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docs: remove outdated and pointless 'Last reviewed' lines from docs

Browse source 
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.
This commit is contained in:
Tim-Philipp Müller 2014-04-26 23:35:17 +01:00
parent c073a6c779
commit c9597298f9
36 changed files with 0 additions and 72 deletions
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2
ext/dv/gstdvdec.c
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@ -34,8 +34,6 @@
* gst-launch-1.0 filesrc location=test.dv ! dvdemux name=demux ! dvdec ! xvimagesink * 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. * ]| This pipeline decodes and renders the raw DV stream to a videosink.
* </refsect2> * </refsect2>
*
* Last reviewed on 2006-02-28 (0.10.3)
*/ */
#ifdef HAVE_CONFIG_H #ifdef HAVE_CONFIG_H

2
ext/dv/gstdvdemux.c
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@ -44,8 +44,6 @@
* gst-launch-1.0 filesrc location=test.dv ! dvdemux name=demux ! queue ! audioconvert ! alsasink demux. ! queue ! dvdec ! xvimagesink * 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. * ]| This pipeline decodes and renders the raw DV stream to an audio and a videosink.
* </refsect2> * </refsect2>
*
* Last reviewed on 2006-02-27 (0.10.3)
*/ */
/* DV output has two modes, normal and wide. The resolution is the same in both /* DV output has two modes, normal and wide. The resolution is the same in both

2
ext/jack/gstjackaudiosink.c
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@ -51,8 +51,6 @@
* gst-launch-1.0 audiotestsrc ! jackaudiosink * gst-launch-1.0 audiotestsrc ! jackaudiosink
* ]| Play a sine wave to using jack. * ]| Play a sine wave to using jack.
* </refsect2> * </refsect2>
*
* Last reviewed on 2006-11-30 (0.10.4)
*/ */
#ifdef HAVE_CONFIG_H #ifdef HAVE_CONFIG_H

2
ext/jack/gstjackaudiosrc.c
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@ -70,8 +70,6 @@
* gst-launch-1.0 jackaudiosrc connect=0 ! jackaudiosink connect=0 * gst-launch-1.0 jackaudiosrc connect=0 ! jackaudiosink connect=0
* ]| Get audio input into gstreamer from jack. * ]| Get audio input into gstreamer from jack.
* </refsect2> * </refsect2>
*
* Last reviewed on 2008-07-22 (0.10.4)
*/ */
#ifdef HAVE_CONFIG_H #ifdef HAVE_CONFIG_H

2
ext/speex/gstspeexdec.c
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@ -34,8 +34,6 @@
* ]| Decode an Ogg/Speex file. To create an Ogg/Speex file refer to the * ]| Decode an Ogg/Speex file. To create an Ogg/Speex file refer to the
* documentation of speexenc. * documentation of speexenc.
* </refsect2> * </refsect2>
*
* Last reviewed on 2006-04-05 (0.10.2)
*/ */
#ifdef HAVE_CONFIG_H #ifdef HAVE_CONFIG_H

2
gst/avi/gstavidemux.c
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@ -37,8 +37,6 @@
* compressed audio or video data, this will only work if you have the * compressed audio or video data, this will only work if you have the
* right decoder elements/plugins installed. * right decoder elements/plugins installed.
* </refsect2> * </refsect2>
*
* Last reviewed on 2006-12-29 (0.10.6)
*/ */
#ifdef HAVE_CONFIG_H #ifdef HAVE_CONFIG_H

2
gst/avi/gstavisubtitle.c
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@ -33,8 +33,6 @@
* This plays an avi file with a video and subtitle stream. * This plays an avi file with a video and subtitle stream.
* </para> * </para>
* </refsect2> * </refsect2>
*
* Last reviewed on 2008-02-01
*/ */
/* example of a subtitle chunk in an avi file /* example of a subtitle chunk in an avi file

2
gst/isomp4/gstqtmoovrecover.c
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@ -57,8 +57,6 @@
* </programlisting> * </programlisting>
* </para> * </para>
* </refsect2> * </refsect2>
*
* Documentation last reviewed on 2011-04-21
*/ */
#ifdef HAVE_CONFIG_H #ifdef HAVE_CONFIG_H

2
gst/isomp4/gstqtmux-doc.c
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@ -89,8 +89,6 @@
* Records a video stream captured from a v4l2 device, encodes it into H.264 * Records a video stream captured from a v4l2 device, encodes it into H.264
* and muxes it into an mp4 file. * and muxes it into an mp4 file.
* </refsect2> * </refsect2>
*
* Documentation last reviewed on 2011-04-21
*/ */
/* ============================= 3gppmux ==================================== */ /* ============================= 3gppmux ==================================== */

2
gst/isomp4/gstqtmux.c
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@ -87,8 +87,6 @@
* ]| * ]|
* Records a video stream captured from a v4l2 device and muxes it into a qt file. * Records a video stream captured from a v4l2 device and muxes it into a qt file.
* </refsect2> * </refsect2>
*
* Last reviewed on 2010-12-03
*/ */
/* /*

2
gst/isomp4/qtdemux.c
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@ -40,8 +40,6 @@
* compressed audio or video data, this will only work if you have the * compressed audio or video data, this will only work if you have the
* right decoder elements/plugins installed. * right decoder elements/plugins installed.
* </refsect2> * </refsect2>
*
* Last reviewed on 2006-12-29 (0.10.5)
*/ */
#ifdef HAVE_CONFIG_H #ifdef HAVE_CONFIG_H

2
gst/multifile/gstmultifilesink.c
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@ -103,8 +103,6 @@
* gst-launch-1.0 videotestsrc ! multifilesink post-messages=true filename="frame%d" * gst-launch-1.0 videotestsrc ! multifilesink post-messages=true filename="frame%d"
* ]| * ]|
* </refsect2> * </refsect2>
*
* Last reviewed on 2009-09-11 (0.10.17)
*/ */
#ifdef HAVE_CONFIG_H #ifdef HAVE_CONFIG_H

2
gst/rtp/gstrtpL16depay.c
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@ -31,8 +31,6 @@
* ]| This example pipeline will depayload an RTP raw audio stream. Refer to * ]| This example pipeline will depayload an RTP raw audio stream. Refer to
* the rtpL16pay example to create the RTP stream. * the rtpL16pay example to create the RTP stream.
* </refsect2> * </refsect2>
*
* Last reviewed on 2013-04-25 (1.1.0)
*/ */
#ifdef HAVE_CONFIG_H #ifdef HAVE_CONFIG_H

2
gst/rtp/gstrtpL16pay.c
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@ -31,8 +31,6 @@
* ]| This example pipeline will payload raw audio. Refer to * ]| This example pipeline will payload raw audio. Refer to
* the rtpL16depay example to depayload and play the RTP stream. * the rtpL16depay example to depayload and play the RTP stream.
* </refsect2> * </refsect2>
*
* Last reviewed on 2013-04-25 (1.1.0)
*/ */
#ifdef HAVE_CONFIG_H #ifdef HAVE_CONFIG_H

2
gst/rtp/gstrtpac3depay.c
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@ -31,8 +31,6 @@
* ]| This example pipeline will depayload and decode an RTP AC3 stream. Refer to * ]| This example pipeline will depayload and decode an RTP AC3 stream. Refer to
* the rtpac3pay example to create the RTP stream. * the rtpac3pay example to create the RTP stream.
* </refsect2> * </refsect2>
*
* Last reviewed on 2013-04-25 (1.1.0)
*/ */
#ifdef HAVE_CONFIG_H #ifdef HAVE_CONFIG_H

2
gst/rtp/gstrtpac3pay.c
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@ -31,8 +31,6 @@
* ]| This example pipeline will encode and payload AC3 stream. Refer to * ]| This example pipeline will encode and payload AC3 stream. Refer to
* the rtpac3depay example to depayload and decode the RTP stream. * the rtpac3depay example to depayload and decode the RTP stream.
* </refsect2> * </refsect2>
*
* Last reviewed on 2013-04-25 (1.1.0)
*/ */
#ifdef HAVE_CONFIG_H #ifdef HAVE_CONFIG_H

2
gst/rtp/gstrtpamrdepay.c
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@ -31,8 +31,6 @@
* ]| This example pipeline will depayload and decode an RTP AMR stream. Refer to * ]| This example pipeline will depayload and decode an RTP AMR stream. Refer to
* the rtpamrpay example to create the RTP stream. * the rtpamrpay example to create the RTP stream.
* </refsect2> * </refsect2>
*
* Last reviewed on 2013-04-25 (1.1.0)
*/ */
/* /*

2
gst/rtp/gstrtpamrpay.c
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@ -31,8 +31,6 @@
* ]| This example pipeline will encode and payload an AMR stream. Refer to * ]| This example pipeline will encode and payload an AMR stream. Refer to
* the rtpamrdepay example to depayload and decode the RTP stream. * the rtpamrdepay example to depayload and decode the RTP stream.
* </refsect2> * </refsect2>
*
* Last reviewed on 2013-04-25 (1.1.0)
*/ */
/* references: /* references:

2
gst/rtp/gstrtpbvdepay.c
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@ -23,8 +23,6 @@
* *
* Extract BroadcomVoice audio from RTP packets according to RFC 4298. * Extract BroadcomVoice audio from RTP packets according to RFC 4298.
* For detailed information see: http://www.rfc-editor.org/rfc/rfc4298.txt * 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 #ifdef HAVE_CONFIG_H

2
gst/rtp/gstrtpbvpay.c
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@ -23,8 +23,6 @@
* *
* Payload BroadcomVoice audio into RTP packets according to RFC 4298. * Payload BroadcomVoice audio into RTP packets according to RFC 4298.
* For detailed information see: http://www.rfc-editor.org/rfc/rfc4298.txt * 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 #ifdef HAVE_CONFIG_H

2
gst/rtpmanager/gstrtpbin.c
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@ -132,8 +132,6 @@
* Send RTCP reports for session 0 on port 5005 and RTCP reports for session 1 * Send RTCP reports for session 0 on port 5005 and RTCP reports for session 1
* on port 5007. * on port 5007.
* </refsect2> * </refsect2>
*
* Last reviewed on 2007-08-30 (0.10.6)
*/ */
#ifdef HAVE_CONFIG_H #ifdef HAVE_CONFIG_H

2
gst/rtpmanager/gstrtpjitterbuffer.c
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@ -91,8 +91,6 @@
* inserted into the pipeline to smooth out network jitter and to reorder the * inserted into the pipeline to smooth out network jitter and to reorder the
* out-of-order RTP packets. * out-of-order RTP packets.
* </refsect2> * </refsect2>
*
* Last reviewed on 2007-05-28 (0.10.5)
*/ */
#ifdef HAVE_CONFIG_H #ifdef HAVE_CONFIG_H

2
gst/rtpmanager/gstrtpmux.c
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@ -46,8 +46,6 @@
* generated, both are encoded into different payload types and muxed together * generated, both are encoded into different payload types and muxed together
* so they can be sent on the same port. * so they can be sent on the same port.
* </refsect2> * </refsect2>
*
* Last reviewed on 2010-09-30 (0.10.21)
*/ */
#ifdef HAVE_CONFIG_H #ifdef HAVE_CONFIG_H

2
gst/rtpmanager/gstrtpptdemux.c
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@ -46,8 +46,6 @@
* ]| Takes an RTP stream and send the RTP packets with the first detected * ]| Takes an RTP stream and send the RTP packets with the first detected
* payload type to fakesink, discarding the other payload types. * payload type to fakesink, discarding the other payload types.
* </refsect2> * </refsect2>
*
* Last reviewed on 2007-05-28 (0.10.5)
*/ */
/* /*

2
gst/rtpmanager/gstrtprtxreceive.c
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@ -109,8 +109,6 @@
* You can play with the drop-probability value for one or both streams. * 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) * You should hear a clear sound. (after a few seconds the two streams wave feel synchronized)
* </refsect2> * </refsect2>
*
* Last reviewed on 2013-11-08 (1.x)
*/ */
#ifdef HAVE_CONFIG_H #ifdef HAVE_CONFIG_H

2
gst/rtpmanager/gstrtpsession.c
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@ -106,8 +106,6 @@
* packets are sent in the PAUSED state). Applications should manually set and * 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. * keep (see gst_element_set_locked_state()) the RTCP udpsink to the PLAYING state.
* </refsect2> * </refsect2>
*
* Last reviewed on 2007-05-28 (0.10.5)
*/ */
#ifdef HAVE_CONFIG_H #ifdef HAVE_CONFIG_H

2
gst/rtpmanager/gstrtpssrcdemux.c
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@ -36,8 +36,6 @@
* ]| Takes an RTP stream and send the RTP packets with the first detected SSRC * ]| Takes an RTP stream and send the RTP packets with the first detected SSRC
* to fakesink, discarding the other SSRCs. * to fakesink, discarding the other SSRCs.
* </refsect2> * </refsect2>
*
* Last reviewed on 2007-05-28 (0.10.5)
*/ */
#ifdef HAVE_CONFIG_H #ifdef HAVE_CONFIG_H

2
gst/rtsp/gstrtpdec.c
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@ -45,8 +45,6 @@
* SECTION:element-rtpdec * SECTION:element-rtpdec
* *
* A simple RTP session manager used internally by rtspsrc. * A simple RTP session manager used internally by rtspsrc.
*
* Last reviewed on 2006-06-20 (0.10.4)
*/ */
/* #define HAVE_RTCP */ /* #define HAVE_RTCP */

2
gst/rtsp/gstrtspsrc.c
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@ -73,8 +73,6 @@
* ]| Establish a connection to an RTSP server and send the raw RTP packets to a * ]| Establish a connection to an RTSP server and send the raw RTP packets to a
* fakesink. * fakesink.
* </refsect2> * </refsect2>
*
* Last reviewed on 2006-08-18 (0.10.5)
*/ */
#ifdef HAVE_CONFIG_H #ifdef HAVE_CONFIG_H

2
gst/spectrum/gstspectrum.c
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@ -94,8 +94,6 @@
* <xi:include xmlns:xi="http://www.w3.org/2003/XInclude" parse="text" href="../../../../tests/examples/spectrum/spectrum-example.c" /> * <xi:include xmlns:xi="http://www.w3.org/2003/XInclude" parse="text" href="../../../../tests/examples/spectrum/spectrum-example.c" />
* </programlisting></informalexample> * </programlisting></informalexample>
* </refsect2> * </refsect2>
*
* Last reviewed on 2011-03-10 (0.10.29)
*/ */
#ifdef HAVE_CONFIG_H #ifdef HAVE_CONFIG_H

2
gst/udp/gstudpsrc.c
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@ -100,8 +100,6 @@
* gst-launch-1.0 -v udpsrc port=0 ! fakesink * gst-launch-1.0 -v udpsrc port=0 ! fakesink
* ]| read udp packets from a free port. * ]| read udp packets from a free port.
* </refsect2> * </refsect2>
*
* Last reviewed on 2007-09-20 (0.10.7)
*/ */
#ifdef HAVE_CONFIG_H #ifdef HAVE_CONFIG_H
#include "config.h" #include "config.h"

2
gst/videofilter/gstgamma.c
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@ -42,8 +42,6 @@
* gst-launch-1.0 videotestsrc ! gamma gamma=0.5 ! videoconvert ! ximagesink * gst-launch-1.0 videotestsrc ! gamma gamma=0.5 ! videoconvert ! ximagesink
* ]| This pipeline will make the image "darker". * ]| This pipeline will make the image "darker".
* </refsect2> * </refsect2>
*
* Last reviewed on 2010-04-18 (0.10.22)
*/ */
#ifdef HAVE_CONFIG_H #ifdef HAVE_CONFIG_H

2
gst/videofilter/gstvideobalance.c
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@ -36,8 +36,6 @@
* ]| This pipeline converts the image to black and white by setting the * ]| This pipeline converts the image to black and white by setting the
* saturation to 0.0. * saturation to 0.0.
* </refsect2> * </refsect2>
*
* Last reviewed on 2010-04-18 (0.10.22)
*/ */
#ifdef HAVE_CONFIG_H #ifdef HAVE_CONFIG_H

2
gst/videofilter/gstvideoflip.c
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@ -35,8 +35,6 @@
* gst-launch-1.0 videotestsrc ! videoflip method=clockwise ! videoconvert ! ximagesink * gst-launch-1.0 videotestsrc ! videoflip method=clockwise ! videoconvert ! ximagesink
* ]| This pipeline flips the test image 90 degrees clockwise. * ]| This pipeline flips the test image 90 degrees clockwise.
* </refsect2> * </refsect2>
*
* Last reviewed on 2010-04-18 (0.10.22)
*/ */

2
gst/wavparse/gstwavparse.c
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@ -37,8 +37,6 @@
* gst-launch-1.0 gnomevfssrc location=http://www.example.org/sine.wav ! queue ! wavparse ! audioconvert ! alsasink * gst-launch-1.0 gnomevfssrc location=http://www.example.org/sine.wav ! queue ! wavparse ! audioconvert ! alsasink
* ]| Stream data from a network url. * ]| Stream data from a network url.
* </refsect2> * </refsect2>
*
* Last reviewed on 2007-02-14 (0.10.6)
*/ */
/* /*

2
sys/osxaudio/gstosxaudiosink.c
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@ -58,8 +58,6 @@
* gst-launch-1.0 filesrc location=sine.ogg ! oggdemux ! vorbisdec ! audioconvert ! audioresample ! osxaudiosink * gst-launch-1.0 filesrc location=sine.ogg ! oggdemux ! vorbisdec ! audioconvert ! audioresample ! osxaudiosink
* ]| Play an Ogg/Vorbis file. * ]| Play an Ogg/Vorbis file.
* </refsect2> * </refsect2>
*
* Last reviewed on 2006-03-01 (0.10.4)
*/ */
#ifdef HAVE_CONFIG_H #ifdef HAVE_CONFIG_H