doc: remove xml from comments

diff --git a/docs/meson.build b/docs/meson.build
index b5ea554..17cc4c1 100644 (file)
--- a/docs/meson.build
+++ b/docs/meson.build
@@ -91,5 +91,6 @@ foreach plugin_name: list_plugin_res.stdout().split(':')
       disable_incremental_build: true,
       gst_cache_file: plugins_cache,
       gst_plugin_name: plugin_name,
+      include_paths: join_paths(meson.current_source_dir(), '..'),
       )]
 endforeach

diff --git a/ext/lame/gstlamemp3enc.c b/ext/lame/gstlamemp3enc.c
index a824c5c..6f6962c 100644 (file)
--- a/ext/lame/gstlamemp3enc.c
+++ b/ext/lame/gstlamemp3enc.c
@@ -30,16 +30,16 @@
  * consideration. See <ulink url="http://www.vorbis.com/">Ogg/Vorbis</ulink>
  * for a royalty free (and often higher quality) alternative.
  *
- * <refsect2>
- * <title>Output sample rate</title>
+ * ## Output sample rate
+ *
  * 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.
  * 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
  * particular sample rate.
- * </refsect2>
- * <refsect2>
- * <title>Example pipelines</title>
+ *
+ * ## Example pipelines
+ *
  * |[
  * gst-launch-1.0 -v audiotestsrc wave=sine num-buffers=100 ! audioconvert ! lamemp3enc ! filesink location=sine.mp3
  * ]| Encode a test sine signal to MP3.
@@ -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
  * ]| Encode to a fixed sample rate
- * </refsect2>
  */
 
 #ifdef HAVE_CONFIG_H

diff --git a/ext/mpg123/gstmpg123audiodec.c b/ext/mpg123/gstmpg123audiodec.c
index fa6743c..3e3c923 100644 (file)
--- a/ext/mpg123/gstmpg123audiodec.c
+++ b/ext/mpg123/gstmpg123audiodec.c
@@ -22,12 +22,11 @@
  *
  * Audio decoder for MPEG-1 layer 1/2/3 audio data.
  *
- * <refsect2>
- * <title>Example pipelines</title>
+ * ## Example pipelines
+ *
  * |[
  * gst-launch-1.0 filesrc location=music.mp3 ! mpegaudioparse ! mpg123audiodec ! audioconvert ! audioresample ! autoaudiosink
  * ]| Decode and play the mp3 file
- * </refsect2>
  */
 
 #ifdef HAVE_CONFIG_H

diff --git a/ext/taglib/gstapev2mux.cc b/ext/taglib/gstapev2mux.cc
index 9659daf..8a145d8 100644 (file)
--- a/ext/taglib/gstapev2mux.cc
+++ b/ext/taglib/gstapev2mux.cc
@@ -30,8 +30,8 @@
  * Tags sent by upstream elements will be picked up automatically (and merged
  * according to the merge mode set via the tag setter interface).
  *
- * <refsect2>
- * <title>Example pipelines</title>
+ * ## Example pipelines
+ *
  * |[
  * 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
@@ -40,7 +40,6 @@
  * |[
  * 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.
- * </refsect2>
  */
 
 #ifdef HAVE_CONFIG_H

diff --git a/ext/taglib/gstid3v2mux.cc b/ext/taglib/gstid3v2mux.cc
index 634c6ef..e342bcc 100644 (file)
--- a/ext/taglib/gstid3v2mux.cc
+++ b/ext/taglib/gstid3v2mux.cc
@@ -31,8 +31,8 @@
  * Tags sent by upstream elements will be picked up automatically (and merged
  * according to the merge mode set via the tag setter interface).
  *
- * <refsect2>
- * <title>Example pipelines</title>
+ * ## Example pipelines
+ *
  * |[
  * 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
@@ -41,7 +41,6 @@
  * |[
  * 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.
- * </refsect2>
  */
 
 #ifdef HAVE_CONFIG_H

diff --git a/ext/twolame/gsttwolamemp2enc.c b/ext/twolame/gsttwolamemp2enc.c
index 31bec6b..09e7a6c 100644 (file)
--- a/ext/twolame/gsttwolamemp2enc.c
+++ b/ext/twolame/gsttwolamemp2enc.c
@@ -30,8 +30,8 @@
  *
  * This element encodes raw integer audio into an MPEG-1 layer 2 (MP2) stream.
  *
- * <refsect2>
- * <title>Example pipelines</title>
+ * ## Example pipelines
+ *
  * |[
  * gst-launch-1.0 -v audiotestsrc wave=sine num-buffers=100 ! audioconvert ! twolame ! filesink location=sine.mp2
  * ]| Encode a test sine signal to MP2.
@@ -44,7 +44,6 @@
  * |[
  * gst-launch-1.0 -v cdda://5 ! audioconvert ! twolame bitrate=192 ! filesink location=track5.mp2
  * ]| Encode Audio CD track 5 to MP2
- * </refsect2>
  *
  */
 

diff --git a/gst/autodetect/gstautoaudiosink.c b/gst/autodetect/gstautoaudiosink.c
index f8f53ac..26d9f4f 100644 (file)
--- a/gst/autodetect/gstautoaudiosink.c
+++ b/gst/autodetect/gstautoaudiosink.c
@@ -25,7 +25,7 @@
  *
  * autoaudiosink is an audio sink that automatically detects an appropriate
  * 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.
  *
  * ## Example launch line

diff --git a/gst/autodetect/gstautoaudiosrc.c b/gst/autodetect/gstautoaudiosrc.c
index 828d6ad..bf3704b 100644 (file)
--- a/gst/autodetect/gstautoaudiosrc.c
+++ b/gst/autodetect/gstautoaudiosrc.c
@@ -26,7 +26,7 @@
  *
  * autoaudiosrc is an audio source that automatically detects an appropriate
  * 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.
  *
  * ## Example launch line

diff --git a/gst/autodetect/gstautovideosink.c b/gst/autodetect/gstautovideosink.c
index 8607b46..d46aa69 100644 (file)
--- a/gst/autodetect/gstautovideosink.c
+++ b/gst/autodetect/gstautovideosink.c
@@ -25,7 +25,7 @@
  *
  * autovideosink is a video sink that automatically detects an appropriate
  * 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.
  *
  * ## Example launch line

diff --git a/gst/autodetect/gstautovideosrc.c b/gst/autodetect/gstautovideosrc.c
index f0d12a6..e5778db 100644 (file)
--- a/gst/autodetect/gstautovideosrc.c
+++ b/gst/autodetect/gstautovideosrc.c
@@ -26,7 +26,7 @@
  *
  * autovideosrc is a video src that automatically detects an appropriate
  * 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.
  *
  * ## Example launch line

diff --git a/gst/dtmf/gstdtmfsrc.c b/gst/dtmf/gstdtmfsrc.c
index c728019..a5bcfcd 100644 (file)
--- a/gst/dtmf/gstdtmfsrc.c
+++ b/gst/dtmf/gstdtmfsrc.c
@@ -37,64 +37,22 @@
  * structure of name "dtmf-event" with fields set according to the following
  * table:
  *
- * <informaltable>
- * <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
- * 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
- * with "method" which specified the output.
- * </entry>
- * </row>
- * <row>
- * <entry>number</entry>
- * <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
- * valid DTMF is from 0 to -36 dBm0. Can be omitted if start is set to FALSE.
- * </entry>
- * </row>
- * <row>
- * <entry>start</entry>
- * <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.
- * </entry>
- * </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
+ *   with "method" which specified the output.
+ *
+ * * `number` (G_TYPE_INT, 0-15): The event number.
+ *
+ * * `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
+ *   valid DTMF is from 0 to -36 dBm0. Can be omitted if start is set to FALSE.
+ *
+ * * `start` (G_TYPE_BOOLEAN, True or False): Whether the event is starting or ending.
+ *
+ * * `method` (G_TYPE_INT, 2): The method used for sending event, this element will react if this
+ *   field is absent or 2.
  *
  * For example, the following code informs the pipeline (and in turn, the
  * DTMFSrc element inside the pipeline) about the start of a DTMF named

diff --git a/gst/dtmf/gstrtpdtmfdepay.c b/gst/dtmf/gstrtpdtmfdepay.c
index 67bb54a..a3a057f 100644 (file)
--- a/gst/dtmf/gstrtpdtmfdepay.c
+++ b/gst/dtmf/gstrtpdtmfdepay.c
@@ -27,58 +27,21 @@
  * This element takes RTP DTMF packets and produces sound. It also emits a
  * message on the #GstBus.
  *
- * The message is called "dtmf-event" and has the following fields
- * <informaltable>
- * <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>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 currently only recognizes events.
- * Do not confuse with "method" which specified the output.
- * </entry>
- * </row>
- * <row>
- * <entry>number</entry>
- * <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
- * valid DTMF is from 0 to -36 dBm0.
- * </entry>
- * </row>
- * <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>
+ * The message is called "dtmf-event" and has the following fields:
+ *
+ * * `type` (G_TYPE_INT, 0-1): 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 currently only recognizes events.
+ *   Do not confuse with "method" which specified the output.
+ *
+ * * `number` (G_TYPE_INT, 0-16): The event number.
+ *
+ * * `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
+ *   valid DTMF is from 0 to -36 dBm0.
+ *
+ * * `method` (G_TYPE_INT, 1): This field will always been 1 (ie RTP event) from this element.
  */
 
 #ifdef HAVE_CONFIG_H

diff --git a/gst/dtmf/gstrtpdtmfsrc.c b/gst/dtmf/gstrtpdtmfsrc.c
index dfd3b9d..652dbdc 100644 (file)
--- a/gst/dtmf/gstrtpdtmfsrc.c
+++ b/gst/dtmf/gstrtpdtmfsrc.c
@@ -35,64 +35,22 @@
  * structure of name "dtmf-event" with fields set according to the following
  * table:
  *
- * <informaltable>
- * <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
- * 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
- * with "method" which specified the output.
- * </entry>
- * </row>
- * <row>
- * <entry>number</entry>
- * <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
- * valid DTMF is from 0 to -36 dBm0. Can be omitted if start is set to FALSE.
- * </entry>
- * </row>
- * <row>
- * <entry>start</entry>
- * <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.
- * </entry>
- * </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
+ *   with "method" which specified the output.
+ *
+ * * `number` (G_TYPE_INT, 0-15): The event number.
+ *
+ * * `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
+ *   valid DTMF is from 0 to -36 dBm0. Can be omitted if start is set to FALSE.
+ *
+ * * `start` (G_TYPE_BOOLEAN, True or False): Whether the event is starting or ending.
+ *
+ * * `method` (G_TYPE_INT, 1): The method used for sending event, this element will react if this
+ *   field is absent or 1.
  *
  * 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

diff --git a/gst/level/gstlevel.c b/gst/level/gstlevel.c
index c386f50..892f4db 100644 (file)
--- a/gst/level/gstlevel.c
+++ b/gst/level/gstlevel.c
@@ -45,9 +45,7 @@
  *
  * ## Example application
  *
- * <informalexample><programlisting language="C">
- * <xi:include xmlns:xi="http://www.w3.org/2003/XInclude" parse="text" href="../../../../tests/examples/level/level-example.c" />
- * </programlisting></informalexample>
+ * {{ tests/examples/level/level-example.c }}
  *
  */
 

diff --git a/gst/rtp/gstrtpL8depay.c b/gst/rtp/gstrtpL8depay.c
index 5b9520a..026f308 100644 (file)
--- a/gst/rtp/gstrtpL8depay.c
+++ b/gst/rtp/gstrtpL8depay.c
@@ -25,13 +25,12 @@
  * Extract raw audio from 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 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

diff --git a/gst/rtp/gstrtpL8pay.c b/gst/rtp/gstrtpL8pay.c
index cf2a3b9..6662cda 100644 (file)
--- a/gst/rtp/gstrtpL8pay.c
+++ b/gst/rtp/gstrtpL8pay.c
@@ -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

diff --git a/gst/rtp/gstrtpreddec.c b/gst/rtp/gstrtpreddec.c
index c7144e6..1e47817 100644 (file)
--- a/gst/rtp/gstrtpreddec.c
+++ b/gst/rtp/gstrtpreddec.c
@@ -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

diff --git a/gst/rtp/gstrtpredenc.c b/gst/rtp/gstrtpredenc.c
index 78d2b97..dba1026 100644 (file)
--- a/gst/rtp/gstrtpredenc.c
+++ b/gst/rtp/gstrtpredenc.c
@@ -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

diff --git a/gst/rtp/gstrtpulpfecdec.c b/gst/rtp/gstrtpulpfecdec.c
index 6248c45..709309a 100644 (file)
--- a/gst/rtp/gstrtpulpfecdec.c
+++ b/gst/rtp/gstrtpulpfecdec.c
@@ -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

diff --git a/gst/rtp/gstrtpulpfecenc.c b/gst/rtp/gstrtpulpfecenc.c
index 6637734..c169075 100644 (file)
--- a/gst/rtp/gstrtpulpfecenc.c
+++ b/gst/rtp/gstrtpulpfecenc.c
@@ -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

diff --git a/gst/spectrum/gstspectrum.c b/gst/spectrum/gstspectrum.c
index b9e064c..16684d4 100644 (file)
--- a/gst/spectrum/gstspectrum.c
+++ b/gst/spectrum/gstspectrum.c
@@ -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 }}
  *
  */
 

diff --git a/sys/v4l2/gstv4l2object.c b/sys/v4l2/gstv4l2object.c
index db9fe88..1d95007 100644 (file)
--- a/sys/v4l2/gstv4l2object.c
+++ b/sys/v4l2/gstv4l2object.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)