From c9597298f922a783309e7754733a35368345eb33 Mon Sep 17 00:00:00 2001 From: =?UTF-8?q?Tim-Philipp=20M=C3=BCller?= Date: Sat, 26 Apr 2014 23:35:17 +0100 Subject: [PATCH] docs: remove outdated and pointless 'Last reviewed' lines from docs They are very confusing for people, and more often than not also just not very accurate. Seeing 'last reviewed: 2005' in your docs is not very confidence-inspiring. Let's just remove those comments. --- ext/dv/gstdvdec.c | 2 -- ext/dv/gstdvdemux.c | 2 -- ext/jack/gstjackaudiosink.c | 2 -- ext/jack/gstjackaudiosrc.c | 2 -- ext/speex/gstspeexdec.c | 2 -- gst/avi/gstavidemux.c | 2 -- gst/avi/gstavisubtitle.c | 2 -- gst/isomp4/gstqtmoovrecover.c | 2 -- gst/isomp4/gstqtmux-doc.c | 2 -- gst/isomp4/gstqtmux.c | 2 -- gst/isomp4/qtdemux.c | 2 -- gst/multifile/gstmultifilesink.c | 2 -- gst/rtp/gstrtpL16depay.c | 2 -- gst/rtp/gstrtpL16pay.c | 2 -- gst/rtp/gstrtpac3depay.c | 2 -- gst/rtp/gstrtpac3pay.c | 2 -- gst/rtp/gstrtpamrdepay.c | 2 -- gst/rtp/gstrtpamrpay.c | 2 -- gst/rtp/gstrtpbvdepay.c | 2 -- gst/rtp/gstrtpbvpay.c | 2 -- gst/rtpmanager/gstrtpbin.c | 2 -- gst/rtpmanager/gstrtpjitterbuffer.c | 2 -- gst/rtpmanager/gstrtpmux.c | 2 -- gst/rtpmanager/gstrtpptdemux.c | 2 -- gst/rtpmanager/gstrtprtxreceive.c | 2 -- gst/rtpmanager/gstrtpsession.c | 2 -- gst/rtpmanager/gstrtpssrcdemux.c | 2 -- gst/rtsp/gstrtpdec.c | 2 -- gst/rtsp/gstrtspsrc.c | 2 -- gst/spectrum/gstspectrum.c | 2 -- gst/udp/gstudpsrc.c | 2 -- gst/videofilter/gstgamma.c | 2 -- gst/videofilter/gstvideobalance.c | 2 -- gst/videofilter/gstvideoflip.c | 2 -- gst/wavparse/gstwavparse.c | 2 -- sys/osxaudio/gstosxaudiosink.c | 2 -- 36 files changed, 72 deletions(-) diff --git a/ext/dv/gstdvdec.c b/ext/dv/gstdvdec.c index 7faeaa55aa..4fd01c9a20 100644 --- a/ext/dv/gstdvdec.c +++ b/ext/dv/gstdvdec.c @@ -34,8 +34,6 @@ * gst-launch-1.0 filesrc location=test.dv ! dvdemux name=demux ! dvdec ! xvimagesink * ]| This pipeline decodes and renders the raw DV stream to a videosink. * - * - * Last reviewed on 2006-02-28 (0.10.3) */ #ifdef HAVE_CONFIG_H diff --git a/ext/dv/gstdvdemux.c b/ext/dv/gstdvdemux.c index 01c8a7dafe..352283a9ab 100644 --- a/ext/dv/gstdvdemux.c +++ b/ext/dv/gstdvdemux.c @@ -44,8 +44,6 @@ * gst-launch-1.0 filesrc location=test.dv ! dvdemux name=demux ! queue ! audioconvert ! alsasink demux. ! queue ! dvdec ! xvimagesink * ]| This pipeline decodes and renders the raw DV stream to an audio and a videosink. * - * - * Last reviewed on 2006-02-27 (0.10.3) */ /* DV output has two modes, normal and wide. The resolution is the same in both diff --git a/ext/jack/gstjackaudiosink.c b/ext/jack/gstjackaudiosink.c index b0d5dd44e0..3a83567844 100644 --- a/ext/jack/gstjackaudiosink.c +++ b/ext/jack/gstjackaudiosink.c @@ -51,8 +51,6 @@ * gst-launch-1.0 audiotestsrc ! jackaudiosink * ]| Play a sine wave to using jack. * - * - * Last reviewed on 2006-11-30 (0.10.4) */ #ifdef HAVE_CONFIG_H diff --git a/ext/jack/gstjackaudiosrc.c b/ext/jack/gstjackaudiosrc.c index 4338a513a6..5a3bfb5fd6 100644 --- a/ext/jack/gstjackaudiosrc.c +++ b/ext/jack/gstjackaudiosrc.c @@ -70,8 +70,6 @@ * gst-launch-1.0 jackaudiosrc connect=0 ! jackaudiosink connect=0 * ]| Get audio input into gstreamer from jack. * - * - * Last reviewed on 2008-07-22 (0.10.4) */ #ifdef HAVE_CONFIG_H diff --git a/ext/speex/gstspeexdec.c b/ext/speex/gstspeexdec.c index 46b054b7ec..8196939f4c 100644 --- a/ext/speex/gstspeexdec.c +++ b/ext/speex/gstspeexdec.c @@ -34,8 +34,6 @@ * ]| Decode an Ogg/Speex file. To create an Ogg/Speex file refer to the * documentation of speexenc. * - * - * Last reviewed on 2006-04-05 (0.10.2) */ #ifdef HAVE_CONFIG_H diff --git a/gst/avi/gstavidemux.c b/gst/avi/gstavidemux.c index 7ffa57dbe2..22d3dbb857 100644 --- a/gst/avi/gstavidemux.c +++ b/gst/avi/gstavidemux.c @@ -37,8 +37,6 @@ * compressed audio or video data, this will only work if you have the * right decoder elements/plugins installed. * - * - * Last reviewed on 2006-12-29 (0.10.6) */ #ifdef HAVE_CONFIG_H diff --git a/gst/avi/gstavisubtitle.c b/gst/avi/gstavisubtitle.c index 7b2fa2dd20..a54ef66f24 100644 --- a/gst/avi/gstavisubtitle.c +++ b/gst/avi/gstavisubtitle.c @@ -33,8 +33,6 @@ * This plays an avi file with a video and subtitle stream. * * - * - * Last reviewed on 2008-02-01 */ /* example of a subtitle chunk in an avi file diff --git a/gst/isomp4/gstqtmoovrecover.c b/gst/isomp4/gstqtmoovrecover.c index 70c7466f0c..80b22ebcb9 100644 --- a/gst/isomp4/gstqtmoovrecover.c +++ b/gst/isomp4/gstqtmoovrecover.c @@ -57,8 +57,6 @@ * * * - * - * Documentation last reviewed on 2011-04-21 */ #ifdef HAVE_CONFIG_H diff --git a/gst/isomp4/gstqtmux-doc.c b/gst/isomp4/gstqtmux-doc.c index 3bde32845c..3b857d84ad 100644 --- a/gst/isomp4/gstqtmux-doc.c +++ b/gst/isomp4/gstqtmux-doc.c @@ -89,8 +89,6 @@ * Records a video stream captured from a v4l2 device, encodes it into H.264 * and muxes it into an mp4 file. * - * - * Documentation last reviewed on 2011-04-21 */ /* ============================= 3gppmux ==================================== */ diff --git a/gst/isomp4/gstqtmux.c b/gst/isomp4/gstqtmux.c index 702fa92fd9..52b60c7707 100644 --- a/gst/isomp4/gstqtmux.c +++ b/gst/isomp4/gstqtmux.c @@ -87,8 +87,6 @@ * ]| * Records a video stream captured from a v4l2 device and muxes it into a qt file. * - * - * Last reviewed on 2010-12-03 */ /* diff --git a/gst/isomp4/qtdemux.c b/gst/isomp4/qtdemux.c index c621eae1cb..a81ad7e7c0 100644 --- a/gst/isomp4/qtdemux.c +++ b/gst/isomp4/qtdemux.c @@ -40,8 +40,6 @@ * compressed audio or video data, this will only work if you have the * right decoder elements/plugins installed. * - * - * Last reviewed on 2006-12-29 (0.10.5) */ #ifdef HAVE_CONFIG_H diff --git a/gst/multifile/gstmultifilesink.c b/gst/multifile/gstmultifilesink.c index edf595fba9..eb896f2d80 100644 --- a/gst/multifile/gstmultifilesink.c +++ b/gst/multifile/gstmultifilesink.c @@ -103,8 +103,6 @@ * gst-launch-1.0 videotestsrc ! multifilesink post-messages=true filename="frame%d" * ]| * - * - * Last reviewed on 2009-09-11 (0.10.17) */ #ifdef HAVE_CONFIG_H diff --git a/gst/rtp/gstrtpL16depay.c b/gst/rtp/gstrtpL16depay.c index 7e96d9dd64..5bbbedc187 100644 --- a/gst/rtp/gstrtpL16depay.c +++ b/gst/rtp/gstrtpL16depay.c @@ -31,8 +31,6 @@ * ]| This example pipeline will depayload an RTP raw audio stream. Refer to * the rtpL16pay example to create the RTP stream. * - * - * Last reviewed on 2013-04-25 (1.1.0) */ #ifdef HAVE_CONFIG_H diff --git a/gst/rtp/gstrtpL16pay.c b/gst/rtp/gstrtpL16pay.c index 4a101ee1de..5d92c25e86 100644 --- a/gst/rtp/gstrtpL16pay.c +++ b/gst/rtp/gstrtpL16pay.c @@ -31,8 +31,6 @@ * ]| This example pipeline will payload raw audio. Refer to * the rtpL16depay example to depayload and play the RTP stream. * - * - * Last reviewed on 2013-04-25 (1.1.0) */ #ifdef HAVE_CONFIG_H diff --git a/gst/rtp/gstrtpac3depay.c b/gst/rtp/gstrtpac3depay.c index 1bcb5430a6..e8a05c932e 100644 --- a/gst/rtp/gstrtpac3depay.c +++ b/gst/rtp/gstrtpac3depay.c @@ -31,8 +31,6 @@ * ]| This example pipeline will depayload and decode an RTP AC3 stream. Refer to * the rtpac3pay example to create the RTP stream. * - * - * Last reviewed on 2013-04-25 (1.1.0) */ #ifdef HAVE_CONFIG_H diff --git a/gst/rtp/gstrtpac3pay.c b/gst/rtp/gstrtpac3pay.c index 175d627225..a6ef9140a7 100644 --- a/gst/rtp/gstrtpac3pay.c +++ b/gst/rtp/gstrtpac3pay.c @@ -31,8 +31,6 @@ * ]| This example pipeline will encode and payload AC3 stream. Refer to * the rtpac3depay example to depayload and decode the RTP stream. * - * - * Last reviewed on 2013-04-25 (1.1.0) */ #ifdef HAVE_CONFIG_H diff --git a/gst/rtp/gstrtpamrdepay.c b/gst/rtp/gstrtpamrdepay.c index 4c6b7224cd..1156ac9252 100644 --- a/gst/rtp/gstrtpamrdepay.c +++ b/gst/rtp/gstrtpamrdepay.c @@ -31,8 +31,6 @@ * ]| This example pipeline will depayload and decode an RTP AMR stream. Refer to * the rtpamrpay example to create the RTP stream. * - * - * Last reviewed on 2013-04-25 (1.1.0) */ /* diff --git a/gst/rtp/gstrtpamrpay.c b/gst/rtp/gstrtpamrpay.c index defc7f4ba2..ed319cc032 100644 --- a/gst/rtp/gstrtpamrpay.c +++ b/gst/rtp/gstrtpamrpay.c @@ -31,8 +31,6 @@ * ]| This example pipeline will encode and payload an AMR stream. Refer to * the rtpamrdepay example to depayload and decode the RTP stream. * - * - * Last reviewed on 2013-04-25 (1.1.0) */ /* references: diff --git a/gst/rtp/gstrtpbvdepay.c b/gst/rtp/gstrtpbvdepay.c index 7b85558d4e..13efebb2aa 100644 --- a/gst/rtp/gstrtpbvdepay.c +++ b/gst/rtp/gstrtpbvdepay.c @@ -23,8 +23,6 @@ * * Extract BroadcomVoice audio from RTP packets according to RFC 4298. * For detailed information see: http://www.rfc-editor.org/rfc/rfc4298.txt - * - * Last reviewed on 2013-04-25 (1.1.0) */ #ifdef HAVE_CONFIG_H diff --git a/gst/rtp/gstrtpbvpay.c b/gst/rtp/gstrtpbvpay.c index be12b38d06..15a7b7f856 100644 --- a/gst/rtp/gstrtpbvpay.c +++ b/gst/rtp/gstrtpbvpay.c @@ -23,8 +23,6 @@ * * Payload BroadcomVoice audio into RTP packets according to RFC 4298. * For detailed information see: http://www.rfc-editor.org/rfc/rfc4298.txt - * - * Last reviewed on 2013-04-25 (1.1.0) */ #ifdef HAVE_CONFIG_H diff --git a/gst/rtpmanager/gstrtpbin.c b/gst/rtpmanager/gstrtpbin.c index 87332fa011..a087c462f4 100644 --- a/gst/rtpmanager/gstrtpbin.c +++ b/gst/rtpmanager/gstrtpbin.c @@ -132,8 +132,6 @@ * Send RTCP reports for session 0 on port 5005 and RTCP reports for session 1 * on port 5007. * - * - * Last reviewed on 2007-08-30 (0.10.6) */ #ifdef HAVE_CONFIG_H diff --git a/gst/rtpmanager/gstrtpjitterbuffer.c b/gst/rtpmanager/gstrtpjitterbuffer.c index 17d15197d6..fc8dfcdaa9 100644 --- a/gst/rtpmanager/gstrtpjitterbuffer.c +++ b/gst/rtpmanager/gstrtpjitterbuffer.c @@ -91,8 +91,6 @@ * inserted into the pipeline to smooth out network jitter and to reorder the * out-of-order RTP packets. * - * - * Last reviewed on 2007-05-28 (0.10.5) */ #ifdef HAVE_CONFIG_H diff --git a/gst/rtpmanager/gstrtpmux.c b/gst/rtpmanager/gstrtpmux.c index 1dc2da4324..1fdfad54c5 100644 --- a/gst/rtpmanager/gstrtpmux.c +++ b/gst/rtpmanager/gstrtpmux.c @@ -46,8 +46,6 @@ * generated, both are encoded into different payload types and muxed together * so they can be sent on the same port. * - * - * Last reviewed on 2010-09-30 (0.10.21) */ #ifdef HAVE_CONFIG_H diff --git a/gst/rtpmanager/gstrtpptdemux.c b/gst/rtpmanager/gstrtpptdemux.c index 2c80ae0337..3051539238 100644 --- a/gst/rtpmanager/gstrtpptdemux.c +++ b/gst/rtpmanager/gstrtpptdemux.c @@ -46,8 +46,6 @@ * ]| Takes an RTP stream and send the RTP packets with the first detected * payload type to fakesink, discarding the other payload types. * - * - * Last reviewed on 2007-05-28 (0.10.5) */ /* diff --git a/gst/rtpmanager/gstrtprtxreceive.c b/gst/rtpmanager/gstrtprtxreceive.c index ba80d61fca..bb0f7e4398 100644 --- a/gst/rtpmanager/gstrtprtxreceive.c +++ b/gst/rtpmanager/gstrtprtxreceive.c @@ -109,8 +109,6 @@ * You can play with the drop-probability value for one or both streams. * You should hear a clear sound. (after a few seconds the two streams wave feel synchronized) * - * - * Last reviewed on 2013-11-08 (1.x) */ #ifdef HAVE_CONFIG_H diff --git a/gst/rtpmanager/gstrtpsession.c b/gst/rtpmanager/gstrtpsession.c index f114f818e1..88293ccb3c 100644 --- a/gst/rtpmanager/gstrtpsession.c +++ b/gst/rtpmanager/gstrtpsession.c @@ -106,8 +106,6 @@ * packets are sent in the PAUSED state). Applications should manually set and * keep (see gst_element_set_locked_state()) the RTCP udpsink to the PLAYING state. * - * - * Last reviewed on 2007-05-28 (0.10.5) */ #ifdef HAVE_CONFIG_H diff --git a/gst/rtpmanager/gstrtpssrcdemux.c b/gst/rtpmanager/gstrtpssrcdemux.c index 75730a4438..ef08290d0d 100644 --- a/gst/rtpmanager/gstrtpssrcdemux.c +++ b/gst/rtpmanager/gstrtpssrcdemux.c @@ -36,8 +36,6 @@ * ]| Takes an RTP stream and send the RTP packets with the first detected SSRC * to fakesink, discarding the other SSRCs. * - * - * Last reviewed on 2007-05-28 (0.10.5) */ #ifdef HAVE_CONFIG_H diff --git a/gst/rtsp/gstrtpdec.c b/gst/rtsp/gstrtpdec.c index f12004803f..b59aa5dfda 100644 --- a/gst/rtsp/gstrtpdec.c +++ b/gst/rtsp/gstrtpdec.c @@ -45,8 +45,6 @@ * SECTION:element-rtpdec * * A simple RTP session manager used internally by rtspsrc. - * - * Last reviewed on 2006-06-20 (0.10.4) */ /* #define HAVE_RTCP */ diff --git a/gst/rtsp/gstrtspsrc.c b/gst/rtsp/gstrtspsrc.c index a57ee7b90f..de456267bb 100644 --- a/gst/rtsp/gstrtspsrc.c +++ b/gst/rtsp/gstrtspsrc.c @@ -73,8 +73,6 @@ * ]| Establish a connection to an RTSP server and send the raw RTP packets to a * fakesink. * - * - * Last reviewed on 2006-08-18 (0.10.5) */ #ifdef HAVE_CONFIG_H diff --git a/gst/spectrum/gstspectrum.c b/gst/spectrum/gstspectrum.c index 5e192c16aa..72242dc3a7 100644 --- a/gst/spectrum/gstspectrum.c +++ b/gst/spectrum/gstspectrum.c @@ -94,8 +94,6 @@ * * * - * - * Last reviewed on 2011-03-10 (0.10.29) */ #ifdef HAVE_CONFIG_H diff --git a/gst/udp/gstudpsrc.c b/gst/udp/gstudpsrc.c index 774e2729cb..b6f7c933fc 100644 --- a/gst/udp/gstudpsrc.c +++ b/gst/udp/gstudpsrc.c @@ -100,8 +100,6 @@ * gst-launch-1.0 -v udpsrc port=0 ! fakesink * ]| read udp packets from a free port. * - * - * Last reviewed on 2007-09-20 (0.10.7) */ #ifdef HAVE_CONFIG_H #include "config.h" diff --git a/gst/videofilter/gstgamma.c b/gst/videofilter/gstgamma.c index 17acd7cb1e..0e143b6482 100644 --- a/gst/videofilter/gstgamma.c +++ b/gst/videofilter/gstgamma.c @@ -42,8 +42,6 @@ * gst-launch-1.0 videotestsrc ! gamma gamma=0.5 ! videoconvert ! ximagesink * ]| This pipeline will make the image "darker". * - * - * Last reviewed on 2010-04-18 (0.10.22) */ #ifdef HAVE_CONFIG_H diff --git a/gst/videofilter/gstvideobalance.c b/gst/videofilter/gstvideobalance.c index 5922e4f7b3..c7298d8af8 100644 --- a/gst/videofilter/gstvideobalance.c +++ b/gst/videofilter/gstvideobalance.c @@ -36,8 +36,6 @@ * ]| This pipeline converts the image to black and white by setting the * saturation to 0.0. * - * - * Last reviewed on 2010-04-18 (0.10.22) */ #ifdef HAVE_CONFIG_H diff --git a/gst/videofilter/gstvideoflip.c b/gst/videofilter/gstvideoflip.c index 1c9eba2ecb..1f82ade604 100644 --- a/gst/videofilter/gstvideoflip.c +++ b/gst/videofilter/gstvideoflip.c @@ -35,8 +35,6 @@ * gst-launch-1.0 videotestsrc ! videoflip method=clockwise ! videoconvert ! ximagesink * ]| This pipeline flips the test image 90 degrees clockwise. * - * - * Last reviewed on 2010-04-18 (0.10.22) */ diff --git a/gst/wavparse/gstwavparse.c b/gst/wavparse/gstwavparse.c index 2ac5fc60a6..83a74fb594 100644 --- a/gst/wavparse/gstwavparse.c +++ b/gst/wavparse/gstwavparse.c @@ -37,8 +37,6 @@ * gst-launch-1.0 gnomevfssrc location=http://www.example.org/sine.wav ! queue ! wavparse ! audioconvert ! alsasink * ]| Stream data from a network url. * - * - * Last reviewed on 2007-02-14 (0.10.6) */ /* diff --git a/sys/osxaudio/gstosxaudiosink.c b/sys/osxaudio/gstosxaudiosink.c index e295f3153f..0479ce1ce5 100644 --- a/sys/osxaudio/gstosxaudiosink.c +++ b/sys/osxaudio/gstosxaudiosink.c @@ -58,8 +58,6 @@ * gst-launch-1.0 filesrc location=sine.ogg ! oggdemux ! vorbisdec ! audioconvert ! audioresample ! osxaudiosink * ]| Play an Ogg/Vorbis file. * - * - * Last reviewed on 2006-03-01 (0.10.4) */ #ifdef HAVE_CONFIG_H