docs: remove outdated and pointless 'Last reviewed' lines from docs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.
2025-01-11 18:05:37 +00:00 · 2014-04-26 23:26:09 +01:00 · 2014-04-26 23:26:09 +01:00 · bcb8068e27
commit bcb8068e27
parent e8d176c20c
41 changed files with 1 additions and 81 deletions
--- a/ext/alsa/gstalsasink.c
+++ b/ext/alsa/gstalsasink.c
@ -32,8 +32,6 @@
 * gst-launch -v filesrc location=sine.ogg ! oggdemux ! vorbisdec ! audioconvert ! audioresample ! alsasink
 * ]| Play an Ogg/Vorbis file.
 * </refsect2>
- *
- * Last reviewed on 2006-03-01 (0.10.4)
 */

 #ifdef HAVE_CONFIG_H
--- a/ext/alsa/gstalsasrc.c
+++ b/ext/alsa/gstalsasrc.c
@ -31,8 +31,6 @@
 * gst-launch -v alsasrc ! audioconvert ! vorbisenc ! oggmux ! filesink location=alsasrc.ogg
 * ]| Record from a sound card using ALSA and encode to Ogg/Vorbis.
 * </refsect2>
- *
- * Last reviewed on 2006-03-01 (0.10.4)
 */

 #ifdef HAVE_CONFIG_H
--- a/ext/ogg/gstoggdemux.c
+++ b/ext/ogg/gstoggdemux.c
@ -31,8 +31,6 @@
 * gst-launch -v filesrc location=test.ogg ! oggdemux ! vorbisdec ! audioconvert ! alsasink
 * ]| Decodes the vorbis audio stored inside an ogg container.
 * </refsect2>
- *
- * Last reviewed on 2006-12-30 (0.10.5)
 */


--- a/ext/ogg/gstoggmux.c
+++ b/ext/ogg/gstoggmux.c
@ -31,8 +31,6 @@
 * ]| Encodes a video stream captured from a v4l2-compatible camera to Ogg/Theora
 * (the encoding will stop automatically after 500 frames)
 * </refsect2>
- *
- * Last reviewed on 2008-02-06 (0.10.17)
 */

 #ifdef HAVE_CONFIG_H
--- a/ext/theora/gsttheoradec.c
+++ b/ext/theora/gsttheoradec.c
@ -36,8 +36,6 @@
 * ]| This example pipeline will decode an ogg stream and decodes the theora video. Refer to
 * the theoraenc example to create the ogg file.
 * </refsect2>
- *
- * Last reviewed on 2006-03-01 (0.10.4)
 */

 #ifdef HAVE_CONFIG_H
--- a/ext/theora/gsttheoraenc.c
+++ b/ext/theora/gsttheoraenc.c
@ -50,8 +50,6 @@
 * ogg container. Refer to the theoradec documentation to decode the create
 * stream.
 * </refsect2>
- *
- * Last reviewed on 2006-03-01 (0.10.4)
 */

 #ifdef HAVE_CONFIG_H
--- a/ext/theora/gsttheoraparse.c
+++ b/ext/theora/gsttheoraparse.c
@ -52,8 +52,6 @@
 * ]| This pipeline shows remuxing. video-remuxed.ogg might not be exactly the same
 * as video.ogg, but they should produce exactly the same decoded data.
 * </refsect2>
- *
- * Last reviewed on 2008-05-28 (0.10.20)
 */

 /* FIXME 0.11: suppress warnings for deprecated API such as GValueArray
--- a/ext/vorbis/gstvorbisdec.c
+++ b/ext/vorbis/gstvorbisdec.c
@ -32,8 +32,6 @@
 * gst-launch -v filesrc location=sine.ogg ! oggdemux ! vorbisdec ! audioconvert ! alsasink
 * ]| Decode an Ogg/Vorbis. To create an Ogg/Vorbis file refer to the documentation of vorbisenc.
 * </refsect2>
- *
- * Last reviewed on 2006-03-01 (0.10.4)
 */

 #ifdef HAVE_CONFIG_H
--- a/ext/vorbis/gstvorbisenc.c
+++ b/ext/vorbis/gstvorbisenc.c
@ -36,8 +36,6 @@
 * gst-launch -v alsasrc ! audioconvert ! vorbisenc ! oggmux ! filesink location=alsasrc.ogg
 * ]| Record from a sound card using ALSA and encode to Ogg/Vorbis.
 * </refsect2>
- *
- * Last reviewed on 2006-03-01 (0.10.4)
 */
 #ifdef HAVE_CONFIG_H
 #include "config.h"
--- a/ext/vorbis/gstvorbisparse.c
+++ b/ext/vorbis/gstvorbisparse.c
@ -45,8 +45,6 @@
 * ]| This pipeline shows remuxing. sine-remuxed.ogg might not be exactly the same
 * as sine.ogg, but they should produce exactly the same decoded data.
 * </refsect2>
- *
- * Last reviewed on 2006-04-01 (0.10.4.1)
 */

 #ifdef HAVE_CONFIG_H
--- a/gst-libs/gst/app/gstappsink.c
+++ b/gst-libs/gst/app/gstappsink.c
@ -57,8 +57,6 @@
 *
 * The eos signal can also be used to be informed when the EOS state is reached
 * to avoid polling.
- *
- * Last reviewed on 2008-12-17 (0.10.22)
 */

 #ifdef HAVE_CONFIG_H
--- a/gst-libs/gst/app/gstappsrc.c
+++ b/gst-libs/gst/app/gstappsrc.c
@ -84,8 +84,6 @@
 * gst_app_src_end_of_stream() or emit the end-of-stream action signal. After
 * this call, no more buffers can be pushed into appsrc until a flushing seek
 * happened or the state of the appsrc has gone through READY.
- *
- * Last reviewed on 2008-12-17 (0.10.10)
 */

 #ifdef HAVE_CONFIG_H
--- a/gst-libs/gst/audio/gstaudiobasesink.c
+++ b/gst-libs/gst/audio/gstaudiobasesink.c
@ -28,8 +28,6 @@
 * This is the base class for audio sinks. Subclasses need to implement the
 * ::create_ringbuffer vmethod. This base class will then take care of
 * writing samples to the ringbuffer, synchronisation, clipping and flushing.
- *
- * Last reviewed on 2006-09-27 (0.10.12)
 */

 #include <string.h>
--- a/gst-libs/gst/audio/gstaudiobasesrc.c
+++ b/gst-libs/gst/audio/gstaudiobasesrc.c
@ -28,8 +28,6 @@
 * This is the base class for audio sources. Subclasses need to implement the
 * ::create_ringbuffer vmethod. This base class will then take care of
 * reading samples from the ringbuffer, synchronisation and flushing.
- *
- * Last reviewed on 2006-09-27 (0.10.12)
 */

 #ifdef HAVE_CONFIG_H
--- a/gst-libs/gst/audio/gstaudioclock.c
+++ b/gst-libs/gst/audio/gstaudioclock.c
@ -29,8 +29,6 @@
 * simply need to provide a function that returns the current clock time.
 *
 * This object is internally used to implement the clock in #GstAudioBaseSink.
- *
- * Last reviewed on 2006-09-27 (0.10.12)
 */

 #ifdef HAVE_CONFIG_H
--- a/gst-libs/gst/audio/gstaudiofilter.c
+++ b/gst-libs/gst/audio/gstaudiofilter.c
@ -36,8 +36,6 @@
 * #GstBaseTransformClass.transform_ip() and/or
 * #GstBaseTransformClass.transform()
 * virtual functions in their class_init function.
- *
- * Last reviewed on 2007-02-03 (0.10.11.1)
 */

 #ifdef HAVE_CONFIG_H
--- a/gst-libs/gst/audio/gstaudioringbuffer.c
+++ b/gst-libs/gst/audio/gstaudioringbuffer.c
@ -35,8 +35,6 @@
 * implementations.
 * </para>
 * </refsect2>
- *
- * Last reviewed on 2006-02-02 (0.10.4)
 */

 #include <string.h>
--- a/gst-libs/gst/audio/gstaudiosink.c
+++ b/gst-libs/gst/audio/gstaudiosink.c
@ -63,8 +63,6 @@
 * All scheduling of samples and timestamps is done in this base class
 * together with #GstAudioBaseSink using a default implementation of a
 * #GstAudioRingBuffer that uses threads.
- *
- * Last reviewed on 2006-09-27 (0.10.12)
 */

 #include <string.h>
--- a/gst-libs/gst/audio/gstaudiosrc.c
+++ b/gst-libs/gst/audio/gstaudiosrc.c
@ -63,8 +63,6 @@
 * All scheduling of samples and timestamps is done in this base class
 * together with #GstAudioBaseSrc using a default implementation of a
 * #GstAudioRingBuffer that uses threads.
- *
- * Last reviewed on 2006-09-27 (0.10.12)
 */

 #include <string.h>
--- a/gst-libs/gst/rtp/gstrtcpbuffer.c
+++ b/gst-libs/gst/rtp/gstrtcpbuffer.c
@ -40,8 +40,6 @@
 * gst_rtcp_packet_move_to_next().
 * </para>
 * </refsect2>
- *
- * Last reviewed on 2007-03-26 (0.10.13)
 */

 #include <string.h>
--- a/gst-libs/gst/rtp/gstrtpbuffer.c
+++ b/gst-libs/gst/rtp/gstrtpbuffer.c
@ -30,8 +30,6 @@
 * 'application/x-rtp' #GstCaps.
 * </para>
 * </refsect2>
- *
- * Last reviewed on 2006-07-17 (0.10.10)
 */

 #include "gstrtpbuffer.h"
--- a/gst-libs/gst/rtp/gstrtphdrext.c
+++ b/gst-libs/gst/rtp/gstrtphdrext.c
@ -26,8 +26,6 @@
 * <para>
 * </para>
 * </refsect2>
- *
- * Last reviewed on 2012-09-24 (1.0)
 */

 #include "gstrtphdrext.h"
--- a/gst-libs/gst/rtp/gstrtppayloads.c
+++ b/gst-libs/gst/rtp/gstrtppayloads.c
@ -32,8 +32,6 @@
 * and get session bandwidth information.
 * </para>
 * </refsect2>
- *
- * Last reviewed on 2007-10-01 (0.10.15)
 */

 #include <string.h>
--- a/gst-libs/gst/rtsp/gstrtspconnection.c
+++ b/gst-libs/gst/rtsp/gstrtspconnection.c
@ -47,8 +47,6 @@
 *
 * This object manages the RTSP connection to the server. It provides function
 * to receive and send bytes and messages.
- *
- * Last reviewed on 2007-07-24 (0.10.14)
 */

 #ifdef HAVE_CONFIG_H
--- a/gst-libs/gst/rtsp/gstrtspdefs.c
+++ b/gst-libs/gst/rtsp/gstrtspdefs.c
@ -46,8 +46,6 @@
 * @see_also: gstrtspurl, gstrtspconnection
 *  
 * Provides common defines for the RTSP library. 
- *  
- * Last reviewed on 2007-07-24 (0.10.14)
 */

 #include <errno.h>
--- a/gst-libs/gst/rtsp/gstrtspextension.c
+++ b/gst-libs/gst/rtsp/gstrtspextension.c
@ -29,8 +29,6 @@
 *  exentension (rtspwms) and the RealMedia RTSP extension (rtspreal).
 * </para>
 * </refsect2>
- *
- * Last reviewed on 2007-07-25 (0.10.14)
 */

 #ifdef HAVE_CONFIG_H
--- a/gst-libs/gst/rtsp/gstrtspmessage.c
+++ b/gst-libs/gst/rtsp/gstrtspmessage.c
@ -47,8 +47,6 @@
 * @see_also: gstrtspconnection
 *  
 * Provides methods for creating and parsing request, response and data messages.
- *  
- * Last reviewed on 2007-07-25 (0.10.14)
 */

 #include <string.h>
--- a/gst-libs/gst/rtsp/gstrtsprange.c
+++ b/gst-libs/gst/rtsp/gstrtsprange.c
@ -45,8 +45,6 @@
 * @short_description: dealing with time ranges
 *  
 * Provides helper functions to deal with time ranges.
- *  
- * Last reviewed on 2007-07-25 (0.10.14)
 */

 #include <math.h>
--- a/gst-libs/gst/rtsp/gstrtsptransport.c
+++ b/gst-libs/gst/rtsp/gstrtsptransport.c
@ -46,8 +46,6 @@
 * @short_description: dealing with RTSP transports
 *  
 * Provides helper functions to deal with RTSP transport strings.
- *  
- * Last reviewed on 2007-07-25 (0.10.14)
 */

 #include <string.h>
--- a/gst-libs/gst/rtsp/gstrtspurl.c
+++ b/gst-libs/gst/rtsp/gstrtspurl.c
@ -45,8 +45,6 @@
 * @short_description: handling RTSP urls
 *  
 * Provides helper functions to handle RTSP urls.
- *  
- * Last reviewed on 2007-07-25 (0.10.14)
 */

 #include <stdlib.h>
--- a/gst-libs/gst/sdp/gstmikey.c
+++ b/gst-libs/gst/sdp/gstmikey.c
@ -29,8 +29,6 @@
 * messages.
 * </para>
 * </refsect2>
- *
- * Last reviewed on 2014-03-20 (1.3.0)
 */

 #include <string.h>
--- a/gst-libs/gst/sdp/gstsdpmessage.c
+++ b/gst-libs/gst/sdp/gstsdpmessage.c
@ -50,8 +50,6 @@
 * messages.
 * </para>
 * </refsect2>
- *
- * Last reviewed on 2007-07-24 (0.10.14)
 */

 #ifdef HAVE_CONFIG_H
--- a/gst/adder/gstadder.c
+++ b/gst/adder/gstadder.c
@ -35,8 +35,6 @@
 * gst-launch audiotestsrc freq=100 ! adder name=mix ! audioconvert ! alsasink audiotestsrc freq=500 ! mix.
 * ]| This pipeline produces two sine waves mixed together.
 * </refsect2>
- *
- * Last reviewed on 2006-05-09 (0.10.7)
 */
 /* Element-Checklist-Version: 5 */

--- a/gst/audioconvert/gstaudioconvert.c
+++ b/gst/audioconvert/gstaudioconvert.c
@ -39,8 +39,6 @@
 * ]| The vorbis encoder takes float audio data instead of the integer data
 * generated by audiotestsrc.
 * </refsect2>
- *
- * Last reviewed on 2006-03-02 (0.10.4)
 */

 /*
--- a/gst/playback/gstplaybin2.c
+++ b/gst/playback/gstplaybin2.c
@ -3790,7 +3790,7 @@ avelements_create (GstPlayBin * playbin, gboolean isaudioelement)
  }

  /* create a list of audio/video elements. Each element in the list
-   * is holding an audio/video decoder and an auido/video sink in which
+   * is holding an audio/video decoder and an audio/video sink in which
   * the decoders srcpad template caps and sink element's sinkpad template
   * caps are compatible */
  dl = dec_list;
--- a/gst/tcp/gstmultifdsink.c
+++ b/gst/tcp/gstmultifdsink.c
@ -97,8 +97,6 @@
 * buffers to the clients. This behaviour can be disabled by setting the sync 
 * property to FALSE. Multifdsink will by default not do QoS and will never
 * drop late buffers.
- *
- * Last reviewed on 2006-09-12 (0.10.10)
 */

 #ifdef HAVE_CONFIG_H
--- a/gst/tcp/gstmultihandlesink.c
+++ b/gst/tcp/gstmultihandlesink.c
@ -96,8 +96,6 @@
 * buffers to the clients. This behaviour can be disabled by setting the sync 
 * property to FALSE. Multisocketsink will by default not do QoS and will never
 * drop late buffers.
- *
- * Last reviewed on 2006-09-12 (0.10.10)
 */

 #ifdef HAVE_CONFIG_H
--- a/gst/tcp/gstmultioutputsink.c
+++ b/gst/tcp/gstmultioutputsink.c
@ -96,8 +96,6 @@
 * buffers to the clients. This behaviour can be disabled by setting the sync 
 * property to FALSE. Multioutputsink will by default not do QoS and will never
 * drop late buffers.
- *
- * Last reviewed on 2006-09-12 (0.10.10)
 */

 #ifdef HAVE_CONFIG_H
--- a/gst/tcp/gstmultisocketsink.c
+++ b/gst/tcp/gstmultisocketsink.c
@ -96,8 +96,6 @@
 * buffers to the clients. This behaviour can be disabled by setting the sync 
 * property to FALSE. Multisocketsink will by default not do QoS and will never
 * drop late buffers.
- *
- * Last reviewed on 2006-09-12 (0.10.10)
 */

 #ifdef HAVE_CONFIG_H
--- a/gst/videorate/gstvideorate.c
+++ b/gst/videorate/gstvideorate.c
@ -63,8 +63,6 @@
 * ]| Capture video from a V4L device, and adjust the stream to 12.5 fps before
 * encoding to Ogg/Theora.
 * </refsect2>
- *
- * Last reviewed on 2006-09-02 (0.10.11)
 */

 #ifdef HAVE_CONFIG_H
--- a/gst/videoscale/gstvideoscale.c
+++ b/gst/videoscale/gstvideoscale.c
@ -44,8 +44,6 @@
 * ]| Decode an Ogg/Theora and display the video using xvimagesink with a width
 * of 50.
 * </refsect2>
- *
- * Last reviewed on 2006-03-02 (0.10.4)
 */

 /*