From f554369ed59fab1d54f63fee424179cce9e4f8ba Mon Sep 17 00:00:00 2001 From: Mathieu Duponchelle Date: Wed, 29 May 2019 22:06:58 +0200 Subject: [PATCH] doc: remove xml from comments --- docs/meson.build | 1 + ext/lame/gstlamemp3enc.c | 11 +++-- ext/mpg123/gstmpg123audiodec.c | 5 +-- ext/taglib/gstapev2mux.cc | 5 +-- ext/taglib/gstid3v2mux.cc | 5 +-- ext/twolame/gsttwolamemp2enc.c | 5 +-- gst/autodetect/gstautoaudiosink.c | 2 +- gst/autodetect/gstautoaudiosrc.c | 2 +- gst/autodetect/gstautovideosink.c | 2 +- gst/autodetect/gstautovideosrc.c | 2 +- gst/dtmf/gstdtmfsrc.c | 74 +++++++------------------------ gst/dtmf/gstrtpdtmfdepay.c | 67 +++++++--------------------- gst/dtmf/gstrtpdtmfsrc.c | 74 +++++++------------------------ gst/level/gstlevel.c | 4 +- gst/rtp/gstrtpL8depay.c | 5 +-- gst/rtp/gstrtpL8pay.c | 5 +-- gst/rtp/gstrtpreddec.c | 5 +-- gst/rtp/gstrtpredenc.c | 5 +-- gst/rtp/gstrtpulpfecdec.c | 10 ++--- gst/rtp/gstrtpulpfecenc.c | 8 ++-- gst/spectrum/gstspectrum.c | 4 +- sys/v4l2/gstv4l2object.c | 2 +- 22 files changed, 83 insertions(+), 220 deletions(-) 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)