Commit f554369e authored by Mathieu Duponchelle's avatar Mathieu Duponchelle 🐸

doc: remove xml from comments

parent 89380bdd
...@@ -91,5 +91,6 @@ foreach plugin_name: list_plugin_res.stdout().split(':') ...@@ -91,5 +91,6 @@ foreach plugin_name: list_plugin_res.stdout().split(':')
disable_incremental_build: true, disable_incremental_build: true,
gst_cache_file: plugins_cache, gst_cache_file: plugins_cache,
gst_plugin_name: plugin_name, gst_plugin_name: plugin_name,
include_paths: join_paths(meson.current_source_dir(), '..'),
)] )]
endforeach endforeach
...@@ -30,16 +30,16 @@ ...@@ -30,16 +30,16 @@
* consideration. See <ulink url="http://www.vorbis.com/">Ogg/Vorbis</ulink> * consideration. See <ulink url="http://www.vorbis.com/">Ogg/Vorbis</ulink>
* for a royalty free (and often higher quality) alternative. * for a royalty free (and often higher quality) alternative.
* *
* <refsect2> * ## Output sample rate
* <title>Output sample rate</title> *
* If no fixed output sample rate is negotiated on the element's src pad, * If no fixed output sample rate is negotiated on the element's src pad,
* the element will choose an optimal sample rate to resample to internally. * the element will choose an optimal sample rate to resample to internally.
* For example, a 16-bit 44.1 KHz mono audio stream encoded at 48 kbit will * For example, a 16-bit 44.1 KHz mono audio stream encoded at 48 kbit will
* get resampled to 32 KHz. Use filter caps on the src pad to force a * get resampled to 32 KHz. Use filter caps on the src pad to force a
* particular sample rate. * particular sample rate.
* </refsect2> *
* <refsect2> * ## Example pipelines
* <title>Example pipelines</title> *
* |[ * |[
* gst-launch-1.0 -v audiotestsrc wave=sine num-buffers=100 ! audioconvert ! lamemp3enc ! filesink location=sine.mp3 * gst-launch-1.0 -v audiotestsrc wave=sine num-buffers=100 ! audioconvert ! lamemp3enc ! filesink location=sine.mp3
* ]| Encode a test sine signal to MP3. * ]| Encode a test sine signal to MP3.
...@@ -55,7 +55,6 @@ ...@@ -55,7 +55,6 @@
* |[ * |[
* gst-launch-1.0 -v audiotestsrc num-buffers=10 ! audio/x-raw,rate=44100,channels=1 ! lamemp3enc target=bitrate cbr=true bitrate=48 ! filesink location=test.mp3 * gst-launch-1.0 -v audiotestsrc num-buffers=10 ! audio/x-raw,rate=44100,channels=1 ! lamemp3enc target=bitrate cbr=true bitrate=48 ! filesink location=test.mp3
* ]| Encode to a fixed sample rate * ]| Encode to a fixed sample rate
* </refsect2>
*/ */
#ifdef HAVE_CONFIG_H #ifdef HAVE_CONFIG_H
......
...@@ -22,12 +22,11 @@ ...@@ -22,12 +22,11 @@
* *
* Audio decoder for MPEG-1 layer 1/2/3 audio data. * Audio decoder for MPEG-1 layer 1/2/3 audio data.
* *
* <refsect2> * ## Example pipelines
* <title>Example pipelines</title> *
* |[ * |[
* gst-launch-1.0 filesrc location=music.mp3 ! mpegaudioparse ! mpg123audiodec ! audioconvert ! audioresample ! autoaudiosink * gst-launch-1.0 filesrc location=music.mp3 ! mpegaudioparse ! mpg123audiodec ! audioconvert ! audioresample ! autoaudiosink
* ]| Decode and play the mp3 file * ]| Decode and play the mp3 file
* </refsect2>
*/ */
#ifdef HAVE_CONFIG_H #ifdef HAVE_CONFIG_H
......
...@@ -30,8 +30,8 @@ ...@@ -30,8 +30,8 @@
* Tags sent by upstream elements will be picked up automatically (and merged * Tags sent by upstream elements will be picked up automatically (and merged
* according to the merge mode set via the tag setter interface). * according to the merge mode set via the tag setter interface).
* *
* <refsect2> * ## Example pipelines
* <title>Example pipelines</title> *
* |[ * |[
* gst-launch-1.0 -v filesrc location=foo.ogg ! decodebin ! audioconvert ! lame ! apev2mux ! filesink location=foo.mp3 * gst-launch-1.0 -v filesrc location=foo.ogg ! decodebin ! audioconvert ! lame ! apev2mux ! filesink location=foo.mp3
* ]| A pipeline that transcodes a file from Ogg/Vorbis to mp3 format with an * ]| A pipeline that transcodes a file from Ogg/Vorbis to mp3 format with an
...@@ -40,7 +40,6 @@ ...@@ -40,7 +40,6 @@
* |[ * |[
* gst-launch-1.0 -m filesrc location=foo.mp3 ! apedemux ! fakesink silent=TRUE 2&gt; /dev/null | grep taglist * gst-launch-1.0 -m filesrc location=foo.mp3 ! apedemux ! fakesink silent=TRUE 2&gt; /dev/null | grep taglist
* ]| Verify that tags have been written. * ]| Verify that tags have been written.
* </refsect2>
*/ */
#ifdef HAVE_CONFIG_H #ifdef HAVE_CONFIG_H
......
...@@ -31,8 +31,8 @@ ...@@ -31,8 +31,8 @@
* Tags sent by upstream elements will be picked up automatically (and merged * Tags sent by upstream elements will be picked up automatically (and merged
* according to the merge mode set via the tag setter interface). * according to the merge mode set via the tag setter interface).
* *
* <refsect2> * ## Example pipelines
* <title>Example pipelines</title> *
* |[ * |[
* gst-launch-1.0 -v filesrc location=foo.ogg ! decodebin ! audioconvert ! lame ! id3v2mux ! filesink location=foo.mp3 * gst-launch-1.0 -v filesrc location=foo.ogg ! decodebin ! audioconvert ! lame ! id3v2mux ! filesink location=foo.mp3
* ]| A pipeline that transcodes a file from Ogg/Vorbis to mp3 format with an * ]| A pipeline that transcodes a file from Ogg/Vorbis to mp3 format with an
...@@ -41,7 +41,6 @@ ...@@ -41,7 +41,6 @@
* |[ * |[
* gst-launch-1.0 -m filesrc location=foo.mp3 ! id3demux ! fakesink silent=TRUE 2&gt; /dev/null | grep taglist * gst-launch-1.0 -m filesrc location=foo.mp3 ! id3demux ! fakesink silent=TRUE 2&gt; /dev/null | grep taglist
* ]| Verify that tags have been written. * ]| Verify that tags have been written.
* </refsect2>
*/ */
#ifdef HAVE_CONFIG_H #ifdef HAVE_CONFIG_H
......
...@@ -30,8 +30,8 @@ ...@@ -30,8 +30,8 @@
* *
* This element encodes raw integer audio into an MPEG-1 layer 2 (MP2) stream. * This element encodes raw integer audio into an MPEG-1 layer 2 (MP2) stream.
* *
* <refsect2> * ## Example pipelines
* <title>Example pipelines</title> *
* |[ * |[
* gst-launch-1.0 -v audiotestsrc wave=sine num-buffers=100 ! audioconvert ! twolame ! filesink location=sine.mp2 * gst-launch-1.0 -v audiotestsrc wave=sine num-buffers=100 ! audioconvert ! twolame ! filesink location=sine.mp2
* ]| Encode a test sine signal to MP2. * ]| Encode a test sine signal to MP2.
...@@ -44,7 +44,6 @@ ...@@ -44,7 +44,6 @@
* |[ * |[
* gst-launch-1.0 -v cdda://5 ! audioconvert ! twolame bitrate=192 ! filesink location=track5.mp2 * gst-launch-1.0 -v cdda://5 ! audioconvert ! twolame bitrate=192 ! filesink location=track5.mp2
* ]| Encode Audio CD track 5 to MP2 * ]| Encode Audio CD track 5 to MP2
* </refsect2>
* *
*/ */
......
...@@ -25,7 +25,7 @@ ...@@ -25,7 +25,7 @@
* *
* autoaudiosink is an audio sink that automatically detects an appropriate * autoaudiosink is an audio sink that automatically detects an appropriate
* audio sink to use. It does so by scanning the registry for all elements * audio sink to use. It does so by scanning the registry for all elements
* that have <quote>Sink</quote> and <quote>Audio</quote> in the class field * that have "Sink" and "Audio" in the class field
* of their element information, and also have a non-zero autoplugging rank. * of their element information, and also have a non-zero autoplugging rank.
* *
* ## Example launch line * ## Example launch line
......
...@@ -26,7 +26,7 @@ ...@@ -26,7 +26,7 @@
* *
* autoaudiosrc is an audio source that automatically detects an appropriate * autoaudiosrc is an audio source that automatically detects an appropriate
* audio source to use. It does so by scanning the registry for all elements * audio source to use. It does so by scanning the registry for all elements
* that have <quote>Source</quote> and <quote>Audio</quote> in the class field * that have "Source" and "Audio" in the class field
* of their element information, and also have a non-zero autoplugging rank. * of their element information, and also have a non-zero autoplugging rank.
* *
* ## Example launch line * ## Example launch line
......
...@@ -25,7 +25,7 @@ ...@@ -25,7 +25,7 @@
* *
* autovideosink is a video sink that automatically detects an appropriate * autovideosink is a video sink that automatically detects an appropriate
* video sink to use. It does so by scanning the registry for all elements * video sink to use. It does so by scanning the registry for all elements
* that have <quote>Sink</quote> and <quote>Video</quote> in the class field * that have "Sink" and "Video" in the class field
* of their element information, and also have a non-zero autoplugging rank. * of their element information, and also have a non-zero autoplugging rank.
* *
* ## Example launch line * ## Example launch line
......
...@@ -26,7 +26,7 @@ ...@@ -26,7 +26,7 @@
* *
* autovideosrc is a video src that automatically detects an appropriate * autovideosrc is a video src that automatically detects an appropriate
* video source to use. It does so by scanning the registry for all elements * video source to use. It does so by scanning the registry for all elements
* that have <quote>Source</quote> and <quote>Video</quote> in the class field * that have "Source" and "Video" in the class field
* of their element information, and also have a non-zero autoplugging rank. * of their element information, and also have a non-zero autoplugging rank.
* *
* ## Example launch line * ## Example launch line
......
...@@ -37,64 +37,22 @@ ...@@ -37,64 +37,22 @@
* structure of name "dtmf-event" with fields set according to the following * structure of name "dtmf-event" with fields set according to the following
* table: * table:
* *
* <informaltable> * * `type` (G_TYPE_INT, 0-1): The application uses this field to specify which of the two methods
* <tgroup cols='4'>
* <colspec colname='Name' />
* <colspec colname='Type' />
* <colspec colname='Possible values' />
* <colspec colname='Purpose' />
* <thead>
* <row>
* <entry>Name</entry>
* <entry>GType</entry>
* <entry>Possible values</entry>
* <entry>Purpose</entry>
* </row>
* </thead>
* <tbody>
* <row>
* <entry>type</entry>
* <entry>G_TYPE_INT</entry>
* <entry>0-1</entry>
* <entry>The application uses this field to specify which of the two methods
* specified in RFC 2833 to use. The value should be 0 for tones and 1 for * specified in RFC 2833 to use. The value should be 0 for tones and 1 for
* named events. Tones are specified by their frequencies and events are specied * named events. Tones are specified by their frequencies and events are specied
* by their number. This element can only take events as input. Do not confuse * by their number. This element can only take events as input. Do not confuse
* with "method" which specified the output. * with "method" which specified the output.
* </entry> *
* </row> * * `number` (G_TYPE_INT, 0-15): The event number.
* <row> *
* <entry>number</entry> * * `volume` (G_TYPE_INT, 0-36): This field describes the power level of the tone, expressed in dBm0
* <entry>G_TYPE_INT</entry>
* <entry>0-15</entry>
* <entry>The event number.</entry>
* </row>
* <row>
* <entry>volume</entry>
* <entry>G_TYPE_INT</entry>
* <entry>0-36</entry>
* <entry>This field describes the power level of the tone, expressed in dBm0
* after dropping the sign. Power levels range from 0 to -63 dBm0. The range of * after dropping the sign. Power levels range from 0 to -63 dBm0. The range of
* valid DTMF is from 0 to -36 dBm0. Can be omitted if start is set to FALSE. * valid DTMF is from 0 to -36 dBm0. Can be omitted if start is set to FALSE.
* </entry> *
* </row> * * `start` (G_TYPE_BOOLEAN, True or False): Whether the event is starting or ending.
* <row> *
* <entry>start</entry> * * `method` (G_TYPE_INT, 2): The method used for sending event, this element will react if this
* <entry>G_TYPE_BOOLEAN</entry>
* <entry>True or False</entry>
* <entry>Whether the event is starting or ending.</entry>
* </row>
* <row>
* <entry>method</entry>
* <entry>G_TYPE_INT</entry>
* <entry>2</entry>
* <entry>The method used for sending event, this element will react if this
* field is absent or 2. * field is absent or 2.
* </entry>
* </row>
* </tbody>
* </tgroup>
* </informaltable>
* *
* For example, the following code informs the pipeline (and in turn, the * For example, the following code informs the pipeline (and in turn, the
* DTMFSrc element inside the pipeline) about the start of a DTMF named * DTMFSrc element inside the pipeline) about the start of a DTMF named
......
...@@ -27,58 +27,21 @@ ...@@ -27,58 +27,21 @@
* This element takes RTP DTMF packets and produces sound. It also emits a * This element takes RTP DTMF packets and produces sound. It also emits a
* message on the #GstBus. * message on the #GstBus.
* *
* The message is called "dtmf-event" and has the following fields * The message is called "dtmf-event" and has the following fields:
* <informaltable> *
* <tgroup cols='4'> * * `type` (G_TYPE_INT, 0-1): Which of the two methods
* <colspec colname='Name' />
* <colspec colname='Type' />
* <colspec colname='Possible values' />
* <colspec colname='Purpose' />
* <thead>
* <row>
* <entry>Name</entry>
* <entry>GType</entry>
* <entry>Possible values</entry>
* <entry>Purpose</entry>
* </row>
* </thead>
* <tbody>
* <row>
* <entry>type</entry>
* <entry>G_TYPE_INT</entry>
* <entry>0-1</entry>
* <entry>Which of the two methods
* specified in RFC 2833 to use. The value should be 0 for tones and 1 for * specified in RFC 2833 to use. The value should be 0 for tones and 1 for
* named events. Tones are specified by their frequencies and events are specied * named events. Tones are specified by their frequencies and events are specied
* by their number. This element currently only recognizes events. * by their number. This element currently only recognizes events.
* Do not confuse with "method" which specified the output. * Do not confuse with "method" which specified the output.
* </entry> *
* </row> * * `number` (G_TYPE_INT, 0-16): The event number.
* <row> *
* <entry>number</entry> * * `volume` (G_TYPE_INT, 0-36): This field describes the power level of the tone, expressed in dBm0
* <entry>G_TYPE_INT</entry>
* <entry>0-16</entry>
* <entry>The event number.</entry>
* </row>
* <row>
* <entry>volume</entry>
* <entry>G_TYPE_INT</entry>
* <entry>0-36</entry>
* <entry>This field describes the power level of the tone, expressed in dBm0
* after dropping the sign. Power levels range from 0 to -63 dBm0. The range of * after dropping the sign. Power levels range from 0 to -63 dBm0. The range of
* valid DTMF is from 0 to -36 dBm0. * valid DTMF is from 0 to -36 dBm0.
* </entry> *
* </row> * * `method` (G_TYPE_INT, 1): This field will always been 1 (ie RTP event) from this element.
* <row>
* <entry>method</entry>
* <entry>G_TYPE_INT</entry>
* <entry>1</entry>
* <entry>This field will always been 1 (ie RTP event) from this element.
* </entry>
* </row>
* </tbody>
* </tgroup>
* </informaltable>
*/ */
#ifdef HAVE_CONFIG_H #ifdef HAVE_CONFIG_H
......
...@@ -35,64 +35,22 @@ ...@@ -35,64 +35,22 @@
* structure of name "dtmf-event" with fields set according to the following * structure of name "dtmf-event" with fields set according to the following
* table: * table:
* *
* <informaltable> * * `type` (G_TYPE_INT, 0-1): The application uses this field to specify which of the two methods
* <tgroup cols='4'>
* <colspec colname='Name' />
* <colspec colname='Type' />
* <colspec colname='Possible values' />
* <colspec colname='Purpose' />
* <thead>
* <row>
* <entry>Name</entry>
* <entry>GType</entry>
* <entry>Possible values</entry>
* <entry>Purpose</entry>
* </row>
* </thead>
* <tbody>
* <row>
* <entry>type</entry>
* <entry>G_TYPE_INT</entry>
* <entry>0-1</entry>
* <entry>The application uses this field to specify which of the two methods
* specified in RFC 2833 to use. The value should be 0 for tones and 1 for * specified in RFC 2833 to use. The value should be 0 for tones and 1 for
* named events. Tones are specified by their frequencies and events are specied * named events. Tones are specified by their frequencies and events are specied
* by their number. This element can only take events as input. Do not confuse * by their number. This element can only take events as input. Do not confuse
* with "method" which specified the output. * with "method" which specified the output.
* </entry> *
* </row> * * `number` (G_TYPE_INT, 0-15): The event number.
* <row> *
* <entry>number</entry> * * `volume` (G_TYPE_INT, 0-36): This field describes the power level of the tone, expressed in dBm0
* <entry>G_TYPE_INT</entry>
* <entry>0-15</entry>
* <entry>The event number.</entry>
* </row>
* <row>
* <entry>volume</entry>
* <entry>G_TYPE_INT</entry>
* <entry>0-36</entry>
* <entry>This field describes the power level of the tone, expressed in dBm0
* after dropping the sign. Power levels range from 0 to -63 dBm0. The range of * after dropping the sign. Power levels range from 0 to -63 dBm0. The range of
* valid DTMF is from 0 to -36 dBm0. Can be omitted if start is set to FALSE. * valid DTMF is from 0 to -36 dBm0. Can be omitted if start is set to FALSE.
* </entry> *
* </row> * * `start` (G_TYPE_BOOLEAN, True or False): Whether the event is starting or ending.
* <row> *
* <entry>start</entry> * * `method` (G_TYPE_INT, 1): The method used for sending event, this element will react if this
* <entry>G_TYPE_BOOLEAN</entry>
* <entry>True or False</entry>
* <entry>Whether the event is starting or ending.</entry>
* </row>
* <row>
* <entry>method</entry>
* <entry>G_TYPE_INT</entry>
* <entry>1</entry>
* <entry>The method used for sending event, this element will react if this
* field is absent or 1. * field is absent or 1.
* </entry>
* </row>
* </tbody>
* </tgroup>
* </informaltable>
* *
* For example, the following code informs the pipeline (and in turn, the * For example, the following code informs the pipeline (and in turn, the
* RTPDTMFSrc element inside the pipeline) about the start of an RTP DTMF named * RTPDTMFSrc element inside the pipeline) about the start of an RTP DTMF named
......
...@@ -45,9 +45,7 @@ ...@@ -45,9 +45,7 @@
* *
* ## Example application * ## Example application
* *
* <informalexample><programlisting language="C"> * {{ tests/examples/level/level-example.c }}
* <xi:include xmlns:xi="http://www.w3.org/2003/XInclude" parse="text" href="../../../../tests/examples/level/level-example.c" />
* </programlisting></informalexample>
* *
*/ */
......
...@@ -25,13 +25,12 @@ ...@@ -25,13 +25,12 @@
* Extract raw audio from RTP packets according to RFC 3551. * Extract raw audio from RTP packets according to RFC 3551.
* For detailed information see: http://www.rfc-editor.org/rfc/rfc3551.txt * For detailed information see: http://www.rfc-editor.org/rfc/rfc3551.txt
* *
* <refsect2> * ## Example pipeline
* <title>Example pipeline</title> *
* |[ * |[
* gst-launch udpsrc caps='application/x-rtp, media=(string)audio, clock-rate=(int)44100, encoding-name=(string)L8, encoding-params=(string)1, channels=(int)1, payload=(int)96' ! rtpL8depay ! pulsesink * gst-launch udpsrc caps='application/x-rtp, media=(string)audio, clock-rate=(int)44100, encoding-name=(string)L8, encoding-params=(string)1, channels=(int)1, payload=(int)96' ! rtpL8depay ! pulsesink
* ]| This example pipeline will depayload an RTP raw audio stream. Refer to * ]| This example pipeline will depayload an RTP raw audio stream. Refer to
* the rtpL8pay example to create the RTP stream. * the rtpL8pay example to create the RTP stream.
* </refsect2>
*/ */
#ifdef HAVE_CONFIG_H #ifdef HAVE_CONFIG_H
......
...@@ -25,13 +25,12 @@ ...@@ -25,13 +25,12 @@
* Payload raw audio into RTP packets according to RFC 3551. * Payload raw audio into RTP packets according to RFC 3551.
* For detailed information see: http://www.rfc-editor.org/rfc/rfc3551.txt * For detailed information see: http://www.rfc-editor.org/rfc/rfc3551.txt
* *
* <refsect2> * ## Example pipeline
* <title>Example pipeline</title> *
* |[ * |[
* gst-launch -v audiotestsrc ! audioconvert ! rtpL8pay ! udpsink * gst-launch -v audiotestsrc ! audioconvert ! rtpL8pay ! udpsink
* ]| This example pipeline will payload raw audio. Refer to * ]| This example pipeline will payload raw audio. Refer to
* the rtpL8depay example to depayload and play the RTP stream. * the rtpL8depay example to depayload and play the RTP stream.
* </refsect2>
*/ */
#ifdef HAVE_CONFIG_H #ifdef HAVE_CONFIG_H
......
...@@ -36,12 +36,11 @@ ...@@ -36,12 +36,11 @@
* When using #GstRtpBin, this element should be inserted through the * When using #GstRtpBin, this element should be inserted through the
* #GstRtpBin::request-aux-receiver signal. * #GstRtpBin::request-aux-receiver signal.
* *
* <refsect2> * ## Example pipeline
* <title>Example pipeline</title> *
* |[ * |[
* gst-launch-1.0 udpsrc port=8888 caps="application/x-rtp, payload=96, clock-rate=90000" ! rtpreddec pt=122 ! rtpstorage size-time=220000000 ! rtpssrcdemux ! application/x-rtp, payload=96, clock-rate=90000, media=video, encoding-name=H264 ! rtpjitterbuffer do-lost=1 latency=200 ! rtpulpfecdec pt=122 ! rtph264depay ! avdec_h264 ! videoconvert ! autovideosink * gst-launch-1.0 udpsrc port=8888 caps="application/x-rtp, payload=96, clock-rate=90000" ! rtpreddec pt=122 ! rtpstorage size-time=220000000 ! rtpssrcdemux ! application/x-rtp, payload=96, clock-rate=90000, media=video, encoding-name=H264 ! rtpjitterbuffer do-lost=1 latency=200 ! rtpulpfecdec pt=122 ! rtph264depay ! avdec_h264 ! videoconvert ! autovideosink
* ]| This example will receive a stream with RED and ULP FEC and try to reconstruct the packets. * ]| This example will receive a stream with RED and ULP FEC and try to reconstruct the packets.
* </refsect2>
* *
* See also: #GstRtpRedEnc, #GstWebRTCBin, #GstRtpBin * See also: #GstRtpRedEnc, #GstWebRTCBin, #GstRtpBin
* Since: 1.14 * Since: 1.14
......
...@@ -38,12 +38,11 @@ ...@@ -38,12 +38,11 @@
* When using #GstRtpBin, this element should be inserted through the * When using #GstRtpBin, this element should be inserted through the
* #GstRtpBin::request-fec-encoder signal. * #GstRtpBin::request-fec-encoder signal.
* *
* <refsect2> * ## Example pipeline
* <title>Example pipeline</title> *
* |[ * |[
* gst-launch-1.0 videotestsrc ! x264enc ! video/x-h264, profile=baseline ! rtph264pay pt=96 ! rtpulpfecenc percentage=100 pt=122 ! rtpredenc pt=122 distance=2 ! identity drop-probability=0.05 ! udpsink port=8888 * gst-launch-1.0 videotestsrc ! x264enc ! video/x-h264, profile=baseline ! rtph264pay pt=96 ! rtpulpfecenc percentage=100 pt=122 ! rtpredenc pt=122 distance=2 ! identity drop-probability=0.05 ! udpsink port=8888
* ]| This example will send a stream with RED and ULP FEC. * ]| This example will send a stream with RED and ULP FEC.
* </refsect2>
* *
* See also: #GstRtpRedDec, #GstWebRTCBin, #GstRtpBin * See also: #GstRtpRedDec, #GstWebRTCBin, #GstRtpBin
* Since: 1.14 * Since: 1.14
......
...@@ -44,18 +44,16 @@ ...@@ -44,18 +44,16 @@
* When using #GstRtpBin, this element should be inserted through the * When using #GstRtpBin, this element should be inserted through the
* #GstRtpBin::request-fec-decoder signal. * #GstRtpBin::request-fec-decoder signal.
* *
* <refsect2> * ## Example pipeline
* <title>Example pipeline</title> *
* |[ * |[
* gst-launch-1.0 udpsrc port=8888 caps="application/x-rtp, payload=96, clock-rate=90000" ! rtpstorage size-time=220000000 ! rtpssrcdemux ! application/x-rtp, payload=96, clock-rate=90000, media=video, encoding-name=H264 ! rtpjitterbuffer do-lost=1 latency=200 ! rtpulpfecdec pt=122 ! rtph264depay ! avdec_h264 ! videoconvert ! autovideosink * gst-launch-1.0 udpsrc port=8888 caps="application/x-rtp, payload=96, clock-rate=90000" ! rtpstorage size-time=220000000 ! rtpssrcdemux ! application/x-rtp, payload=96, clock-rate=90000, media=video, encoding-name=H264 ! rtpjitterbuffer do-lost=1 latency=200 ! rtpulpfecdec pt=122 ! rtph264depay ! avdec_h264 ! videoconvert ! autovideosink
* ]| This example will receive a stream with FEC and try to reconstruct the packets. * ]| This example will receive a stream with FEC and try to reconstruct the packets.
* *
* Example programs are available at * Example programs are available at
* <ulink url="https://gitlab.freedesktop.org/gstreamer/gstreamer-rs/blob/master/examples/src/bin/rtpfecserver.rs">rtpfecserver.rs</ulink> * <https://gitlab.freedesktop.org/gstreamer/gstreamer-rs/blob/master/examples/src/bin/rtpfecserver.rs>
* and * and
* <ulink url="https://gitlab.freedesktop.org/gstreamer/gstreamer-rs/blob/master/examples/src/bin/rtpfecclient.rs">rtpfecclient.rs</ulink> * <https://gitlab.freedesktop.org/gstreamer/gstreamer-rs/blob/master/examples/src/bin/rtpfecclient.rs>
*
* </refsect2>
* *
* See also: #GstRtpUlpFecEnc, #GstRtpBin, #GstRtpStorage * See also: #GstRtpUlpFecEnc, #GstRtpBin, #GstRtpStorage
* Since: 1.14 * Since: 1.14
......
...@@ -69,18 +69,16 @@ ...@@ -69,18 +69,16 @@
* When using #GstRtpBin, this element should be inserted through the * When using #GstRtpBin, this element should be inserted through the
* #GstRtpBin::request-fec-encoder signal. * #GstRtpBin::request-fec-encoder signal.
* *
* ## Example pipeline
* *
* <refsect2>
* <title>Example pipeline</title>
* |[ * |[
* gst-launch-1.0 videotestsrc ! x264enc ! video/x-h264, profile=baseline ! rtph264pay pt=96 ! rtpulpfecenc percentage=100 pt=122 ! udpsink port=8888 * gst-launch-1.0 videotestsrc ! x264enc ! video/x-h264, profile=baseline ! rtph264pay pt=96 ! rtpulpfecenc percentage=100 pt=122 ! udpsink port=8888
* ]| This example will receive a stream with FEC and try to reconstruct the packets. * ]| This example will receive a stream with FEC and try to reconstruct the packets.
* *
* Example programs are available at * Example programs are available at
* <ulink url="https://gitlab.freedesktop.org/gstreamer/gstreamer-rs/blob/master/examples/src/bin/rtpfecserver.rs">rtpfecserver.rs</ulink> * <https://gitlab.freedesktop.org/gstreamer/gstreamer-rs/blob/master/examples/src/bin/rtpfecserver.rs>
* and * and
* <ulink url="https://gitlab.freedesktop.org/gstreamer/gstreamer-rs/blob/master/examples/src/bin/rtpfecclient.rs">rtpfecclient.rs</ulink> * <https://gitlab.freedesktop.org/gstreamer/gstreamer-rs/blob/master/examples/src/bin/rtpfecclient.rs>
* </refsect2>
* *
* See also: #GstRtpUlpFecDec, #GstRtpBin * See also: #GstRtpUlpFecDec, #GstRtpBin
* Since: 1.14 * Since: 1.14
......
...@@ -49,9 +49,7 @@ ...@@ -49,9 +49,7 @@
* *
* ## Example application * ## Example application
* *
* <informalexample><programlisting language="C"> * {{ tests/examples/spectrum/spectrum-example.c }}
* <xi:include xmlns:xi="http://www.w3.org/2003/XInclude" parse="text" href="../../../../tests/examples/spectrum/spectrum-example.c" />
* </programlisting></informalexample>
* *
*/ */
......
...@@ -1231,7 +1231,7 @@ failed: ...@@ -1231,7 +1231,7 @@ failed:
/* /*
* Get the list of supported capture formats, a list of * Get the list of supported capture formats, a list of
* <code>struct v4l2_fmtdesc</code>. * `struct v4l2_fmtdesc`.
*/ */
static GSList * static GSList *
gst_v4l2_object_get_format_list (GstV4l2Object * v4l2object) gst_v4l2_object_get_format_list (GstV4l2Object * v4l2object)
......
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