docs: remove outdated and pointless 'Last reviewed' lines from docs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

diff --git a/ext/dv/gstdvdec.c b/ext/dv/gstdvdec.c
index 7faeaa5..4fd01c9 100644 (file)
--- 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.
  * </refsect2>
- *
- * 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 01c8a7d..352283a 100644 (file)
--- 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.
  * </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

diff --git a/ext/jack/gstjackaudiosink.c b/ext/jack/gstjackaudiosink.c
index b0d5dd4..3a83567 100644 (file)
--- 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.
  * </refsect2>
- *
- * 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 4338a51..5a3bfb5 100644 (file)
--- 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.
  * </refsect2>
- *
- * 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 46b054b..8196939 100644 (file)
--- 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.
  * </refsect2>
- *
- * 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 7ffa57d..22d3dbb 100644 (file)
--- 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.
  * </refsect2>
- *
- * 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 7b2fa2d..a54ef66 100644 (file)
--- 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.
  * </para>
  * </refsect2>
- *
- * 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 70c7466..80b22eb 100644 (file)
--- a/gst/isomp4/gstqtmoovrecover.c
+++ b/gst/isomp4/gstqtmoovrecover.c
@@ -57,8 +57,6 @@
  * </programlisting>
  * </para>
  * </refsect2>
- *
- * 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 3bde328..3b857d8 100644 (file)
--- 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.
  * </refsect2>
- *
- * Documentation last reviewed on 2011-04-21
  */
 
 /* ============================= 3gppmux ==================================== */

diff --git a/gst/isomp4/gstqtmux.c b/gst/isomp4/gstqtmux.c
index 702fa92..52b60c7 100644 (file)
--- 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.
  * </refsect2>
- *
- * Last reviewed on 2010-12-03
  */
 
 /*

diff --git a/gst/isomp4/qtdemux.c b/gst/isomp4/qtdemux.c
index c621eae..a81ad7e 100644 (file)
--- 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.
  * </refsect2>
- *
- * 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 edf595f..eb896f2 100644 (file)
--- 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"
  * ]|
  * </refsect2>
- *
- * 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 7e96d9d..5bbbedc 100644 (file)
--- 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.
  * </refsect2>
- *
- * 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 4a101ee..5d92c25 100644 (file)
--- 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.
  * </refsect2>
- *
- * 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 1bcb543..e8a05c9 100644 (file)
--- 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.
  * </refsect2>
- *
- * 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 175d627..a6ef914 100644 (file)
--- 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.
  * </refsect2>
- *
- * 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 4c6b722..1156ac9 100644 (file)
--- 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.
  * </refsect2>
- *
- * Last reviewed on 2013-04-25 (1.1.0)
  */
 
 /*

diff --git a/gst/rtp/gstrtpamrpay.c b/gst/rtp/gstrtpamrpay.c
index defc7f4..ed319cc 100644 (file)
--- 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.
  * </refsect2>
- *
- * Last reviewed on 2013-04-25 (1.1.0)
  */
 
 /* references:

diff --git a/gst/rtp/gstrtpbvdepay.c b/gst/rtp/gstrtpbvdepay.c
index 7b85558..13efebb 100644 (file)
--- 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 be12b38..15a7b7f 100644 (file)
--- 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 87332fa..a087c46 100644 (file)
--- 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.
  * </refsect2>
- *
- * 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 17d1519..fc8dfcd 100644 (file)
--- 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.
  * </refsect2>
- *
- * 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 1dc2da4..1fdfad5 100644 (file)
--- 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.
  * </refsect2>
- *
- * 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 2c80ae0..3051539 100644 (file)
--- 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.
  * </refsect2>
- *
- * Last reviewed on 2007-05-28 (0.10.5)
  */
 
 /*

diff --git a/gst/rtpmanager/gstrtprtxreceive.c b/gst/rtpmanager/gstrtprtxreceive.c
index ba80d61..bb0f7e4 100644 (file)
--- 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)
  * </refsect2>
- *
- * 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 f114f81..88293cc 100644 (file)
--- 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.
  * </refsect2>
- *
- * 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 75730a4..ef08290 100644 (file)
--- 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.
  * </refsect2>
- *
- * 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 f120048..b59aa5d 100644 (file)
--- 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 a57ee7b..de45626 100644 (file)
--- 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.
  * </refsect2>
- *
- * 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 5e192c1..72242dc 100644 (file)
--- a/gst/spectrum/gstspectrum.c
+++ b/gst/spectrum/gstspectrum.c
@@ -94,8 +94,6 @@
  * <xi:include xmlns:xi="http://www.w3.org/2003/XInclude" parse="text" href="../../../../tests/examples/spectrum/spectrum-example.c" />
  * </programlisting></informalexample>
  * </refsect2>
- *
- * 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 774e272..b6f7c93 100644 (file)
--- 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.
  * </refsect2>
- *
- * 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 17acd7c..0e143b6 100644 (file)
--- 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".
  * </refsect2>
- *
- * 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 5922e4f..c7298d8 100644 (file)
--- 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.
  * </refsect2>
- *
- * 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 1c9eba2..1f82ade 100644 (file)
--- 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.
  * </refsect2>
- *
- * Last reviewed on 2010-04-18 (0.10.22)
  */
 
 

diff --git a/gst/wavparse/gstwavparse.c b/gst/wavparse/gstwavparse.c
index 2ac5fc6..83a74fb 100644 (file)
--- 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.
  * </refsect2>
- *
- * Last reviewed on 2007-02-14 (0.10.6)
  */
 
 /*

diff --git a/sys/osxaudio/gstosxaudiosink.c b/sys/osxaudio/gstosxaudiosink.c
index e295f31..0479ce1 100644 (file)
--- 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.
  * </refsect2>
- *
- * Last reviewed on 2006-03-01 (0.10.4)
  */
 
 #ifdef HAVE_CONFIG_H