mirror of
https://gitlab.freedesktop.org/gstreamer/gstreamer.git
synced 2024-12-27 18:50:48 +00:00
doc: remove xml from comments
This commit is contained in:
Mathieu Duponchelle
parent
89380bddea
commit
f554369ed5
22 changed files with 83 additions and 220 deletions
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@ -91,5 +91,6 @@ foreach plugin_name: list_plugin_res.stdout().split(':')
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disable_incremental_build: true,
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gst_cache_file: plugins_cache,
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gst_plugin_name: plugin_name,
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include_paths: join_paths(meson.current_source_dir(), '..'),
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)]
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endforeach
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@ -30,16 +30,16 @@
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* consideration. See <ulink url="http://www.vorbis.com/">Ogg/Vorbis</ulink>
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* for a royalty free (and often higher quality) alternative.
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*
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* <refsect2>
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* <title>Output sample rate</title>
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* ## Output sample rate
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*
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* If no fixed output sample rate is negotiated on the element's src pad,
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* the element will choose an optimal sample rate to resample to internally.
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* For example, a 16-bit 44.1 KHz mono audio stream encoded at 48 kbit will
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* get resampled to 32 KHz. Use filter caps on the src pad to force a
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* particular sample rate.
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* </refsect2>
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* <refsect2>
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* <title>Example pipelines</title>
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*
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* ## Example pipelines
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*
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* |[
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* gst-launch-1.0 -v audiotestsrc wave=sine num-buffers=100 ! audioconvert ! lamemp3enc ! filesink location=sine.mp3
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* ]| Encode a test sine signal to MP3.
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@ -55,7 +55,6 @@
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* |[
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* 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
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* ]| Encode to a fixed sample rate
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* </refsect2>
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*/
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#ifdef HAVE_CONFIG_H
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@ -22,12 +22,11 @@
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*
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* Audio decoder for MPEG-1 layer 1/2/3 audio data.
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*
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* <refsect2>
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* <title>Example pipelines</title>
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* ## Example pipelines
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*
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* |[
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* gst-launch-1.0 filesrc location=music.mp3 ! mpegaudioparse ! mpg123audiodec ! audioconvert ! audioresample ! autoaudiosink
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* ]| Decode and play the mp3 file
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* </refsect2>
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*/
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#ifdef HAVE_CONFIG_H
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@ -30,8 +30,8 @@
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* Tags sent by upstream elements will be picked up automatically (and merged
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* according to the merge mode set via the tag setter interface).
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*
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* <refsect2>
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* <title>Example pipelines</title>
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* ## Example pipelines
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*
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* |[
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* gst-launch-1.0 -v filesrc location=foo.ogg ! decodebin ! audioconvert ! lame ! apev2mux ! filesink location=foo.mp3
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* ]| A pipeline that transcodes a file from Ogg/Vorbis to mp3 format with an
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@ -40,7 +40,6 @@
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* |[
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* gst-launch-1.0 -m filesrc location=foo.mp3 ! apedemux ! fakesink silent=TRUE 2> /dev/null | grep taglist
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* ]| Verify that tags have been written.
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* </refsect2>
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*/
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#ifdef HAVE_CONFIG_H
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@ -31,8 +31,8 @@
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* Tags sent by upstream elements will be picked up automatically (and merged
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* according to the merge mode set via the tag setter interface).
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*
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* <refsect2>
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* <title>Example pipelines</title>
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* ## Example pipelines
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*
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* |[
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* gst-launch-1.0 -v filesrc location=foo.ogg ! decodebin ! audioconvert ! lame ! id3v2mux ! filesink location=foo.mp3
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* ]| A pipeline that transcodes a file from Ogg/Vorbis to mp3 format with an
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@ -41,7 +41,6 @@
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* |[
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* gst-launch-1.0 -m filesrc location=foo.mp3 ! id3demux ! fakesink silent=TRUE 2> /dev/null | grep taglist
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* ]| Verify that tags have been written.
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* </refsect2>
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*/
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#ifdef HAVE_CONFIG_H
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@ -30,8 +30,8 @@
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*
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* This element encodes raw integer audio into an MPEG-1 layer 2 (MP2) stream.
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*
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* <refsect2>
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* <title>Example pipelines</title>
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* ## Example pipelines
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*
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* |[
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* gst-launch-1.0 -v audiotestsrc wave=sine num-buffers=100 ! audioconvert ! twolame ! filesink location=sine.mp2
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* ]| Encode a test sine signal to MP2.
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@ -44,7 +44,6 @@
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* |[
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* gst-launch-1.0 -v cdda://5 ! audioconvert ! twolame bitrate=192 ! filesink location=track5.mp2
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* ]| Encode Audio CD track 5 to MP2
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* </refsect2>
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*
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*/
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@ -25,7 +25,7 @@
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*
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* autoaudiosink is an audio sink that automatically detects an appropriate
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* audio sink to use. It does so by scanning the registry for all elements
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* that have <quote>Sink</quote> and <quote>Audio</quote> in the class field
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* that have "Sink" and "Audio" in the class field
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* of their element information, and also have a non-zero autoplugging rank.
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*
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* ## Example launch line
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@ -26,7 +26,7 @@
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*
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* autoaudiosrc is an audio source that automatically detects an appropriate
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* audio source to use. It does so by scanning the registry for all elements
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* that have <quote>Source</quote> and <quote>Audio</quote> in the class field
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* that have "Source" and "Audio" in the class field
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* of their element information, and also have a non-zero autoplugging rank.
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*
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* ## Example launch line
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@ -25,7 +25,7 @@
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*
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* autovideosink is a video sink that automatically detects an appropriate
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* video sink to use. It does so by scanning the registry for all elements
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* that have <quote>Sink</quote> and <quote>Video</quote> in the class field
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* that have "Sink" and "Video" in the class field
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* of their element information, and also have a non-zero autoplugging rank.
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*
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* ## Example launch line
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@ -26,7 +26,7 @@
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*
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* autovideosrc is a video src that automatically detects an appropriate
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* video source to use. It does so by scanning the registry for all elements
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* that have <quote>Source</quote> and <quote>Video</quote> in the class field
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* that have "Source" and "Video" in the class field
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* of their element information, and also have a non-zero autoplugging rank.
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*
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* ## Example launch line
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@ -37,64 +37,22 @@
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* structure of name "dtmf-event" with fields set according to the following
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* table:
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*
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* <informaltable>
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* <tgroup cols='4'>
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* <colspec colname='Name' />
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* <colspec colname='Type' />
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* <colspec colname='Possible values' />
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* <colspec colname='Purpose' />
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* <thead>
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* <row>
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* <entry>Name</entry>
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* <entry>GType</entry>
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* <entry>Possible values</entry>
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* <entry>Purpose</entry>
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* </row>
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* </thead>
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* <tbody>
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* <row>
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* <entry>type</entry>
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* <entry>G_TYPE_INT</entry>
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* <entry>0-1</entry>
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* <entry>The application uses this field to specify which of the two methods
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* specified in RFC 2833 to use. The value should be 0 for tones and 1 for
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* named events. Tones are specified by their frequencies and events are specied
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* by their number. This element can only take events as input. Do not confuse
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* with "method" which specified the output.
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* </entry>
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* </row>
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* <row>
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* <entry>number</entry>
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* <entry>G_TYPE_INT</entry>
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* <entry>0-15</entry>
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* <entry>The event number.</entry>
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* </row>
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* <row>
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* <entry>volume</entry>
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* <entry>G_TYPE_INT</entry>
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* <entry>0-36</entry>
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* <entry>This field describes the power level of the tone, expressed in dBm0
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* after dropping the sign. Power levels range from 0 to -63 dBm0. The range of
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* valid DTMF is from 0 to -36 dBm0. Can be omitted if start is set to FALSE.
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* </entry>
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* </row>
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* <row>
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* <entry>start</entry>
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* <entry>G_TYPE_BOOLEAN</entry>
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* <entry>True or False</entry>
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* <entry>Whether the event is starting or ending.</entry>
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* </row>
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* <row>
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* <entry>method</entry>
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* <entry>G_TYPE_INT</entry>
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* <entry>2</entry>
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* <entry>The method used for sending event, this element will react if this
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* field is absent or 2.
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* </entry>
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* </row>
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* </tbody>
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* </tgroup>
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* </informaltable>
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* * `type` (G_TYPE_INT, 0-1): The application uses this field to specify which of the two methods
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* specified in RFC 2833 to use. The value should be 0 for tones and 1 for
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* named events. Tones are specified by their frequencies and events are specied
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* by their number. This element can only take events as input. Do not confuse
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* with "method" which specified the output.
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*
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* * `number` (G_TYPE_INT, 0-15): The event number.
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*
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* * `volume` (G_TYPE_INT, 0-36): This field describes the power level of the tone, expressed in dBm0
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* after dropping the sign. Power levels range from 0 to -63 dBm0. The range of
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* valid DTMF is from 0 to -36 dBm0. Can be omitted if start is set to FALSE.
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*
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* * `start` (G_TYPE_BOOLEAN, True or False): Whether the event is starting or ending.
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*
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* * `method` (G_TYPE_INT, 2): The method used for sending event, this element will react if this
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* field is absent or 2.
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*
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* For example, the following code informs the pipeline (and in turn, the
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* DTMFSrc element inside the pipeline) about the start of a DTMF named
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@ -27,58 +27,21 @@
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* This element takes RTP DTMF packets and produces sound. It also emits a
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* message on the #GstBus.
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*
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* The message is called "dtmf-event" and has the following fields
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* <informaltable>
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* <tgroup cols='4'>
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* <colspec colname='Name' />
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* <colspec colname='Type' />
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* <colspec colname='Possible values' />
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* <colspec colname='Purpose' />
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* <thead>
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* <row>
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* <entry>Name</entry>
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* <entry>GType</entry>
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* <entry>Possible values</entry>
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* <entry>Purpose</entry>
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* </row>
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* </thead>
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* <tbody>
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* <row>
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* <entry>type</entry>
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* <entry>G_TYPE_INT</entry>
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* <entry>0-1</entry>
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* <entry>Which of the two methods
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* specified in RFC 2833 to use. The value should be 0 for tones and 1 for
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* named events. Tones are specified by their frequencies and events are specied
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* by their number. This element currently only recognizes events.
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* Do not confuse with "method" which specified the output.
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* </entry>
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* </row>
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* <row>
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* <entry>number</entry>
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* <entry>G_TYPE_INT</entry>
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* <entry>0-16</entry>
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* <entry>The event number.</entry>
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* </row>
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* <row>
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* <entry>volume</entry>
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* <entry>G_TYPE_INT</entry>
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* <entry>0-36</entry>
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* <entry>This field describes the power level of the tone, expressed in dBm0
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* after dropping the sign. Power levels range from 0 to -63 dBm0. The range of
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* valid DTMF is from 0 to -36 dBm0.
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* </entry>
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* </row>
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* <row>
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* <entry>method</entry>
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* <entry>G_TYPE_INT</entry>
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* <entry>1</entry>
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* <entry>This field will always been 1 (ie RTP event) from this element.
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* </entry>
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* </row>
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* </tbody>
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* </tgroup>
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* </informaltable>
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* The message is called "dtmf-event" and has the following fields:
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*
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* * `type` (G_TYPE_INT, 0-1): Which of the two methods
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* specified in RFC 2833 to use. The value should be 0 for tones and 1 for
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* named events. Tones are specified by their frequencies and events are specied
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* by their number. This element currently only recognizes events.
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* Do not confuse with "method" which specified the output.
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*
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* * `number` (G_TYPE_INT, 0-16): The event number.
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*
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* * `volume` (G_TYPE_INT, 0-36): This field describes the power level of the tone, expressed in dBm0
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* after dropping the sign. Power levels range from 0 to -63 dBm0. The range of
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* valid DTMF is from 0 to -36 dBm0.
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*
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* * `method` (G_TYPE_INT, 1): This field will always been 1 (ie RTP event) from this element.
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*/
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#ifdef HAVE_CONFIG_H
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@ -35,64 +35,22 @@
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* structure of name "dtmf-event" with fields set according to the following
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* table:
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*
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* <informaltable>
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* <tgroup cols='4'>
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* <colspec colname='Name' />
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* <colspec colname='Type' />
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* <colspec colname='Possible values' />
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* <colspec colname='Purpose' />
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* <thead>
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* <row>
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* <entry>Name</entry>
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* <entry>GType</entry>
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* <entry>Possible values</entry>
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* <entry>Purpose</entry>
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* </row>
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* </thead>
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* <tbody>
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* <row>
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* <entry>type</entry>
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* <entry>G_TYPE_INT</entry>
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* <entry>0-1</entry>
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* <entry>The application uses this field to specify which of the two methods
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* specified in RFC 2833 to use. The value should be 0 for tones and 1 for
|
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* named events. Tones are specified by their frequencies and events are specied
|
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* by their number. This element can only take events as input. Do not confuse
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* with "method" which specified the output.
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* </entry>
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* </row>
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* <row>
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* <entry>number</entry>
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* <entry>G_TYPE_INT</entry>
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* <entry>0-15</entry>
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* <entry>The event number.</entry>
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* </row>
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* <row>
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* <entry>volume</entry>
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* <entry>G_TYPE_INT</entry>
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* <entry>0-36</entry>
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* <entry>This field describes the power level of the tone, expressed in dBm0
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* after dropping the sign. Power levels range from 0 to -63 dBm0. The range of
|
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* valid DTMF is from 0 to -36 dBm0. Can be omitted if start is set to FALSE.
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* </entry>
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* </row>
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* <row>
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* <entry>start</entry>
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* <entry>G_TYPE_BOOLEAN</entry>
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* <entry>True or False</entry>
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* <entry>Whether the event is starting or ending.</entry>
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* </row>
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* <row>
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* <entry>method</entry>
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||||
* <entry>G_TYPE_INT</entry>
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* <entry>1</entry>
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||||
* <entry>The method used for sending event, this element will react if this
|
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* field is absent or 1.
|
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* </entry>
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||||
* </row>
|
||||
* </tbody>
|
||||
* </tgroup>
|
||||
* </informaltable>
|
||||
* * `type` (G_TYPE_INT, 0-1): 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
|
||||
* 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
|
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* with "method" which specified the output.
|
||||
*
|
||||
* * `number` (G_TYPE_INT, 0-15): The event number.
|
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*
|
||||
* * `volume` (G_TYPE_INT, 0-36): 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
|
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* valid DTMF is from 0 to -36 dBm0. Can be omitted if start is set to FALSE.
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*
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* * `start` (G_TYPE_BOOLEAN, True or False): Whether the event is starting or ending.
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*
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* * `method` (G_TYPE_INT, 1): The method used for sending event, this element will react if this
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* field is absent or 1.
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*
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* For example, the following code informs the pipeline (and in turn, the
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* RTPDTMFSrc element inside the pipeline) about the start of an RTP DTMF named
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|
|
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@ -45,9 +45,7 @@
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*
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* ## Example application
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*
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* <informalexample><programlisting language="C">
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* <xi:include xmlns:xi="http://www.w3.org/2003/XInclude" parse="text" href="../../../../tests/examples/level/level-example.c" />
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* </programlisting></informalexample>
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* {{ tests/examples/level/level-example.c }}
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*
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*/
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|
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@ -25,13 +25,12 @@
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* Extract raw audio from RTP packets according to RFC 3551.
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* For detailed information see: http://www.rfc-editor.org/rfc/rfc3551.txt
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*
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||||
* <refsect2>
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||||
* <title>Example pipeline</title>
|
||||
* ## Example pipeline
|
||||
*
|
||||
* |[
|
||||
* 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
|
||||
* the rtpL8pay example to create the RTP stream.
|
||||
* </refsect2>
|
||||
*/
|
||||
|
||||
#ifdef HAVE_CONFIG_H
|
||||
|
|
|
|||
|
|
@ -25,13 +25,12 @@
|
|||
* Payload raw audio into RTP packets according to RFC 3551.
|
||||
* For detailed information see: http://www.rfc-editor.org/rfc/rfc3551.txt
|
||||
*
|
||||
* <refsect2>
|
||||
* <title>Example pipeline</title>
|
||||
* ## Example pipeline
|
||||
*
|
||||
* |[
|
||||
* gst-launch -v audiotestsrc ! audioconvert ! rtpL8pay ! udpsink
|
||||
* ]| This example pipeline will payload raw audio. Refer to
|
||||
* the rtpL8depay example to depayload and play the RTP stream.
|
||||
* </refsect2>
|
||||
*/
|
||||
|
||||
#ifdef HAVE_CONFIG_H
|
||||
|
|
|
|||
|
|
@ -36,12 +36,11 @@
|
|||
* When using #GstRtpBin, this element should be inserted through the
|
||||
* #GstRtpBin::request-aux-receiver signal.
|
||||
*
|
||||
* <refsect2>
|
||||
* <title>Example pipeline</title>
|
||||
* ## Example pipeline
|
||||
*
|
||||
* |[
|
||||
* 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.
|
||||
* </refsect2>
|
||||
*
|
||||
* See also: #GstRtpRedEnc, #GstWebRTCBin, #GstRtpBin
|
||||
* Since: 1.14
|
||||
|
|
|
|||
|
|
@ -38,12 +38,11 @@
|
|||
* When using #GstRtpBin, this element should be inserted through the
|
||||
* #GstRtpBin::request-fec-encoder signal.
|
||||
*
|
||||
* <refsect2>
|
||||
* <title>Example pipeline</title>
|
||||
* ## Example pipeline
|
||||
*
|
||||
* |[
|
||||
* 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.
|
||||
* </refsect2>
|
||||
*
|
||||
* See also: #GstRtpRedDec, #GstWebRTCBin, #GstRtpBin
|
||||
* Since: 1.14
|
||||
|
|
|
|||
|
|
@ -44,18 +44,16 @@
|
|||
* When using #GstRtpBin, this element should be inserted through the
|
||||
* #GstRtpBin::request-fec-decoder signal.
|
||||
*
|
||||
* <refsect2>
|
||||
* <title>Example pipeline</title>
|
||||
* ## Example pipeline
|
||||
*
|
||||
* |[
|
||||
* 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.
|
||||
*
|
||||
* 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
|
||||
* <ulink url="https://gitlab.freedesktop.org/gstreamer/gstreamer-rs/blob/master/examples/src/bin/rtpfecclient.rs">rtpfecclient.rs</ulink>
|
||||
*
|
||||
* </refsect2>
|
||||
* <https://gitlab.freedesktop.org/gstreamer/gstreamer-rs/blob/master/examples/src/bin/rtpfecclient.rs>
|
||||
*
|
||||
* See also: #GstRtpUlpFecEnc, #GstRtpBin, #GstRtpStorage
|
||||
* Since: 1.14
|
||||
|
|
|
|||
|
|
@ -69,18 +69,16 @@
|
|||
* When using #GstRtpBin, this element should be inserted through the
|
||||
* #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
|
||||
* ]| This example will receive a stream with FEC and try to reconstruct the packets.
|
||||
*
|
||||
* 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
|
||||
* <ulink url="https://gitlab.freedesktop.org/gstreamer/gstreamer-rs/blob/master/examples/src/bin/rtpfecclient.rs">rtpfecclient.rs</ulink>
|
||||
* </refsect2>
|
||||
* <https://gitlab.freedesktop.org/gstreamer/gstreamer-rs/blob/master/examples/src/bin/rtpfecclient.rs>
|
||||
*
|
||||
* See also: #GstRtpUlpFecDec, #GstRtpBin
|
||||
* Since: 1.14
|
||||
|
|
|
|||
|
|
@ -49,9 +49,7 @@
|
|||
*
|
||||
* ## Example application
|
||||
*
|
||||
* <informalexample><programlisting language="C">
|
||||
* <xi:include xmlns:xi="http://www.w3.org/2003/XInclude" parse="text" href="../../../../tests/examples/spectrum/spectrum-example.c" />
|
||||
* </programlisting></informalexample>
|
||||
* {{ tests/examples/spectrum/spectrum-example.c }}
|
||||
*
|
||||
*/
|
||||
|
||||
|
|
|
|||
|
|
@ -1231,7 +1231,7 @@ failed:
|
|||
|
||||
/*
|
||||
* Get the list of supported capture formats, a list of
|
||||
* <code>struct v4l2_fmtdesc</code>.
|
||||
* `struct v4l2_fmtdesc`.
|
||||
*/
|
||||
static GSList *
|
||||
gst_v4l2_object_get_format_list (GstV4l2Object * v4l2object)
|
||||
|
|
|
|||
Loading…
Reference in a new issue