTSP response(ctx->response) via a call to gst_rtsp_message_init_response() @@ -1646,6 +1678,8 @@ possibility to adjust the error code. + called when a response to the GET request is about to + be sent for a tunneled connection. The response can be modified. Since: 1.4 @@ -1742,7 +1776,7 @@ The connection object returned remains valid until the client is freed. - + Get the #GstRTSPMountPoints object that @client uses to manage its sessions. @@ -1756,7 +1790,7 @@ The connection object returned remains valid until the client is freed. - + Get the #GstRTSPSessionPool object that @client uses to manage its sessions. @@ -1941,7 +1975,7 @@ limit with response status 413 Request Entity Too Large - + Set @mounts as the mount points for @client which it will use to map urls to media streams. These mount points are usually inherited from the server that created the client but can be overriden later. @@ -2026,7 +2060,7 @@ but not both at the same time. - + Set @pool as the sessionpool for @client which it will use to find or allocate sessions. the sessionpool is usually inherited from the server that created the client but can be overridden later. @@ -2062,16 +2096,16 @@ that created the client but can be overridden later. - + - + - + - + @@ -2394,6 +2428,7 @@ that created the client but can be overridden later. + called when the SDP needs to be created for media. @@ -2410,6 +2445,9 @@ that created the client but can be overridden later. + called when the stream in media needs to be configured. + The default implementation will configure the blocksize on the payloader when + spcified in the request headers. @@ -2432,6 +2470,8 @@ that created the client but can be overridden later. + called when the client transport needs to be + configured. @@ -2451,6 +2491,8 @@ that created the client but can be overridden later. + set parameters. This function should also initialize the + RTSP response(ctx->response) via a call to gst_rtsp_message_init_response() @@ -2467,6 +2509,8 @@ that created the client but can be overridden later. + get parameters. This function should also initialize the + RTSP response(ctx->response) via a call to gst_rtsp_message_init_response() @@ -2483,6 +2527,7 @@ that created the client but can be overridden later. + called to create path from uri. @@ -2499,6 +2544,8 @@ that created the client but can be overridden later. + called to give the application the possibility to adjust + the range, seek flags, rate and rate-control. Since 1.18 @@ -2530,6 +2577,9 @@ that created the client but can be overridden later. + called to give the implementation the possibility to + adjust the response to a play request, for example if extra headers were + parsed when #GstRTSPClientClass.adjust_play_mode was called. Since 1.18 @@ -2719,6 +2769,8 @@ that created the client but can be overridden later. + called when a response to the GET request is about to + be sent for a tunneled connection. The response can be modified. Since: 1.4 @@ -3018,7 +3070,7 @@ that created the client but can be overridden later. - + @@ -3287,6 +3339,7 @@ Ownership is taken of @element. + convert a range to the given unit @@ -3315,6 +3368,7 @@ Ownership is taken of @element. + handle a message @@ -3410,6 +3464,7 @@ such as the duration. + query the current position in the pipeline @@ -3424,6 +3479,7 @@ such as the duration. + query when playback will stop @@ -3697,7 +3753,7 @@ g_object_unref() after usage. - + Get the kernel UDP buffer size. @@ -3711,7 +3767,7 @@ g_object_unref() after usage. - + Get the clock that is used by the pipeline in @media. @media must be prepared before this method returns a valid clock object. @@ -3739,7 +3795,7 @@ g_object_unref() after usage. - + Get the configured DSCP QoS of attached media. @@ -3753,7 +3809,7 @@ g_object_unref() after usage. - + Get the element that was used when constructing @media. @@ -3767,7 +3823,7 @@ g_object_unref() after usage. - + Get ensure-keyunit-on-start flag. @@ -3781,7 +3837,7 @@ g_object_unref() after usage. - + Get ensure-keyunit-on-start-timeout time. @@ -3795,7 +3851,7 @@ g_object_unref() after usage. - + Get the latency that is used for receiving media. @@ -3809,7 +3865,7 @@ g_object_unref() after usage. - + Get the the maximum time-to-live value of outgoing multicast packets. @@ -3852,7 +3908,7 @@ g_free() after usage. - + Get the allowed profiles of @media. @@ -3866,7 +3922,7 @@ g_free() after usage. - + Get the allowed protocols of @media. @@ -4002,7 +4058,7 @@ until @media is prepared or in error. - + Get how @media will be suspended. @@ -4016,7 +4072,7 @@ until @media is prepared or in error. - + Get the #GstNetTimeProvider for the clock used by @media. The time provider will listen on @address and @port for client time requests. @@ -4039,7 +4095,7 @@ will listen on @address and @port for client time requests. - + Check if the pipeline for @media can be used for PLAY or RECORD methods. @@ -4359,7 +4415,7 @@ and > 0 to indicate the longest duration between any two random access points - + Decide whether the multicast socket should be bound to a multicast address or INADDR_ANY. @@ -4377,7 +4433,7 @@ INADDR_ANY. - + Set the kernel UDP buffer size. @@ -4394,7 +4450,7 @@ INADDR_ANY. - + Configure the clock used for the media. @@ -4426,7 +4482,7 @@ INADDR_ANY. - + Configure the dscp qos of attached streams to @dscp_qos. @@ -4443,7 +4499,7 @@ INADDR_ANY. - + Set whether or not a keyunit should be ensured when a client connects. It will also configure the streams to drop delta units to ensure that they start on a keyunit. @@ -4464,7 +4520,7 @@ Note that this will only affect non-shared medias for now. - + Sets the maximum allowed time before the first keyunit is considered expired. @@ -4485,7 +4541,7 @@ enabled. - + Set or unset if an EOS event will be sent to the pipeline for @media before it is unprepared. @@ -4503,7 +4559,7 @@ it is unprepared. - + Configure the latency used for receiving media. @@ -4520,7 +4576,7 @@ it is unprepared. - + Set the maximum time-to-live value of outgoing multicast packets. @@ -4589,7 +4645,7 @@ it is unprepared. - + Configure the allowed lower transport for @media. @@ -4606,7 +4662,7 @@ it is unprepared. - + Configure the allowed lower transport for @media. @@ -4673,7 +4729,7 @@ in the ONVIF replay spec. - + Set or unset if the pipeline for @media can be reused after the pipeline has been unprepared. @@ -4691,7 +4747,7 @@ been unprepared. - + Set or unset if the pipeline for @media can be shared will multiple clients. When @shared is %TRUE, client requests for this media will share the media pipeline. @@ -4737,7 +4793,7 @@ a #GPtrArray of #GstRTSPStreamTransport pointers - + Set or unset if the pipeline for @media should be stopped when a client disconnects without sending TEARDOWN. @@ -4755,7 +4811,7 @@ client disconnects without sending TEARDOWN. - + Control how @ media will be suspended after the SDP has been generated and after a PAUSE request has been performed. @@ -4775,7 +4831,7 @@ Media must be unprepared when setting the suspend mode. - + Sets if the media pipeline can work in PLAY or RECORD mode @@ -4912,22 +4968,22 @@ when the media was not in the suspended state. - + - + - + - + - + - + Whether or not a keyunit should be ensured when a client connects. It will also configure the streams to drop delta units to ensure that they start on a keyunit. @@ -4935,7 +4991,7 @@ on a keyunit. Note that this will only affect non-shared medias for now. - + The maximum allowed time before the first keyunit is considered expired. @@ -4943,37 +4999,37 @@ Note that this will only have an effect when ensure-keyunit-on-start is enabled. - + - + - + - + - + - + - + - + - + - + - + @@ -5058,6 +5114,7 @@ enabled. + handle a message @@ -5074,6 +5131,9 @@ enabled. + the default implementation adds all elements and sets the + pipeline's state to GST_STATE_PAUSED (or GST_STATE_PLAYING + in case of NO_PREROLL elements). @@ -5094,6 +5154,8 @@ enabled. + the default implementation sets the pipeline's state + to GST_STATE_NULL and removes all elements. @@ -5109,6 +5171,9 @@ enabled. + the default implementation sets the pipeline's state to + GST_STATE_NULL GST_STATE_PAUSED depending on the selected + suspend mode. @@ -5124,6 +5189,9 @@ enabled. + the default implementation reverts the suspend operation. + The pipeline will be prerolled again if it's state was + set to GST_STATE_NULL in suspend. @@ -5139,6 +5207,7 @@ enabled. + convert a range to the given unit @@ -5158,6 +5227,7 @@ enabled. + query the current position in the pipeline @@ -5174,6 +5244,7 @@ enabled. + query when playback will stop @@ -5369,6 +5440,8 @@ can contain multiple streams like audio and video. + configure the media created with @construct. The default + implementation will configure the 'shared' property of the media. @@ -5434,6 +5507,8 @@ launch parameter. + create a new pipeline or re-use an existing one and + add the #GstRTSPMedia's element created by @construct to the pipeline. @@ -5448,6 +5523,9 @@ launch parameter. + convert @url to a key for caching shared #GstRTSPMedia objects. + The default implementation of this function will use the complete URL + including the query parameters to return a key. @@ -5462,6 +5540,7 @@ launch parameter. + signal emitted when a media should be configured @@ -5476,6 +5555,7 @@ launch parameter. + signal emitted when a media was constructed @@ -5599,7 +5679,7 @@ usage. - + Get the kernel UDP buffer size. @@ -5613,7 +5693,7 @@ usage. - + Returns the clock that is going to be used by the pipelines of all medias created from this factory. @@ -5640,7 +5720,7 @@ of all medias created from this factory. - + Get the configured media DSCP QoS. @@ -5654,7 +5734,7 @@ of all medias created from this factory. - + Get ensure-keyunit-on-start flag. @@ -5668,7 +5748,7 @@ of all medias created from this factory. - + Get ensure-keyunit-on-start-timeout time. @@ -5682,7 +5762,7 @@ of all medias created from this factory. - + Get the latency that is used for receiving media @@ -5696,7 +5776,7 @@ of all medias created from this factory. - + Get the gst_parse_launch() pipeline description that will be used in the default prepare vmethod. @@ -5712,7 +5792,7 @@ usage. - + Get the the maximum time-to-live value of outgoing multicast packets. @@ -5769,7 +5849,7 @@ usage. - + Get the allowed profiles of @factory. @@ -5783,7 +5863,7 @@ usage. - + Get the allowed protocols of @factory. @@ -5825,7 +5905,7 @@ usage. - + Get how media created from this factory will be suspended. @@ -5839,7 +5919,7 @@ usage. - + Get if media created from this factory can be used for PLAY or RECORD methods. @@ -5939,7 +6019,7 @@ pipeline before shutdown. - + Decide whether the multicast socket should be bound to a multicast address or INADDR_ANY. @@ -5957,7 +6037,7 @@ INADDR_ANY. - + Set the kernel UDP buffer size. @@ -5974,7 +6054,7 @@ INADDR_ANY. - + Configures a specific clock to be used by the pipelines of all medias created from this factory. @@ -6008,7 +6088,7 @@ receiving media - + Configure the media dscp qos to @dscp_qos. @@ -6025,7 +6105,7 @@ receiving media - + Decide whether the created media should send and receive RTCP @@ -6042,7 +6122,7 @@ receiving media - + If media from this factory should ensure a key unit when a client connects. @@ -6059,7 +6139,7 @@ receiving media - + Configures medias from this factory to consider keyunits older than timeout to be expired. Expired keyunits will be discarded. @@ -6077,7 +6157,7 @@ to be expired. Expired keyunits will be discarded. - + Configure if media created from this factory will have an EOS sent to the pipeline before shutdown. @@ -6095,7 +6175,7 @@ pipeline before shutdown. - + Configure the latency used for receiving media @@ -6112,7 +6192,7 @@ pipeline before shutdown. - + The gst_parse_launch() line to use for constructing the pipeline in the default prepare vmethod. @@ -6137,7 +6217,7 @@ etc.. Each of the payloaders will result in a stream. - + Set the maximum time-to-live value of outgoing multicast packets. @@ -6208,7 +6288,7 @@ may of course do something different) - + Configure the allowed profiles for @factory. @@ -6225,7 +6305,7 @@ may of course do something different) - + Configure the allowed lower transport for @factory. @@ -6276,7 +6356,7 @@ may of course do something different) - + Configure if media created from this factory can be shared between clients. @@ -6293,7 +6373,7 @@ may of course do something different) - + Configure if media created from this factory should be stopped when a client disconnects without sending TEARDOWN. @@ -6311,7 +6391,7 @@ when a client disconnects without sending TEARDOWN. - + Configure how media created from this factory will be suspended. @@ -6328,7 +6408,7 @@ when a client disconnects without sending TEARDOWN. - + Configure if this factory creates media for PLAY or RECORD modes. @@ -6345,23 +6425,23 @@ when a client disconnects without sending TEARDOWN. - + - + - + - + - + Whether the created media should send and receive RTCP - + If media from this factory should ensure a key unit when a client connects. This property will ensure that the stream always starts on a key unit @@ -6370,7 +6450,7 @@ instead of a delta unit which the client would not be able to decode. Note that this will only affect non-shared medias for now. - + Timeout in milliseconds used to determine if a keyunit should be discarded when a client connects. @@ -6380,34 +6460,34 @@ the currently blocking keyframe will be used. This options is only relevant when ensure-keyunit-on-start is enabled. - + - + - + - + - + - + - + - + - + - + @@ -6449,6 +6529,9 @@ This options is only relevant when ensure-keyunit-on-start is enabled. + convert @url to a key for caching shared #GstRTSPMedia objects. + The default implementation of this function will use the complete URL + including the query parameters to return a key. @@ -6465,6 +6548,10 @@ This options is only relevant when ensure-keyunit-on-start is enabled. + Construct and return a #GstElement that is a #GstBin containing + the elements to use for streaming the media. The bin should contain + payloaders pay\%d for each stream. The default implementation of this + function returns the bin created from the launch parameter. @@ -6484,6 +6571,10 @@ This options is only relevant when ensure-keyunit-on-start is enabled. + the vmethod that will be called when the factory has to create the + #GstRTSPMedia for @url. The default implementation of this + function calls create_element to retrieve an element and then looks for + pay\%d to create the streams. @@ -6503,6 +6594,8 @@ This options is only relevant when ensure-keyunit-on-start is enabled. + create a new pipeline or re-use an existing one and + add the #GstRTSPMedia's element created by @construct to the pipeline. @@ -6519,6 +6612,8 @@ This options is only relevant when ensure-keyunit-on-start is enabled. + configure the media created with @construct. The default + implementation will configure the 'shared' property of the media. @@ -6535,6 +6630,7 @@ This options is only relevant when ensure-keyunit-on-start is enabled. + signal emitted when a media was constructed @@ -6551,6 +6647,7 @@ This options is only relevant when ensure-keyunit-on-start is enabled. + signal emitted when a media should be configured @@ -6572,7 +6669,7 @@ This options is only relevant when ensure-keyunit-on-start is enabled. - + @@ -6586,7 +6683,7 @@ This options is only relevant when ensure-keyunit-on-start is enabled. - + Get the URI that will provide media for this factory. @@ -6600,7 +6697,7 @@ This options is only relevant when ensure-keyunit-on-start is enabled. - + Set the URI of the resource that will be streamed by this factory. @@ -6617,10 +6714,10 @@ This options is only relevant when ensure-keyunit-on-start is enabled. - + - + @@ -6647,10 +6744,10 @@ This options is only relevant when ensure-keyunit-on-start is enabled. - + - + @@ -6843,6 +6940,7 @@ g_object_unref() after usage. + make a path from the given url. @@ -6867,7 +6965,7 @@ g_object_unref() after usage. - + @@ -7178,10 +7276,10 @@ prepare. - + - + @@ -7406,13 +7504,13 @@ remains valid for as long as @permissions is valid. Whether the clock and possibly RTP/clock offset should be published according to RFC7273. - + Publish nothing - + Publish the clock but not the offset - + Publish the clock and offset @@ -7498,6 +7596,7 @@ new connection on @socket or @server. + emitted when a new client connected. @@ -7512,6 +7611,10 @@ new connection on @socket or @server. + Create, configure a new GstRTSPClient + object that handles the new connection on @socket. The default + implementation will create a GstRTSPClient and will configure the + mount-points, auth, session-pool and thread-pool on the client. @@ -7637,7 +7740,7 @@ occurred. Free with g_source_unref () - + Get the address on which the server will accept connections. @@ -7666,7 +7769,7 @@ usage. - + The maximum amount of queued requests for the server. @@ -7680,7 +7783,7 @@ usage. - + Get the port number where the server was bound to. @@ -7694,7 +7797,7 @@ usage. - + Get the Content-Length limit of @server. @@ -7708,7 +7811,7 @@ usage. - + Get the #GstRTSPMountPoints used as the mount points of @server. @@ -7723,7 +7826,7 @@ usage. - + Get the service on which the server will accept connections. @@ -7737,7 +7840,7 @@ usage. - + Get the #GstRTSPSessionPool used as the session pool of @server. @@ -7767,7 +7870,7 @@ usage. - + Configure @server to accept connections on the given address. This function must be called before the server is bound. @@ -7803,7 +7906,7 @@ This function must be called before the server is bound. - + configure the maximum amount of requests that may be queued for the server. @@ -7823,7 +7926,7 @@ This function must be called before the server is bound. - + Define an appropriate request size limit and reject requests exceeding the limit. @@ -7841,7 +7944,7 @@ Configure @server to use the specified Content-Length limit. - + configure @mounts to be used as the mount points of @server. @@ -7858,7 +7961,7 @@ Configure @server to use the specified Content-Length limit. - + Configure @server to accept connections on the given service. @service should be a string containing the service name (see services(5)) or a string containing a port number between 1 and 65535. @@ -7883,7 +7986,7 @@ This function must be called before the server is bound. - + configure @pool to be used as the session pool of @server. @@ -7950,25 +8053,25 @@ that the HTTP server read from the socket while parsing the HTTP header. - + - + - + - + - + - + - + @@ -8000,6 +8103,10 @@ that the HTTP server read from the socket while parsing the HTTP header. + Create, configure a new GstRTSPClient + object that handles the new connection on @socket. The default + implementation will create a GstRTSPClient and will configure the + mount-points, auth, session-pool and thread-pool on the client. @@ -8013,6 +8120,7 @@ that the HTTP server read from the socket while parsing the HTTP header. + emitted when a new client connected. @@ -8066,7 +8174,7 @@ gst_rtsp_server_client_filter(). - + @@ -8203,7 +8311,7 @@ characters of @path. - + Get the sessionid of @session. @@ -8218,7 +8326,7 @@ The value remains valid as long as @session is alive. - + Get the timeout value of @session. @@ -8362,7 +8470,7 @@ Ownership is taken from @media. - + Configure @session for a timeout of @timeout seconds. The session will be cleaned up when there is no activity for @timeout seconds. @@ -8393,16 +8501,16 @@ cleaned up when there is no activity for @timeout seconds. - + - + - + - + @@ -8701,7 +8809,7 @@ matched characters is returned in @matched. - + @@ -8718,6 +8826,7 @@ usage. + make a new session object. @@ -8732,6 +8841,8 @@ usage. + create a new random session id. Subclasses can create + custom session ids and should not check if the session exists. @@ -8743,6 +8854,7 @@ usage. + a session was removed from the pool @@ -8859,7 +8971,7 @@ or %NULL when the session did not exist. g_object_unref() after usage. - + Get the maximum allowed number of sessions in @pool. 0 means an unlimited amount of sessions. @@ -8906,7 +9018,7 @@ amount of sessions. - + Configure the maximum allowed number of sessions in @pool to @max. A value of 0 means an unlimited amount of sessions. @@ -8924,7 +9036,7 @@ A value of 0 means an unlimited amount of sessions. - + @@ -8955,6 +9067,8 @@ A value of 0 means an unlimited amount of sessions. + create a new random session id. Subclasses can create + custom session ids and should not check if the session exists. @@ -8968,6 +9082,7 @@ A value of 0 means an unlimited amount of sessions. + make a new session object. @@ -8984,6 +9099,7 @@ A value of 0 means an unlimited amount of sessions. + a session was removed from the pool @@ -9058,10 +9174,10 @@ more sessions timed out. - + - + @@ -9236,7 +9352,7 @@ use gst_caps_unref() after usage. - + Get the control string to identify this stream. @@ -9382,7 +9498,7 @@ g_free() after usage. - + Get the allowed profiles of @stream. @@ -9396,7 +9512,7 @@ g_free() after usage. - + Get the allowed protocols of @stream. @@ -10250,7 +10366,7 @@ an RTSP connection. - + Set the control string in @stream. @@ -10336,7 +10452,7 @@ an RTSP connection. - + Configure the allowed profiles for @stream. @@ -10353,7 +10469,7 @@ an RTSP connection. - + Configure the allowed lower transport for @stream. @@ -10611,13 +10727,13 @@ be removed from @stream. - + - + - + @@ -10673,7 +10789,7 @@ be removed from @stream. - + @@ -11159,19 +11275,19 @@ gst_rtsp_stream_transport_filter(). - + The suspend mode of the media pipeline. A media pipeline is suspended right after creating the SDP and when the client performs a PAUSED request. - + Media is not suspended - + Media is PAUSED in suspend - + The media is set to NULL when suspended @@ -11258,6 +11374,8 @@ structures. + configure a thread object. this vmethod is called when + a new thread has been created and should be configured. @@ -11298,6 +11416,7 @@ gst_rtsp_thread_stop() after usage + called from the thread when it is entered @@ -11312,6 +11431,7 @@ gst_rtsp_thread_stop() after usage + called from the thread when it is left @@ -11325,7 +11445,7 @@ gst_rtsp_thread_stop() after usage - + Get the maximum number of threads used for client connections. See gst_rtsp_thread_pool_set_max_threads(). @@ -11363,7 +11483,7 @@ gst_rtsp_thread_stop() after usage - + Set the maximum threads used by the pool to handle client requests. A value of 0 will use the pool mainloop, a value of -1 will use an unlimited number of threads. @@ -11382,7 +11502,7 @@ unlimited number of threads. - + @@ -11408,6 +11528,8 @@ unlimited number of threads. + this function should make or reuse an existing thread that runs + a mainloop. @@ -11432,6 +11554,8 @@ gst_rtsp_thread_stop() after usage + configure a thread object. this vmethod is called when + a new thread has been created and should be configured. @@ -11451,6 +11575,7 @@ gst_rtsp_thread_stop() after usage + called from the thread when it is entered @@ -11467,6 +11592,7 @@ gst_rtsp_thread_stop() after usage + called from the thread when it is left @@ -11488,7 +11614,7 @@ gst_rtsp_thread_stop() after usage - + @@ -11674,10 +11800,10 @@ MT safe. The supported modes of the media. - + Transport supports PLAY mode - + Transport supports RECORD mode diff --git a/girs/GstSdp-1.0.gir b/girs/GstSdp-1.0.gir index f45145681d..0eaff482ac 100644 --- a/girs/GstSdp-1.0.gir +++ b/girs/GstSdp-1.0.gir @@ -20,7 +20,7 @@ and/or use gtk-doc annotations. --> to be used for the specific CSB. - + @@ -39,7 +39,7 @@ and/or use gtk-doc annotations. --> AES-GCM using a 128-bit key (Since: 1.16) - + diff --git a/girs/GstTag-1.0.gir b/girs/GstTag-1.0.gir index e0f2e37cd5..29abd2a9b4 100644 --- a/girs/GstTag-1.0.gir +++ b/girs/GstTag-1.0.gir @@ -400,6 +400,9 @@ Subclasses have to do four things: 128 bytes. + identify tag and determine the size required to parse the +tag. Buffer may be larger than the specified minimum size. +Subclassed MUST override this vfunc in their class_init function. @@ -420,6 +423,10 @@ Subclasses have to do four things: + merge start and end tags. Subclasses may want to override this +vfunc to allow prioritising of start or end tag according to user +preference. Note that both start_tags and end_tags may be NULL. By default +start tags are preferred over end tags. @@ -437,6 +444,12 @@ Subclasses have to do four things: + parse the tag. Buffer will be exactly of the size determined by +the identify_tag vfunc before. The parse_tag vfunc may change the size +stored in *tag_size and return GST_TAG_DEMUX_RESULT_AGAIN to request a +larger or smaller buffer. It is also permitted to adjust the tag_size to a +smaller value and then return GST_TAG_DEMUX_RESULT_OK in one go. +Subclassed MUST override the parse_tag vfunc in their class_init function. @@ -493,6 +506,9 @@ Subclasses should set this in their class_init function. + identify tag and determine the size required to parse the +tag. Buffer may be larger than the specified minimum size. +Subclassed MUST override this vfunc in their class_init function. @@ -515,6 +531,12 @@ Subclasses should set this in their class_init function. + parse the tag. Buffer will be exactly of the size determined by +the identify_tag vfunc before. The parse_tag vfunc may change the size +stored in *tag_size and return GST_TAG_DEMUX_RESULT_AGAIN to request a +larger or smaller buffer. It is also permitted to adjust the tag_size to a +smaller value and then return GST_TAG_DEMUX_RESULT_OK in one go. +Subclassed MUST override the parse_tag vfunc in their class_init function. @@ -540,6 +562,10 @@ Subclasses should set this in their class_init function. + merge start and end tags. Subclasses may want to override this +vfunc to allow prioritising of start or end tag according to user +preference. Note that both start_tags and end_tags may be NULL. By default +start tags are preferred over end tags. @@ -564,146 +590,146 @@ 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) @@ -726,6 +752,8 @@ Subclasses have to do the following things: + create a tag buffer to add to the end of the + input stream given a tag list, or NULL @@ -740,6 +768,8 @@ Subclasses have to do the following things: + create a tag buffer to add to the beginning of the + input stream given a tag list, or NULL @@ -775,6 +805,8 @@ of the two render vfuncs. + create a tag buffer to add to the beginning of the + input stream given a tag list, or NULL @@ -791,6 +823,8 @@ of the two render vfuncs. + create a tag buffer to add to the end of the + input stream given a tag list, or NULL @@ -812,7 +846,7 @@ of the two render vfuncs. - + diff --git a/girs/GstTranscoder-1.0.gir b/girs/GstTranscoder-1.0.gir index b3848b659f..3614d02526 100644 --- a/girs/GstTranscoder-1.0.gir +++ b/girs/GstTranscoder-1.0.gir @@ -68,7 +68,7 @@ about the serialization format. - + %TRUE if the transcoder tries to avoid reencoding streams where @@ -82,7 +82,7 @@ reencoding is not strictly needed, %FALSE otherwise. - + Gets the URI of the destination of the transcoded stream. @@ -97,7 +97,7 @@ destination of the transcoded stream. g_free() after usage. - + Retrieves the duration of the media stream that self represents. @@ -137,7 +137,7 @@ fill memory. To avoid that, the bus has to be set "flushing". - + The internal uritranscodebin instance @@ -150,7 +150,7 @@ fill memory. To avoid that, the bus has to be set "flushing". - + the absolute position time, in nanoseconds, of the @@ -164,7 +164,7 @@ transcoding stream. - + current position update interval in milliseconds @@ -265,7 +265,7 @@ notified about any error. - + @@ -302,7 +302,7 @@ should try to use. It takes into account the number of cores available. - + Set interval in milliseconds between two position-updated signals. Pass 0 to stop updating the position. @@ -320,29 +320,29 @@ Pass 0 to stop updating the position. - + See #encodebin:avoid-reencoding - + - + - + - + - + - + @@ -353,7 +353,7 @@ Pass 0 to stop updating the position. - + generic error. @@ -380,22 +380,22 @@ Pass 0 to stop updating the position. Types of messages that will be posted on the transcoder API bus. See also #gst_transcoder_get_message_bus() - + Sink position changed - + Duration of stream changed - + Pipeline state changed - + Transcoding is done - + Message contains an error - + Message contains an error @@ -508,7 +508,7 @@ See also #gst_transcoder_get_message_bus() Transforms #GstTranscoder bus messages to signals from the adapter object. - + The #GstTranscoder @self is tracking @@ -521,7 +521,7 @@ See also #gst_transcoder_get_message_bus() - + The #GstTranscoder tracked by the adapter. @@ -595,13 +595,13 @@ See also #gst_transcoder_get_message_bus() High level representation of the transcoder pipeline state. - + the transcoder is stopped. - + the transcoder is paused. - + the transcoder is currently transcoding a stream. diff --git a/girs/GstVa-1.0.gir b/girs/GstVa-1.0.gir index 454573e799..73ca3570df 100644 --- a/girs/GstVa-1.0.gir +++ b/girs/GstVa-1.0.gir @@ -528,7 +528,7 @@ GstVaDisplay descendants. - + @@ -582,11 +582,11 @@ and operate the device in @path. - + - + @@ -614,7 +614,7 @@ pipeline is not in NULL state. - + @@ -742,13 +742,13 @@ later it populates the @buffer with those DMABufs. - + The feature is disabled. - + The feature is enabled. - + The feature is enabled automatically. @@ -1037,7 +1037,7 @@ to it. - + It imports the array of @mem, representing a single frame, into a VASurfaceID and it's attached into every @mem. @@ -1066,7 +1066,7 @@ VASurfaceID and it's attached into every @mem. array of DMABuf file descriptors. - + diff --git a/girs/GstValidate-1.0.gir b/girs/GstValidate-1.0.gir index 15af42c755..04e9574956 100644 --- a/girs/GstValidate-1.0.gir +++ b/girs/GstValidate-1.0.gir @@ -258,7 +258,7 @@ value to a GstClockTime. - + @@ -269,26 +269,26 @@ GST_VALIDATE_EXECUTE_ACTION_ERROR_REPORTED: GST_VALIDATE_EXECUTE_ACTION_IN_PROGRESS: GST_VALIDATE_EXECUTE_ACTION_NONE: GST_VALIDATE_EXECUTE_ACTION_DONE: - + - + - + - + The action will be executed asynchronously without blocking further actions to be executed - + Use #GST_VALIDATE_EXECUTE_ACTION_NON_BLOCKING instead. - + - + - + - + @@ -352,54 +352,54 @@ actions to be executed - + No special flag - + The action is a config - + The action can be executed ASYNC - + The action can be executed asynchronously but without blocking further actions execution. - + Use #GST_VALIDATE_ACTION_TYPE_NON_BLOCKING instead. - + The action will be executed on 'element-added' for a particular element type if no playback-time is specified - + The pipeline will need to be synchronized with the clock for that action type to be used. - + Do not consider the non execution of the action as a fatal error. - + The action can use the 'optional' keyword. Such action instances will have the #GST_VALIDATE_ACTION_TYPE_NO_EXECUTION_NOT_FATAL flag set and won't be considered as fatal if they fail. - + - + The action can be used in config files even if it is not strictly a config action (ie. it needs a scenario to run). - + The action is checking some state from objects in the pipeline. It means that it can be used as 'check' in different action which have a `check` "sub action", such as the 'wait' action type. This implies that the action can be executed from any thread and not only from the scenario thread as other types. - + @@ -443,7 +443,7 @@ under @monitor watch - + @@ -490,19 +490,19 @@ GST_VALIDATE_FATAL_CRITICALS: GST_VALIDATE_PRINT_ISSUES: GST_VALIDATE_PRINT_WARNINGS: GST_VALIDATE_PRINT_CRITICALS: - + - + - + - + - + - + - + @@ -651,14 +651,14 @@ Class that wraps a #GstElement for Validate checks - + The report will be completely ignored. - + The report will be kept by the reporter, but not reported to the runner. - + The report will be kept by the reporter and reported to the runner. @@ -804,13 +804,13 @@ Class that wraps a #GstElement for Validate checks GST_VALIDATE_ISSUE_FLAGS_NONE: No special flags for the issue type GST_VALIDATE_ISSUE_FLAGS_FULL_DETAILS: Always show all occurrences of the issue in full details GST_VALIDATE_ISSUE_FLAGS_NO_BACKTRACE: Do not generate backtrace for the issue type - + - + - + - + Always generate backtrace, even if not a critical issue @@ -1117,10 +1117,10 @@ GST_VALIDATE_ISSUE_FLAGS_NO_BACKTRACE: Do not generate backtrace for the issue t - + - + @@ -1318,16 +1318,16 @@ GST_VALIDATE_ISSUE_FLAGS_NO_BACKTRACE: Do not generate backtrace for the issue t - + - + - + - + - + @@ -1580,7 +1580,7 @@ Class that wraps a #GObject for Validate checks - + The pipeline in which @monitor @@ -1624,7 +1624,7 @@ target is in. - + @@ -1633,7 +1633,7 @@ target is in. - + @@ -1739,11 +1739,11 @@ target is in. - + The property is optional, if it is not found on the object, nothing happens. - + Do not check that after setting the property, the value is the one we set. @@ -2255,7 +2255,7 @@ setting the property, the value is the one we set. - + @@ -2561,7 +2561,7 @@ Class that wraps a #GstPad for Validate checks - + @@ -2676,6 +2676,36 @@ equations in the action structure. + + Reports a new issue in the GstValidate reporting system with @m +as the source of that issue. + + + + The #GstValidateReporter where the issue happened + + + The #GstValidateIssueId of the issue + + + The format of the message describing the issue in a printf + format, followed by the parameters. + + + + + + + + + + + + + + + + @@ -3088,19 +3118,19 @@ You can also use #GST_VALIDATE_REPORT instead. - + - + - + - + - + - + - + @@ -3430,26 +3460,26 @@ and levels. No object category / name sets the global level. Examples: GST_VALIDATE_REPORTING_DETAILS=synthetic,h264parse:all GST_VALIDATE_REPORTING_DETAILS=none,h264parse::sink_0:synthetic - + No reporting level known, reporting will default to the global reporting level. - + No debugging level specified or desired. Used to deactivate debugging output. - + Summary of the issues found, with no details. - + If set as the default level, similar issues can be reported multiple times for different subchains. If set as the level for a particular object (my_object:subchain), validate will report the issues where the object is the first to report an issue for a subchain. - + If set as the default level, all the distinct issues for all the monitors will be reported. If set as the level for a particular object, all the distinct issues for this object @@ -3457,16 +3487,16 @@ will be reported. Note that if the same issue happens twice on the same object, up until this level that issue is only reported once. - + All the issues will be reported, even those that repeat themselves inside the same object. This can be *very* verbose if set globally. - + Sythetic for not fatal issues and detailed for others - + @@ -3581,7 +3611,7 @@ exit code of the application. - + @@ -3624,7 +3654,7 @@ exit code of the application. - + @@ -3765,10 +3795,10 @@ against - + - + @@ -3821,20 +3851,20 @@ against - + - + - + - + - + - + @@ -3847,17 +3877,17 @@ against Defines the level of verbosity of -validate (ie, printing on stdout). - + - + - + - + - + - + diff --git a/girs/GstVideo-1.0.gir b/girs/GstVideo-1.0.gir index 00aaa6d0c8..5e5ae56694 100644 --- a/girs/GstVideo-1.0.gir +++ b/girs/GstVideo-1.0.gir @@ -70,13 +70,13 @@ and/or use gtk-doc annotations. --> Location of a @GstAncillaryMeta. - + Progressive or no field specified (default) - + Interlaced first field - + Interlaced second field @@ -411,6 +411,7 @@ for modifying the color balance implemented by an element providing the #GstColorBalance interface. For example, Hue or Saturation. + default handler for value changed notification @@ -465,6 +466,7 @@ for modifying the color balance implemented by an element providing the + default handler for value changed notification @@ -494,6 +496,7 @@ for modifying the color balance implemented by an element providing the + list handled channels @@ -514,6 +517,7 @@ for modifying the color balance implemented by an element providing the + set a channel value @@ -536,6 +540,7 @@ for modifying the color balance implemented by an element providing the + get a channel value @@ -555,6 +560,7 @@ for modifying the color balance implemented by an element providing the + implementation type @@ -570,6 +576,7 @@ for modifying the color balance implemented by an element providing the + default handler for value changed notification @@ -602,11 +609,11 @@ for modifying the color balance implemented by an element providing the 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. @@ -1967,62 +1974,62 @@ For convenience in handling DVD navigation, the MENU commands are aliased as: 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. @@ -2031,65 +2038,65 @@ feature. Enum values for the various events that an element implementing the GstNavigation interface might send up the pipeline. Touch events have been inspired by the libinput API, and have the same meaning here. - + 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. - + A mouse scroll event. Use gst_navigation_event_parse_mouse_scroll_event() to extract the details from the event. - + An event describing a new touch point, which will be assigned an identifier that is unique to it for the duration of its movement on the screen. Use gst_navigation_event_parse_touch_event() to extract the details from the event. - + An event describing the movement of an active touch point across the screen. Use gst_navigation_event_parse_touch_event() to extract the details from the event. - + An event describing a removed touch point. After this event, its identifier may be reused for any new touch points. Use gst_navigation_event_parse_touch_up_event() to extract the details from the event. - + An event signaling the end of a sequence of simultaneous touch events. - + An event cancelling all currently active touch points. - + A mouse button double click event. Use gst_navigation_event_parse_mouse_button_event() to extract the details from the event. @@ -2103,6 +2110,7 @@ from the event. + sending a navigation event @@ -2119,6 +2127,7 @@ from the event. + sending a navigation event (Since: 1.22) @@ -2140,25 +2149,25 @@ from the event. 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) @@ -2169,71 +2178,71 @@ in events. Typical modifier keys are Shift, Control, Meta, Super, Hyper, Alt, Compose, Apple, CapsLock or ShiftLock. - + - + the Shift key. - + - + the Control key. - + the third modifier key - + the fourth modifier key - + the fifth modifier key - + the sixth modifier key - + the seventh modifier key - + the first mouse button (usually the left button). - + the second mouse button (usually the right button). - + the third mouse button (usually the mouse wheel button or middle button). - + the fourth mouse button (typically the "Back" button). - + the fifth mouse button (typically the "forward" button). - + the Super modifier - + the Hyper modifier - + the Meta modifier - + A mask covering all entries in #GdkModifierType. - + the Meta modifier Types of navigation interface queries. - + invalid query - + command query - + viewing angle query @@ -4320,13 +4329,13 @@ https://www.etsi.org/deliver/etsi_ts/101100_101199/101154/02.01.01_60/ts_101154v https://www.atsc.org/wp-content/uploads/2015/03/a_53-Part-4-2009.pdf 2) SMPTE ST2016-1: - + AFD value is from DVB/ETSI standard - + AFD value is from ATSC A/53 standard - + @@ -4362,53 +4371,53 @@ of the active image. 2) AFD 0 is reserved for DVB/ETSI 3) values 1, 5, 6, 7, and 12 are reserved for both ATSC and DVB/ETSI 4) values 2 and 3 are not recommended for ATSC, but are valid for DVB/ETSI - + Unavailable (see note 0 below). - + For 4:3 coded frame, letterbox 16:9 image, at top of the coded frame. For 16:9 coded frame, full frame 16:9 image, the same as the coded frame. - + For 4:3 coded frame, letterbox 14:9 image, at top of the coded frame. For 16:9 coded frame, pillarbox 14:9 image, horizontally centered in the coded frame. - + For 4:3 coded frame, letterbox image with an aspect ratio greater than 16:9, vertically centered in the coded frame. For 16:9 coded frame, letterbox image with an aspect ratio greater than 16:9. - + For 4:3 coded frame, full frame 4:3 image, the same as the coded frame. For 16:9 coded frame, full frame 16:9 image, the same as the coded frame. - + For 4:3 coded frame, full frame 4:3 image, the same as the coded frame. For 16:9 coded frame, pillarbox 4:3 image, horizontally centered in the coded frame. - + For 4:3 coded frame, letterbox 16:9 image, vertically centered in the coded frame with all image areas protected. For 16:9 coded frame, full frame 16:9 image, with all image areas protected. - + For 4:3 coded frame, letterbox 14:9 image, vertically centered in the coded frame. For 16:9 coded frame, pillarbox 14:9 image, horizontally centered in the coded frame. - + For 4:3 coded frame, full frame 4:3 image, with alternative 14:9 center. For 16:9 coded frame, pillarbox 4:3 image, with alternative 14:9 center. - + For 4:3 coded frame, letterbox 16:9 image, with alternative 14:9 center. For 16:9 coded frame, full frame 16:9 image, with alternative 14:9 center. - + For 4:3 coded frame, letterbox 16:9 image, with alternative 4:3 center. For 16:9 coded frame, full frame 16:9 image, with alternative 4:3 center. @@ -4488,6 +4497,12 @@ Zorder for each input stream can be configured on the #GstVideoAggregatorPad. + Lets subclasses aggregate frames that are ready. Subclasses + should iterate the GstElement.sinkpads and use the already + mapped #GstVideoFrame from gst_video_aggregator_pad_get_prepared_frame() + or directly use the #GstBuffer from gst_video_aggregator_pad_get_current_buffer() + if it needs to map the buffer in a special way. The result of the + aggregation should land in @outbuffer. @@ -4502,6 +4517,9 @@ Zorder for each input stream can be configured on the + Optional. + Lets subclasses provide a #GstBuffer to be used as @outbuffer of + the #aggregate_frames vmethod. @@ -4536,6 +4554,9 @@ Zorder for each input stream can be configured on the + Optional. + Lets subclasses update the #GstCaps representing + the src pad caps before usage. Return %NULL to indicate failure. @@ -4568,7 +4589,7 @@ Subclasses can add their own operation to perform using the returned - + Causes the element to aggregate on a timeout even when no live source is connected to its sinks. See #GstAggregator:min-upstream-latency for a companion property: in the vast majority of cases where you plan to plug in @@ -4598,6 +4619,9 @@ srcpad caps. + Optional. + Lets subclasses update the #GstCaps representing + the src pad caps before usage. Return %NULL to indicate failure. @@ -4614,6 +4638,12 @@ srcpad caps. + Lets subclasses aggregate frames that are ready. Subclasses + should iterate the GstElement.sinkpads and use the already + mapped #GstVideoFrame from gst_video_aggregator_pad_get_prepared_frame() + or directly use the #GstBuffer from gst_video_aggregator_pad_get_current_buffer() + if it needs to map the buffer in a special way. The result of the + aggregation should land in @outbuffer. @@ -4630,6 +4660,9 @@ srcpad caps. + Optional. + Lets subclasses provide a #GstBuffer to be used as @outbuffer of + the #aggregate_frames vmethod. @@ -4646,6 +4679,8 @@ srcpad caps. + Optional. + Lets subclasses decide of the best common format to use. @@ -4754,12 +4789,13 @@ update for any changes that have happened. - + + clean the frame previously prepared in prepare_frame @@ -4777,6 +4813,9 @@ update for any changes that have happened. + Prepare the frame from the pad buffer and sets it to prepared_frame. + Implementations should always return TRUE. Returning FALSE will cease + iteration over subsequent pads. @@ -4847,6 +4886,8 @@ If overriden, `prepare_frame_finish` must also be overriden. + Called when either the input or output formats + have changed. @@ -4934,13 +4975,13 @@ or from the #GstVideoAggregatorPadClass::prepare_frame virtual method of the agg - + - + - + @@ -4965,6 +5006,8 @@ or from the #GstVideoAggregatorPadClass::prepare_frame virtual method of the agg + Called when either the input or output formats + have changed. @@ -4978,6 +5021,9 @@ or from the #GstVideoAggregatorPadClass::prepare_frame virtual method of the agg + Prepare the frame from the pad buffer and sets it to prepared_frame. + Implementations should always return TRUE. Returning FALSE will cease + iteration over subsequent pads. @@ -5000,6 +5046,7 @@ or from the #GstVideoAggregatorPadClass::prepare_frame virtual method of the agg + clean the frame previously prepared in prepare_frame @@ -5072,7 +5119,7 @@ or from the #GstVideoAggregatorPadClass::prepare_frame virtual method of the agg - + @@ -5095,7 +5142,7 @@ See #GstVideoAggregator for more details. - + @@ -5141,16 +5188,16 @@ structure is usually used to configure the bufferpool if it supports the 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 @@ -5190,40 +5237,40 @@ the parity check bits). - + - + - + - + - + - + - + - + - + - + - + - + Some know types of Ancillary Data identifiers. - + CEA 708 Ancillary data according to SMPTE 334 - + CEA 608 Ancillary data according to SMPTE 334 - + AFD/Bar Ancillary data according to SMPTE 2016-3 (Since: 1.18) @@ -5280,27 +5327,27 @@ buffers carrying closed caption data, or video data - even encoded data. Note that these are only valid for #GstCaps of type: video/... and caption/... 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_FLAG_TFF flag setting) is repeated. - + If the #GstBuffer is interlaced, then only the first field (as defined by the %GST_VIDEO_BUFFER_FLAG_TFF flag setting) is to be displayed (Since: 1.16). - + 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 @@ -5308,33 +5355,33 @@ They can conflict with other extended buffer flags. mono / non-mono streams, the absence 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. - + The video frame has the top field only. This is the same as GST_VIDEO_BUFFER_FLAG_TFF | GST_VIDEO_BUFFER_FLAG_ONEFIELD (Since: 1.16). Use GST_VIDEO_BUFFER_IS_TOP_FIELD() to check for this flag. - + If the #GstBuffer is interlaced, then only the first field (as defined by the %GST_VIDEO_BUFFER_FLAG_TFF flag setting) is to be displayed (Since: 1.16). - + The video frame has the bottom field only. This is the same as GST_VIDEO_BUFFER_FLAG_ONEFIELD (GST_VIDEO_BUFFER_FLAG_TFF flag unset) (Since: 1.16). Use GST_VIDEO_BUFFER_IS_BOTTOM_FIELD() to check for this flag. - + The #GstBuffer contains the end of a video field or frame boundary such as the last subframe or packet (Since: 1.18). - + Offset to define more flags @@ -5362,7 +5409,7 @@ supports all the video bufferpool options. - + @@ -5395,10 +5442,10 @@ supports all the video bufferpool options. The various known types of Closed Caption (CC). - + Unknown type of CC - + CEA-608 as byte pairs. Note that this format is not recommended since is does not specify to which field the caption comes from and therefore assumes @@ -5407,7 +5454,7 @@ supports all the video bufferpool options. if you wish to store CEA-608 from two fields and prefix each byte pair with 0xFC for the first field and 0xFD for the second field. - + CEA-608 as byte triplets as defined in SMPTE S334-1 Annex A. The second and third byte of the byte triplet is the raw CEA608 data, the first byte is a bitfield: The top/7th bit is @@ -5417,12 +5464,12 @@ supports all the video bufferpool options. for 525-line field 1, line 272 for 525-line field 2, line 5 for 625-line field 1 and line 318 for 625-line field 2). - + CEA-708 as cc_data byte triplets. They can also contain 608-in-708 and the first byte of each triplet has to be inspected for detecting the type. - + CEA-708 (and optionally CEA-608) in a CDP (Caption Distribution Packet) defined by SMPTE S-334-2. Contains the whole CDP (starting with 0x9669). @@ -5459,40 +5506,40 @@ type, or %GST_VIDEO_CAPTION_TYPE_UNKNOWN. 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. @@ -5590,31 +5637,31 @@ performed. 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 @@ -5836,19 +5883,19 @@ will be freed. 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 - + The buffer data is corrupted. @@ -5929,27 +5976,27 @@ will be freed. The color matrix is used to convert between Y'PbPr and non-linear RGB (R'G'B') - + unknown matrix - + identity matrix. Order of coefficients is actually GBR, also IEC 61966-2-1 (sRGB) - + FCC Title 47 Code of Federal Regulations 73.682 (a)(20) - + ITU-R BT.709 color matrix, also ITU-R BT1361 / IEC 61966-2-4 xvYCC709 / SMPTE RP177 Annex B - + ITU-R BT.601 color matrix, also SMPTE170M / ITU-R BT1358 525 / ITU-R BT1700 NTSC - + SMPTE 240M color matrix - + ITU-R BT.2020 color matrix. Since: 1.6 @@ -6037,51 +6084,51 @@ and "ITU-T H.273 Table 4". The color primaries define the how to transform linear RGB values to and from the CIE XYZ colorspace. - + unknown color primaries - + BT709 primaries, also ITU-R BT1361 / IEC 61966-2-4 / SMPTE RP177 Annex B - + BT470M primaries, also FCC Title 47 Code of Federal Regulations 73.682 (a)(20) - + BT470BG primaries, also ITU-R BT601-6 625 / ITU-R BT1358 625 / ITU-R BT1700 625 PAL & SECAM - + SMPTE170M primaries, also ITU-R BT601-6 525 / ITU-R BT1358 525 / ITU-R BT1700 NTSC - + SMPTE240M primaries - + Generic film (colour filters using Illuminant C) - + ITU-R BT2020 primaries. Since: 1.6 - + Adobe RGB primaries. Since: 1.8 - + SMPTE ST 428 primaries (CIE 1931 XYZ). Since: 1.16 - + SMPTE RP 431 primaries (ST 431-2 (2011) / DCI P3). Since: 1.16 - + SMPTE EG 432 primaries (ST 432-1 (2010) / P3 D65). Since: 1.16 - + EBU 3213 primaries (JEDEC P22 phosphors). Since: 1.16 @@ -6197,13 +6244,13 @@ XYZ colorspace. 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. @@ -6499,7 +6546,7 @@ with the parsed values. - + Convert the pixels of @src into @dest using @convert. @@ -6828,6 +6875,9 @@ The bare minimum that a functional subclass needs to implement is: @gst_video_decoder_finish_frame, or call @gst_video_decoder_drop_frame. + Optional. + Called when the element changes to GST_STATE_NULL. + Allows closing external resources. @@ -6839,6 +6889,12 @@ The bare minimum that a functional subclass needs to implement is: + Optional. + Setup the allocation parameters for allocating output + buffers. The passed in query contains the result of the + downstream allocation query. + Subclasses should chain up to the parent implementation to + invoke the default handler. @@ -6853,6 +6909,11 @@ The bare minimum that a functional subclass needs to implement is: + Optional. + Called to request subclass to decode any data it can at this + point, but that more data may arrive after. (e.g. at segment end). + Sub-classes should be prepared to handle new data afterward, + or seamless segment processing will break. Since: 1.6 @@ -6864,6 +6925,9 @@ The bare minimum that a functional subclass needs to implement is: + Optional. + Called to request subclass to dispatch any pending remaining + data at EOS. Sub-classes can refuse to decode new data after. @@ -6875,6 +6939,9 @@ The bare minimum that a functional subclass needs to implement is: + Optional. + Flush all remaining data from the decoder without + pushing it downstream. Since: 1.2 @@ -6886,6 +6953,11 @@ The bare minimum that a functional subclass needs to implement is: + Optional. + Allows for a custom sink getcaps implementation. + If not implemented, default returns + gst_video_decoder_proxy_getcaps + applied to sink template caps. @@ -6953,6 +7025,9 @@ negotiate fails. + Optional. + Called when the element changes to GST_STATE_READY. + Allows opening external resources. @@ -6964,6 +7039,9 @@ negotiate fails. + Required for non-packetized input. + Allows chopping incoming data into manageable units (frames) + for subsequent decoding. @@ -6984,6 +7062,10 @@ negotiate fails. + Optional. + Propose buffer allocation parameters for upstream elements. + Subclasses should chain up to the parent implementation to + invoke the default handler. @@ -6998,6 +7080,9 @@ negotiate fails. + Optional. + Allows subclass (decoder) to perform post-seek semantics reset. + Deprecated. @@ -7012,6 +7097,7 @@ negotiate fails. + Notifies subclass of incoming data format (caps). @@ -7026,6 +7112,12 @@ negotiate fails. + Optional. + Event handler on the sink pad. This function should return + TRUE if the event was handled and should be discarded + (i.e. not unref'ed). + Subclasses should chain up to the parent implementation to + invoke the default handler. @@ -7040,6 +7132,11 @@ negotiate fails. + Optional. + Query handler on the sink pad. This function should + return TRUE if the query could be performed. Subclasses + should chain up to the parent implementation to invoke the + default handler. Since: 1.4 @@ -7054,6 +7151,12 @@ negotiate fails. + Optional. + Event handler on the source pad. This function should return + TRUE if the event was handled and should be discarded + (i.e. not unref'ed). + Subclasses should chain up to the parent implementation to + invoke the default handler. @@ -7068,6 +7171,11 @@ negotiate fails. + Optional. + Query handler on the source pad. This function should + return TRUE if the query could be performed. Subclasses + should chain up to the parent implementation to invoke the + default handler. Since: 1.4 @@ -7082,6 +7190,9 @@ negotiate fails. + Optional. + Called when the element starts processing. + Allows opening external resources. @@ -7093,6 +7204,9 @@ negotiate fails. + Optional. + Called when the element stops processing. + Allows closing external resources. @@ -7104,6 +7218,11 @@ negotiate fails. + Optional. Transform the metadata on the input buffer to the + output buffer. By default this method is copies all meta without + tags and meta with only the "video" tag. subclasses can + implement this method and return %TRUE if the metadata is to be + copied. Since: 1.6 @@ -7439,7 +7558,7 @@ and should therefore occur as soon/skippy as possible. - + currently configured decoder tolerated error count. @@ -7831,7 +7950,7 @@ global 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 @@ -8012,28 +8131,28 @@ handler with %GST_PAD_SET_ACCEPT_INTERSECT and - + GstVideoDecoderRequestSyncPointFlags to use for the automatically requested sync points if `automatic-request-sync-points` is enabled. - + If set to %TRUE the decoder will automatically request sync points when it seems like a good idea, e.g. if the first frames are not key frames or if packet loss was reported by upstream. - + If set to %TRUE the decoder will discard frames that are marked as corrupted instead of outputting them. - + Maximum number of tolerated consecutive decode errors. See gst_video_decoder_set_max_errors() for more details. - + Minimum interval between force-key-unit events sent upstream by the decoder. Setting this to 0 will cause every event to be handled, setting this to %GST_CLOCK_TIME_NONE will cause every event to be ignored. @@ -8042,7 +8161,7 @@ See gst_video_event_new_upstream_force_key_unit() for more details about force-key-unit events. - + If set to %TRUE the decoder will handle QoS events received from downstream elements. This includes dropping output frames which are detected as late @@ -8086,6 +8205,9 @@ and likely as well. If non-packetized input is supported or expected, + Optional. + Called when the element changes to GST_STATE_READY. + Allows opening external resources. @@ -8099,6 +8221,9 @@ and likely as well. If non-packetized input is supported or expected, + Optional. + Called when the element changes to GST_STATE_NULL. + Allows closing external resources. @@ -8112,6 +8237,9 @@ and likely as well. If non-packetized input is supported or expected, + Optional. + Called when the element starts processing. + Allows opening external resources. @@ -8125,6 +8253,9 @@ and likely as well. If non-packetized input is supported or expected, + Optional. + Called when the element stops processing. + Allows closing external resources. @@ -8138,6 +8269,9 @@ and likely as well. If non-packetized input is supported or expected, + Required for non-packetized input. + Allows chopping incoming data into manageable units (frames) + for subsequent decoding. @@ -8160,6 +8294,7 @@ and likely as well. If non-packetized input is supported or expected, + Notifies subclass of incoming data format (caps). @@ -8176,6 +8311,9 @@ and likely as well. If non-packetized input is supported or expected, + Optional. + Allows subclass (decoder) to perform post-seek semantics reset. + Deprecated. @@ -8192,6 +8330,9 @@ and likely as well. If non-packetized input is supported or expected, + Optional. + Called to request subclass to dispatch any pending remaining + data at EOS. Sub-classes can refuse to decode new data after. @@ -8205,6 +8346,9 @@ and likely as well. If non-packetized input is supported or expected, + Provides input data frame to subclass. In subframe mode, the subclass needs + to take ownership of @GstVideoCodecFrame.input_buffer as it will be modified + by the base class on the next subframe buffer receiving. @@ -8223,6 +8367,12 @@ and likely as well. If non-packetized input is supported or expected, + Optional. + Event handler on the sink pad. This function should return + TRUE if the event was handled and should be discarded + (i.e. not unref'ed). + Subclasses should chain up to the parent implementation to + invoke the default handler. @@ -8239,6 +8389,12 @@ and likely as well. If non-packetized input is supported or expected, + Optional. + Event handler on the source pad. This function should return + TRUE if the event was handled and should be discarded + (i.e. not unref'ed). + Subclasses should chain up to the parent implementation to + invoke the default handler. @@ -8255,6 +8411,10 @@ and likely as well. If non-packetized input is supported or expected, + Optional. + Negotiate with downstream and configure buffer pools, etc. + Subclasses should chain up to the parent implementation to + invoke the default handler. @@ -8270,6 +8430,12 @@ and likely as well. If non-packetized input is supported or expected, + Optional. + Setup the allocation parameters for allocating output + buffers. The passed in query contains the result of the + downstream allocation query. + Subclasses should chain up to the parent implementation to + invoke the default handler. @@ -8286,6 +8452,10 @@ and likely as well. If non-packetized input is supported or expected, + Optional. + Propose buffer allocation parameters for upstream elements. + Subclasses should chain up to the parent implementation to + invoke the default handler. @@ -8302,6 +8472,9 @@ and likely as well. If non-packetized input is supported or expected, + Optional. + Flush all remaining data from the decoder without + pushing it downstream. Since: 1.2 @@ -8315,6 +8488,11 @@ and likely as well. If non-packetized input is supported or expected, + Optional. + Query handler on the sink pad. This function should + return TRUE if the query could be performed. Subclasses + should chain up to the parent implementation to invoke the + default handler. Since: 1.4 @@ -8331,6 +8509,11 @@ and likely as well. If non-packetized input is supported or expected, + Optional. + Query handler on the source pad. This function should + return TRUE if the query could be performed. Subclasses + should chain up to the parent implementation to invoke the + default handler. Since: 1.4 @@ -8347,6 +8530,11 @@ and likely as well. If non-packetized input is supported or expected, + Optional. + Allows for a custom sink getcaps implementation. + If not implemented, default returns + gst_video_decoder_proxy_getcaps + applied to sink template caps. @@ -8363,6 +8551,11 @@ and likely as well. If non-packetized input is supported or expected, + Optional. + Called to request subclass to decode any data it can at this + point, but that more data may arrive after. (e.g. at segment end). + Sub-classes should be prepared to handle new data afterward, + or seamless segment processing will break. Since: 1.6 @@ -8376,6 +8569,11 @@ and likely as well. If non-packetized input is supported or expected, + Optional. Transform the metadata on the input buffer to the + output buffer. By default this method is copies all meta without + tags and meta with only the "video" tag. subclasses can + implement this method and return %TRUE if the metadata is to be + copied. Since: 1.6 @@ -8423,17 +8621,17 @@ and likely as well. If non-packetized input is supported or expected, - + Flags to be used in combination with gst_video_decoder_request_sync_point(). See the function documentation for more details. - + discard all following input until the next sync point. - + discard all following output until the next sync point. @@ -8442,7 +8640,7 @@ See the function documentation for more details. The interface allows unified access to control flipping and rotation operations of video-sources or operators. - + @@ -8454,7 +8652,7 @@ operations of video-sources or operators. - + GstVideoDither provides implementations of several dithering algorithms that can be applied to lines of video pixels to quantize and dither them. @@ -8541,31 +8739,31 @@ performance is achieved when @quantizer is a power of 2. 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 @@ -8631,6 +8829,9 @@ pipeline to catch up. + Optional. + Called when the element changes to GST_STATE_NULL. + Allows closing external resources. @@ -8642,6 +8843,12 @@ pipeline to catch up. + Optional. + Setup the allocation parameters for allocating output + buffers. The passed in query contains the result of the + downstream allocation query. + Subclasses should chain up to the parent implementation to + invoke the default handler. @@ -8656,6 +8863,9 @@ pipeline to catch up. + Optional. + Called to request subclass to dispatch any pending remaining + data (e.g. at EOS). @@ -8667,6 +8877,9 @@ pipeline to catch up. + Optional. + Flush all remaining data from the encoder without + pushing it downstream. Since: 1.2 @@ -8678,6 +8891,11 @@ pipeline to catch up. + Optional. + Allows for a custom sink getcaps implementation (e.g. + for multichannel input specification). If not implemented, + default returns gst_video_encoder_proxy_getcaps + applied to sink template caps. @@ -8692,6 +8910,7 @@ pipeline to catch up. + Provides input frame to subclass. @@ -8722,6 +8941,9 @@ negotiate fails. + Optional. + Called when the element changes to GST_STATE_READY. + Allows opening external resources. @@ -8733,6 +8955,10 @@ negotiate fails. + Optional. + Allows subclass to push frame downstream in whatever + shape or form it deems appropriate. If not provided, + provided encoded frame data is simply pushed downstream. @@ -8747,6 +8973,10 @@ negotiate fails. + Optional. + Propose buffer allocation parameters for upstream elements. + Subclasses should chain up to the parent implementation to + invoke the default handler. @@ -8761,6 +8991,9 @@ negotiate fails. + Optional. + Allows subclass (encoder) to perform post-seek semantics reset. + Deprecated. @@ -8775,6 +9008,10 @@ negotiate fails. + Optional. + Notifies subclass of incoming data format. + GstVideoCodecState fields have already been + set according to provided caps. @@ -8789,6 +9026,12 @@ negotiate fails. + Optional. + Event handler on the sink pad. This function should return + TRUE if the event was handled and should be discarded + (i.e. not unref'ed). + Subclasses should chain up to the parent implementation to + invoke the default handler. @@ -8803,6 +9046,11 @@ negotiate fails. + Optional. + Query handler on the sink pad. This function should + return TRUE if the query could be performed. Subclasses + should chain up to the parent implementation to invoke the + default handler. Since: 1.4 @@ -8817,6 +9065,12 @@ negotiate fails. + Optional. + Event handler on the source pad. This function should return + TRUE if the event was handled and should be discarded + (i.e. not unref'ed). + Subclasses should chain up to the parent implementation to + invoke the default handler. @@ -8831,6 +9085,11 @@ negotiate fails. + Optional. + Query handler on the source pad. This function should + return TRUE if the query could be performed. Subclasses + should chain up to the parent implementation to invoke the + default handler. Since: 1.4 @@ -8845,6 +9104,9 @@ negotiate fails. + Optional. + Called when the element starts processing. + Allows opening external resources. @@ -8856,6 +9118,9 @@ negotiate fails. + Optional. + Called when the element stops processing. + Allows closing external resources. @@ -8867,6 +9132,11 @@ negotiate fails. + Optional. Transform the metadata on the input buffer to the + output buffer. By default this method is copies all meta without + tags and meta with only the "video" tag. subclasses can + implement this method and return %TRUE if the metadata is to be + copied. Since: 1.6 @@ -9115,7 +9385,7 @@ If no QoS events have been received from downstream, or if - + Returns the minimum force-keyunit interval, see gst_video_encoder_set_min_force_key_unit_interval() for more details. @@ -9304,7 +9574,7 @@ so the pipeline can reconfigure its global latency. - + Sets the minimum interval for requesting keyframes based on force-keyunit events. Setting this to 0 will allow to handle every event, setting this to %GST_CLOCK_TIME_NONE causes force-keyunit events to be ignored. @@ -9399,12 +9669,12 @@ from the next call to #gst_video_encoder_finish_frame(). - + Minimum interval between force-keyunit requests in nanoseconds. See gst_video_encoder_set_min_force_key_unit_interval() for more details. - + @@ -9443,6 +9713,9 @@ and @get_caps are likely needed as well. + Optional. + Called when the element changes to GST_STATE_READY. + Allows opening external resources. @@ -9456,6 +9729,9 @@ and @get_caps are likely needed as well. + Optional. + Called when the element changes to GST_STATE_NULL. + Allows closing external resources. @@ -9469,6 +9745,9 @@ and @get_caps are likely needed as well. + Optional. + Called when the element starts processing. + Allows opening external resources. @@ -9482,6 +9761,9 @@ and @get_caps are likely needed as well. + Optional. + Called when the element stops processing. + Allows closing external resources. @@ -9495,6 +9777,10 @@ and @get_caps are likely needed as well. + Optional. + Notifies subclass of incoming data format. + GstVideoCodecState fields have already been + set according to provided caps. @@ -9511,6 +9797,7 @@ and @get_caps are likely needed as well. + Provides input frame to subclass. @@ -9527,6 +9814,9 @@ and @get_caps are likely needed as well. + Optional. + Allows subclass (encoder) to perform post-seek semantics reset. + Deprecated. @@ -9543,6 +9833,9 @@ and @get_caps are likely needed as well. + Optional. + Called to request subclass to dispatch any pending remaining + data (e.g. at EOS). @@ -9556,6 +9849,10 @@ and @get_caps are likely needed as well. + Optional. + Allows subclass to push frame downstream in whatever + shape or form it deems appropriate. If not provided, + provided encoded frame data is simply pushed downstream. @@ -9572,6 +9869,11 @@ and @get_caps are likely needed as well. + Optional. + Allows for a custom sink getcaps implementation (e.g. + for multichannel input specification). If not implemented, + default returns gst_video_encoder_proxy_getcaps + applied to sink template caps. @@ -9588,6 +9890,12 @@ and @get_caps are likely needed as well. + Optional. + Event handler on the sink pad. This function should return + TRUE if the event was handled and should be discarded + (i.e. not unref'ed). + Subclasses should chain up to the parent implementation to + invoke the default handler. @@ -9604,6 +9912,12 @@ and @get_caps are likely needed as well. + Optional. + Event handler on the source pad. This function should return + TRUE if the event was handled and should be discarded + (i.e. not unref'ed). + Subclasses should chain up to the parent implementation to + invoke the default handler. @@ -9620,6 +9934,10 @@ and @get_caps are likely needed as well. + Optional. + Negotiate with downstream and configure buffer pools, etc. + Subclasses should chain up to the parent implementation to + invoke the default handler. @@ -9635,6 +9953,12 @@ and @get_caps are likely needed as well. + Optional. + Setup the allocation parameters for allocating output + buffers. The passed in query contains the result of the + downstream allocation query. + Subclasses should chain up to the parent implementation to + invoke the default handler. @@ -9651,6 +9975,10 @@ and @get_caps are likely needed as well. + Optional. + Propose buffer allocation parameters for upstream elements. + Subclasses should chain up to the parent implementation to + invoke the default handler. @@ -9667,6 +9995,9 @@ and @get_caps are likely needed as well. + Optional. + Flush all remaining data from the encoder without + pushing it downstream. Since: 1.2 @@ -9680,6 +10011,11 @@ and @get_caps are likely needed as well. + Optional. + Query handler on the sink pad. This function should + return TRUE if the query could be performed. Subclasses + should chain up to the parent implementation to invoke the + default handler. Since: 1.4 @@ -9696,6 +10032,11 @@ and @get_caps are likely needed as well. + Optional. + Query handler on the source pad. This function should + return TRUE if the query could be performed. Subclasses + should chain up to the parent implementation to invoke the + default handler. Since: 1.4 @@ -9712,6 +10053,11 @@ and @get_caps are likely needed as well. + Optional. Transform the metadata on the input buffer to the + output buffer. By default this method is copies all meta without + tags and meta with only the "video" tag. subclasses can + implement this method and return %TRUE if the metadata is to be + copied. Since: 1.6 @@ -9736,7 +10082,7 @@ and @get_caps are likely needed as well. - + @@ -9744,14 +10090,14 @@ and @get_caps are likely needed as well. 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 @@ -9792,6 +10138,7 @@ The videofilter will by default enable QoS on the parent GstBaseTransform to implement frame dropping. + function to be called with the negotiated caps and video infos @@ -9815,6 +10162,7 @@ to implement frame dropping. + transform a video frame @@ -9832,6 +10180,7 @@ to implement frame dropping. + transform a video frame in place @@ -9871,6 +10220,7 @@ to implement frame dropping. + function to be called with the negotiated caps and video infos @@ -9896,6 +10246,7 @@ to implement frame dropping. + transform a video frame @@ -9915,6 +10266,7 @@ to implement frame dropping. + transform a video frame in place @@ -9938,14 +10290,14 @@ to implement frame dropping. 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. @@ -9955,439 +10307,439 @@ to implement frame dropping. See the [GStreamer raw video format design document](https://gstreamer.freedesktop.org/documentation/additional/design/mediatype-video-raw.html#formats) for details about the layout and packing of these formats in memory. - + 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 packed into 24 bits without padding (`R-G-B-R-G-B`) - + reverse RGB packed into 24 bits without padding (`B-G-R-B-G-R`) - + 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 (native endianness) per channel - + packed 4:4:4 YUV with alpha channel, 16 bits (native endianness) 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) - + Fully packed variant of NV12_10LE32 (Since: 1.16) - + packed 4:2:2 YUV, 10 bits per channel (Since: 1.16) - + packed 4:4:4 YUV, 10 bits per channel(A-V-Y-U...) (Since: 1.16) - + packed 4:4:4 YUV with alpha channel (V0-U0-Y0-A0...) (Since: 1.16) - + packed 4:4:4 RGB with alpha channel(B-G-R-A), 10 bits for R/G/B channel and MSB 2 bits for alpha channel (Since: 1.16) - + packed 4:4:4 RGB with alpha channel(R-G-B-A), 10 bits for R/G/B channel and MSB 2 bits for alpha channel (Since: 1.18) - + planar 4:4:4 YUV, 16 bits per channel (Since: 1.18) - + planar 4:4:4 YUV, 16 bits per channel (Since: 1.18) - + planar 4:2:0 YUV with interleaved UV plane, 16 bits per channel (Since: 1.18) - + planar 4:2:0 YUV with interleaved UV plane, 16 bits per channel (Since: 1.18) - + planar 4:2:0 YUV with interleaved UV plane, 12 bits per channel (Since: 1.18) - + planar 4:2:0 YUV with interleaved UV plane, 12 bits per channel (Since: 1.18) - + packed 4:2:2 YUV, 12 bits per channel (Y-U-Y-V) (Since: 1.18) - + packed 4:2:2 YUV, 12 bits per channel (Y-U-Y-V) (Since: 1.18) - + packed 4:4:4:4 YUV, 12 bits per channel(U-Y-V-A...) (Since: 1.18) - + packed 4:4:4:4 YUV, 12 bits per channel(U-Y-V-A...) (Since: 1.18) - + NV12 with 4x4 tiles in linear order. - + NV12 with 32x32 tiles in linear order. - + Planar 4:4:4 RGB, R-G-B order - + Planar 4:4:4 RGB, B-G-R order - + Planar 4:2:0 YUV with interleaved UV plane with alpha as 3rd plane. - + RGB with alpha channel first, 16 bits (little endian) per channel. - + RGB with alpha channel first, 16 bits (big endian) per channel. - + RGB with alpha channel last, 16 bits (little endian) per channel. - + RGB with alpha channel last, 16 bits (big endian) per channel. - + Reverse RGB with alpha channel last, 16 bits (little endian) per channel. - + Reverse RGB with alpha channel last, 16 bits (big endian) per channel. - + Reverse RGB with alpha channel first, 16 bits (little endian) per channel. - + Reverse RGB with alpha channel first, 16 bits (big endian) per channel. - + NV12 with 16x32 Y tiles and 16x16 UV tiles. - + NV12 with 8x128 tiles in linear order. - + NV12 10bit big endian with 8x128 tiles in linear order. - + @GST_VIDEO_FORMAT_NV12_10LE40 with 4x4 pixels tiles (5 bytes per tile row). This format is produced by Verisilicon/Hantro decoders. - + @GST_VIDEO_FORMAT_DMA_DRM represent the DMA DRM special format. It's only used with memory:DMABuf #GstCapsFeatures, where an extra parameter (drm-format) is required to define the image format and its memory layout. - + Mediatek 10bit NV12 little endian with 16x32 tiles in linear order, tile 2 bits. - + Mediatek 10bit NV12 little endian with 16x32 tiles in linear order, raster 2 bits. - + planar 4:4:2:2 YUV, 8 bits per channel - + planar 4:4:4:4 YUV, 8 bits per channel - + planar 4:4:4:4 YUV, 12 bits per channel - + planar 4:4:4:4 YUV, 12 bits per channel - + planar 4:4:2:2 YUV, 12 bits per channel - + planar 4:4:2:2 YUV, 12 bits per channel - + planar 4:4:2:0 YUV, 12 bits per channel - + planar 4:4:2:0 YUV, 12 bits per channel - + planar 4:4:4:4 YUV, 16 bits per channel - + planar 4:4:4:4 YUV, 16 bits per channel - + planar 4:4:2:2 YUV, 16 bits per channel - + planar 4:4:2:2 YUV, 16 bits per channel - + planar 4:4:2:0 YUV, 16 bits per channel - + planar 4:4:2:0 YUV, 16 bits per channel - + planar 4:4:4 RGB, 16 bits per channel - + planar 4:4:4 RGB, 16 bits per channel - + packed RGB with alpha, 8 bits per channel - + packed 4:2:2 YUV, 16 bits per channel (Y-U-Y-V) - + packed 4:2:2 YUV, 16 bits per channel (Y-U-Y-V) - + packed 4:4:4:4 YUV, 16 bits per channel(U-Y-V-A) - + packed 4:4:4:4 YUV, 16 bits per channel(U-Y-V-A) - + 10-bit grayscale, packed into 16bit words (6 bits left padding) @@ -10531,43 +10883,43 @@ versions were printing a critical warning and returned %NULL. 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. - + The tile size varies per plane according to the subsampling. @@ -11060,40 +11412,40 @@ All video planes of @buffer will be mapped and the pointers will be set in 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. - + The video frame has the top field only. This is the same as GST_VIDEO_FRAME_FLAG_TFF | GST_VIDEO_FRAME_FLAG_ONEFIELD (Since: 1.16). - + The video frame has one field - + The video frame has the bottom field only. This is the same as GST_VIDEO_FRAME_FLAG_ONEFIELD (GST_VIDEO_FRAME_FLAG_TFF flag unset) (Since: 1.16). @@ -11101,52 +11453,52 @@ All video planes of @buffer will be mapped and the pointers will be set in 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 @@ -11229,10 +11581,10 @@ to upload something to an OpenGL texture. - + disable gamma handling - + convert between input and output gamma Different gamma conversion modes @@ -11757,18 +12109,18 @@ valid 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 @@ -11777,7 +12129,7 @@ mode of the stream. height property. This mode requires multiple GstVideoMeta metadata to describe the fields. - + 1 field is stored in one buffer, @GST_VIDEO_BUFFER_FLAG_TF or @GST_VIDEO_BUFFER_FLAG_BF indicates if the buffer is carrying the top or bottom field, respectively. The top and @@ -11962,18 +12314,18 @@ mode of the stream. 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. @@ -12045,6 +12397,7 @@ gst_meta_deserialize(). + map the memory of a plane @@ -12073,6 +12426,7 @@ gst_meta_deserialize(). + unmap the memory of a plane @@ -12254,30 +12608,30 @@ defined in @meta and will fail to update if they are not. GstVideoMultiviewFlags are used to indicate extra properties of a stereo/multiview stream beyond the frame layout and buffer mapping that is conveyed in the #GstVideoMultiviewMode. - + 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 @@ -12288,7 +12642,7 @@ that is conveyed in the #GstVideoMultiviewMode. 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 @@ -12308,42 +12662,42 @@ 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. @@ -12353,60 +12707,60 @@ 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` @@ -12765,6 +13119,7 @@ operation of video-sources or operators. + virtual method to get horizontal flipping state @@ -12784,6 +13139,7 @@ operation of video-sources or operators. + virtual method to get vertical flipping state @@ -12803,6 +13159,7 @@ operation of video-sources or operators. + virtual method to get horizontal centering state @@ -12822,6 +13179,7 @@ operation of video-sources or operators. + virtual method to get vertical centering state @@ -12841,6 +13199,7 @@ operation of video-sources or operators. + virtual method to set horizontal flipping state @@ -12860,6 +13219,7 @@ operation of video-sources or operators. + virtual method to set vertical flipping state @@ -12879,6 +13239,7 @@ operation of video-sources or operators. + virtual method to set horizontal centering state @@ -12898,6 +13259,7 @@ operation of video-sources or operators. + virtual method to set vertical centering state @@ -12919,34 +13281,34 @@ operation of video-sources or operators. The different video orientation methods. - + Identity (no rotation) - + Rotate clockwise 90 degrees - + Rotate 180 degrees - + Rotate counter-clockwise 90 degrees - + Flip horizontally - + Flip vertically - + Rotate counter-clockwise 90 degrees and flip vertically - + Rotate clockwise 90 degrees and flip vertically - + Select flip method based on image-orientation tag - + Current status depends on plugin internal setup @@ -13295,6 +13657,7 @@ from the #GstVideoOverlay. + virtual method to set the render rectangle @@ -13469,7 +13832,7 @@ tell the overlay to stop using that window and create an internal one. - + Functions to create and handle overlay compositions on video buffers. An overlay composition describes one or more overlay rectangles to be @@ -13665,13 +14028,13 @@ contained in the rectangles are not copied. Overlay format flags. - + no flags - + RGB are premultiplied by A/255. - + a global-alpha value != 1 is set. @@ -13683,6 +14046,7 @@ contained in the rectangles are not copied. + virtual method to handle expose events @@ -13697,6 +14061,7 @@ contained in the rectangles are not copied. + virtual method to handle events @@ -13715,6 +14080,7 @@ contained in the rectangles are not copied. + virtual method to set the render rectangle @@ -13740,6 +14106,7 @@ contained in the rectangles are not copied. + virtual method to configure the window handle @@ -13758,7 +14125,7 @@ contained in the rectangles are not copied. - + An opaque video overlay rectangle object. A rectangle contains a single overlay rectangle which can be added to a composition. @@ -14138,17 +14505,17 @@ gst_video_overlay_composition_copy(). 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 slightly 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) @@ -14156,14 +14523,14 @@ gst_video_overlay_composition_copy(). 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 @@ -14374,10 +14741,10 @@ required to perform various kinds of resampling filtering. 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 @@ -14385,21 +14752,21 @@ required to perform various kinds of resampling filtering. 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 @@ -14432,7 +14799,7 @@ required to perform various kinds of resampling filtering. - + #GstVideoScaler is a utility object for rescaling and resampling video frames using various interpolation / sampling methods. @@ -14690,10 +15057,10 @@ on the @method and @in_size/@out_size. Different scale flags. - + no flags - + Set up a scaler for interlaced content @@ -14750,6 +15117,10 @@ observing out-of-sync frames. + render a video frame. Maps to #GstBaseSinkClass.render() and + #GstBaseSinkClass.preroll() vfuncs. Rendering during preroll will be + suppressed if the #GstVideoSink:show-preroll-frame property is set to + %FALSE. @@ -14763,7 +15134,7 @@ observing out-of-sync frames. - + Whether to show video frames during preroll. If set to %FALSE, video frames will only be rendered in PLAYING state. @@ -14797,6 +15168,10 @@ frames will only be rendered in PLAYING state. + render a video frame. Maps to #GstBaseSinkClass.render() and + #GstBaseSinkClass.preroll() vfuncs. Rendering during preroll will be + suppressed if the #GstVideoSink:show-preroll-frame property is set to + %FALSE. @@ -14839,7 +15214,7 @@ frames will only be rendered in PLAYING state. - + @@ -14874,22 +15249,22 @@ when older APIs are being used to expose this format. 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. - + Tiles are in row 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. @@ -15416,13 +15791,13 @@ The configuration of the time code. 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 @@ -15579,70 +15954,70 @@ automatically incremented/interpolated. 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, also ITU-R BT470M / ITU-R BT1700 625 PAL & SECAM / ITU-R BT1361 - + Gamma 2.2 curve with a linear segment in the lower range - + Gamma 2.4 curve with a linear segment in the lower range. IEC 61966-2-1 (sRGB or sYCC) - + Gamma 2.8 curve, also ITU-R BT470BG - + Logarithmic transfer characteristic 100:1 range - + Logarithmic transfer characteristic 316.22777:1 range (100 * sqrt(10) : 1) - + 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 - + Rec. ITU-R BT.2020-2 with 10 bits per component. (functionally the same as the values GST_VIDEO_TRANSFER_BT709 and GST_VIDEO_TRANSFER_BT601). Since: 1.18 - + SMPTE ST 2084 for 10, 12, 14, and 16-bit systems. Known as perceptual quantization (PQ) Since: 1.18 - + Association of Radio Industries and Businesses (ARIB) STD-B67 and Rec. ITU-R BT.2100-1 hybrid loggamma (HLG) system Since: 1.18 - + also known as SMPTE170M / ITU-R BT1358 525 or 625 / ITU-R BT1700 NTSC @@ -15768,7 +16143,7 @@ and "ITU-T H.273 Table 3". - + An encoder for writing ancillary data to the Vertical Blanking Interval lines of component signals. @@ -15872,7 +16247,7 @@ the parity check bits). - + A parser for detecting and extracting @GstVideoAncillary data from Vertical Blanking Interval lines of component signals. @@ -15962,13 +16337,13 @@ data. Return values for #GstVideoVBIParser - + No line were provided, or no more Ancillary data was found. - + A #GstVideoAncillary was found. - + An error occurred diff --git a/girs/GstVulkan-1.0.gir b/girs/GstVulkan-1.0.gir index 3f3122d95b..fb287437cf 100644 --- a/girs/GstVulkan-1.0.gir +++ b/girs/GstVulkan-1.0.gir @@ -864,7 +864,7 @@ and/or use gtk-doc annotations. --> - + @@ -919,14 +919,14 @@ and/or use gtk-doc annotations. --> - + no barrier type - + - + - + @@ -1149,7 +1149,7 @@ multiple times. This must be called before any other #GstVulkanBufferMemory ope - + @@ -1206,7 +1206,7 @@ multiple times. This must be called before any other #GstVulkanBufferMemory ope - + Decreases the refcount of the buffer. If the refcount reaches 0, the buffer will be freed. @@ -1219,7 +1219,7 @@ will be freed. - + @@ -1311,7 +1311,7 @@ need to use this function. - + @@ -1379,7 +1379,7 @@ need to use this function. - + @@ -1485,7 +1485,7 @@ need to use this function. - + @@ -1560,7 +1560,7 @@ need to use this function. - + Decreases the refcount of the buffer. If the refcount reaches 0, the buffer will be freed. @@ -1573,7 +1573,7 @@ will be freed. - + @@ -1743,7 +1743,7 @@ only have an effect before the call to gst_vulkan_device_open(). - + the #GstVulkanInstance used to create this @device @@ -1756,7 +1756,7 @@ only have an effect before the call to gst_vulkan_device_open(). - + The VkPhysicalDevice used to create @device @@ -1891,10 +1891,10 @@ only have an effect before the call to gst_vulkan_device_open(). - + - + @@ -1948,7 +1948,7 @@ only have an effect before the call to gst_vulkan_device_open(). - + @@ -2048,6 +2048,7 @@ surrounding elements of @element. + create a window a new #GstVulkanWindow for @display or @@ -2062,6 +2063,7 @@ surrounding elements of @element. + get the native handle to the display the winsys specific handle of @display @@ -2192,6 +2194,7 @@ second argument is @data. + get the native handle to the display @@ -2207,6 +2210,7 @@ second argument is @data. + create a window @@ -2228,31 +2232,31 @@ second argument is @data. - + - + no display - + XCB display - + wayland display - + cocoa display for macOS - + ios display - + win32 display - + - + any display type @@ -2287,7 +2291,7 @@ second argument is @data. - + undetermined error @@ -2432,19 +2436,19 @@ the error - + is a YUV format - + is a RGB format - + has an alpha channel - + data is stored in little-endiate byte order - + data is stored complex and cannot be read/write only using the information in the #GstVulkanFormatInfo @@ -2550,25 +2554,25 @@ the error - + [0, 2^n - 1] -> [0.0, 1.0] - + [-2^(n-1), 2^(n-1) - 1] -> [-1.0, 1.0] - + [0, 2^n - 1] -> [0.0, float(2^n - 1)] - + [-2^(n-1), 2^(n-1) - 1] -> [float(-2^(n-1)), float(2^(n-1) - 1)] - + [0, 2^n - 1] -> [0, 2^n - 1] - + [-2^(n-1), 2^(n-1) - 1] -> [-2^(n-1), 2^(n-1) - 1] - + @GST_VULKAN_FORMAT_SCALING_UNORM but the first three components are gamma corrected for the sRGB colour space. @@ -2994,7 +2998,7 @@ See also: gst_vulkan_full_screen_quad_set_blend_factors(). - + @@ -3190,7 +3194,7 @@ more specific implementations (#GstVulkanCommandBuffer, #GstVulkanCommandPool).< - + Decreases the refcount of the buffer. If the refcount reaches 0, the buffer will be freed. @@ -3203,7 +3207,7 @@ will be freed. - + @@ -3287,6 +3291,7 @@ inside @handle. + acquire a handle for usage @@ -3298,6 +3303,7 @@ inside @handle. + allocate a new handle @@ -3309,6 +3315,7 @@ inside @handle. + free a handle @@ -3323,6 +3330,7 @@ inside @handle. + release a handle for possible reuse at the next call to @acquire @@ -3405,6 +3413,7 @@ inside @handle. + allocate a new handle @@ -3418,6 +3427,7 @@ inside @handle. + acquire a handle for usage @@ -3431,6 +3441,7 @@ inside @handle. + release a handle for possible reuse at the next call to @acquire @@ -3447,6 +3458,7 @@ inside @handle. + free a handle @@ -3469,34 +3481,34 @@ inside @handle. - + descripter set layout - + pipeline layout - + pipeline - + render pass - + sampler - + framebuffer - + shader - + video session - + video session parameters - + sampler with YCBCR conversion @@ -3642,7 +3654,7 @@ src or DPB images. - + @@ -4029,7 +4041,7 @@ criteria. - + Decreases the refcount of the trash object. If the refcount reaches 0, the trash will be freed. @@ -4042,7 +4054,7 @@ trash will be freed. - + @@ -4366,10 +4378,10 @@ version with this function. - + - + @@ -4413,7 +4425,7 @@ It can be called from any thread. - + @@ -5015,7 +5027,7 @@ gst_vulkan_operation_add_frame_barrier(). - + @@ -5087,7 +5099,7 @@ called. - + The #GstVulkanInstance associated with this physical device @@ -5133,13 +5145,13 @@ called. - + - + - + @@ -5200,7 +5212,7 @@ called. - + @@ -5364,7 +5376,7 @@ See gst_vulkan_queue_submit_lock() for details on when this call is needed. - + @@ -5465,10 +5477,10 @@ See gst_vulkan_queue_submit_lock() for details on when this call is needed. - + - + @@ -5509,7 +5521,7 @@ See gst_vulkan_queue_submit_lock() for details on when this call is needed. - + @@ -5589,7 +5601,7 @@ to call when @fence is signalled. - + Decreases the refcount of the trash object. If the refcount reaches 0, the trash will be freed. @@ -5602,7 +5614,7 @@ trash will be freed. - + A #GstVulkanTrashNotify implementation for unreffing a #GstMiniObject when the associated #GstVulkanFence is signalled @@ -5660,6 +5672,7 @@ associated #GstVulkanFence is signalled + the #GstVulkanTrashListAdd functions whether @trash could be added to @trash_list @@ -5677,6 +5690,7 @@ associated #GstVulkanFence is signalled + the #GstVulkanTrashListGC function @@ -5689,6 +5703,7 @@ associated #GstVulkanFence is signalled + the #GstVulkanTrashListWait function whether all objects were signalled and freed within the @timeout @@ -5881,7 +5896,28 @@ signalled and freed. - + + + + + + + + + + + + + + + + + + + + + + @@ -5983,19 +6019,50 @@ signalled and freed. The type of video operation. - + decode operation - + encode operation - + unknown - + + the generic vulkan video profile + + + + + + + + + + + + + + + + + + + + + + + + + + + + + + @@ -6100,6 +6167,7 @@ provided api. + retrieve whether this window supports presentation whether the given combination of @window, @device and @@ -6122,6 +6190,7 @@ provided api. + retrieve the vulkan surface for this window the VkSurface for displaying into. The caller is responsible for @@ -6136,6 +6205,7 @@ provided api. + retrieve the current size of the window @@ -6177,6 +6247,7 @@ from the @window. + open the connection to the display whether @window could be successfully opened @@ -6190,6 +6261,7 @@ from the @window. + set the external window handle to render into @@ -6216,7 +6288,7 @@ from the @window. - + the #GstVulkanDisplay for @window @@ -6409,7 +6481,7 @@ Currently intended for subclasses to update internal state. - + @@ -6494,6 +6566,7 @@ Currently intended for subclasses to update internal state. + open the connection to the display @@ -6509,6 +6582,7 @@ Currently intended for subclasses to update internal state. + close the connection to the display @@ -6523,6 +6597,7 @@ Currently intended for subclasses to update internal state. + retrieve the vulkan surface for this window @@ -6539,6 +6614,7 @@ Currently intended for subclasses to update internal state. + retrieve whether this window supports presentation @@ -6563,6 +6639,7 @@ Currently intended for subclasses to update internal state. + set the external window handle to render into @@ -6579,6 +6656,7 @@ Currently intended for subclasses to update internal state. + retrieve the current size of the window @@ -6625,13 +6703,13 @@ Currently intended for subclasses to update internal state. - + failed - + old libraries - + resource unavailable @@ -6640,7 +6718,7 @@ Currently intended for subclasses to update internal state. - + diff --git a/girs/GstWebRTC-1.0.gir b/girs/GstWebRTC-1.0.gir index e930006605..76a9410d9a 100644 --- a/girs/GstWebRTC-1.0.gir +++ b/girs/GstWebRTC-1.0.gir @@ -333,71 +333,71 @@ and/or use gtk-doc annotations. --> See https://tools.ietf.org/html/draft-ietf-rtcweb-jsep-24#section-4.1.1 for more information. - + none - + balanced - + max-compat - + max-bundle - + none - + actpass - + sendonly - + recvonly - + - + - + - + - + - + - + new - + closed - + failed - + connecting - + connected @@ -486,37 +486,37 @@ for more information. - + - + - + - + - + - + - + - + - + - + @@ -596,57 +596,57 @@ for more information. - + See <http://w3c.github.io/webrtc-pc/#dom-rtcdatachannelstate> - + connecting - + open - + closing - + closed See <https://www.w3.org/TR/webrtc/#dom-rtcerrordetailtype> for more information. - + data-channel-failure - + dtls-failure - + fingerprint-failure - + sctp-failure - + sdp-syntax-error - + hardware-encoder-not-available - + encoder-error - + invalid-state (part of WebIDL specification) - + GStreamer-specific failure, not matching any other value from the specification - + invalid-modification (part of WebIDL specification) - + type-error (maps to JavaScript TypeError) @@ -656,10 +656,10 @@ for more information. - + none - + ulpfec + red @@ -1436,12 +1436,12 @@ Get HTTP Proxy to be used when connecting to TURN server. - + Maximum port for local rtp port range. min-rtp-port must be <= max-rtp-port - + Minimum port for local rtp port range. min-rtp-port must be <= max-rtp-port @@ -1973,46 +1973,46 @@ Get HTTP Proxy to be used when connecting to TURN server. - + RTP component - + RTCP component See <http://w3c.github.io/webrtc-pc/#dom-rtciceconnectionstate> - + new - + checking - + connected - + completed - + failed - + disconnected - + closed See <http://w3c.github.io/webrtc-pc/#dom-rtcicegatheringstate> - + new - + gathering - + complete @@ -2042,10 +2042,10 @@ Get HTTP Proxy to be used when connecting to TURN server. - + controlled - + controlling @@ -2111,7 +2111,7 @@ Get HTTP Proxy to be used when connecting to TURN server. - + @@ -2233,13 +2233,13 @@ Get HTTP Proxy to be used when connecting to TURN server. - + - + - + @@ -2311,58 +2311,58 @@ Get HTTP Proxy to be used when connecting to TURN server. See https://tools.ietf.org/html/draft-ietf-rtcweb-jsep-24#section-4.1.1 for more information. - + all - + relay https://w3c.github.io/mediacapture-main/#dom-mediastreamtrack-kind - + Kind has not yet been set - + Kind is audio - + Kind is video See <http://w3c.github.io/webrtc-pc/#dom-rtcpeerconnectionstate> - + new - + connecting - + connected - + disconnected - + failed - + closed See <http://w3c.github.io/webrtc-pc/#dom-rtcprioritytype> - + very-low - + low - + medium - + high @@ -2376,7 +2376,7 @@ Mostly matches the WebRTC RTCRtpReceiver interface. - + @@ -2384,7 +2384,7 @@ Mostly matches the WebRTC RTCRtpReceiver interface. Mostly matches the WebRTC RTCRtpSender interface. - + Sets the content of the IPv4 Type of Service (ToS), also known as DSCP (Differentiated Services Code Point). This also sets the Traffic Class field of IPv6. @@ -2403,7 +2403,7 @@ This also sets the Traffic Class field of IPv6. - + The priority from which to set the DSCP field on packets @@ -2412,7 +2412,7 @@ This also sets the Traffic Class field of IPv6. - + @@ -2422,22 +2422,22 @@ This also sets the Traffic Class field of IPv6. Caps representing the codec preferences. - + The transceiver's current directionality, or none if the transceiver is stopped or has never participated in an exchange of offers and answers. To change the transceiver's directionality, set the value of the direction property. - + Direction of the transceiver. - + The kind of media this transceiver transports - + The media ID of the m-line associated with this transceiver. This association is established, when possible, whenever either a local or remote description is applied. This field is null if @@ -2446,7 +2446,7 @@ associated m-line is rejected by either a remote offer or any answer. - + @@ -2456,71 +2456,71 @@ answer. - + - + none - + inactive - + sendonly - + recvonly - + sendrecv - + - + - + - + See <http://w3c.github.io/webrtc-pc/#dom-rtcsctptransportstate> - + new - + connecting - + connected - + closed See <http://w3c.github.io/webrtc-pc/#rtcsdptype> - + offer - + pranswer - + answer - + rollback @@ -2596,67 +2596,67 @@ answer. See <http://w3c.github.io/webrtc-pc/#dom-rtcsignalingstate> - + stable - + closed - + have-local-offer - + have-remote-offer - + have-local-pranswer - + have-remote-pranswer See <https://w3c.github.io/webrtc-stats/#dom-rtcstatstype> - + codec - + inbound-rtp - + outbound-rtp - + remote-inbound-rtp - + remote-outbound-rtp - + csrc - + peer-connection - + data-channel - + stream - + transport - + candidate-pair - + local-candidate - + remote-candidate - + certificate