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Merge branch 'master' into 0.11

Browse source 
Conflicts:
	gst-libs/gst/video/convertframe.c
This commit is contained in:
Wim Taymans 2011-05-24 09:47:15 +02:00
commit a87c021237
36 changed files with 500 additions and 288 deletions
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2
docs/libs/gst-plugins-base-libs-docs.sgml
View file

@ -189,6 +189,7 @@
<xi:include href="xml/gsttagvorbis.xml" />
<xi:include href="xml/gsttagid3.xml" />
<xi:include href="xml/gsttagxmp.xml" />
<xi:include href="xml/gsttagxmpwriter.xml" />
<xi:include href="xml/gsttagdemux.xml" />
<xi:include href="xml/gsttaglanguagecodes.xml" />
</chapter>
@ -202,6 +203,7 @@
flags.
</para>
<xi:include href="xml/gstpbutils.xml" />
<xi:include href="xml/gstpluginsbaseversion.xml" />
<xi:include href="xml/gstpbutilscodecutils.xml" />
<xi:include href="xml/gstpbutilsdescriptions.xml" />
<xi:include href="xml/gstpbutilsmissingplugins.xml" />

85
docs/libs/gst-plugins-base-libs-sections.txt
View file

@ -3,7 +3,6 @@
<FILE>gstappsrc</FILE>
<TITLE>appsrc</TITLE>
<INCLUDE>gst/app/gstappsrc.h</INCLUDE>
GstAppSrc
GstAppStreamType
gst_app_src_set_caps
gst_app_src_get_caps
@ -35,7 +34,9 @@ GST_APP_BUFFER_CLASS
GST_IS_APP_BUFFER
GST_IS_APP_BUFFER_CLASS
GST_TYPE_APP_BUFFER
GST_TYPE_APP_STREAM_TYPE
<SUBSECTION Private>
GstAppSrc
GstAppSrcPrivate
GstAppBuffer
GstAppBufferClass
@ -48,7 +49,6 @@ gst_app_buffer_new
<FILE>gstappsink</FILE>
<TITLE>appsink</TITLE>
<INCLUDE>gst/app/gstappsink.h</INCLUDE>
GstAppSink
gst_app_sink_set_caps
gst_app_sink_get_caps
gst_app_sink_is_eos
@ -64,6 +64,7 @@ gst_app_sink_pull_buffer_list
GstAppSinkCallbacks
gst_app_sink_set_callbacks
<SUBSECTION Standard>
GstAppSink
GstAppSinkPrivate
GstAppSinkClass
GST_APP_SINK
@ -859,6 +860,22 @@ gst_netbuffer_get_type
<SECTION>
<FILE>gstriff</FILE>
<INCLUDE>gst/riff/riff-media.h</INCLUDE>
gst_riff_create_audio_caps
gst_riff_create_audio_template_caps
gst_riff_create_iavs_caps
gst_riff_create_iavs_template_caps
gst_riff_create_video_caps
gst_riff_create_video_template_caps
gst_riff_init
gst_riff_parse_chunk
gst_riff_parse_file_header
gst_riff_parse_info
gst_riff_parse_strf_auds
gst_riff_parse_strf_iavs
gst_riff_parse_strf_vids
gst_riff_parse_strh
gst_riff_read_chunk
<SUBSECTION Standard>
GST_RIFF_00
GST_RIFF_0021
GST_RIFF_0031
@ -1043,28 +1060,12 @@ GST_RIFF_yuy2
GST_RIFF_yv12
gst_riff_acid
gst_riff_create_audio_caps
gst_riff_create_audio_template_caps
gst_riff_create_iavs_caps
gst_riff_create_iavs_template_caps
gst_riff_create_video_caps
gst_riff_create_video_template_caps
gst_riff_dmlh
gst_riff_index_entry
gst_riff_init
gst_riff_parse_chunk
gst_riff_parse_file_header
gst_riff_parse_info
gst_riff_parse_strf_auds
gst_riff_parse_strf_iavs
gst_riff_parse_strf_vids
gst_riff_parse_strh
gst_riff_read_chunk
gst_riff_strf_auds
gst_riff_strf_iavs
gst_riff_strf_vids
gst_riff_strh
<SUBSECTION Standard>
</SECTION>
@ -1392,6 +1393,7 @@ gst_rtp_buffer_list_add_extension_twobytes_header
<FILE>gstrtspdefs</FILE>
<INCLUDE>gst/rtsp/gstrtspdefs.h</INCLUDE>
GST_RTSP_CHECK
GST_RTSP_AUTH_MAX
GstRTSPEvent
GstRTSPResult
GstRTSPFamily
@ -1399,7 +1401,6 @@ GstRTSPState
GstRTSPVersion
GstRTSPMethod
GstRTSPAuthMethod
GST_RTSP_AUTH_MAX
GstRTSPHeaderField
GstRTSPStatusCode
gst_rtsp_strresult
@ -1739,6 +1740,7 @@ GST_TAG_CAPTURING_FLASH_FIRED
GST_TAG_CAPTURING_FLASH_MODE
GST_TAG_CAPTURING_METERING_MODE
GST_TAG_CAPTURING_SOURCE
GST_TAG_CAPTURING_EXPOSURE_COMPENSATION
GST_TAG_IMAGE_HORIZONTAL_PPI
GST_TAG_IMAGE_VERTICAL_PPI
gst_tag_register_musicbrainz_tags
@ -1786,6 +1788,8 @@ gst_tag_list_add_id3_image
<INCLUDE>gst/tag/tag.h</INCLUDE>
gst_tag_list_from_xmp_buffer
gst_tag_list_to_xmp_buffer
gst_tag_list_to_xmp_buffer_full
gst_tag_xmp_list_schemas
<SUBSECTION Standard>
</SECTION>
@ -1820,7 +1824,6 @@ gst_tag_demux_result_get_type
<SECTION>
<FILE>gsttaglanguagecodes</FILE>
<INCLUDE>gst/tag/tag.h</INCLUDE>
<SUBSECTION>
gst_tag_get_language_codes
gst_tag_get_language_name
gst_tag_get_language_code
@ -1829,6 +1832,26 @@ gst_tag_get_language_code_iso_639_2B
gst_tag_get_language_code_iso_639_2T
</SECTION>
<SECTION>
<FILE>gsttagxmpwriter</FILE>
gst_tag_xmp_writer_add_all_schemas
gst_tag_xmp_writer_add_schema
gst_tag_xmp_writer_has_schema
gst_tag_xmp_writer_remove_schema
gst_tag_xmp_writer_remove_all_schemas
gst_tag_xmp_writer_tag_list_to_xmp_buffer
<SUBSECTION Standard>
GstTagXmpWriter
GstTagXmpWriterInterface
GST_TYPE_TAG_XMP_WRITER
GST_TAG_XMP_WRITER
GST_TAG_XMP_WRITER_INTERFACE
GST_IS_TAG_XMP_WRITER
GST_IS_TAG_XMP_WRITER_INTERFACE
GST_TAG_XMP_WRITER_GET_INTERFACE
gst_tag_xmp_writer_get_type
</SECTION>
# base utils
<SECTION>
@ -1836,6 +1859,11 @@ gst_tag_get_language_code_iso_639_2T
<INCLUDE>gst/pbutils/pbutils.h</INCLUDE>
<SUBSECTION>
gst_pb_utils_init
</SECTION>
<SECTION>
<FILE>gstpluginsbaseversion</FILE>
<INCLUDE>gst/pbutils/gstpluginsbaseversion.h</INCLUDE>
<SUBSECTION>
GST_PLUGINS_BASE_VERSION_MAJOR
GST_PLUGINS_BASE_VERSION_MINOR
@ -2054,6 +2082,20 @@ GST_VIDEO_CAPS_BGR_16
GST_VIDEO_CAPS_RGB8_PALETTED
GST_VIDEO_CAPS_GRAY8
GST_VIDEO_CAPS_GRAY16
GST_VIDEO_CAPS_ARGB_64
GST_VIDEO_CAPS_r210
GST_VIDEO_COMP1_MASK_15
GST_VIDEO_COMP1_MASK_15_INT
GST_VIDEO_COMP1_MASK_16
GST_VIDEO_COMP1_MASK_16_INT
GST_VIDEO_COMP2_MASK_15
GST_VIDEO_COMP2_MASK_15_INT
GST_VIDEO_COMP2_MASK_16
GST_VIDEO_COMP2_MASK_16_INT
GST_VIDEO_COMP3_MASK_15
GST_VIDEO_COMP3_MASK_15_INT
GST_VIDEO_COMP3_MASK_16
GST_VIDEO_COMP3_MASK_16_INT
GST_VIDEO_FPS_RANGE
GST_VIDEO_GREEN_MASK_15
GST_VIDEO_GREEN_MASK_15_INT
@ -2071,12 +2113,15 @@ GstVideoFormat
gst_video_calculate_display_ratio
gst_video_frame_rate
gst_video_get_size
gst_video_get_size_from_caps
gst_video_format_convert
gst_video_format_new_caps
gst_video_format_new_caps_interlaced
gst_video_format_new_template_caps
gst_video_format_get_component_height
gst_video_format_get_component_offset
gst_video_format_get_component_width
gst_video_format_get_component_depth
gst_video_format_get_pixel_stride
gst_video_format_get_row_stride
gst_video_format_get_size

54
gst-libs/gst/app/gstappsink.c
View file

@ -17,22 +17,6 @@
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
/**
* SECTION:element-appsink
*
* Appsink is a sink plugin that supports many different methods for making
* the application get a handle on the GStreamer data in a pipeline. Unlike
* most GStreamer elements, Appsink provides external API functions.
*
* For the documentation of the API, please see the
* <link linkend="gst-plugins-base-libs-appsink">libgstapp</link> section in
* the GStreamer Plugins Base Libraries documentation.
*
* Since: 0.10.22
*/
/**
* SECTION:gstappsink
* @short_description: Easy way for applications to extract buffers from a
@ -281,10 +265,10 @@ gst_app_sink_class_init (GstAppSinkClass * klass)
* GstAppSink::new-preroll:
* @appsink: the appsink element that emited the signal
*
* Signal that a new preroll buffer is available.
* Signal that a new preroll buffer is available.
*
* This signal is emited from the steaming thread and only when the
* "emit-signals" property is %TRUE.
* "emit-signals" property is %TRUE.
*
* The new preroll buffer can be retrieved with the "pull-preroll" action
* signal or gst_app_sink_pull_preroll() either from this signal callback
@ -304,7 +288,7 @@ gst_app_sink_class_init (GstAppSinkClass * klass)
* Signal that a new buffer is available.
*
* This signal is emited from the steaming thread and only when the
* "emit-signals" property is %TRUE.
* "emit-signals" property is %TRUE.
*
* The new buffer can be retrieved with the "pull-buffer" action
* signal or gst_app_sink_pull_buffer() either from this signal callback
@ -324,7 +308,7 @@ gst_app_sink_class_init (GstAppSinkClass * klass)
* Signal that a new bufferlist is available.
*
* This signal is emited from the steaming thread and only when the
* "emit-signals" property is %TRUE.
* "emit-signals" property is %TRUE.
*
* The new buffer can be retrieved with the "pull-buffer-list" action
* signal or gst_app_sink_pull_buffer_list() either from this signal callback
@ -354,10 +338,10 @@ gst_app_sink_class_init (GstAppSinkClass * klass)
* when calling gst_app_sink_pull_buffer() or the "pull-buffer" action signal.
*
* If an EOS event was received before any buffers, this function returns
* %NULL. Use gst_app_sink_is_eos () to check for the EOS condition.
* %NULL. Use gst_app_sink_is_eos () to check for the EOS condition.
*
* This function blocks until a preroll buffer or EOS is received or the appsink
* element is set to the READY/NULL state.
* element is set to the READY/NULL state.
*
* Returns: a #GstBuffer or NULL when the appsink is stopped or EOS.
*/
@ -371,11 +355,11 @@ gst_app_sink_class_init (GstAppSinkClass * klass)
* @appsink: the appsink element to emit this signal on
*
* This function blocks until a buffer or EOS becomes available or the appsink
* element is set to the READY/NULL state.
* element is set to the READY/NULL state.
*
* This function will only return buffers when the appsink is in the PLAYING
* state. All rendered buffers will be put in a queue so that the application
* can pull buffers at its own rate.
* can pull buffers at its own rate.
*
* Note that when the application does not pull buffers fast enough, the
* queued buffers could consume a lot of memory, especially when dealing with
@ -383,7 +367,7 @@ gst_app_sink_class_init (GstAppSinkClass * klass)
* the "drop" and "max-buffers" properties.
*
* If an EOS event was received before any buffers, this function returns
* %NULL. Use gst_app_sink_is_eos () to check for the EOS condition.
* %NULL. Use gst_app_sink_is_eos () to check for the EOS condition.
*
* Returns: a #GstBuffer or NULL when the appsink is stopped or EOS.
*/
@ -397,11 +381,11 @@ gst_app_sink_class_init (GstAppSinkClass * klass)
* @appsink: the appsink element to emit this signal on
*
* This function blocks until a buffer list or EOS becomes available or the appsink
* element is set to the READY/NULL state.
* element is set to the READY/NULL state.
*
* This function will only return bufferlists when the appsink is in the PLAYING
* state. All rendered bufferlists will be put in a queue so that the application
* can pull bufferlists at its own rate.
* can pull bufferlists at its own rate.
*
* Note that when the application does not pull bufferlists fast enough, the
* queued bufferlists could consume a lot of memory, especially when dealing with
@ -409,7 +393,7 @@ gst_app_sink_class_init (GstAppSinkClass * klass)
* the "drop" and "max-buffers" properties.
*
* If an EOS event was received before any buffers, this function returns
* %NULL. Use gst_app_sink_is_eos () to check for the EOS condition.
* %NULL. Use gst_app_sink_is_eos () to check for the EOS condition.
*
* Returns: a #GstBufferList or NULL when the appsink is stopped or EOS.
*/
@ -924,7 +908,7 @@ not_started:
* 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, you must check the caps
* on the buffers to get the actual used caps.
* on the buffers to get the actual used caps.
*
* Since: 0.10.22
*/
@ -1209,10 +1193,10 @@ gst_app_sink_get_drop (GstAppSink * appsink)
* when calling gst_app_sink_pull_buffer().
*
* If an EOS event was received before any buffers, this function returns
* %NULL. Use gst_app_sink_is_eos () to check for the EOS condition.
* %NULL. Use gst_app_sink_is_eos () to check for the EOS condition.
*
* This function blocks until a preroll buffer or EOS is received or the appsink
* element is set to the READY/NULL state.
* element is set to the READY/NULL state.
*
* Returns: a #GstBuffer or NULL when the appsink is stopped or EOS.
*
@ -1271,7 +1255,7 @@ not_started:
* @appsink: a #GstAppSink
*
* This function blocks until a buffer or EOS becomes available or the appsink
* element is set to the READY/NULL state.
* element is set to the READY/NULL state.
*
* This function will only return buffers when the appsink is in the PLAYING
* state. All rendered buffers will be put in a queue so that the application
@ -1280,7 +1264,7 @@ not_started:
* especially when dealing with raw video frames.
*
* If an EOS event was received before any buffers, this function returns
* %NULL. Use gst_app_sink_is_eos () to check for the EOS condition.
* %NULL. Use gst_app_sink_is_eos () to check for the EOS condition.
*
* Returns: a #GstBuffer or NULL when the appsink is stopped or EOS.
*
@ -1299,7 +1283,7 @@ gst_app_sink_pull_buffer (GstAppSink * appsink)
* @appsink: a #GstAppSink
*
* This function blocks until a buffer list or EOS becomes available or the
* appsink element is set to the READY/NULL state.
* appsink element is set to the READY/NULL state.
*
* This function will only return buffer lists when the appsink is in the
* PLAYING state. All rendered buffer lists will be put in a queue so that
@ -1309,7 +1293,7 @@ gst_app_sink_pull_buffer (GstAppSink * appsink)
* video frames.
*
* If an EOS event was received before any buffer lists, this function returns
* %NULL. Use gst_app_sink_is_eos () to check for the EOS condition.
* %NULL. Use gst_app_sink_is_eos () to check for the EOS condition.
*
* Returns: a #GstBufferList or NULL when the appsink is stopped or EOS.
*/

51
gst-libs/gst/app/gstappsrc.c
View file

@ -17,21 +17,6 @@
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
/**
* SECTION:element-appsrc
*
* The appsrc element can be used by applications to insert data into a
* GStreamer pipeline. Unlike most GStreamer elements, Appsrc provides
* external API functions.
*
* For the documentation of the API, please see the
* <link linkend="gst-plugins-base-libs-appsrc">libgstapp</link> section in the
* GStreamer Plugins Base Libraries documentation.
*
* Since: 0.10.22
*/
/**
* SECTION:gstappsrc
* @short_description: Easy way for applications to inject buffers into a
@ -95,7 +80,7 @@
* For the stream and seekable modes, setting this property is optional but
* recommended.
*
* When the application is finished pushing data into appsrc, it should call
* When the application is finished pushing data into appsrc, it should call
* gst_app_src_end_of_stream() or emit the end-of-stream action signal. After
* this call, no more buffers can be pushed into appsrc until a flushing seek
* happened or the state of the appsrc has gone through READY.
@ -467,7 +452,7 @@ gst_app_src_class_init (GstAppSrcClass * klass)
* GstAppSrc::end-of-stream:
* @appsrc: the appsrc
*
* Notify @appsrc that no more buffer are available.
* Notify @appsrc that no more buffer are available.
*/
gst_app_src_signals[SIGNAL_END_OF_STREAM] =
g_signal_new ("end-of-stream", G_TYPE_FROM_CLASS (klass),
@ -1063,7 +1048,7 @@ seek_error:
* a copy of the caps structure. After calling this method, the source will
* only produce caps that match @caps. @caps must be fixed and the caps on the
* buffers must match the caps or left NULL.
*
*
* Since: 0.10.22
*/
void
@ -1096,7 +1081,7 @@ gst_app_src_set_caps (GstAppSrc * appsrc, const GstCaps * caps)
* Get the configured caps on @appsrc.
*
* Returns: the #GstCaps produced by the source. gst_caps_unref() after usage.
*
*
* Since: 0.10.22
*/
GstCaps *
@ -1124,8 +1109,8 @@ gst_app_src_get_caps (GstAppSrc * appsrc)
* @size: the size to set
*
* Set the size of the stream in bytes. A value of -1 means that the size is
* not known.
*
* not known.
*
* Since: 0.10.22
*/
void
@ -1148,10 +1133,10 @@ gst_app_src_set_size (GstAppSrc * appsrc, gint64 size)
* @appsrc: a #GstAppSrc
*
* Get the size of the stream in bytes. A value of -1 means that the size is
* not known.
* not known.
*
* Returns: the size of the stream previously set with gst_app_src_set_size();
*
*
* Since: 0.10.22
*/
gint64
@ -1180,8 +1165,8 @@ gst_app_src_get_size (GstAppSrc * appsrc)
* Set the stream type on @appsrc. For seekable streams, the "seek" signal must
* be connected to.
*
* A stream_type stream
*
* A stream_type stream
*
* Since: 0.10.22
*/
void
@ -1207,7 +1192,7 @@ gst_app_src_set_stream_type (GstAppSrc * appsrc, GstAppStreamType type)
* with gst_app_src_set_stream_type().
*
* Returns: the stream type.
*
*
* Since: 0.10.22
*/
GstAppStreamType
@ -1236,7 +1221,7 @@ gst_app_src_get_stream_type (GstAppSrc * appsrc)
* 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.
*
*
* Since: 0.10.22
*/
void
@ -1265,7 +1250,7 @@ gst_app_src_set_max_bytes (GstAppSrc * appsrc, guint64 max)
* Get the maximum amount of bytes that can be queued in @appsrc.
*
* Returns: The maximum amount of bytes that can be queued.
*
*
* Since: 0.10.22
*/
guint64
@ -1319,7 +1304,7 @@ gst_app_src_set_latencies (GstAppSrc * appsrc, gboolean do_min, guint64 min,
*
* Configure the @min and @max latency in @src. If @min is set to -1, the
* default latency calculations for pseudo-live sources will be used.
*
*
* Since: 0.10.22
*/
void
@ -1335,7 +1320,7 @@ gst_app_src_set_latency (GstAppSrc * appsrc, guint64 min, guint64 max)
* @max: the min latency
*
* Retrieve the min and max latencies in @min and @max respectively.
*
*
* Since: 0.10.22
*/
void
@ -1511,7 +1496,7 @@ eos:
* Returns: #GST_FLOW_OK when the buffer was successfuly queued.
* #GST_FLOW_WRONG_STATE when @appsrc is not PAUSED or PLAYING.
* #GST_FLOW_UNEXPECTED when EOS occured.
*
*
* Since: 0.10.22
*/
GstFlowReturn
@ -1537,7 +1522,7 @@ gst_app_src_push_buffer_action (GstAppSrc * appsrc, GstBuffer * buffer)
*
* Returns: #GST_FLOW_OK when the EOS was successfuly queued.
* #GST_FLOW_WRONG_STATE when @appsrc is not PAUSED or PLAYING.
*
*
* Since: 0.10.22
*/
GstFlowReturn
@ -1550,7 +1535,7 @@ gst_app_src_end_of_stream (GstAppSrc * appsrc)
priv = appsrc->priv;
g_mutex_lock (priv->mutex);
/* can't accept buffers when we are flushing. We can accept them when we are
/* can't accept buffers when we are flushing. We can accept them when we are
* EOS although it will not do anything. */
if (priv->flushing)
goto flushing;

3
gst-libs/gst/audio/gstaudioclock.h
View file

@ -60,14 +60,13 @@ typedef GstClockTime (*GstAudioClockGetTimeFunc) (GstClock *clock, gpointer user
/**
* GstAudioClock:
* @clock: parent #GstSystemClock
*
* Opaque #GstAudioClock.
*/
struct _GstAudioClock {
GstSystemClock clock;
/* --- protected --- */
/*< protected >*/
GstAudioClockGetTimeFunc func;
gpointer user_data;

1
gst-libs/gst/audio/gstaudiofilter.h
View file

@ -47,7 +47,6 @@ typedef struct _GstAudioFilterClass GstAudioFilterClass;
/**
* GstAudioFilter:
* @basetransform: Element parent class
*
* Base class for audio filters with the same format for input and output.
*

26
gst-libs/gst/audio/gstaudioiec61937.c
View file

@ -22,11 +22,11 @@
/**
* SECTION:gstaudioiec61937
* @short_description: Utility functions for IEC 61937 payloading
* @since: 0.10.35
*
* This module contains some helper functions for encapsulating various
* audio formats in IEC 61937 headers and padding.
*
* Since: 0.10.35
*/
#ifdef HAVE_CONFIG_H
@ -60,12 +60,14 @@ caps_get_string_field (const GstCaps * caps, const gchar * field)
}
/**
* gst_audio_iec61937_frame_size
* @type: the type of data to be payloaded as a #GstBufferFormatType
* gst_audio_iec61937_frame_size:
* @spec: the ringbufer spec
*
* Returns 0 if the given @type is not supported or cannot be payloaded, else
* returns the size of the buffer expected by gst_audio_iec61937_payload() for
* payloading @type.
* Calculated the size of the buffer expected by gst_audio_iec61937_payload() for
* payloading type from @spec.
*
* Returns: the size or 0 if the given @type is not supported or cannot be
* payloaded.
*
* Since: 0.10.35
*/
@ -128,19 +130,19 @@ gst_audio_iec61937_frame_size (const GstRingBufferSpec * spec)
}
/**
* gst_audio_iec61937_payload
* gst_audio_iec61937_payload:
* @src: a buffer containing the data to payload
* @src_n: size of @src in bytes
* @dst: the destination buffer to store the payloaded contents in. Should not
* overlap with @src
* @dst_n: size of @dst in bytes
* @type: the type of data in @src
* @spec: the ringbufer spec for @src
*
* Payloads @src in the form specified by IEC 61937 for @type and stores
* the result in @dst. @src must contain exactly one frame of data and the
* frame is not checked for errors.
* Payloads @src in the form specified by IEC 61937 for the type from @spec and
* stores the result in @dst. @src must contain exactly one frame of data and
* the frame is not checked for errors.
*
* Returns: transfer-full: #TRUE if the payloading was successful, #FALSE
* Returns: transfer-full: %TRUE if the payloading was successful, %FALSE
* otherwise.
*
* Since: 0.10.35

7
gst-libs/gst/audio/gstaudioiec61937.h
View file

@ -18,13 +18,6 @@
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
/**
* SECTION:gstaudioiec61937
* @short_description: Utility functions for IEC 61937 payloading
*
* This module contains some helper functions for encapsulating various
* audio formats in IEC 61937 headers and padding.
*/
#ifndef __GST_AUDIO_IEC61937_H__
#define __GST_AUDIO_IEC61937_H__

1
gst-libs/gst/audio/gstaudiosrc.h
View file

@ -40,7 +40,6 @@ typedef struct _GstAudioSrcClass GstAudioSrcClass;
/**
* GstAudioSrc:
* @element: parent class
*
* Base class for simple audio sources.
*/

8
gst-libs/gst/audio/gstbaseaudiosink.c
View file

@ -704,6 +704,7 @@ gst_base_audio_sink_setcaps (GstBaseSink * bsink, GstCaps * caps)
GstBaseAudioSink *sink = GST_BASE_AUDIO_SINK (bsink);
GstRingBufferSpec *spec;
GstClockTime now;
GstClockTime crate_num, crate_denom;
if (!sink->ringbuffer)
return FALSE;
@ -745,6 +746,13 @@ gst_base_audio_sink_setcaps (GstBaseSink * bsink, GstCaps * caps)
gst_ring_buffer_activate (sink->ringbuffer, TRUE);
}
/* due to possible changes in the spec file we should recalibrate the clock */
gst_clock_get_calibration (sink->provided_clock, NULL, NULL,
&crate_num, &crate_denom);
gst_clock_set_calibration (sink->provided_clock,
gst_clock_get_internal_time (sink->provided_clock), now, crate_num,
crate_denom);
/* calculate actual latency and buffer times.
* FIXME: In 0.11, store the latency_time internally in ns */
spec->latency_time = gst_util_uint64_scale (spec->segsize,

47
gst-libs/gst/audio/gstringbuffer.h
View file

@ -116,6 +116,53 @@ typedef enum
GST_BUFTYPE_MPEG4_AAC,
} GstBufferFormatType;
/**
* GstBufferFormat:
* @GST_UNKNOWN: unspecified
* @GST_S8: integer signed 8 bit
* @GST_U8: integer unsigned 8 bit
* @GST_S16_LE: integer signed 16 bit little endian
* @GST_S16_BE: integer signed 16 bit big endian
* @GST_U16_LE: integer unsigned 16 bit little endian
* @GST_U16_BE: integer unsigned 16 bit big endian
* @GST_S24_LE: integer signed 24 bit little endian
* @GST_S24_BE: integer signed 24 bit big endian
* @GST_U24_LE: integer unsigned 24 bit little endian
* @GST_U24_BE: integer unsigned 24 bit big endian
* @GST_S32_LE: integer signed 32 bit little endian
* @GST_S32_BE: integer signed 32 bit big endian
* @GST_U32_LE: integer unsigned 32 bit little endian
* @GST_U32_BE: integer unsigned 32 bit big endian
* @GST_S24_3LE: integer signed 24 bit little endian packed in 3 bytes
* @GST_S24_3BE: integer signed 24 bit big endian packed in 3 bytes
* @GST_U24_3LE: integer unsigned 24 bit little endian packed in 3 bytes
* @GST_U24_3BE: integer unsigned 24 bit big endian packed in 3 bytes
* @GST_S20_3LE: integer signed 20 bit little endian packed in 3 bytes
* @GST_S20_3BE: integer signed 20 bit big endian packed in 3 bytes
* @GST_U20_3LE: integer unsigned 20 bit little endian packed in 3 bytes
* @GST_U20_3BE: integer unsigned 20 bit big endian packed in 3 bytes
* @GST_S18_3LE: integer signed 18 bit little endian packed in 3 bytes
* @GST_S18_3BE: integer signed 18 bit big endian packed in 3 bytes
* @GST_U18_3LE: integer unsigned 18 bit little endian packed in 3 bytes
* @GST_U18_3BE: integer unsigned 18 bit big endian packed in 3 bytes
* @GST_FLOAT32_LE: floating 32 bit little endian
* @GST_FLOAT32_BE: floating 32 bit big endian
* @GST_FLOAT64_LE: floating 64 bit little endian
* @GST_FLOAT64_BE: floating 64 bit big endian
* @GST_MU_LAW: mu-law
* @GST_A_LAW: a-law
* @GST_IMA_ADPCM: ima adpcm
* @GST_MPEG: mpeg audio (but not aac)
* @GST_GSM: gsm
* @GST_IEC958: IEC958 frames
* @GST_AC3: ac3
* @GST_EAC3: eac3
* @GST_DTS: dts
* @GST_MPEG2_AAC: mpeg-2 aac
* @GST_MPEG4_AAC: mpeg-4 aac
*
* The detailed format of the samples in the ringbuffer.
*/
typedef enum
{
GST_UNKNOWN,

4
gst-libs/gst/audio/multichannel.h
View file

@ -42,6 +42,9 @@ G_BEGIN_DECLS
* @GST_AUDIO_CHANNEL_POSITION_NONE: used for position-less channels, e.g.
* from a sound card that records 1024 channels; mutually exclusive with
* any other channel position
* @GST_AUDIO_CHANNEL_POSITION_INVALID: invalid position
*
* Audio channel positions.
*/
typedef enum {
GST_AUDIO_CHANNEL_POSITION_INVALID = -1,
@ -74,6 +77,7 @@ typedef enum {
* are defined or all positions are undefined, but can't mix'n'match */
GST_AUDIO_CHANNEL_POSITION_NONE,
/*< private >*/
/* don't use - counter */
GST_AUDIO_CHANNEL_POSITION_NUM
} GstAudioChannelPosition;

11
gst-libs/gst/cdda/gstcddabasesrc.h
View file

@ -115,6 +115,17 @@ struct _GstCddaBaseSrc {
gpointer _gst_reserved2[GST_PADDING/2];
};
/**
* GstCddaBaseSrcClass:
* @pushsrc_class: the parent class
* @open: opening the device
* @close: closing the device
* @read_sector: reading a sector
* @get_default_device: getting the default device
* @probe_devices: probing possible devices
*
* Cdda source base class.
*/
struct _GstCddaBaseSrcClass {
GstPushSrcClass pushsrc_class;

12
gst-libs/gst/interfaces/colorbalance.h
View file

@ -66,6 +66,17 @@ typedef enum
GST_COLOR_BALANCE_SOFTWARE
} GstColorBalanceType;
/**
* GstColorBalanceClass:
* @klass: the parent interface
* @balance_type: implementation type
* @list_channels: list handled channels
* @set_value: set a channel value
* @get_value: get a channel value
* @value_changed: default handler for value changed notification
*
* Color-balance interface.
*/
struct _GstColorBalanceClass {
GTypeInterface klass;
@ -85,6 +96,7 @@ struct _GstColorBalanceClass {
GstColorBalanceChannel *channel,
gint value);
/*< private >*/
gpointer _gst_reserved[GST_PADDING];
};

9
gst-libs/gst/interfaces/colorbalancechannel.h
View file

@ -57,7 +57,13 @@ struct _GstColorBalanceChannel {
gint max_value;
};
/**
* GstColorBalanceChannelClass:
* @parent: the parent interface
* @value_changed: default handler for value changed notification
*
* Color-balance channel class.
*/
struct _GstColorBalanceChannelClass {
GObjectClass parent;
@ -65,6 +71,7 @@ struct _GstColorBalanceChannelClass {
void (* value_changed) (GstColorBalanceChannel *channel,
gint value);
/*< private >*/
gpointer _gst_reserved[GST_PADDING];
};

7
gst-libs/gst/interfaces/mixer.h
View file

@ -47,6 +47,13 @@ G_BEGIN_DECLS
typedef struct _GstMixer GstMixer;
typedef struct _GstMixerClass GstMixerClass;
/**
* GstMixerType:
* @GST_MIXER_HARDWARE: mixing is implemented with dedicated hardware.
* @GST_MIXER_SOFTWARE: mixing is implemented via software processing.
*
* Mixer classification.
*/
typedef enum
{
GST_MIXER_HARDWARE,

7
gst-libs/gst/interfaces/mixeroptions.h
View file

@ -49,22 +49,25 @@ typedef struct _GstMixerOptionsClass GstMixerOptionsClass;
/**
* GstMixerOptions:
* @parent: Parent object
* @values: List of option strings. Do not access this member directly,
* always use gst_mixer_options_get_values() instead.
*
* Mixer control object.
*/
struct _GstMixerOptions {
GstMixerTrack parent;
/*< public >*/
/* list of strings (do not access directly) (FIXME 0.11: make private) */
GList *values;
/*< private >*/
gpointer _gst_reserved[GST_PADDING];
};
/**
* GstMixerOptionsClass:
* @parent: Parent class
* @parent: the parent interface
* @get_values: Optional implementation of gst_mixer_options_get_values().
* (Since: 0.10.18)
*/

24
gst-libs/gst/interfaces/navigation.h
View file

@ -39,12 +39,20 @@ G_BEGIN_DECLS
typedef struct _GstNavigation GstNavigation;
typedef struct _GstNavigationInterface GstNavigationInterface;
/**
* GstNavigationInterface:
* @g_iface: the parent interface
* @send_event: sending a navigation event
*
* Color-balance interface.
*/
struct _GstNavigationInterface {
GTypeInterface g_iface;
/* virtual functions */
void (*send_event) (GstNavigation *navigation, GstStructure *structure);
/*< private >*/
gpointer _gst_reserved[GST_PADDING];
};
@ -109,7 +117,7 @@ typedef enum {
GST_NAVIGATION_COMMAND_MENU5 = 5,
GST_NAVIGATION_COMMAND_MENU6 = 6,
GST_NAVIGATION_COMMAND_MENU7 = 7,
GST_NAVIGATION_COMMAND_LEFT = 20,
GST_NAVIGATION_COMMAND_RIGHT = 21,
GST_NAVIGATION_COMMAND_UP = 22,
@ -130,6 +138,14 @@ typedef enum {
#define GST_NAVIGATION_COMMAND_DVD_CHAPTER_MENU GST_NAVIGATION_COMMAND_MENU7
/* Queries */
/**
* GstNavigationQueryType:
* @GST_NAVIGATION_QUERY_INVALID: invalid query
* @GST_NAVIGATION_QUERY_COMMANDS: command query
* @GST_NAVIGATION_QUERY_ANGLES: viewing angle query
*
* Tyoes of navigation interface queries.
*/
typedef enum
{
GST_NAVIGATION_QUERY_INVALID = 0,
@ -245,9 +261,9 @@ gboolean gst_navigation_event_parse_command (GstEvent *event,
/* interface virtual function wrappers */
void gst_navigation_send_event (GstNavigation *navigation,
GstStructure *structure);
void gst_navigation_send_key_event (GstNavigation *navigation,
void gst_navigation_send_key_event (GstNavigation *navigation,
const char *event, const char *key);
void gst_navigation_send_mouse_event (GstNavigation *navigation,
void gst_navigation_send_mouse_event (GstNavigation *navigation,
const char *event, int button, double x, double y);
void gst_navigation_send_command (GstNavigation *navigation,
GstNavigationCommand command);

20
gst-libs/gst/interfaces/tuner.h
View file

@ -45,6 +45,25 @@ G_BEGIN_DECLS
typedef struct _GstTuner GstTuner;
typedef struct _GstTunerClass GstTunerClass;
/**
* GstTunerClass:
* @klass: the parent interface
* @list_channels: list available channels
* @set_channel: set to a channel
* @get_channel: return the current channel
* @list_norms: list available norms
* @set_norm: set a norm
* @get_norm: return the current norm
* @set_frequency: set the frequency
* @get_frequency: return the current frequency
* @signal_strength: get the signal strength
* @channel_changed: default handler for channel changed notification
* @norm_changed: default handler for norm changed notification
* @frequency_changed: default handler for frequency changed notification
* @signal_changed: default handler for signal-strength changed notification
*
* Tuner interface.
*/
struct _GstTunerClass {
GTypeInterface klass;
@ -80,6 +99,7 @@ struct _GstTunerClass {
GstTunerChannel *channel,
gint signal);
/*< private >*/
gpointer _gst_reserved[GST_PADDING];
};

70
gst-libs/gst/interfaces/xoverlay.c
View file

@ -66,23 +66,23 @@
* // ignore anything but 'prepare-xwindow-id' element messages
* if (GST_MESSAGE_TYPE (message) != GST_MESSAGE_ELEMENT)
* return GST_BUS_PASS;
*
*
* if (!gst_structure_has_name (message-&gt;structure, "prepare-xwindow-id"))
* return GST_BUS_PASS;
*
*
* win = XCreateSimpleWindow (disp, root, 0, 0, 320, 240, 0, 0, 0);
*
*
* XSetWindowBackgroundPixmap (disp, win, None);
*
*
* XMapRaised (disp, win);
*
*
* XSync (disp, FALSE);
*
*
* gst_x_overlay_set_window_handle (GST_X_OVERLAY (GST_MESSAGE_SRC (message)),
* win);
*
*
* gst_message_unref (message);
*
*
* return GST_BUS_DROP;
* }
* ...
@ -158,17 +158,17 @@
* return GST_BUS_PASS;
* if (!gst_structure_has_name (message-&gt;structure, "prepare-xwindow-id"))
* return GST_BUS_PASS;
*
*
* if (video_window_xid != 0) {
* GstXOverlay *xoverlay;
*
*
* // GST_MESSAGE_SRC (message) will be the video sink element
* xoverlay = GST_X_OVERLAY (GST_MESSAGE_SRC (message));
* gst_x_overlay_set_window_handle (xoverlay, video_window_xid);
* } else {
* g_warning ("Should have obtained video_window_xid by now!");
* }
*
*
* gst_message_unref (message);
* return GST_BUS_DROP;
* }
@ -182,7 +182,7 @@
* if (!gdk_window_ensure_native (widget->window))
* g_error ("Couldn't create native window needed for GstXOverlay!");
* #endif
*
*
* #ifdef GDK_WINDOWING_X11
* video_window_xid = GDK_WINDOW_XID (video_window->window);
* #endif
@ -208,12 +208,12 @@
* ...
* // show the GUI
* gtk_widget_show_all (app_window);
*
*
* // realize window now so that the video window gets created and we can
* // obtain its XID before the pipeline is started up and the videosink
* // asks for the XID of the window to render onto
* gtk_widget_realize (window);
*
*
* // we should have the XID now
* g_assert (video_window_xid != 0);
* ...
@ -235,39 +235,39 @@
* #include &lt;glib.h&gt;
* #include &lt;gst/gst.h&gt;
* #include &lt;gst/interfaces/xoverlay.h&gt;
*
*
* #include &lt;QApplication&gt;
* #include &lt;QTimer&gt;
* #include &lt;QWidget&gt;
*
*
* int main(int argc, char *argv[])
* {
* if (!g_thread_supported ())
* g_thread_init (NULL);
*
*
* gst_init (&argc, &argv);
* QApplication app(argc, argv);
* app.connect(&app, SIGNAL(lastWindowClosed()), &app, SLOT(quit ()));
*
*
* // prepare the pipeline
*
*
* GstElement *pipeline = gst_pipeline_new ("xvoverlay");
* GstElement *src = gst_element_factory_make ("videotestsrc", NULL);
* GstElement *sink = gst_element_factory_make ("xvimagesink", NULL);
* gst_bin_add_many (GST_BIN (pipeline), src, sink, NULL);
* gst_element_link (src, sink);
*
*
* // prepare the ui
*
*
* QWidget window;
* window.resize(320, 240);
* window.show();
*
*
* WId xwinid = window.winId();
* gst_x_overlay_set_window_handle (GST_X_OVERLAY (sink), xwinid);
*
*
* // run the pipeline
*
*
* GstStateChangeReturn sret = gst_element_set_state (pipeline,
* GST_STATE_PLAYING);
* if (sret == GST_STATE_CHANGE_FAILURE) {
@ -276,13 +276,13 @@
* // Exit application
* QTimer::singleShot(0, QApplication::activeWindow(), SLOT(quit()));
* }
*
*
* int ret = app.exec();
*
*
* window.hide();
* gst_element_set_state (pipeline, GST_STATE_NULL);
* gst_object_unref (pipeline);
*
*
* return ret;
* }
* ]|
@ -333,12 +333,12 @@ gst_x_overlay_base_init (gpointer g_class)
/**
* gst_x_overlay_set_xwindow_id:
* @overlay: a #GstXOverlay to set the XWindow on.
* @xwindow_id: a #XID referencing the XWindow.
* @overlay: a #GstXOverlay to set the window on.
* @xwindow_id: a XID referencing the XWindow.
*
* This will call the video overlay's set_xwindow_id method. You should
* use this method to tell to a XOverlay to display video output to a
* specific XWindow. Passing 0 as the xwindow_id will tell the overlay to
* specific XWindow. Passing 0 as the @xwindow_id will tell the overlay to
* stop using that window and create an internal one.
*
* Deprecated: Use gst_x_overlay_set_window_handle() instead.
@ -358,13 +358,13 @@ gst_x_overlay_set_xwindow_id (GstXOverlay * overlay, gulong xwindow_id)
/**
* gst_x_overlay_set_window_handle:
* @overlay: a #GstXOverlay to set the XWindow on.
* @xwindow_id: a #XID referencing the XWindow.
* @overlay: a #GstXOverlay to set the window on.
* @handle: a handle referencing the window.
*
* This will call the video overlay's set_window_handle method. You
* should use this method to tell to a XOverlay to display video output to a
* specific XWindow. Passing 0 as the xwindow_id will tell the overlay to
* stop using that window and create an internal one.
* specific window (e.g. an XWindow on X11). Passing 0 as the @handle will
* tell the overlay to stop using that window and create an internal one.
*
* Since: 0.10.31
*/
@ -452,7 +452,7 @@ gst_x_overlay_got_window_handle (GstXOverlay * overlay, guintptr handle)
* @overlay: a #GstXOverlay which does not yet have an XWindow.
*
* This will post a "prepare-xwindow-id" element message on the bus
* to give applications an opportunity to call
* to give applications an opportunity to call
* gst_x_overlay_set_xwindow_id() before a plugin creates its own
* window.
*

13
gst-libs/gst/interfaces/xoverlay.h
View file

@ -52,10 +52,11 @@ typedef struct _GstXOverlayClass GstXOverlayClass;
/**
* GstXOverlayClass:
* @klass: parent interface type.
* @set_xwindow_id: virtual method to configure the XWindow id
* @set_xwindow_id: (deprecated) virtual method to configure the XWindow handle
* @expose: virtual method to handle expose events
* @handle_events: virtual method to handle events
* @set_render_rectangle: virtual method to set the render rectangle (since 0.10.29)
* @set_window_handle: virtual method to configure the window handle
*
* #GstXOverlay interface
*/
@ -67,14 +68,16 @@ struct _GstXOverlayClass {
void (* set_xwindow_id) (GstXOverlay *overlay,
gulong xwindow_id);
#else
#ifndef __GTK_DOC_IGNORE__
void (* set_xwindow_id_disabled) (GstXOverlay *overlay,
gulong xwindow_id);
#endif
#endif /* not GST_DISABLE_DEPRECATED */
void (* expose) (GstXOverlay *overlay);
void (* handle_events) (GstXOverlay *overlay,
gboolean handle_events);
gboolean handle_events);
void (* set_render_rectangle) (GstXOverlay *overlay,
gint x, gint y,
@ -90,7 +93,7 @@ GType gst_x_overlay_get_type (void);
/* virtual class function wrappers */
#ifndef GST_DISABLE_DEPRECATED
void gst_x_overlay_set_xwindow_id (GstXOverlay *overlay,
void gst_x_overlay_set_xwindow_id (GstXOverlay *overlay,
gulong xwindow_id);
#endif
@ -103,7 +106,7 @@ void gst_x_overlay_expose (GstXOverlay *overlay);
void gst_x_overlay_handle_events (GstXOverlay *overlay,
gboolean handle_events);
void gst_x_overlay_set_window_handle (GstXOverlay *overlay,
void gst_x_overlay_set_window_handle (GstXOverlay *overlay,
guintptr handle);
/* public methods to dispatch bus messages */

2
gst-libs/gst/pbutils/gstpluginsbaseversion.c
View file

@ -26,7 +26,7 @@
* if you need to check at runtime what version of the gst-plugins-base
* libraries are being used / you are currently linked against.
*
* The version macros get defined by including <gst/pbutils/pbutils.h>.
* The version macros get defined by including &lt;gst/pbutils/pbutils.h&gt;.
*/
#include "gstpluginsbaseversion.h"

8
gst-libs/gst/rtp/gstbasertpaudiopayload.c
View file

@ -21,12 +21,9 @@
* SECTION:gstbasertpaudiopayload
* @short_description: Base class for audio RTP payloader
*
* <refsect2>
* <para>
* Provides a base class for audio RTP payloaders for frame or sample based
* audio codecs (constant bitrate)
* </para>
* <para>
*
* This class derives from GstBaseRTPPayload. It can be used for payloading
* audio codecs. It will only work with constant bitrate codecs. It supports
* both frame based and sample based codecs. It takes care of packing up the
@ -38,7 +35,8 @@
* equal to min-ptime (if set). If min-ptime is not set, any residual data is
* sent in a last RTP packet. In the case of frame based codecs, the resulting
* RTP packets always contain full frames.
* </para>
*
* <refsect2>
* <title>Usage</title>
* <para>
* To use this base class, your child element needs to call either

7
gst-libs/gst/rtp/gstbasertpaudiopayload.h
View file

@ -61,10 +61,17 @@ struct _GstBaseRTPAudioPayload
gpointer _gst_reserved[GST_PADDING];
};
/**
* GstBaseRTPAudioPayloadClass:
* @parent_class: the parent class
*
* Base class for audio RTP payloader.
*/
struct _GstBaseRTPAudioPayloadClass
{
GstBaseRTPPayloadClass parent_class;
/*< private >*/
gpointer _gst_reserved[GST_PADDING];
};

6
gst-libs/gst/rtp/gstbasertpdepayload.c
View file

@ -1,5 +1,5 @@
/* GStreamer
* Copyright (C) <2005> Philippe Khalaf <burger@speedy.org>
* Copyright (C) <2005> Philippe Khalaf <burger@speedy.org>
* Copyright (C) <2005> Nokia Corporation <kai.vehmanen@nokia.com>
*
* This library is free software; you can redistribute it and/or
@ -22,11 +22,7 @@
* SECTION:gstbasertpdepayload
* @short_description: Base class for RTP depayloader
*
* <refsect2>
* <para>
* Provides a base class for RTP depayloaders
* </para>
* </refsect2>
*/
#include "gstbasertpdepayload.h"

14
gst-libs/gst/rtp/gstbasertpdepayload.h
View file

@ -1,5 +1,5 @@
/* GStreamer
* Copyright (C) <2005> Philippe Khalaf <burger@speedy.org>
* Copyright (C) <2005> Philippe Khalaf <burger@speedy.org>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
@ -62,6 +62,18 @@ struct _GstBaseRTPDepayload
gpointer _gst_reserved[GST_PADDING-1];
};
/**
* GstBaseRTPDepayloadClass:
* @parent_class: the parent class
* @set_caps: configure the depayloader
* @add_to_queue: (deprecated)
* @process: process incoming rtp packets
* @set_gst_timestamp: convert from RTP timestamp to GST timestamp
* @packet_lost: signal the depayloader about packet loss
* @handle_event: custom event handling
*
* Base class for audio RTP payloader.
*/
struct _GstBaseRTPDepayloadClass
{
GstElementClass parent_class;

4
gst-libs/gst/rtp/gstbasertppayload.c
View file

@ -16,11 +16,7 @@
* SECTION:gstbasertppayload
* @short_description: Base class for RTP payloader
*
* <refsect2>
* <para>
* Provides a base class for RTP payloaders
* </para>
* </refsect2>
*/
#ifdef HAVE_CONFIG_H

10
gst-libs/gst/rtp/gstbasertppayload.h
View file

@ -120,6 +120,16 @@ struct _GstBaseRTPPayload
} abidata;
};
/**
* GstBaseRTPPayloadClass:
* @parent_class: the parent class
* @set_caps: configure the payloader
* @handle_buffer: process data
* @handle_event: custom event handling
* @get_caps: get desired caps
*
* Base class for audio RTP payloader.
*/
struct _GstBaseRTPPayloadClass
{
GstElementClass parent_class;

104
gst-libs/gst/rtsp/gstrtspdefs.h
View file

@ -185,7 +185,7 @@ typedef enum {
* @GST_RTSP_GET: the GET method (HTTP). Since 0.10.25
* @GST_RTSP_POST: the POST method (HTTP). Since 0.10.25
*
* The different supported RTSP methods.
* The different supported RTSP methods.
*/
typedef enum {
GST_RTSP_INVALID = 0,
@ -221,11 +221,17 @@ typedef enum {
/**
* GST_RTSP_AUTH_MAX:
*
* Strongest available authentication method
* Strongest available authentication method
*/
#define GST_RTSP_AUTH_MAX GST_RTSP_AUTH_DIGEST
/**
* GstRTSPHeaderField:
*
* Enumeration of rtsp header fields.
*/
typedef enum {
/*< protected >*/
GST_RTSP_HDR_INVALID,
/*
@ -330,51 +336,57 @@ typedef enum {
GST_RTSP_HDR_LAST
} GstRTSPHeaderField;
/**
* GstRTSPStatusCode:
*
* Enumeration of rtsp status codes.
*/
typedef enum {
GST_RTSP_STS_INVALID = 0,
GST_RTSP_STS_CONTINUE = 100,
GST_RTSP_STS_OK = 200,
GST_RTSP_STS_CREATED = 201,
GST_RTSP_STS_LOW_ON_STORAGE = 250,
GST_RTSP_STS_MULTIPLE_CHOICES = 300,
GST_RTSP_STS_MOVED_PERMANENTLY = 301,
GST_RTSP_STS_MOVE_TEMPORARILY = 302,
GST_RTSP_STS_SEE_OTHER = 303,
GST_RTSP_STS_NOT_MODIFIED = 304,
GST_RTSP_STS_USE_PROXY = 305,
GST_RTSP_STS_BAD_REQUEST = 400,
GST_RTSP_STS_UNAUTHORIZED = 401,
GST_RTSP_STS_PAYMENT_REQUIRED = 402,
GST_RTSP_STS_FORBIDDEN = 403,
GST_RTSP_STS_NOT_FOUND = 404,
GST_RTSP_STS_METHOD_NOT_ALLOWED = 405,
GST_RTSP_STS_NOT_ACCEPTABLE = 406,
GST_RTSP_STS_PROXY_AUTH_REQUIRED = 407,
GST_RTSP_STS_REQUEST_TIMEOUT = 408,
GST_RTSP_STS_GONE = 410,
GST_RTSP_STS_LENGTH_REQUIRED = 411,
GST_RTSP_STS_PRECONDITION_FAILED = 412,
GST_RTSP_STS_REQUEST_ENTITY_TOO_LARGE = 413,
GST_RTSP_STS_REQUEST_URI_TOO_LARGE = 414,
GST_RTSP_STS_UNSUPPORTED_MEDIA_TYPE = 415,
GST_RTSP_STS_PARAMETER_NOT_UNDERSTOOD = 451,
GST_RTSP_STS_CONFERENCE_NOT_FOUND = 452,
GST_RTSP_STS_NOT_ENOUGH_BANDWIDTH = 453,
GST_RTSP_STS_SESSION_NOT_FOUND = 454,
GST_RTSP_STS_METHOD_NOT_VALID_IN_THIS_STATE = 455,
GST_RTSP_STS_HEADER_FIELD_NOT_VALID_FOR_RESOURCE = 456,
GST_RTSP_STS_INVALID_RANGE = 457,
GST_RTSP_STS_PARAMETER_IS_READONLY = 458,
GST_RTSP_STS_AGGREGATE_OPERATION_NOT_ALLOWED = 459,
GST_RTSP_STS_ONLY_AGGREGATE_OPERATION_ALLOWED = 460,
GST_RTSP_STS_UNSUPPORTED_TRANSPORT = 461,
GST_RTSP_STS_DESTINATION_UNREACHABLE = 462,
GST_RTSP_STS_INTERNAL_SERVER_ERROR = 500,
GST_RTSP_STS_NOT_IMPLEMENTED = 501,
GST_RTSP_STS_BAD_GATEWAY = 502,
GST_RTSP_STS_SERVICE_UNAVAILABLE = 503,
GST_RTSP_STS_GATEWAY_TIMEOUT = 504,
GST_RTSP_STS_RTSP_VERSION_NOT_SUPPORTED = 505,
/*< protected >*/
GST_RTSP_STS_INVALID = 0,
GST_RTSP_STS_CONTINUE = 100,
GST_RTSP_STS_OK = 200,
GST_RTSP_STS_CREATED = 201,
GST_RTSP_STS_LOW_ON_STORAGE = 250,
GST_RTSP_STS_MULTIPLE_CHOICES = 300,
GST_RTSP_STS_MOVED_PERMANENTLY = 301,
GST_RTSP_STS_MOVE_TEMPORARILY = 302,
GST_RTSP_STS_SEE_OTHER = 303,
GST_RTSP_STS_NOT_MODIFIED = 304,
GST_RTSP_STS_USE_PROXY = 305,
GST_RTSP_STS_BAD_REQUEST = 400,
GST_RTSP_STS_UNAUTHORIZED = 401,
GST_RTSP_STS_PAYMENT_REQUIRED = 402,
GST_RTSP_STS_FORBIDDEN = 403,
GST_RTSP_STS_NOT_FOUND = 404,
GST_RTSP_STS_METHOD_NOT_ALLOWED = 405,
GST_RTSP_STS_NOT_ACCEPTABLE = 406,
GST_RTSP_STS_PROXY_AUTH_REQUIRED = 407,
GST_RTSP_STS_REQUEST_TIMEOUT = 408,
GST_RTSP_STS_GONE = 410,
GST_RTSP_STS_LENGTH_REQUIRED = 411,
GST_RTSP_STS_PRECONDITION_FAILED = 412,
GST_RTSP_STS_REQUEST_ENTITY_TOO_LARGE = 413,
GST_RTSP_STS_REQUEST_URI_TOO_LARGE = 414,
GST_RTSP_STS_UNSUPPORTED_MEDIA_TYPE = 415,
GST_RTSP_STS_PARAMETER_NOT_UNDERSTOOD = 451,
GST_RTSP_STS_CONFERENCE_NOT_FOUND = 452,
GST_RTSP_STS_NOT_ENOUGH_BANDWIDTH = 453,
GST_RTSP_STS_SESSION_NOT_FOUND = 454,
GST_RTSP_STS_METHOD_NOT_VALID_IN_THIS_STATE = 455,
GST_RTSP_STS_HEADER_FIELD_NOT_VALID_FOR_RESOURCE = 456,
GST_RTSP_STS_INVALID_RANGE = 457,
GST_RTSP_STS_PARAMETER_IS_READONLY = 458,
GST_RTSP_STS_AGGREGATE_OPERATION_NOT_ALLOWED = 459,
GST_RTSP_STS_ONLY_AGGREGATE_OPERATION_ALLOWED = 460,
GST_RTSP_STS_UNSUPPORTED_TRANSPORT = 461,
GST_RTSP_STS_DESTINATION_UNREACHABLE = 462,
GST_RTSP_STS_INTERNAL_SERVER_ERROR = 500,
GST_RTSP_STS_NOT_IMPLEMENTED = 501,
GST_RTSP_STS_BAD_GATEWAY = 502,
GST_RTSP_STS_SERVICE_UNAVAILABLE = 503,
GST_RTSP_STS_GATEWAY_TIMEOUT = 504,
GST_RTSP_STS_RTSP_VERSION_NOT_SUPPORTED = 505,
GST_RTSP_STS_OPTION_NOT_SUPPORTED = 551
} GstRTSPStatusCode;

8
gst-libs/gst/tag/gstxmptag.c
View file

@ -162,10 +162,10 @@ xmp_tag_get_type_name (XmpTag * xmptag)
switch (xmptag->type) {
case GstXmpTagTypeSeq:
return "rdf:Seq";
default:
g_assert_not_reached ();
case GstXmpTagTypeBag:
return "rdf:Bag";
default:
g_assert_not_reached ();
}
}
@ -844,7 +844,7 @@ deserialize_xmp_rating (XmpTag * xmptag, GstTagList * taglist,
return;
}
if (value < 0 || value > 100) {
if (value > 100) {
GST_WARNING ("Unsupported Rating tag %u (should be from 0 to 100), "
"ignoring", value);
return;
@ -1073,7 +1073,7 @@ read_one_tag (GstTagList * list, XmpTag * xmptag,
g_return_if_fail (tag != NULL);
if (xmptag && xmptag->deserialize) {
if (xmptag->deserialize) {
xmptag->deserialize (xmptag, list, tag, xmptag->tag_name, v, pending_tags);
return;
}

32
gst-libs/gst/video/convertframe.c
View file

@ -236,7 +236,7 @@ link_failed:
* @from_caps: the #GstCaps to convert from
* @to_caps: the #GstCaps to convert to
* @timeout: the maximum amount of time allowed for the processing.
* @err: pointer to a #GError. Can be %NULL.
* @error: pointer to a #GError. Can be %NULL.
*
* Converts a raw video buffer into the specified output caps.
*
@ -256,7 +256,7 @@ gst_video_convert_frame (GstBuffer * buf, GstCaps * from_caps,
{
GstMessage *msg;
GstBuffer *result = NULL;
GError *error = NULL;
GError *err = NULL;
GstBus *bus;
GstCaps *to_caps_copy = NULL;
GstFlowReturn ret;
@ -278,8 +278,7 @@ gst_video_convert_frame (GstBuffer * buf, GstCaps * from_caps,
}
pipeline =
build_convert_frame_pipeline (&src, &sink, from_caps, to_caps_copy,
&error);
build_convert_frame_pipeline (&src, &sink, from_caps, to_caps_copy, &err);
if (!pipeline)
goto no_pipeline;
@ -317,14 +316,14 @@ gst_video_convert_frame (GstBuffer * buf, GstCaps * from_caps,
case GST_MESSAGE_ERROR:{
gchar *dbg = NULL;
gst_message_parse_error (msg, &error, &dbg);
if (error) {
GST_ERROR ("Could not convert video frame: %s", error->message);
GST_DEBUG ("%s [debug: %s]", error->message, GST_STR_NULL (dbg));
if (err)
*err = error;
gst_message_parse_error (msg, &err, &dbg);
if (err) {
GST_ERROR ("Could not convert video frame: %s", err->message);
GST_DEBUG ("%s [debug: %s]", err->message, GST_STR_NULL (dbg));
if (error)
*error = err;
else
g_error_free (error);
g_error_free (err);
}
g_free (dbg);
break;
@ -336,8 +335,8 @@ gst_video_convert_frame (GstBuffer * buf, GstCaps * from_caps,
gst_message_unref (msg);
} else {
GST_ERROR ("Could not convert video frame: timeout during conversion");
if (err)
*err = g_error_new (GST_CORE_ERROR, GST_CORE_ERROR_FAILED,
if (error)
*error = g_error_new (GST_CORE_ERROR, GST_CORE_ERROR_FAILED,
"Could not convert video frame: timeout during conversion");
}
@ -353,10 +352,10 @@ no_pipeline:
{
gst_caps_unref (to_caps_copy);
if (err)
*err = error;
if (error)
*error = err;
else
g_error_free (error);
g_error_free (err);
return NULL;
}
@ -576,6 +575,7 @@ done:
* @to_caps: the #GstCaps to convert to
* @timeout: the maximum amount of time allowed for the processing.
* @callback: %GstVideoConvertFrameCallback that will be called after conversion.
* @user_data: extra data that will be passed to the @callback
* @destroy_notify: %GDestroyNotify to be called after @user_data is not needed anymore
*
* Converts a raw video buffer into the specified output caps.

6
gst-libs/gst/video/gstvideofilter.h
View file

@ -47,6 +47,12 @@ struct _GstVideoFilter {
gboolean inited;
};
/**
* GstVideoFilterClass:
* @parent_class: the parent class structure
*
* The video filter class structure.
*/
struct _GstVideoFilterClass {
GstBaseTransformClass parent_class;
};

10
gst-libs/gst/video/gstvideosink.h
View file

@ -27,7 +27,7 @@
#include <gst/base/gstbasesink.h>
G_BEGIN_DECLS
#define GST_TYPE_VIDEO_SINK (gst_video_sink_get_type())
#define GST_VIDEO_SINK(obj) \
(G_TYPE_CHECK_INSTANCE_CAST ((obj), GST_TYPE_VIDEO_SINK, GstVideoSink))
@ -60,7 +60,7 @@ G_BEGIN_DECLS
#define GST_VIDEO_SINK_WIDTH(obj) (GST_VIDEO_SINK_CAST (obj)->width)
#define GST_VIDEO_SINK_HEIGHT(obj) (GST_VIDEO_SINK_CAST (obj)->height)
typedef struct _GstVideoSink GstVideoSink;
typedef struct _GstVideoSinkClass GstVideoSinkClass;
typedef struct _GstVideoRectangle GstVideoRectangle;
@ -84,7 +84,6 @@ struct _GstVideoRectangle {
/**
* GstVideoSink:
* @element: the parent object structure (which is GstBaseSink)
* @height: video height (derived class needs to set this)
* @width: video width (derived class needs to set this)
*
@ -93,9 +92,10 @@ struct _GstVideoRectangle {
*/
struct _GstVideoSink {
GstBaseSink element; /* FIXME 0.11: this should not be called 'element' */
/*< public >*/
gint width, height;
/*< private >*/
GstVideoSinkPrivate *priv;

75
gst-libs/gst/video/video.c
View file

@ -31,7 +31,7 @@
*
* <refsect2>
* <para>
* This library contains some helper functions and includes the
* This library contains some helper functions and includes the
* videosink and videofilter base classes.
* </para>
* </refsect2>
@ -53,7 +53,7 @@ static GstVideoFormat gst_video_format_from_rgb16_masks (int red_mask,
*
* A convenience function to retrieve a GValue holding the framerate
* from the caps on a pad.
*
*
* The pad needs to have negotiated caps containing a framerate property.
*
* Returns: NULL if the pad has no configured caps or the configured caps
@ -121,7 +121,7 @@ no_fraction:
*
* Inspect the caps of the provided pad and retrieve the width and height of
* the video frames it is configured for.
*
*
* The pad needs to have negotiated caps containing width and height properties.
*
* Returns: TRUE if the width and height could be retrieved.
@ -181,13 +181,13 @@ no_size:
* @display_par_n: Numerator of the pixel aspect ratio of the display device
* @display_par_d: Denominator of the pixel aspect ratio of the display device
*
* Given the Pixel Aspect Ratio and size of an input video frame, and the
* pixel aspect ratio of the intended display device, calculates the actual
* Given the Pixel Aspect Ratio and size of an input video frame, and the
* pixel aspect ratio of the intended display device, calculates the actual
* display ratio the video will be rendered with.
*
* Returns: A boolean indicating success and a calculated Display Ratio in the
* dar_n and dar_d parameters.
* The return value is FALSE in the case of integer overflow or other error.
* Returns: A boolean indicating success and a calculated Display Ratio in the
* dar_n and dar_d parameters.
* The return value is FALSE in the case of integer overflow or other error.
*
* Since: 0.10.7
*/
@ -299,7 +299,7 @@ gst_video_parse_caps_color_matrix (GstCaps * caps)
* halfway-sited vertically), "jpeg" for JPEG and Theora style
* chroma siting (halfway-sited both horizontally and vertically).
* Other chroma site values are possible, but uncommon.
*
*
* When no chroma site is specified in the caps, it should be assumed
* to be "mpeg2".
*
@ -681,10 +681,10 @@ gst_video_format_new_caps_raw (GstVideoFormat format)
blue_mask = GST_VIDEO_COMP1_MASK_15_INT;
break;
default:
return NULL;
g_assert_not_reached ();
}
} else if (bpp != 8) {
return NULL;
g_assert_not_reached ();
}
caps = gst_caps_new_simple ("video/x-raw-rgb",
@ -977,7 +977,7 @@ gst_video_format_to_fourcc (GstVideoFormat format)
* @blue_mask: blue bit mask
*
* Converts red, green, blue bit masks into the corresponding
* #GstVideoFormat.
* #GstVideoFormat.
*
* Since: 0.10.16
*
@ -1216,7 +1216,7 @@ gst_video_format_is_gray (GstVideoFormat format)
/**
* gst_video_format_has_alpha:
* @format: a #GstVideoFormat
*
*
* Returns TRUE or FALSE depending on if the video format provides an
* alpha channel.
*
@ -1278,9 +1278,10 @@ gst_video_format_has_alpha (GstVideoFormat format)
/**
* gst_video_format_get_component_depth:
* @format: a #GstVideoFormat
*
* @component: the video component (e.g. 0 for 'R' in RGB)
*
* Returns the number of bits used to encode an individual pixel of
* a given component. Typically this is 8, although higher and lower
* a given @component. Typically this is 8, although higher and lower
* values are possible for some formats.
*
* Since: 0.10.33
@ -1767,7 +1768,7 @@ gst_video_format_get_component_offset (GstVideoFormat format,
GST_ROUND_UP_4 (GST_ROUND_UP_2 (width) / 2) *
(GST_ROUND_UP_2 (height) / 2);
}
return 0;
break;
case GST_VIDEO_FORMAT_YV12: /* same as I420, but components 1+2 swapped */
if (component == 0)
return 0;
@ -1778,7 +1779,7 @@ gst_video_format_get_component_offset (GstVideoFormat format,
GST_ROUND_UP_4 (GST_ROUND_UP_2 (width) / 2) *
(GST_ROUND_UP_2 (height) / 2);
}
return 0;
break;
case GST_VIDEO_FORMAT_YUY2:
if (component == 0)
return 0;
@ -1786,7 +1787,7 @@ gst_video_format_get_component_offset (GstVideoFormat format,
return 1;
if (component == 2)
return 3;
return 0;
break;
case GST_VIDEO_FORMAT_YVYU:
if (component == 0)
return 0;
@ -1794,7 +1795,7 @@ gst_video_format_get_component_offset (GstVideoFormat format,
return 3;
if (component == 2)
return 1;
return 0;
break;
case GST_VIDEO_FORMAT_UYVY:
if (component == 0)
return 1;
@ -1802,7 +1803,7 @@ gst_video_format_get_component_offset (GstVideoFormat format,
return 0;
if (component == 2)
return 2;
return 0;
break;
case GST_VIDEO_FORMAT_AYUV:
if (component == 0)
return 1;
@ -1812,7 +1813,7 @@ gst_video_format_get_component_offset (GstVideoFormat format,
return 3;
if (component == 3)
return 0;
return 0;
break;
case GST_VIDEO_FORMAT_RGBx:
case GST_VIDEO_FORMAT_RGBA:
if (component == 0)
@ -1823,7 +1824,7 @@ gst_video_format_get_component_offset (GstVideoFormat format,
return 2;
if (component == 3)
return 3;
return 0;
break;
case GST_VIDEO_FORMAT_BGRx:
case GST_VIDEO_FORMAT_BGRA:
if (component == 0)
@ -1834,7 +1835,7 @@ gst_video_format_get_component_offset (GstVideoFormat format,
return 0;
if (component == 3)
return 3;
return 0;
break;
case GST_VIDEO_FORMAT_xRGB:
case GST_VIDEO_FORMAT_ARGB:
if (component == 0)
@ -1845,7 +1846,7 @@ gst_video_format_get_component_offset (GstVideoFormat format,
return 3;
if (component == 3)
return 0;
return 0;
break;
case GST_VIDEO_FORMAT_xBGR:
case GST_VIDEO_FORMAT_ABGR:
if (component == 0)
@ -1856,7 +1857,7 @@ gst_video_format_get_component_offset (GstVideoFormat format,
return 1;
if (component == 3)
return 0;
return 0;
break;
case GST_VIDEO_FORMAT_RGB:
case GST_VIDEO_FORMAT_v308:
if (component == 0)
@ -1865,7 +1866,7 @@ gst_video_format_get_component_offset (GstVideoFormat format,
return 1;
if (component == 2)
return 2;
return 0;
break;
case GST_VIDEO_FORMAT_BGR:
if (component == 0)
return 2;
@ -1873,7 +1874,7 @@ gst_video_format_get_component_offset (GstVideoFormat format,
return 1;
if (component == 2)
return 0;
return 0;
break;
case GST_VIDEO_FORMAT_Y41B:
if (component == 0)
return 0;
@ -1882,7 +1883,7 @@ gst_video_format_get_component_offset (GstVideoFormat format,
if (component == 2)
return (GST_ROUND_UP_4 (width) +
(GST_ROUND_UP_16 (width) / 4)) * height;
return 0;
break;
case GST_VIDEO_FORMAT_Y42B:
if (component == 0)
return 0;
@ -1890,7 +1891,7 @@ gst_video_format_get_component_offset (GstVideoFormat format,
return GST_ROUND_UP_4 (width) * height;
if (component == 2)
return (GST_ROUND_UP_4 (width) + (GST_ROUND_UP_8 (width) / 2)) * height;
return 0;
break;
case GST_VIDEO_FORMAT_Y444:
return GST_ROUND_UP_4 (width) * height * component;
case GST_VIDEO_FORMAT_v210:
@ -1904,7 +1905,7 @@ gst_video_format_get_component_offset (GstVideoFormat format,
return 2;
if (component == 2)
return 6;
return 0;
break;
case GST_VIDEO_FORMAT_NV12:
if (component == 0)
return 0;
@ -1912,6 +1913,7 @@ gst_video_format_get_component_offset (GstVideoFormat format,
return GST_ROUND_UP_4 (width) * GST_ROUND_UP_2 (height);
if (component == 2)
return GST_ROUND_UP_4 (width) * GST_ROUND_UP_2 (height) + 1;
break;
case GST_VIDEO_FORMAT_NV21:
if (component == 0)
return 0;
@ -1919,6 +1921,7 @@ gst_video_format_get_component_offset (GstVideoFormat format,
return GST_ROUND_UP_4 (width) * GST_ROUND_UP_2 (height) + 1;
if (component == 2)
return GST_ROUND_UP_4 (width) * GST_ROUND_UP_2 (height);
break;
case GST_VIDEO_FORMAT_GRAY8:
case GST_VIDEO_FORMAT_GRAY16_BE:
case GST_VIDEO_FORMAT_GRAY16_LE:
@ -1943,6 +1946,7 @@ gst_video_format_get_component_offset (GstVideoFormat format,
2 * GST_ROUND_UP_4 (GST_ROUND_UP_2 (width) / 2) *
(GST_ROUND_UP_2 (height) / 2);
}
break;
case GST_VIDEO_FORMAT_RGB8_PALETTED:
return 0;
case GST_VIDEO_FORMAT_YUV9:
@ -1955,7 +1959,7 @@ gst_video_format_get_component_offset (GstVideoFormat format,
GST_ROUND_UP_4 (GST_ROUND_UP_4 (width) / 4) *
(GST_ROUND_UP_4 (height) / 4);
}
return 0;
break;
case GST_VIDEO_FORMAT_YVU9:
if (component == 0)
return 0;
@ -1966,7 +1970,7 @@ gst_video_format_get_component_offset (GstVideoFormat format,
}
if (component == 2)
return GST_ROUND_UP_4 (width) * height;
return 0;
break;
case GST_VIDEO_FORMAT_IYU1:
if (component == 0)
return 1;
@ -1974,6 +1978,7 @@ gst_video_format_get_component_offset (GstVideoFormat format,
return 0;
if (component == 2)
return 4;
break;
case GST_VIDEO_FORMAT_ARGB64:
case GST_VIDEO_FORMAT_AYUV64:
if (component == 0)
@ -1984,10 +1989,12 @@ gst_video_format_get_component_offset (GstVideoFormat format,
return 6;
if (component == 3)
return 0;
return 0;
break;
default:
return 0;
break;
}
GST_WARNING ("unhandled format %d or component %d", format, component);
return 0;
}
/**

26
gst/app/gstapp.c
View file

@ -16,6 +16,32 @@
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
/**
* SECTION:element-appsrc
*
* The appsrc element can be used by applications to insert data into a
* GStreamer pipeline. Unlike most GStreamer elements, Appsrc provides
* external API functions.
*
* For the documentation of the API, please see the
* <link linkend="gst-plugins-base-libs-appsrc">libgstapp</link> section in the
* GStreamer Plugins Base Libraries documentation.
*
* Since: 0.10.22
*/
/**
* SECTION:element-appsink
*
* Appsink is a sink plugin that supports many different methods for making
* the application get a handle on the GStreamer data in a pipeline. Unlike
* most GStreamer elements, Appsink provides external API functions.
*
* For the documentation of the API, please see the
* <link linkend="gst-plugins-base-libs-appsink">libgstapp</link> section in
* the GStreamer Plugins Base Libraries documentation.
*
* Since: 0.10.22
*/
#ifdef HAVE_CONFIG_H
#include "config.h"

14
gst/subparse/gstsubparse.c
View file

@ -1203,7 +1203,8 @@ gst_sub_parse_data_format_autodetect_regex_once (GstSubParseRegex regtype)
switch (regtype) {
case GST_SUB_PARSE_REGEX_MDVDSUB:
result =
(gpointer) g_regex_new ("^\\{[0-9]+\\}\\{[0-9]+\\}", 0, 0, &gerr);
(gpointer) g_regex_new ("^\\{[0-9]+\\}\\{[0-9]+\\}",
G_REGEX_RAW | G_REGEX_OPTIMIZE, 0, &gerr);
if (result == NULL) {
g_warning ("Compilation of mdvd regex failed: %s", gerr->message);
g_error_free (gerr);
@ -1213,7 +1214,7 @@ gst_sub_parse_data_format_autodetect_regex_once (GstSubParseRegex regtype)
result = (gpointer) g_regex_new ("^([ 0-9]){0,3}[0-9]\\s*(\x0d)?\x0a"
"[ 0-9][0-9]:[ 0-9][0-9]:[ 0-9][0-9][,.][ 0-9]{0,2}[0-9]"
" +--> +([ 0-9])?[0-9]:[ 0-9][0-9]:[ 0-9][0-9][,.][ 0-9]{0,2}[0-9]",
0, 0, &gerr);
G_REGEX_RAW | G_REGEX_OPTIMIZE, 0, &gerr);
if (result == NULL) {
g_warning ("Compilation of subrip regex failed: %s", gerr->message);
g_error_free (gerr);
@ -1221,7 +1222,7 @@ gst_sub_parse_data_format_autodetect_regex_once (GstSubParseRegex regtype)
break;
case GST_SUB_PARSE_REGEX_DKS:
result = (gpointer) g_regex_new ("^\\[[0-9]+:[0-9]+:[0-9]+\\].*",
0, 0, &gerr);
G_REGEX_RAW | G_REGEX_OPTIMIZE, 0, &gerr);
if (result == NULL) {
g_warning ("Compilation of dks regex failed: %s", gerr->message);
g_error_free (gerr);
@ -1741,12 +1742,7 @@ gst_subparse_type_find (GstTypeFind * tf, gpointer private)
}
}
converted_str = gst_convert_to_utf8 (str, 128, enc, &tmp, &err);
if (converted_str == NULL) {
GST_DEBUG ("Charset conversion failed: %s", err->message);
g_error_free (err);
g_free (str);
return;
} else {
if (converted_str != NULL) {
g_free (str);
str = converted_str;
}