Commit a87c0212 authored by Wim Taymans's avatar Wim Taymans

Merge branch 'master' into 0.11

Conflicts:
	gst-libs/gst/video/convertframe.c
parents 45309113 8da23e9d
......@@ -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" />
......
......@@ -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
......
......@@ -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.
*/
......
......@@ -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;
......
......@@ -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;
......
......@@ -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.
*
......
......@@ -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
*
* Calculated the size of the buffer expected by gst_audio_iec61937_payload() for
* payloading type from @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.
* 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
......
......@@ -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__
......
......@@ -40,7 +40,6 @@ typedef struct _GstAudioSrcClass GstAudioSrcClass;
/**
* GstAudioSrc:
* @element: parent class
*
* Base class for simple audio sources.