diff --git a/ext/alsa/gstalsasink.c b/ext/alsa/gstalsasink.c
index 38957499ca..1985cfb2e2 100644
--- 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
diff --git a/ext/alsa/gstalsasrc.c b/ext/alsa/gstalsasrc.c
index 741f9f9fa2..43b82b12fc 100644
--- 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
diff --git a/ext/ogg/gstoggdemux.c b/ext/ogg/gstoggdemux.c
index 343f9973ca..902410b611 100644
--- 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)
  */
 
 
diff --git a/ext/ogg/gstoggmux.c b/ext/ogg/gstoggmux.c
index fb96d7e661..cc0af3a9ef 100644
--- 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
diff --git a/ext/theora/gsttheoradec.c b/ext/theora/gsttheoradec.c
index fee82bdc27..11ecffd1a0 100644
--- 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
diff --git a/ext/theora/gsttheoraenc.c b/ext/theora/gsttheoraenc.c
index 5196ae9e23..48b3533871 100644
--- 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
diff --git a/ext/theora/gsttheoraparse.c b/ext/theora/gsttheoraparse.c
index 062cacd75d..f3558e80e6 100644
--- 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
diff --git a/ext/vorbis/gstvorbisdec.c b/ext/vorbis/gstvorbisdec.c
index 7273d2c194..3a5a334ce1 100644
--- 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
diff --git a/ext/vorbis/gstvorbisenc.c b/ext/vorbis/gstvorbisenc.c
index 655dd15889..6579321ec2 100644
--- 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"
diff --git a/ext/vorbis/gstvorbisparse.c b/ext/vorbis/gstvorbisparse.c
index eff07b8eb2..57b792c69b 100644
--- 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
diff --git a/gst-libs/gst/app/gstappsink.c b/gst-libs/gst/app/gstappsink.c
index be6237f5f0..ecec3b56f1 100644
--- 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
diff --git a/gst-libs/gst/app/gstappsrc.c b/gst-libs/gst/app/gstappsrc.c
index 34b8a169ee..a3be7741f7 100644
--- 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
diff --git a/gst-libs/gst/audio/gstaudiobasesink.c b/gst-libs/gst/audio/gstaudiobasesink.c
index 0e2884b5d5..26f71347ea 100644
--- 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>
diff --git a/gst-libs/gst/audio/gstaudiobasesrc.c b/gst-libs/gst/audio/gstaudiobasesrc.c
index 872116b0fc..38b22b54c9 100644
--- 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
diff --git a/gst-libs/gst/audio/gstaudioclock.c b/gst-libs/gst/audio/gstaudioclock.c
index a996b2d3f9..70f507434d 100644
--- 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
diff --git a/gst-libs/gst/audio/gstaudiofilter.c b/gst-libs/gst/audio/gstaudiofilter.c
index 71aed1bbaa..94b3a586d0 100644
--- 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
diff --git a/gst-libs/gst/audio/gstaudioringbuffer.c b/gst-libs/gst/audio/gstaudioringbuffer.c
index c0b060f8f4..7f4b17b017 100644
--- 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>
diff --git a/gst-libs/gst/audio/gstaudiosink.c b/gst-libs/gst/audio/gstaudiosink.c
index 66bb476082..a978030805 100644
--- 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>
diff --git a/gst-libs/gst/audio/gstaudiosrc.c b/gst-libs/gst/audio/gstaudiosrc.c
index 71dc1e9fc1..197ab1722e 100644
--- 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>
diff --git a/gst-libs/gst/rtp/gstrtcpbuffer.c b/gst-libs/gst/rtp/gstrtcpbuffer.c
index 2935b9ba31..3ab97e843d 100644
--- 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>
diff --git a/gst-libs/gst/rtp/gstrtpbuffer.c b/gst-libs/gst/rtp/gstrtpbuffer.c
index 01abd2adae..5f0fe8fe0c 100644
--- 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"
diff --git a/gst-libs/gst/rtp/gstrtphdrext.c b/gst-libs/gst/rtp/gstrtphdrext.c
index ed8f33adb6..9314ba4996 100644
--- 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"
diff --git a/gst-libs/gst/rtp/gstrtppayloads.c b/gst-libs/gst/rtp/gstrtppayloads.c
index 5717ec6762..4a38f808c6 100644
--- 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>
diff --git a/gst-libs/gst/rtsp/gstrtspconnection.c b/gst-libs/gst/rtsp/gstrtspconnection.c
index 7a0e31795f..ac53f15c43 100644
--- 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
diff --git a/gst-libs/gst/rtsp/gstrtspdefs.c b/gst-libs/gst/rtsp/gstrtspdefs.c
index 667538f990..f1d98e2797 100644
--- 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>
diff --git a/gst-libs/gst/rtsp/gstrtspextension.c b/gst-libs/gst/rtsp/gstrtspextension.c
index 89936aac58..5b5eed3b5a 100644
--- 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
diff --git a/gst-libs/gst/rtsp/gstrtspmessage.c b/gst-libs/gst/rtsp/gstrtspmessage.c
index 3eece20fec..841aa19709 100644
--- 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>
diff --git a/gst-libs/gst/rtsp/gstrtsprange.c b/gst-libs/gst/rtsp/gstrtsprange.c
index f523afe7bd..96efd451da 100644
--- 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>
diff --git a/gst-libs/gst/rtsp/gstrtsptransport.c b/gst-libs/gst/rtsp/gstrtsptransport.c
index 827a4f66b2..81c7431086 100644
--- 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>
diff --git a/gst-libs/gst/rtsp/gstrtspurl.c b/gst-libs/gst/rtsp/gstrtspurl.c
index 700ac0b9e0..6c7e8c4faa 100644
--- 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>
diff --git a/gst-libs/gst/sdp/gstmikey.c b/gst-libs/gst/sdp/gstmikey.c
index 2193b82d8b..712941a9fd 100644
--- 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>
diff --git a/gst-libs/gst/sdp/gstsdpmessage.c b/gst-libs/gst/sdp/gstsdpmessage.c
index 9e2565c5c7..6d8e4dc9c3 100644
--- 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
diff --git a/gst/adder/gstadder.c b/gst/adder/gstadder.c
index fb280ffa07..6e9298e9f5 100644
--- 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 */
 
diff --git a/gst/audioconvert/gstaudioconvert.c b/gst/audioconvert/gstaudioconvert.c
index 01dc950d4d..21510fa912 100644
--- 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)
  */
 
 /*
diff --git a/gst/playback/gstplaybin2.c b/gst/playback/gstplaybin2.c
index 89bb951dc2..ab482000bf 100644
--- 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;
diff --git a/gst/tcp/gstmultifdsink.c b/gst/tcp/gstmultifdsink.c
index 2d6128b51e..ab95985d40 100644
--- 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
diff --git a/gst/tcp/gstmultihandlesink.c b/gst/tcp/gstmultihandlesink.c
index 46a618964b..b9ba36b39b 100644
--- 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
diff --git a/gst/tcp/gstmultioutputsink.c b/gst/tcp/gstmultioutputsink.c
index 3d6d56c63c..e0503b5530 100644
--- 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
diff --git a/gst/tcp/gstmultisocketsink.c b/gst/tcp/gstmultisocketsink.c
index 06e97790d4..01647b1d4a 100644
--- 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
diff --git a/gst/videorate/gstvideorate.c b/gst/videorate/gstvideorate.c
index 3284738816..14d628ff11 100644
--- 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
diff --git a/gst/videoscale/gstvideoscale.c b/gst/videoscale/gstvideoscale.c
index e7529318e8..97d5648b21 100644
--- 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)
  */
 
 /*