diff --git a/docs/meson.build b/docs/meson.build
index b5ea554e22..17cc4c19d5 100644
--- 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 a824c5c3d7..6f6962c047 100644
--- a/ext/lame/gstlamemp3enc.c
+++ b/ext/lame/gstlamemp3enc.c
@@ -30,16 +30,16 @@
* consideration. See Ogg/Vorbis
* for a royalty free (and often higher quality) alternative.
*
- *
- * Output sample rate
+ * ## 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.
- *
- *
- * Example pipelines
+ *
+ * ## 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
- *
*/
#ifdef HAVE_CONFIG_H
diff --git a/ext/mpg123/gstmpg123audiodec.c b/ext/mpg123/gstmpg123audiodec.c
index fa6743cb90..3e3c923a0e 100644
--- 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.
*
- *
- * Example pipelines
+ * ## Example pipelines
+ *
* |[
* gst-launch-1.0 filesrc location=music.mp3 ! mpegaudioparse ! mpg123audiodec ! audioconvert ! audioresample ! autoaudiosink
* ]| Decode and play the mp3 file
- *
*/
#ifdef HAVE_CONFIG_H
diff --git a/ext/taglib/gstapev2mux.cc b/ext/taglib/gstapev2mux.cc
index 9659dafca0..8a145d845d 100644
--- 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).
*
- *
- * Example pipelines
+ * ## 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> /dev/null | grep taglist
* ]| Verify that tags have been written.
- *
*/
#ifdef HAVE_CONFIG_H
diff --git a/ext/taglib/gstid3v2mux.cc b/ext/taglib/gstid3v2mux.cc
index 634c6ef2ba..e342bccae0 100644
--- 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).
*
- *
- * Example pipelines
+ * ## 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> /dev/null | grep taglist
* ]| Verify that tags have been written.
- *
*/
#ifdef HAVE_CONFIG_H
diff --git a/ext/twolame/gsttwolamemp2enc.c b/ext/twolame/gsttwolamemp2enc.c
index 31bec6b78c..09e7a6c06c 100644
--- 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.
*
- *
- * Example pipelines
+ * ## 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
- *
*
*/
diff --git a/gst/autodetect/gstautoaudiosink.c b/gst/autodetect/gstautoaudiosink.c
index f8f53ac4cc..26d9f4f31a 100644
--- 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 Sink
and Audio
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 828d6ad079..bf3704ba56 100644
--- 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 Source
and Audio
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 8607b46cf3..d46aa69120 100644
--- 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 Sink
and Video
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 f0d12a6b40..e5778dba08 100644
--- 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 Source
and Video
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 c72801975a..a5bcfcd444 100644
--- 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:
*
- *
- *
- *
- *
- *
- *
- *
- *
- * Name
- * GType
- * Possible values
- * Purpose
- *
- *
- *
- *
- * 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.
- *
- *
- *
- *
- *
+ * * `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 67bb54af3f..a3a057fb67 100644
--- 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
- *
- *
- *
- *
- *
- *
- *
- *
- * Name
- * GType
- * Possible values
- * Purpose
- *
- *
- *
- *
- * 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.
- *
- *
- *
- *
- *
+ * 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 dfd3b9d1f0..652dbdc01f 100644
--- 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:
*
- *
- *
- *
- *
- *
- *
- *
- *
- * Name
- * GType
- * Possible values
- * Purpose
- *
- *
- *
- *
- * 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.
- *
- *
- *
- *
- *
+ * * `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 c386f50bd4..892f4db667 100644
--- a/gst/level/gstlevel.c
+++ b/gst/level/gstlevel.c
@@ -45,9 +45,7 @@
*
* ## Example application
*
- *
- *
- *
+ * {{ tests/examples/level/level-example.c }}
*
*/
diff --git a/gst/rtp/gstrtpL8depay.c b/gst/rtp/gstrtpL8depay.c
index 5b9520a8da..026f308cbb 100644
--- 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
*
- *
- * Example pipeline
+ * ## 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.
- *
*/
#ifdef HAVE_CONFIG_H
diff --git a/gst/rtp/gstrtpL8pay.c b/gst/rtp/gstrtpL8pay.c
index cf2a3b95fd..6662cda916 100644
--- 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
*
- *
- * Example pipeline
+ * ## 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.
- *
*/
#ifdef HAVE_CONFIG_H
diff --git a/gst/rtp/gstrtpreddec.c b/gst/rtp/gstrtpreddec.c
index c7144e6e4b..1e47817278 100644
--- 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.
*
- *
- * Example pipeline
+ * ## 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.
- *
*
* See also: #GstRtpRedEnc, #GstWebRTCBin, #GstRtpBin
* Since: 1.14
diff --git a/gst/rtp/gstrtpredenc.c b/gst/rtp/gstrtpredenc.c
index 78d2b9716c..dba10261e7 100644
--- 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.
*
- *
- * Example pipeline
+ * ## 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.
- *
*
* See also: #GstRtpRedDec, #GstWebRTCBin, #GstRtpBin
* Since: 1.14
diff --git a/gst/rtp/gstrtpulpfecdec.c b/gst/rtp/gstrtpulpfecdec.c
index 6248c45588..709309ad16 100644
--- 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.
*
- *
- * Example pipeline
+ * ## 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
- * rtpfecserver.rs
+ *
* and
- * rtpfecclient.rs
- *
- *
+ *
*
* See also: #GstRtpUlpFecEnc, #GstRtpBin, #GstRtpStorage
* Since: 1.14
diff --git a/gst/rtp/gstrtpulpfecenc.c b/gst/rtp/gstrtpulpfecenc.c
index 6637734e18..c16907531e 100644
--- 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
*
- *
- * Example pipeline
* |[
* 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
- * rtpfecserver.rs
+ *
* and
- * rtpfecclient.rs
- *
+ *
*
* See also: #GstRtpUlpFecDec, #GstRtpBin
* Since: 1.14
diff --git a/gst/spectrum/gstspectrum.c b/gst/spectrum/gstspectrum.c
index b9e064c0d5..16684d45d1 100644
--- a/gst/spectrum/gstspectrum.c
+++ b/gst/spectrum/gstspectrum.c
@@ -49,9 +49,7 @@
*
* ## Example application
*
- *
- *
- *
+ * {{ tests/examples/spectrum/spectrum-example.c }}
*
*/
diff --git a/sys/v4l2/gstv4l2object.c b/sys/v4l2/gstv4l2object.c
index db9fe8858e..1d95007a36 100644
--- 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
- * struct v4l2_fmtdesc
.
+ * `struct v4l2_fmtdesc`.
*/
static GSList *
gst_v4l2_object_get_format_list (GstV4l2Object * v4l2object)