Release 0.10.16

1 files changed, 5719 insertions, 11 deletions

diff --git a/ChangeLog b/ChangeLog
index 3321dce6..7c566ef8 100644
--- a/ChangeLog
+++ b/ChangeLog
@@ -1,9 +1,5723 @@
+=== release 0.10.16 ===
+
+2009-08-29  Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+	* configure.ac:
+	  releasing 0.10.16, "Secret Handshakes"
+
+2009-08-26 00:58:45 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+	* configure.ac:
+	  0.10.15.5 pre-release
+
+2009-08-25 16:53:29 +0200  Wim Taymans <wim.taymans@collabora.co.uk>
+
+	* ext/pulse/pulsesink.c:
+	  pulsesink: don't use relative seeks
+	  Don't use relative seeks, it's too hard to track where we are after a flush
+	  etc.
+	  fixes #593015
+
+2009-08-24 17:50:29 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+	* configure.ac:
+	* po/LINGUAS:
+	* po/af.po:
+	* po/az.po:
+	* po/bg.po:
+	* po/ca.po:
+	* po/cs.po:
+	* po/da.po:
+	* po/de.po:
+	* po/en_GB.po:
+	* po/es.po:
+	* po/eu.po:
+	* po/fi.po:
+	* po/fr.po:
+	* po/hu.po:
+	* po/id.po:
+	* po/it.po:
+	* po/ja.po:
+	* po/lt.po:
+	* po/lv.po:
+	* po/mt.po:
+	* po/nb.po:
+	* po/nl.po:
+	* po/or.po:
+	* po/pl.po:
+	* po/pt_BR.po:
+	* po/ru.po:
+	* po/sk.po:
+	* po/sq.po:
+	* po/sr.po:
+	* po/sv.po:
+	* po/tr.po:
+	* po/uk.po:
+	* po/vi.po:
+	* po/zh_CN.po:
+	* po/zh_HK.po:
+	* po/zh_TW.po:
+	  0.10.15.4 pre-release
+
+2009-08-24 16:22:47 +0200  Wim Taymans <wim.taymans@collabora.co.uk>
+
+	* ext/pulse/pulsesrc.c:
+	  pulsesrc: don't discard the result of _set_caps()
+	  Use the result of gst_pad_set_caps() instead of assuming success.
+	  See #590678
+
+2009-08-21 11:44:43 +0200  Wim Taymans <wim.taymans@collabora.co.uk>
+
+	* gst/qtdemux/qtdemux.c:
+	* gst/qtdemux/qtdemux_fourcc.h:
+	  qtdemux: add support for agsm
+	  Fixes #592530
+
+2009-08-18 17:16:11 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
+
+	* gst/qtdemux/qtdemux.c:
+	  qtdemux: fix qt style string tag extraction
+	  QT style tags are tested on starting with (C) symbol using >>,
+	  and (unsigned) int (may) have different >> behaviour.
+	  Fixes #592232.
+
+2009-08-17 15:48:20 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+	* ext/jpeg/smokecodec.c:
+	  smokeenc: don't crash when compiled against libjpeg7
+	  Set parameters so that we don't crash with libjpeg7. Based on
+	  Stefan Kost's fix for jpegenc. Fixes #591951.
+
+2009-08-14 20:18:04 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+	* configure.ac:
+	* po/af.po:
+	* po/az.po:
+	* po/bg.po:
+	* po/ca.po:
+	* po/cs.po:
+	* po/da.po:
+	* po/de.po:
+	* po/en_GB.po:
+	* po/es.po:
+	* po/eu.po:
+	* po/fi.po:
+	* po/fr.po:
+	* po/hu.po:
+	* po/id.po:
+	* po/it.po:
+	* po/ja.po:
+	* po/lt.po:
+	* po/mt.po:
+	* po/nb.po:
+	* po/nl.po:
+	* po/or.po:
+	* po/pl.po:
+	* po/pt_BR.po:
+	* po/ru.po:
+	* po/sk.po:
+	* po/sq.po:
+	* po/sr.po:
+	* po/sv.po:
+	* po/tr.po:
+	* po/uk.po:
+	* po/vi.po:
+	* po/zh_CN.po:
+	* po/zh_HK.po:
+	* po/zh_TW.po:
+	  0.10.15.3 pre-release
+
+2009-08-14 13:45:08 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+	* tests/check/elements/rtpbin.c:
+	  checks: add test for leak to rtpbin unit test
+	  See #591476.
+
+2009-08-11 14:47:12 -0400  Olivier Crête <olivier.crete@collabora.co.uk>
+
+	* gst/rtpmanager/gstrtpbin.c:
+	  rtpbin: Fix reference leak
+	  Fixes #591476.
+
+2009-08-14 13:34:53 +0100  Zaheer Merali <zaheerabbas@merali.org>
+
+	* ext/dv/gstdvdec.c:
+	  dvdec: set bottom field first on PAL interlaced content, not top field first
+	  DV interlaced content is always bottom field first. Fixes #591712.
+
+2009-08-14 12:44:06 +0100  Hans de Goede <jwrdegoede@fedoraproject.org>
+
+	* sys/v4l2/gstv4l2src.c:
+	  v4l2src: fix 'hang' with some cameras caused by bad timestamping if no framerate is available
+	  For cameras/drivers that don't support e.g. VIDIOC_G_PARM we'd end up without
+	  a framerate and would try to divide by 0, causing run-time warnings and all
+	  frames to be timestamped with 0, which makes sinks that sync against the clock
+	  drop them, causing 'hangs' (observed with the pwc driver and a Logitech QuickCam
+	  Pro 4000). So if we do not know the framerate, simply don't adjust the
+	  timestamps. Fixes #591451.
+
+2009-08-14 10:11:25 +0200  Filippo Argiolas <filippo.argiolas@gmail.com>
+
+	* sys/v4l2/gstv4l2object.c:
+	* sys/v4l2/gstv4l2src.c:
+	  v4l2src: clear format list in READY->NULL
+	  Clear format list and probed caps when going to NULL so if a new device
+	  is set we'll probe the formats again instead of using previously
+	  detected ones. Fixes bug #591747.
+
+2009-08-11 17:30:41 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+	* configure.ac:
+	* po/LINGUAS:
+	* po/af.po:
+	* po/az.po:
+	* po/bg.po:
+	* po/ca.po:
+	* po/cs.po:
+	* po/da.po:
+	* po/de.po:
+	* po/en_GB.po:
+	* po/es.po:
+	* po/eu.po:
+	* po/fi.po:
+	* po/fr.po:
+	* po/hu.po:
+	* po/id.po:
+	* po/it.po:
+	* po/ja.po:
+	* po/lt.po:
+	* po/mt.po:
+	* po/nb.po:
+	* po/nl.po:
+	* po/or.po:
+	* po/pl.po:
+	* po/pt_BR.po:
+	* po/ru.po:
+	* po/sk.po:
+	* po/sq.po:
+	* po/sr.po:
+	* po/sv.po:
+	* po/tr.po:
+	* po/uk.po:
+	* po/vi.po:
+	* po/zh_CN.po:
+	* po/zh_HK.po:
+	* po/zh_TW.po:
+	  0.10.15.2 pre-release
+
+2009-08-11 15:25:39 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+	* MAINTAINERS:
+	  Add myself to MAINTAINERS file and update Wim's e-mail.
+
+2009-08-11 03:08:01 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+	* sys/v4l2/Makefile.am:
+	  v4l2: fix make distcheck by disting some more headers
+
+2009-08-11 02:42:16 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+	* docs/plugins/gst-plugins-good-plugins.args:
+	* docs/plugins/gst-plugins-good-plugins.hierarchy:
+	* docs/plugins/gst-plugins-good-plugins.interfaces:
+	* docs/plugins/gst-plugins-good-plugins.prerequisites:
+	* docs/plugins/gst-plugins-good-plugins.signals:
+	* docs/plugins/inspect/plugin-avi.xml:
+	* docs/plugins/inspect/plugin-cairo.xml:
+	* docs/plugins/inspect/plugin-matroska.xml:
+	* docs/plugins/inspect/plugin-pulseaudio.xml:
+	* docs/plugins/inspect/plugin-rtp.xml:
+	* docs/plugins/inspect/plugin-video4linux2.xml:
+	* docs/plugins/inspect/plugin-wavparse.xml:
+	  docs: update
+
+2009-08-11 02:31:44 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+	* configure.ac:
+	* docs/plugins/Makefile.am:
+	* docs/plugins/gst-plugins-good-plugins-docs.sgml:
+	* docs/plugins/gst-plugins-good-plugins-sections.txt:
+	* docs/plugins/inspect/plugin-gstrtpmanager.xml:
+	* gst-plugins-good.spec.in:
+	* tests/check/Makefile.am:
+	* tests/check/elements/.gitignore:
+	* tests/check/pipelines/.gitignore:
+	  Move rtpmanager from -bad to -good.
+	  Hook up build infrastructure (autotools, docs, unit test).
+
+2009-08-06 19:26:21 +0200  ric <csxnju at sogou.com>
+
+	* gst/rtpmanager/rtpsource.c:
+	  rtpsource: avoid buffer leak on bad seqnum
+	  Fixes #590797
+
+2009-07-28 18:18:20 +0200  Wim Taymans <wim.taymans@collabora.co.uk>
+
+	* gst/rtpmanager/rtpsource.c:
+	  rtpsource: allow for NULL caps on buffers
+	  Add the NULL caps check where it matters and also cover another case of
+	  potential NULL caps.
+	  Fixes #590030
+
+2009-07-28 11:59:56 -0400  Olivier Crête <olivier.crete@collabora.co.uk>
+
+	* gst/rtpmanager/rtpsource.c:
+	  rtpsource: Incoming buffers do not always have caps
+
+2009-07-27 15:46:23 +0200  Wim Taymans <wim.taymans@collabora.co.uk>
+
+	* gst/rtpmanager/rtpsession.c:
+	  rtpsession: avoid doing lip-sync in BYE
+	  When we get a BYE packet, don't do lip-sync with the SR inside because some
+	  senders have trouble constructing valid SR packets after BYE.
+
+2009-07-27 13:17:20 +0200  Wim Taymans <wim.taymans@collabora.co.uk>
+
+	* gst/rtpmanager/rtpsession.c:
+	  rtpbin: don't do lip-sync after a BYE
+	  After a BYE packet from a source, stop forwarding the SR packets for lip-sync
+	  to rtpbin. Some senders don't update their SR packets correctly after sending a
+	  BYE and then we break lip-sync. We prefer to let the jitterbuffers drain with
+	  the current lip-sync instead.
+
+2009-07-27 12:43:02 +0200  Wim Taymans <wim.taymans@collabora.co.uk>
+
+	* gst/rtpmanager/rtpsession.c:
+	  rtpbin: only reconsider once for BYE
+	  When iterating the sources of a BYE packet, don't signal a reconsideration for
+	  each of them but signal after we handled all sources.
+
+2009-07-21 15:33:41 -0400  Olivier Crête <olivier.crete@collabora.co.uk>
+
+	* gst/rtpmanager/rtpsession.c:
+	  rtpsession: Free conflicting addresses on finalize
+
+2009-07-01 12:55:03 +0200  Wim Taymans <wim.taymans@collabora.co.uk>
+
+	* gst/rtpmanager/rtpsource.c:
+	  rtpbin: use new method for netaddress to string
+
+2009-06-29 18:48:33 +0200  Wim Taymans <wim.taymans@collabora.co.uk>
+
+	* gst/rtpmanager/gstrtpbin.c:
+	* tests/check/elements/rtpbin.c:
+	  rtpbin: do better cleanup of the src ghostpads
+	  Connect to the pad-removed signal of the ptdemux elements so that we remove the
+	  ghostpads for them. Fixes cleanup when going to NULL as well as when releasing
+	  the sinkpads.
+	  Fixes #561752
+
+2009-05-28 19:08:40 +0200  Wim Taymans <wim.taymans@collabora.co.uk>
+
+	* gst/rtpmanager/rtpsession.c:
+	  rtpsession: add a comment
+
+2009-06-29 16:37:54 +0200  Wim Taymans <wim.taymans@collabora.co.uk>
+
+	* gst/rtpmanager/gstrtpbin.c:
+	* gst/rtpmanager/gstrtpbin.h:
+	* gst/rtpmanager/gstrtpsession.c:
+	  rtpbin: add SDES property
+	  Remove all individual SDES properties and use one sdes property that takes a
+	  GstStructure instead. This will allow us to add more custom stuff to the SDES
+	  messages later.
+
+2009-06-29 16:21:05 +0200  Wim Taymans <wim.taymans@collabora.co.uk>
+
+	* gst/rtpmanager/rtpsession.c:
+	* gst/rtpmanager/rtpsession.h:
+	* gst/rtpmanager/rtpsource.c:
+	* gst/rtpmanager/rtpsource.h:
+	  rtpbin: add SDES property that takes GstStructure
+	  Remove all individual SDES properties and use one sdes property that takes a
+	  GstStructure instead. This will allow us to add more custom stuff to the SDES
+	  messages later.
+
+2009-06-02 17:46:08 +0200  Wim Taymans <wim.taymans@collabora.co.uk>
+
+	* gst/rtpmanager/Makefile.am:
+	* gst/rtpmanager/gstrtpclient.c:
+	* gst/rtpmanager/gstrtpclient.h:
+	* gst/rtpmanager/gstrtpmanager.c:
+	  rtpbin: removed old gstrtpclient
+
+2009-06-19 19:09:19 +0200  Branko Subasic <branko.subasic at axis.com>
+
+	* gst/rtpmanager/gstrtpsession.c:
+	* gst/rtpmanager/rtpsession.c:
+	* gst/rtpmanager/rtpsession.h:
+	* gst/rtpmanager/rtpsource.c:
+	* gst/rtpmanager/rtpsource.h:
+	* tests/check/elements/rtpbin_buffer_list.c:
+	  rtpbin: add support for buffer-list
+	  Add support for sending buffer-lists.
+	  Add unit test for testing that the buffer-list passed through rtpbin.
+	  fixes #585839
+
+2009-06-19 16:21:28 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+	* gst/rtpmanager/gstrtpjitterbuffer.c:
+	  Make build without warnings with debugging disabled
+
+2009-05-28 17:37:44 -0400  Olivier Crête <olivier.crete@collabora.co.uk>
+
+	* gst/rtpmanager/gstrtpbin.c:
+	  rtpbin: Transform the right session sdes message
+	  Fixes #584165
+
+2009-05-28 17:33:10 -0400  Olivier Crête <olivier.crete@collabora.co.uk>
+
+	* gst/rtpmanager/rtpsource.c:
+	  Add ssrc to application/x-rtp-source-sdes structure
+
+2009-05-27 11:03:14 +0200  Wim Taymans <wim.taymans@collabora.co.uk>
+
+	* gst/rtpmanager/rtpsource.c:
+	  rtpsouce: the network address is in network order
+	  Bring the network address in netowkr byte order to the host order.
+
+2009-05-26 15:40:52 +0200  Wim Taymans <wim.taymans@collabora.co.uk>
+
+	* gst/rtpmanager/rtpsource.c:
+	  rtpsource: byteswap the port from GstNetAddress
+	  Since the port in GstNetAddress is in network order we might need to byteswap it
+	  before adding it to the source statistics.
+
+2009-05-25 13:46:29 +0200  Wim Taymans <wim.taymans@collabora.co.uk>
+
+	* gst/rtpmanager/gstrtpbin.c:
+	  rtpbin: remove ptdemux ghostpads
+
+2009-05-25 13:33:20 +0200  Wim Taymans <wim.taymans@collabora.co.uk>
+
+	* tests/check/elements/rtpbin.c:
+	  tests: add receive rtpbin unit test
+
+2009-05-22 16:41:19 +0200  Wim Taymans <wim.taymans@collabora.co.uk>
+
+	* gst/rtpmanager/gstrtpbin.c:
+	  rtpbin: add to new signal to remove SSRC pads
+
+2009-05-22 16:35:20 +0200  Ali Sabil <ali.sabil at gmail.com>
+
+	* gst/rtpmanager/gstrtpbin-marshal.list:
+	* gst/rtpmanager/gstrtpssrcdemux.c:
+	* gst/rtpmanager/gstrtpssrcdemux.h:
+	  ssrcdemux: emit signal when pads are removed
+	  Add action signal to clear an SSRC in the ssrc demuxer.
+	  Add signal to notify of removed ssrc.
+	  See #554839
+
+2009-05-22 15:45:19 +0200  Wim Taymans <wim.taymans@collabora.co.uk>
+
+	* gst/rtpmanager/gstrtpbin.c:
+	  rtpbin: use our ghostpads instead of its target
+	  Since we keep a reference to our ghostpads, we can use them to track sessions.
+	  This avoid us having to mess with the target of the ghostpad.
+
+2009-05-22 15:37:29 +0200  Wim Taymans <wim.taymans@collabora.co.uk>
+
+	* tests/check/elements/rtpbin.c:
+	  tests: more rtpbin checks
+
+2009-05-22 15:36:17 +0200  Wim Taymans <wim.taymans@collabora.co.uk>
+
+	* gst/rtpmanager/gstrtpbin.c:
+	  rtpbin: don't warn when getting request pads twice
+	  Allow getting the request pads multiple times, just return the previously
+	  created pads.
+
+2009-05-22 13:47:30 +0200  Wim Taymans <wim.taymans@collabora.co.uk>
+
+	* gst/rtpmanager/rtpsource.c:
+	  rtpsource: add RTP and RTCP source address
+	  Add the RTP and RTCP sender addresses in the stats structure.
+
+2009-05-22 13:45:15 +0200  Wim Taymans <wim.taymans@collabora.co.uk>
+
+	* gst/rtpmanager/gstrtpsession.c:
+	  rtpsession: reuse source code for SDES
+	  Reuse the RTPSource object property instead of duplicating code.
+
+2009-05-22 13:44:17 +0200  Wim Taymans <wim.taymans@collabora.co.uk>
+
+	* tests/check/elements/rtpbin.c:
+	  tests: add more rtpbin tests
+
+2009-05-22 12:23:27 +0200  Wim Taymans <wim.taymans@collabora.co.uk>
+
+	* tests/check/elements/rtpbin.c:
+	  tests: add rtpbin unit test
+	  Add the beginnings of an rtpbin unit test
+	  Add some more stuff to .gitignore
+
+2009-05-22 12:20:13 +0200  Wim Taymans <wim.taymans@collabora.co.uk>
+
+	* gst/rtpmanager/gstrtpbin.c:
+	  rtpbin: set target state on new elements
+	  Set the state on newly added elements to the state of the parent.
+	  Add some debug info and do some cleanups
+
+2009-05-22 11:59:17 +0200  Wim Taymans <wim.taymans@collabora.co.uk>
+
+	* gst/rtpmanager/gstrtpbin.c:
+	  rtpbin: unref requests pads after releasing
+
+2009-05-22 01:43:50 +0200  Olivier Crête <olivier.crete@collabora.co.uk>
+
+	* gst/rtpmanager/gstrtpbin.c:
+	  rtpbin: Implement releasing the streams
+	  See #561752
+
+2009-05-22 01:16:11 +0200  Olivier Crête <olivier.crete@collabora.co.uk>
+
+	* gst/rtpmanager/gstrtpbin.c:
+	  rtpbin: Keep jb signals handler
+	  Keep the signal handlers so they can be disconnected at release time
+	  See #561752
+
+2009-05-22 01:12:57 +0200  Wim Taymans <wim.taymans@collabora.co.uk>
+
+	* gst/rtpmanager/gstrtpbin.c:
+	  rtpbin: use the right lock for the sessions
+	  Use the right lock when iterating the sessions.
+
+2009-05-22 01:03:55 +0200  Olivier Crête <olivier.crete@collabora.co.uk>
+
+	* gst/rtpmanager/gstrtpbin.c:
+	  rtpbin: Free session if request pads are released
+	  Free the session when all the request pads are released.
+	  Don't mess with the session list in free_session as it is called from a foreach
+	  on that list.
+	  Set the state of the upstream element to NULL first.
+	  See #561752
+
+2009-05-22 00:51:53 +0200  Olivier Crête <olivier.crete@collabora.co.uk>
+
+	* gst/rtpmanager/gstrtpbin.c:
+	  rtpbin: Implement relasing of the rtp recv pad
+
+2009-05-22 00:44:51 +0200  Olivier Crête <olivier.crete@collabora.co.uk>
+
+	* gst/rtpmanager/gstrtpbin.c:
+	  rtpbin: Implement releasing of rtp send pads
+
+2009-05-22 00:34:36 +0200  Olivier Crête <olivier.crete@collabora.co.uk>
+
+	* gst/rtpmanager/gstrtpbin.c:
+	  rtpbin: Implement release of the recv rtcp pad
+	  See #561752
+
+2009-05-22 00:16:19 +0200  Olivier Crête <olivier.crete@collabora.co.uk>
+
+	* gst/rtpmanager/gstrtpbin.c:
+	  rtpbin: Implement releasing of rtcp src pad
+	  See #561752
+
+2009-05-05 16:48:37 +0200  Wim Taymans <wim.taymans@collabora.co.uk>
+
+	* gst/rtpmanager/gstrtpssrcdemux.c:
+	  rtpssrcdemux: drop unexpected RTCP packets
+	  We usually only get SR packets in our chain function but if an invalid packet
+	  contains the SR packet after the RR packet, we must not fail but simply ignore
+	  the malformed packet.
+	  Fixes #581375
+
+2009-04-27 11:09:08 +0200  Olivier Crete <olivier.crete@collabora.co.uk>
+
+	* gst/rtpmanager/rtpsource.c:
+	  rtpsouce: make WARNING into LOG
+	  Since neither rtpmanager nor any of the payloaders properly implement
+	  pad allocation, there is no way for the rtpmanager to inform downstream elements
+	  of the new SSRC if there is an SSRC collision. So the warning is emitted all the
+	  time and it is confusing.
+	  Fixes #580144
+
+2009-04-27 11:06:01 +0200  Olivier Crete <olivier.crete@collabora.co.uk>
+
+	* gst/rtpmanager/rtpsession.c:
+	  rtpsession: notify when SSRC changes
+	  Emit a g_object_notify when the SSRc changes because of a collision.
+	  Fixes #580144
+
+2009-04-17 16:16:29 +0200  Wim Taymans <wim.taymans@collabora.co.uk>
+
+	* gst/rtpmanager/gstrtpsession.c:
+	  rtpsession: join the RTCP thread
+	  Avoid a case where a joinable thread would be left unjoined, which leaked the
+	  thread structure.
+	  Fixes #577318.
+
+2009-04-15 18:14:48 +0200  Wim Taymans <wim.taymans@collabora.co.uk>
+
+	* gst/rtpmanager/gstrtpjitterbuffer.c:
+	  jitterbuffer: prevent overflow in EOS estimation
+	  Use a guint64 instead of a guint to hold a 64bit value to prevent completely
+	  bogues EOS estimation values due to overflows.
+
+2009-04-15 17:44:17 +0200  Wim Taymans <wim.taymans@collabora.co.uk>
+
+	* gst/rtpmanager/gstrtpbin.c:
+	* gst/rtpmanager/gstrtpbin.h:
+	  rtpbin: we should not provide a clock
+	  There is no need to provide a clock.
+
+2009-04-15 17:28:56 +0200  Wim Taymans <wim.taymans@collabora.co.uk>
+
+	* gst/rtpmanager/gstrtpjitterbuffer.c:
+	  jitterbuffer: more estimated EOS fixes
+	  Do more accurate EOS estimate and guard against backward timestamps.
+
+2009-04-15 17:25:02 +0200  Wim Taymans <wim.taymans@collabora.co.uk>
+
+	* gst/rtpmanager/gstrtpjitterbuffer.c:
+	  jitterbuffer: release lock before pushing EOS
+	  Make sure we release the jitterbuffer lock before we start pushing out data
+	  because else we might deadlock.
+
+2009-03-27 17:44:57 +0100  Wim Taymans <wim.taymans@collabora.co.uk>
+
+	* gst/rtpmanager/gstrtpbin.c:
+	* gst/rtpmanager/gstrtpbin.h:
+	* gst/rtpmanager/gstrtpjitterbuffer.c:
+	* gst/rtpmanager/gstrtpjitterbuffer.h:
+	  rtpbin: add on_npt_stop signal
+	  Add the on_npt_stop signal to rtpbin and rtpjitterbuffer to notify the
+	  application that the NPT stop position has been reached.
+
+2009-03-13 15:59:37 +0100  Wim Taymans <wim.taymans@collabora.co.uk>
+
+	* gst/rtpmanager/gstrtpsession.c:
+	  rtpbin: don't return FALSE on seek events
+	  Silently ignore the seek event instead of returning FALSE.
+
+2009-02-26 13:10:29 +0100  Olivier Crête <olivier.crete@collabora.co.uk>
+
+	* gst/rtpmanager/gstrtpsession.c:
+	  gstrtpbin: Don't forward revc events to sender
+	  Don't send events from the receiver to the sender side.
+	  Fixes #572900.
+
+2009-02-25 11:45:05 +0200  Stefan Kost <ensonic@users.sf.net>
+
+	* gst/rtpmanager/rtpjitterbuffer.c:
+	  docs: various doc fixes
+	  No short-desc as we have them in the element details.
+	  Also keep things (Makefile.am and sections.txt) sorted.
+	  Reword ambigous returns. No text after since please.
+
+2009-01-23 12:13:00 +0100  Wim Taymans <wim.taymans@collabora.co.uk>
+
+	* gst/rtpmanager/rtpstats.c:
+	  Send BYE packets immediatly for small sessions
+	  When the number of participants is less than 50, the RFC allows for sending the
+	  BYE packet immediatly instead of using the regular BYE timeout.
+	  Fixes #567828.
+
+2009-01-22 13:33:14 +0100  Wim Taymans <wim.taymans@collabora.co.uk>
+
+	* gst/rtpmanager/gstrtpjitterbuffer.c:
+	  Unlock the jitterbuffer before pushing out the packet-lost events. Move some code before we do the unlock to make the jitterbuffer state consistent while we are unlocked.
+
+2009-01-02 17:40:06 +0000  Olivier Crete <tester@tester.ca>
+
+	  gst/rtpmanager/: When an SSRC is found on the caps of the sender RTP, use this as the internal SSRC. Fixes #565910.
+	  Original commit message from CVS:
+	  Patch by: Olivier Crete <tester at tester dot ca>
+	  * gst/rtpmanager/gstrtpsession.c:
+	  (gst_rtp_session_setcaps_send_rtp), (create_send_rtp_sink):
+	  * gst/rtpmanager/rtpsession.c: (rtp_session_set_internal_ssrc):
+	  When an SSRC is found on the caps of the sender RTP, use this as the
+	  internal SSRC. Fixes #565910.
+
+2009-01-02 16:50:53 +0000  Wim Taymans <wim.taymans@gmail.com>
+
+	  gst/rtpmanager/: Rename a method to better reflect what it really does.
+	  Original commit message from CVS:
+	  * gst/rtpmanager/gstrtpsession.c:
+	  (gst_rtp_session_event_send_rtp_sink),
+	  (gst_rtp_session_getcaps_send_rtp):
+	  * gst/rtpmanager/rtpsession.c: (check_collision),
+	  (rtp_session_schedule_bye_locked), (rtp_session_schedule_bye):
+	  * gst/rtpmanager/rtpsession.h:
+	  Rename a method to better reflect what it really does.
+
+2008-12-29 15:49:37 +0000  Wim Taymans <wim.taymans@gmail.com>
+
+	  gst/rtpmanager/gstrtpsession.c: Use method to get the internal SSRC.
+	  Original commit message from CVS:
+	  * gst/rtpmanager/gstrtpsession.c:
+	  (gst_rtp_session_getcaps_send_rtp):
+	  Use method to get the internal SSRC.
+	  * gst/rtpmanager/rtpsession.c: (rtp_session_class_init),
+	  (rtp_session_set_property), (rtp_session_get_property):
+	  Add property to congiure the internal SSRC of the session.
+	  Fixes #565910.
+
+2008-12-29 15:21:58 +0000  Wim Taymans <wim.taymans@gmail.com>
+
+	  gst/rtpmanager/rtpsession.c: Only change the SSRC of the session and reset the internal source when the SSRC actually...
+	  Original commit message from CVS:
+	  * gst/rtpmanager/rtpsession.c: (rtp_session_set_internal_ssrc):
+	  Only change the SSRC of the session and reset the internal source when
+	  the SSRC actually changed. See #565910.
+
+2008-12-29 14:21:47 +0000  Wim Taymans <wim.taymans@gmail.com>
+
+	  gst/rtpmanager/rtpsource.*: When no payload was specified on the caps but there was a clock-rate, assume the clock-ra...
+	  Original commit message from CVS:
+	  * gst/rtpmanager/rtpsource.c: (rtp_source_init),
+	  (rtp_source_update_caps), (get_clock_rate):
+	  * gst/rtpmanager/rtpsource.h:
+	  When no payload was specified on the caps but there was a clock-rate,
+	  assume the clock-rate corresponds to the first payload type found in the
+	  RTP packets. Fixes #565509.
+
+2008-12-23 11:39:59 +0000  Arnout Vandecappelle <arnout@mind.be>
+
+	  gst/rtpmanager/rtpjitterbuffer.*: Keep track of the last outgoing timestamp and of the last sender-side time.  Timest...
+	  Original commit message from CVS:
+	  Patch by: Arnout Vandecappelle <arnout at mind dot be>
+	  * gst/rtpmanager/rtpjitterbuffer.c: (rtp_jitter_buffer_reset_skew),
+	  (calculate_skew):
+	  * gst/rtpmanager/rtpjitterbuffer.h:
+	  Keep track of the last outgoing timestamp and of the last sender-side
+	  time.  Timestamps can only go forward if they do at the sender
+	  side, can only go back if they do at the sender side, and remain the
+	  same if they remain the same at the sender side. Fixes #565319.
+
+2008-11-26 12:40:18 +0000  Wim Taymans <wim.taymans@gmail.com>
+
+	  gst/rtpmanager/rtpsession.c: Make obtain_source return an aditional ref so that we don't lose our ref to it when a se...
+	  Original commit message from CVS:
+	  * gst/rtpmanager/rtpsession.c: (obtain_source),
+	  (rtp_session_create_source), (rtp_session_process_rtp),
+	  (rtp_session_process_sr), (rtp_session_process_rr),
+	  (rtp_session_process_sdes), (rtp_session_process_bye):
+	  Make obtain_source return an aditional ref so that we don't lose our ref
+	  to it when a session cleanup occurs when we are emiting a signal.
+	  Emit the on_new_ssrc signal for the CSRC, not the SSRC.
+	  Fixes #562319.
+
+2008-11-26 12:02:21 +0000  Wim Taymans <wim.taymans@gmail.com>
+
+	  gst/rtpmanager/gstrtpbin.c: Reset the sync parameters when clearing the payload type map too.
+	  Original commit message from CVS:
+	  * gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_reset_sync),
+	  (gst_rtp_bin_clear_pt_map):
+	  Reset the sync parameters when clearing the payload type map too.
+	  Fixes #562312.
+
+2008-11-26 11:44:37 +0000  Wim Taymans <wim.taymans@gmail.com>
+
+	  gst/rtpmanager/gstrtpbin.*: Remove a lot of per stream state that is not needed and pass new info in the method call.
+	  Original commit message from CVS:
+	  * gst/rtpmanager/gstrtpbin.c: (get_client),
+	  (gst_rtp_bin_reset_sync), (gst_rtp_bin_associate),
+	  (gst_rtp_bin_handle_sync), (create_stream),
+	  (gst_rtp_bin_class_init), (new_ssrc_pad_found):
+	  * gst/rtpmanager/gstrtpbin.h:
+	  Remove a lot of per stream state that is not needed and pass new info in
+	  the method call.
+	  Add signal to reset sync parameters.
+	  Avoid parsing the caps to get a clock_base, we get this from the sync
+	  signal now.
+
+2008-11-25 15:12:06 +0000  Wim Taymans <wim.taymans@gmail.com>
+
+	  gst/rtpmanager/gstrtpsession.c: Fix event leak.
+	  Original commit message from CVS:
+	  * gst/rtpmanager/gstrtpsession.c:
+	  (gst_rtp_session_event_send_rtcp_src):
+	  Fix event leak.
+
+2008-11-22 15:31:36 +0000  Wim Taymans <wim.taymans@gmail.com>
+
+	  gst/rtpmanager/rtpsession.c: Add property to configure the RTCP MTU.
+	  Original commit message from CVS:
+	  * gst/rtpmanager/rtpsession.c: (rtp_session_class_init),
+	  (rtp_session_init), (rtp_session_set_property),
+	  (rtp_session_get_property):
+	  Add property to configure the RTCP MTU.
+
+2008-11-22 15:24:47 +0000  Wim Taymans <wim.taymans@gmail.com>
+
+	  gst/rtpmanager/rtpsession.c: Add G_PARAM_STATIC_STRINGS.
+	  Original commit message from CVS:
+	  * gst/rtpmanager/rtpsession.c: (rtp_session_class_init),
+	  (copy_source), (rtp_session_create_sources),
+	  (rtp_session_get_property):
+	  Add G_PARAM_STATIC_STRINGS.
+	  Add property to return a GValueArray of all known RTPSources in the
+	  session.
+	  * gst/rtpmanager/rtpsource.c: (rtp_source_class_init),
+	  (rtp_source_create_sdes), (rtp_source_set_property),
+	  (rtp_source_get_property):
+	  Remove properties to set the various SDES items, an application is never
+	  supposed to change the RTPSource data.
+	  Change the SDES getter properties to one SDES property that returns all
+	  SDES items in a GstStructure.
+
+2008-11-22 13:17:24 +0000  Wim Taymans <wim.taymans@gmail.com>
+
+	  gst/rtpmanager/gstrtpbin.c: Also unref the target pad for unknown pads.
+	  Original commit message from CVS:
+	  * gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_release_pad):
+	  Also unref the target pad for unknown pads.
+
+2008-11-21 16:17:22 +0000  Olivier Crete <tester@tester.ca>
+
+	  gst/rtpmanager/gstrtpbin.c: Release the right pads on rtpbin. Fixes #561752.
+	  Original commit message from CVS:
+	  Patch by: Olivier Crete <tester at tester dot ca>
+	  * gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_release_pad):
+	  Release the right pads on rtpbin. Fixes #561752.
+
+2008-11-20 18:41:34 +0000  Wim Taymans <wim.taymans@gmail.com>
+
+	  gst/rtpmanager/gstrtpsession.c: Pass the running time to the session when processing RTP packets.
+	  Original commit message from CVS:
+	  * gst/rtpmanager/gstrtpsession.c: (get_current_times),
+	  (rtcp_thread), (gst_rtp_session_chain_recv_rtp):
+	  Pass the running time to the session when processing RTP packets.
+	  Improve the time function to provide more info.
+	  * gst/rtpmanager/rtpsession.c: (rtp_session_class_init),
+	  (rtp_session_init), (update_arrival_stats),
+	  (rtp_session_process_rtp), (rtp_session_process_sdes),
+	  (rtp_session_process_rtcp), (session_start_rtcp),
+	  (rtp_session_on_timeout):
+	  * gst/rtpmanager/rtpsession.h:
+	  Mark the internal source with a flag.
+	  Use running_time instead of the more useless timestamp.
+	  Validate a source when a valid SDES has been received.
+	  Pass the current system time when processing SR packets.
+	  * gst/rtpmanager/rtpsource.c: (rtp_source_class_init),
+	  (rtp_source_init), (rtp_source_create_stats),
+	  (rtp_source_get_property), (rtp_source_send_rtp),
+	  (rtp_source_process_rb), (rtp_source_get_new_rb),
+	  (rtp_source_get_last_rb):
+	  * gst/rtpmanager/rtpsource.h:
+	  Add property to get source stats.
+	  Mark params as STATIC_STRINGS.
+	  Calculate the bitrate at the sender SSRC.
+	  Avoid negative values in the round trip time calculations.
+	  * gst/rtpmanager/rtpstats.h:
+	  Update some docs and change some variable name to more closely reflect
+	  what it contains.
+
+2008-11-20 08:19:15 +0000  Sebastian Dröge <slomo@circular-chaos.org>
+
+	  gst/rtpmanager/gstrtpjitterbuffer.c: Initialize return value to fix compiler warning about uninitialized variable.
+	  Original commit message from CVS:
+	  * gst/rtpmanager/gstrtpjitterbuffer.c:
+	  (gst_rtp_jitter_buffer_chain_rtcp):
+	  Initialize return value to fix compiler warning about uninitialized
+	  variable.
+
+2008-11-19 16:48:38 +0000  Wim Taymans <wim.taymans@gmail.com>
+
+	  gst/rtpmanager/gstrtpjitterbuffer.c: Mark signal arg as static scope.
+	  Original commit message from CVS:
+	  * gst/rtpmanager/gstrtpjitterbuffer.c:
+	  (gst_rtp_jitter_buffer_class_init):
+	  Mark signal arg as static scope.
+
+2008-11-19 09:06:29 +0000  Wim Taymans <wim.taymans@gmail.com>
+
+	  gst/rtpmanager/gstrtpbin.c: Remove internal sync pad, use signals instead to get lip-sync notifications.
+	  Original commit message from CVS:
+	  * gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_associate),
+	  (gst_rtp_bin_handle_sync), (create_stream), (free_stream),
+	  (new_ssrc_pad_found):
+	  Remove internal sync pad, use signals instead to get lip-sync
+	  notifications.
+	  * gst/rtpmanager/gstrtpjitterbuffer.c:
+	  (gst_rtp_jitter_buffer_base_init),
+	  (gst_rtp_jitter_buffer_class_init),
+	  (gst_rtp_jitter_buffer_internal_links), (create_rtcp_sink),
+	  (remove_rtcp_sink), (gst_rtp_jitter_buffer_request_new_pad),
+	  (gst_rtp_jitter_buffer_release_pad),
+	  (gst_rtp_jitter_buffer_sink_rtcp_event),
+	  (gst_rtp_jitter_buffer_chain_rtcp),
+	  (gst_rtp_jitter_buffer_get_property):
+	  * gst/rtpmanager/gstrtpjitterbuffer.h:
+	  Make it possible to send SR packets to the jitterbuffer.
+	  Check if the SR timestamps are valid by comparing them to the RTP
+	  timestamps.
+	  Signal the SR packet and the timing information to listeners.
+	  * gst/rtpmanager/gstrtpssrcdemux.c: (create_demux_pad_for_ssrc),
+	  (gst_rtp_ssrc_demux_rtcp_chain), (gst_rtp_ssrc_demux_src_query):
+	  Remove some unused code.
+	  * gst/rtpmanager/rtpjitterbuffer.c: (rtp_jitter_buffer_reset_skew),
+	  (calculate_skew), (rtp_jitter_buffer_get_sync):
+	  * gst/rtpmanager/rtpjitterbuffer.h:
+	  Keep track of the last seen RTP timestamp so that we can filter out
+	  invalid SR packets.
+
+2008-11-17 19:47:32 +0000  Sebastian Dröge <slomo@circular-chaos.org>
+
+	  gst/rtpmanager/rtpsource.c: Fix GST_DEBUG call to only have as many arguments as required by the format string. Fixes...
+	  Original commit message from CVS:
+	  * gst/rtpmanager/rtpsource.c: (get_clock_rate):
+	  Fix GST_DEBUG call to only have as many arguments as required
+	  by the format string. Fixes a compiler warning.
+
+2008-11-17 15:17:52 +0000  Wim Taymans <wim.taymans@gmail.com>
+
+	  gst/rtpmanager/gstrtpbin.c: Do not try to keep track of the clock-rate ourselves but simply get the value from the ji...
+	  Original commit message from CVS:
+	  * gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_associate),
+	  (gst_rtp_bin_sync_chain), (create_stream), (new_ssrc_pad_found):
+	  Do not try to keep track of the clock-rate ourselves but simply get the
+	  value from the jitterbuffer.
+	  * gst/rtpmanager/gstrtpjitterbuffer.c:
+	  (gst_jitter_buffer_sink_parse_caps), (gst_rtp_jitter_buffer_chain),
+	  (gst_rtp_jitter_buffer_get_sync):
+	  * gst/rtpmanager/gstrtpjitterbuffer.h:
+	  Add some debug info.
+	  Pass the clock-rate to the jitterbuffer.
+	  Also pass the clock-rate along with the rtp timestamp when getting the
+	  sync parameters.
+	  * gst/rtpmanager/gstrtpptdemux.c: (gst_rtp_pt_demux_chain):
+	  Fix some debug.
+	  * gst/rtpmanager/rtpjitterbuffer.c: (rtp_jitter_buffer_reset_skew),
+	  (calculate_skew), (rtp_jitter_buffer_get_sync):
+	  * gst/rtpmanager/rtpjitterbuffer.h:
+	  Keep track of clock-rate changes and return the clock-rate together with
+	  the rtp timestamps used for sync.
+	  Don't try to construct timestamps when we have no base_time.
+	  * gst/rtpmanager/rtpsource.c: (get_clock_rate):
+	  Request a new clock-rate when the payload type changes.
+	  Reset the jitter calculation when the clock-rate changes.
+
+2008-11-13 15:48:54 +0000  Wim Taymans <wim.taymans@gmail.com>
+
+	  gst/rtpmanager/: Small cleanups and some more debug info.
+	  Original commit message from CVS:
+	  * gst/rtpmanager/gstrtpjitterbuffer.c:
+	  (gst_jitter_buffer_sink_parse_caps),
+	  (gst_rtp_jitter_buffer_flush_stop), (gst_rtp_jitter_buffer_chain):
+	  * gst/rtpmanager/rtpjitterbuffer.c: (rtp_jitter_buffer_reset_skew),
+	  (calculate_skew):
+	  Small cleanups and some more debug info.
+
+2008-11-10 15:26:40 +0000  Wim Taymans <wim.taymans@gmail.com>
+
+	  gst/rtpmanager/gstrtpjitterbuffer.c: Also configure the next expected output seqnum when we get a seqnum-base on the ...
+	  Original commit message from CVS:
+	  * gst/rtpmanager/gstrtpjitterbuffer.c:
+	  (gst_jitter_buffer_sink_parse_caps), (gst_rtp_jitter_buffer_chain):
+	  Also configure the next expected output seqnum when we get a seqnum-base
+	  on the caps.
+
+2008-11-04 12:42:30 +0000  Stefan Kost <ensonic@users.sourceforge.net>
+
+	  Don't install static libs for plugins. Fixes #550851 for -bad.
+	  Original commit message from CVS:
+	  * ext/alsaspdif/Makefile.am:
+	  * ext/amrwb/Makefile.am:
+	  * ext/apexsink/Makefile.am:
+	  * ext/arts/Makefile.am:
+	  * ext/artsd/Makefile.am:
+	  * ext/audiofile/Makefile.am:
+	  * ext/audioresample/Makefile.am:
+	  * ext/bz2/Makefile.am:
+	  * ext/cdaudio/Makefile.am:
+	  * ext/celt/Makefile.am:
+	  * ext/dc1394/Makefile.am:
+	  * ext/dirac/Makefile.am:
+	  * ext/directfb/Makefile.am:
+	  * ext/divx/Makefile.am:
+	  * ext/dts/Makefile.am:
+	  * ext/faac/Makefile.am:
+	  * ext/faad/Makefile.am:
+	  * ext/gsm/Makefile.am:
+	  * ext/hermes/Makefile.am:
+	  * ext/ivorbis/Makefile.am:
+	  * ext/jack/Makefile.am:
+	  * ext/jp2k/Makefile.am:
+	  * ext/ladspa/Makefile.am:
+	  * ext/lcs/Makefile.am:
+	  * ext/libfame/Makefile.am:
+	  * ext/libmms/Makefile.am:
+	  * ext/metadata/Makefile.am:
+	  * ext/mpeg2enc/Makefile.am:
+	  * ext/mplex/Makefile.am:
+	  * ext/musepack/Makefile.am:
+	  * ext/musicbrainz/Makefile.am:
+	  * ext/mythtv/Makefile.am:
+	  * ext/nas/Makefile.am:
+	  * ext/neon/Makefile.am:
+	  * ext/ofa/Makefile.am:
+	  * ext/polyp/Makefile.am:
+	  * ext/resindvd/Makefile.am:
+	  * ext/sdl/Makefile.am:
+	  * ext/shout/Makefile.am:
+	  * ext/snapshot/Makefile.am:
+	  * ext/sndfile/Makefile.am:
+	  * ext/soundtouch/Makefile.am:
+	  * ext/spc/Makefile.am:
+	  * ext/swfdec/Makefile.am:
+	  * ext/tarkin/Makefile.am:
+	  * ext/theora/Makefile.am:
+	  * ext/timidity/Makefile.am:
+	  * ext/twolame/Makefile.am:
+	  * ext/x264/Makefile.am:
+	  * ext/xine/Makefile.am:
+	  * ext/xvid/Makefile.am:
+	  * gst-libs/gst/app/Makefile.am:
+	  * gst-libs/gst/dshow/Makefile.am:
+	  * gst/aiffparse/Makefile.am:
+	  * gst/app/Makefile.am:
+	  * gst/audiobuffer/Makefile.am:
+	  * gst/bayer/Makefile.am:
+	  * gst/cdxaparse/Makefile.am:
+	  * gst/chart/Makefile.am:
+	  * gst/colorspace/Makefile.am:
+	  * gst/dccp/Makefile.am:
+	  * gst/deinterlace/Makefile.am:
+	  * gst/deinterlace2/Makefile.am:
+	  * gst/dvdspu/Makefile.am:
+	  * gst/festival/Makefile.am:
+	  * gst/filter/Makefile.am:
+	  * gst/flacparse/Makefile.am:
+	  * gst/flv/Makefile.am:
+	  * gst/games/Makefile.am:
+	  * gst/h264parse/Makefile.am:
+	  * gst/librfb/Makefile.am:
+	  * gst/mixmatrix/Makefile.am:
+	  * gst/modplug/Makefile.am:
+	  * gst/mpeg1sys/Makefile.am:
+	  * gst/mpeg4videoparse/Makefile.am:
+	  * gst/mpegdemux/Makefile.am:
+	  * gst/mpegtsmux/Makefile.am:
+	  * gst/mpegvideoparse/Makefile.am:
+	  * gst/mve/Makefile.am:
+	  * gst/nsf/Makefile.am:
+	  * gst/nuvdemux/Makefile.am:
+	  * gst/overlay/Makefile.am:
+	  * gst/passthrough/Makefile.am:
+	  * gst/pcapparse/Makefile.am:
+	  * gst/playondemand/Makefile.am:
+	  * gst/rawparse/Makefile.am:
+	  * gst/real/Makefile.am:
+	  * gst/rtjpeg/Makefile.am:
+	  * gst/rtpmanager/Makefile.am:
+	  * gst/scaletempo/Makefile.am:
+	  * gst/sdp/Makefile.am:
+	  * gst/selector/Makefile.am:
+	  * gst/smooth/Makefile.am:
+	  * gst/smoothwave/Makefile.am:
+	  * gst/speed/Makefile.am:
+	  * gst/speexresample/Makefile.am:
+	  * gst/stereo/Makefile.am:
+	  * gst/subenc/Makefile.am:
+	  * gst/tta/Makefile.am:
+	  * gst/vbidec/Makefile.am:
+	  * gst/videodrop/Makefile.am:
+	  * gst/videosignal/Makefile.am:
+	  * gst/virtualdub/Makefile.am:
+	  * gst/vmnc/Makefile.am:
+	  * gst/y4m/Makefile.am:
+	  * sys/acmenc/Makefile.am:
+	  * sys/cdrom/Makefile.am:
+	  * sys/dshowdecwrapper/Makefile.am:
+	  * sys/dshowsrcwrapper/Makefile.am:
+	  * sys/dvb/Makefile.am:
+	  * sys/dxr3/Makefile.am:
+	  * sys/fbdev/Makefile.am:
+	  * sys/oss4/Makefile.am:
+	  * sys/qcam/Makefile.am:
+	  * sys/qtwrapper/Makefile.am:
+	  * sys/vcd/Makefile.am:
+	  * sys/wininet/Makefile.am:
+	  * win32/common/config.h:
+	  Don't install static libs for plugins. Fixes #550851 for -bad.
+
+2008-10-16 13:05:37 +0000  Wim Taymans <wim.taymans@gmail.com>
+
+	  gst/rtpmanager/gstrtpjitterbuffer.c: Fix problem with using the output seqnum counter to check for input seqnum disco...
+	  Original commit message from CVS:
+	  * gst/rtpmanager/gstrtpjitterbuffer.c:
+	  (gst_jitter_buffer_sink_parse_caps),
+	  (gst_rtp_jitter_buffer_flush_start),
+	  (gst_rtp_jitter_buffer_flush_stop), (gst_rtp_jitter_buffer_chain),
+	  (gst_rtp_jitter_buffer_loop):
+	  Fix problem with using the output seqnum counter to check for input
+	  seqnum discontinuities.
+	  Improve gap detection and recovery, reset and flush the jitterbuffer on
+	  seqnum restart. Fixes #556520.
+	  * gst/rtpmanager/rtpjitterbuffer.c: (rtp_jitter_buffer_insert):
+	  Fix wrong G_LIKELY.
+
+2008-10-16 09:51:28 +0000  Wim Taymans <wim.taymans@gmail.com>
+
+	  gst/rtpmanager/gstrtpsession.c: Install event handler on the rtcp_src pad, make LATENCY event return
+	  Original commit message from CVS:
+	  * gst/rtpmanager/gstrtpsession.c:
+	  (gst_rtp_session_event_send_rtcp_src), (create_send_rtcp_src):
+	  Install event handler on the rtcp_src pad, make LATENCY event return
+	  TRUE.
+
+2008-10-07 18:54:41 +0000  Håvard Graff <havard.graff@tandberg.com>
+
+	  gst/rtpmanager/gstrtpbin-marshal.list: Add marshaller for new action signal.
+	  Original commit message from CVS:
+	  Patch by: Håvard Graff <havard dot graff at tandberg dot com>
+	  * gst/rtpmanager/gstrtpbin-marshal.list:
+	  Add marshaller for new action signal.
+	  * gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_get_internal_session),
+	  (gst_rtp_bin_class_init):
+	  * gst/rtpmanager/gstrtpbin.h:
+	  Add action signal to retrieve the internal RTPSession object.
+	  * gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_class_init),
+	  (gst_rtp_session_get_property), (gst_rtp_session_release_pad):
+	  Add property to access the internal RTPSession object.
+	  * gst/rtpmanager/rtpsession.c: (rtp_session_class_init),
+	  (check_collision):
+	  * gst/rtpmanager/rtpsession.h:
+	  Add action signal to retrieve an RTPSource object by SSRC.
+	  See #555396.
+
+2008-10-07 11:33:10 +0000  Wim Taymans <wim.taymans@gmail.com>
+
+	  gst/rtpmanager/gstrtpbin.c: Release pads of the session manager.
+	  Original commit message from CVS:
+	  * gst/rtpmanager/gstrtpbin.c: (find_session_by_pad),
+	  (free_session), (gst_rtp_bin_dispose), (remove_recv_rtp),
+	  (remove_recv_rtcp), (remove_send_rtp), (remove_rtcp),
+	  (gst_rtp_bin_release_pad):
+	  Release pads of the session manager.
+	  Start implementing releasing pads of gstrtpbin.
+	  * gst/rtpmanager/gstrtpsession.c: (remove_recv_rtp_sink),
+	  (remove_recv_rtcp_sink), (remove_send_rtp_sink),
+	  (remove_send_rtcp_src), (gst_rtp_session_release_pad):
+	  Implement releasing pads in gstrtpsession.
+
+2008-10-07 10:02:20 +0000  Wim Taymans <wim.taymans@gmail.com>
+
+	  gst/rtpmanager/gstrtpjitterbuffer.c: Only update the seqnum-base when it was not already configured for the streams.
+	  Original commit message from CVS:
+	  * gst/rtpmanager/gstrtpjitterbuffer.c:
+	  (gst_jitter_buffer_sink_parse_caps):
+	  Only update the seqnum-base when it was not already configured for the
+	  streams.
+
+2008-09-30 15:08:52 +0000  Wim Taymans <wim.taymans@gmail.com>
+
+	  gst/rtpmanager/rtpsession.c: Ref the rtpsource object before we release the session lock when we emit the signals.
+	  Original commit message from CVS:
+	  * gst/rtpmanager/rtpsession.c: (on_new_ssrc), (on_ssrc_collision),
+	  (on_ssrc_validated), (on_ssrc_active), (on_ssrc_sdes),
+	  (on_bye_ssrc), (on_bye_timeout), (on_timeout), (on_sender_timeout):
+	  Ref the rtpsource object before we release the session lock when we emit
+	  the signals.
+
+2008-09-23 18:13:31 +0000  Wim Taymans <wim.taymans@gmail.com>
+
+	  gst/rtpmanager/: Fix some docs.
+	  Original commit message from CVS:
+	  * gst/rtpmanager/rtpjitterbuffer.c: (rtp_jitter_buffer_insert),
+	  (rtp_jitter_buffer_get_sync):
+	  * gst/rtpmanager/rtpsession.c: (on_sender_timeout),
+	  (session_cleanup):
+	  * gst/rtpmanager/rtpsource.c:
+	  Fix some docs.
+
+2008-09-17 13:59:21 +0000  Jan Schmidt <thaytan@mad.scientist.com>
+
+	  Fix compiler warnings on OS/X
+	  Original commit message from CVS:
+	  * ext/jack/gstjackaudiosink.c: (jack_process_cb):
+	  * gst/rtpmanager/rtpjitterbuffer.c: (calculate_skew):
+	  Fix compiler warnings on OS/X
+
+2008-09-13 01:37:50 +0000  Wim Taymans <wim.taymans@gmail.com>
+
+	  gst/rtpmanager/gstrtpbin.c: Do not try to adjust the offset of streams for which we have not yet seen an SR packet. A...
+	  Original commit message from CVS:
+	  * gst/rtpmanager/gstrtpbin.c: (create_session),
+	  (gst_rtp_bin_associate), (gst_rtp_bin_sync_chain):
+	  Do not try to adjust the offset of streams for which we have not yet
+	  seen an SR packet. Avoids large ts-offsets in some cases.
+
+2008-09-05 13:52:34 +0000  Wim Taymans <wim.taymans@gmail.com>
+
+	  gst/rtpmanager/gstrtpbin.*: Add signal to notify listeners when a sender becomes a receiver.
+	  Original commit message from CVS:
+	  * gst/rtpmanager/gstrtpbin.c: (on_sender_timeout),
+	  (create_session), (gst_rtp_bin_associate),
+	  (gst_rtp_bin_sync_chain), (gst_rtp_bin_class_init),
+	  (gst_rtp_bin_request_new_pad):
+	  * gst/rtpmanager/gstrtpbin.h:
+	  Add signal to notify listeners when a sender becomes a receiver.
+	  Tweak lip-sync code, don't store our own copy of the ts-offset of the
+	  jitterbuffer, don't adjust sync if the change is less than 4msec.
+	  Get the RTP timestamp <-> GStreamer timestamp relation directly from
+	  the jitterbuffer instead of our inaccurate version from the source.
+	  * gst/rtpmanager/gstrtpjitterbuffer.c:
+	  (gst_rtp_jitter_buffer_chain), (gst_rtp_jitter_buffer_loop),
+	  (gst_rtp_jitter_buffer_get_sync):
+	  * gst/rtpmanager/gstrtpjitterbuffer.h:
+	  Add G_LIKELY macros, use global defines for max packet reorder and
+	  dropouts.
+	  Reset the jitterbuffer clock skew detection when packets seqnums are
+	  changed unexpectedly.
+	  * gst/rtpmanager/gstrtpsession.c: (on_sender_timeout),
+	  (gst_rtp_session_class_init), (gst_rtp_session_init):
+	  * gst/rtpmanager/gstrtpsession.h:
+	  Add sender timeout signal.
+	  * gst/rtpmanager/rtpjitterbuffer.c: (rtp_jitter_buffer_reset_skew),
+	  (calculate_skew), (rtp_jitter_buffer_insert),
+	  (rtp_jitter_buffer_get_sync):
+	  * gst/rtpmanager/rtpjitterbuffer.h:
+	  Add some G_LIKELY macros.
+	  Keep track of the extended RTP timestamp so that we can report the RTP
+	  timestamp <-> GStreamer timestamp relation for lip-sync.
+	  Remove server timestamp gap detection code, the server can sometimes
+	  make a huge gap in timestamps (talk spurts,...) see #549774.
+	  Detect timetamp weirdness instead by observing the sender/receiver
+	  timestamp relation and resync if it changes more than 1 second.
+	  Add method to report about the current rtp <-> gst timestamp relation
+	  which is needed for lip-sync.
+	  * gst/rtpmanager/rtpsession.c: (rtp_session_class_init),
+	  (on_sender_timeout), (check_collision), (rtp_session_process_sr),
+	  (session_cleanup):
+	  * gst/rtpmanager/rtpsession.h:
+	  Add sender timeout signal.
+	  Remove inaccurate rtp <-> gst timestamp relation code, the
+	  jitterbuffer can now do an accurate reporting about this.
+	  * gst/rtpmanager/rtpsource.c: (rtp_source_init),
+	  (rtp_source_update_caps), (calculate_jitter),
+	  (rtp_source_process_rtp):
+	  * gst/rtpmanager/rtpsource.h:
+	  Remove inaccurate rtp <-> gst timestamp relation code.
+	  * gst/rtpmanager/rtpstats.h:
+	  Define global max-reorder and max-dropout constants for use in various
+	  subsystems.
+
+2008-08-28 15:21:45 +0000  Wim Taymans <wim.taymans@gmail.com>
+
+	  gst/rtpmanager/gstrtpsession.c: Send EOS when the session object instructs us to.
+	  Original commit message from CVS:
+	  * gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_send_rtcp),
+	  (gst_rtp_session_event_send_rtp_sink):
+	  Send EOS when the session object instructs us to.
+	  * gst/rtpmanager/rtpsession.c: (rtp_session_on_timeout):
+	  * gst/rtpmanager/rtpsession.h:
+	  Make it possible for the session manager to instruct us to send EOS. We
+	  currently will EOS when the session is a sender and when the sender part
+	  goes EOS. This is not entirely correct behaviour because the session
+	  could still participate as a receiver.
+	  Fixes #549409.
+
+2008-08-13 14:31:02 +0000  Wim Taymans <wim.taymans@gmail.com>
+
+	  gst/rtpmanager/gstrtpbin.c: Reset rtp timestamp interpollation when we detect a gap when the clock_base changed.
+	  Original commit message from CVS:
+	  * gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_associate),
+	  (gst_rtp_bin_sync_chain), (new_ssrc_pad_found):
+	  Reset rtp timestamp interpollation when we detect a gap when the
+	  clock_base changed.
+	  Don't try to adjust the ts-offset when it's too big (> 3seconds)
+	  * gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_set_ssrc):
+	  * gst/rtpmanager/gstrtpsession.h:
+	  Add method to set session SSRC.
+	  * gst/rtpmanager/rtpsession.c: (check_collision),
+	  (rtp_session_set_internal_ssrc), (rtp_session_get_internal_ssrc),
+	  (rtp_session_on_timeout):
+	  * gst/rtpmanager/rtpsession.h:
+	  Added debugging for the collision checks.
+	  Add method to change the internal SSRC of the session.
+	  * gst/rtpmanager/rtpsource.c: (rtp_source_process_rtp):
+	  Reset the clock base when we detect large jumps in the seqnums.
+
+2008-08-11 07:20:15 +0000  Stefan Kost <ensonic@users.sourceforge.net>
+
+	  gst/rtpmanager/gstrtpbin.c: Print the pad-name in debug log.
+	  Original commit message from CVS:
+	  * gst/rtpmanager/gstrtpbin.c:
+	  Print the pad-name in debug log.
+	  * sys/dshowsrcwrapper/gstdshowaudiosrc.c:
+	  * sys/dshowsrcwrapper/gstdshowvideosrc.c:
+	  Use "-" instead of "_" in property names. Can we call them just
+	  "device" like everywhere else?
+
+2008-08-05 09:42:53 +0000  Olivier Crete <tester@tester.ca>
+
+	  gst/rtpmanager/gstrtpjitterbuffer.c: Make the buffer metadata writable before inserting it in the jitterbuffer becaus...
+	  Original commit message from CVS:
+	  Based on patch by: Olivier Crete <tester at tester dot ca>
+	  * gst/rtpmanager/gstrtpjitterbuffer.c:
+	  (gst_rtp_jitter_buffer_chain), (gst_rtp_jitter_buffer_loop):
+	  Make the buffer metadata writable before inserting it in the
+	  jitterbuffer because the jitterbuffer will modify the timestamps.
+	  * gst/rtpmanager/rtpjitterbuffer.c:
+	  Update method comment about requiring writable metadata on buffers.
+	  * gst/rtpmanager/rtpsession.c: (rtp_session_process_sr),
+	  (rtp_session_process_rtcp):
+	  Make the RTCP buffer metadata writable because we want to modify the
+	  metadata.
+	  Fixes #546312.
+
+2008-08-05 09:00:50 +0000  Håvard Graff <havard.graff@tandberg.com>
+
+	  gst/rtpmanager/gstrtpjitterbuffer.c: Fix debug by logging the right seqnum.
+	  Original commit message from CVS:
+	  Patch by: Håvard Graff <havard dot graff at tandberg dot com>
+	  * gst/rtpmanager/gstrtpjitterbuffer.c:
+	  (gst_rtp_jitter_buffer_chain):
+	  Fix debug by logging the right seqnum.
+
+2008-08-05 08:58:27 +0000  Olivier Crete <tester@tester.ca>
+
+	  gst/rtpmanager/gstrtpbin.c: Release lock before emitting the request-pt-map signal.
+	  Original commit message from CVS:
+	  Patch by: Olivier Crete <tester at tester dot ca>
+	  * gst/rtpmanager/gstrtpbin.c: (get_pt_map):
+	  Release lock before emitting the request-pt-map signal.
+	  Fixes #543480.
+
+2008-07-03 14:44:51 +0000  Peter Kjellerstedt <pkj@axis.com>
+
+	  gst/rtpmanager/: Corrected a typo (interpollate -> interpolate).
+	  Original commit message from CVS:
+	  * ChangeLog:
+	  * gst/rtpmanager/gstrtpjitterbuffer.c: (gst_rtp_jitter_buffer_loop):
+	  * gst/rtpmanager/rtpsource.c: (rtp_source_get_new_sr):
+	  Corrected a typo (interpollate -> interpolate).
+
+2008-07-03 14:31:10 +0000  Peter Kjellerstedt <pkj@axis.com>
+
+	  gst/rtpmanager/: Changed some GST_DEBUG() to GST_LOG() to reduce the spam when a pipeline is running normally.
+	  Original commit message from CVS:
+	  * gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_process_rtp),
+	  (gst_rtp_session_send_rtp), (gst_rtp_session_send_rtcp),
+	  (gst_rtp_session_sync_rtcp), (gst_rtp_session_chain_recv_rtp),
+	  (gst_rtp_session_chain_recv_rtcp), (gst_rtp_session_chain_send_rtp):
+	  * gst/rtpmanager/rtpsession.c: (source_push_rtp),
+	  (rtp_session_send_rtp):
+	  * gst/rtpmanager/rtpsource.c: (push_packet), (calculate_jitter),
+	  (rtp_source_process_rtp), (rtp_source_send_rtp):
+	  Changed some GST_DEBUG() to GST_LOG() to reduce the spam when a
+	  pipeline is running normally.
+
+2008-07-03 13:47:19 +0000  Peter Kjellerstedt <pkj@axis.com>
+
+	  gst/rtpmanager/: Do not mix the use of g_get_current_time() with gst_clock_get_time().
+	  Original commit message from CVS:
+	  * gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_init),
+	  (gst_rtp_session_finalize), (rtcp_thread),
+	  (gst_rtp_session_chain_recv_rtp), (gst_rtp_session_chain_recv_rtcp),
+	  (gst_rtp_session_event_send_rtp_sink),
+	  (gst_rtp_session_chain_send_rtp):
+	  * gst/rtpmanager/rtpsession.c: (check_collision),
+	  (update_arrival_stats), (rtp_session_process_rtp),
+	  (rtp_session_process_rtcp), (rtp_session_send_rtp),
+	  (rtp_session_send_bye_locked), (rtp_session_send_bye),
+	  (rtp_session_next_timeout), (session_report_blocks), (session_cleanup),
+	  (is_rtcp_time), (rtp_session_on_timeout):
+	  * gst/rtpmanager/rtpsession.h:
+	  Do not mix the use of g_get_current_time() with gst_clock_get_time().
+
+2008-06-16 07:30:34 +0000  Stefan Kost <ensonic@users.sourceforge.net>
+
+	  Final round of doc updates.
+	  Original commit message from CVS:
+	  * gst/rtpmanager/gstrtpjitterbuffer.c:
+	  * gst/speed/gstspeed.c:
+	  * gst/speexresample/gstspeexresample.c:
+	  * gst/videosignal/gstvideoanalyse.c:
+	  * gst/videosignal/gstvideodetect.c:
+	  * gst/videosignal/gstvideomark.c:
+	  * sys/dvb/gstdvbsrc.c:
+	  * sys/oss4/oss4-mixer.c:
+	  * sys/oss4/oss4-sink.c:
+	  * sys/oss4/oss4-source.c:
+	  * sys/wininet/gstwininetsrc.c:
+	  Final round of doc updates.
+
+2008-06-16 07:03:58 +0000  Stefan Kost <ensonic@users.sourceforge.net>
+
+	  gst/: More doc updates. More xrefs.
+	  Original commit message from CVS:
+	  * gst/deinterlace/gstdeinterlace.c:
+	  * gst/rtpmanager/gstrtpbin.c:
+	  * gst/rtpmanager/gstrtpclient.c:
+	  * gst/rtpmanager/gstrtpjitterbuffer.c:
+	  * gst/rtpmanager/gstrtpptdemux.c:
+	  * gst/rtpmanager/gstrtpsession.c:
+	  * gst/rtpmanager/gstrtpssrcdemux.c:
+	  * gst/sdp/gstsdpdemux.c:
+	  More doc updates. More xrefs.
+
+2008-06-12 14:49:18 +0000  Stefan Kost <ensonic@users.sourceforge.net>
+
+	  Do not use short_description in section docs for elements. We extract them from element details and there will be war...
+	  Original commit message from CVS:
+	  * ext/dc1394/gstdc1394.c:
+	  * ext/ivorbis/vorbisdec.c:
+	  * ext/jack/gstjackaudiosink.c:
+	  * ext/metadata/gstmetadatademux.c:
+	  * ext/mythtv/gstmythtvsrc.c:
+	  * ext/theora/theoradec.c:
+	  * gst-libs/gst/app/gstappsink.c:
+	  * gst/bayer/gstbayer2rgb.c:
+	  * gst/deinterlace/gstdeinterlace.c:
+	  * gst/rawparse/gstaudioparse.c:
+	  * gst/rawparse/gstvideoparse.c:
+	  * gst/rtpmanager/gstrtpbin.c:
+	  * gst/rtpmanager/gstrtpclient.c:
+	  * gst/rtpmanager/gstrtpjitterbuffer.c:
+	  * gst/rtpmanager/gstrtpptdemux.c:
+	  * gst/rtpmanager/gstrtpsession.c:
+	  * gst/rtpmanager/gstrtpssrcdemux.c:
+	  * gst/selector/gstinputselector.c:
+	  * gst/selector/gstoutputselector.c:
+	  * gst/videosignal/gstvideoanalyse.c:
+	  * gst/videosignal/gstvideodetect.c:
+	  * gst/videosignal/gstvideomark.c:
+	  * sys/oss4/oss4-mixer.c:
+	  * sys/oss4/oss4-sink.c:
+	  * sys/oss4/oss4-source.c:
+	  Do not use short_description in section docs for elements. We extract
+	  them from element details and there will be warnings if they differ.
+	  Also fixing up the ChangeLog order.
+
+2008-06-06 13:01:05 +0000  Wim Taymans <wim.taymans@gmail.com>
+
+	  gst/rtpmanager/gstrtpbin.c: Fix deadlock when shutting down, use a new lock instead to properly shutdown.
+	  Original commit message from CVS:
+	  * gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_init),
+	  (gst_rtp_bin_finalize), (gst_rtp_bin_change_state):
+	  Fix deadlock when shutting down, use a new lock instead to properly
+	  shutdown.
+
+2008-05-27 16:48:10 +0000  Wim Taymans <wim.taymans@gmail.com>
+
+	  gst/rtpmanager/gstrtpbin.c: Break out of callbacks when we are shutting down.
+	  Original commit message from CVS:
+	  * gst/rtpmanager/gstrtpbin.c:
+	  (gst_rtp_bin_propagate_property_to_jitterbuffer),
+	  (gst_rtp_bin_change_state), (new_payload_found),
+	  (new_ssrc_pad_found):
+	  Break out of callbacks when we are shutting down.
+	  Make sure no state changes can happen when we reconfigure.
+
+2008-05-26 10:09:29 +0000  Wim Taymans <wim.taymans@gmail.com>
+
+	  gst/rtpmanager/gstrtpjitterbuffer.c: When checking the seqnum, reset the jitterbuffer if the gap is too big, we need ...
+	  Original commit message from CVS:
+	  * gst/rtpmanager/gstrtpjitterbuffer.c:
+	  (gst_rtp_jitter_buffer_chain), (gst_rtp_jitter_buffer_loop):
+	  When checking the seqnum, reset the jitterbuffer if the gap is too big,
+	  we need to do this so that we can better handle a restarted source.
+	  Fix some comments.
+	  * gst/rtpmanager/rtpjitterbuffer.c: (calculate_skew),
+	  (rtp_jitter_buffer_insert):
+	  Tweak the skew resync diff.
+	  Use our working seqnum compare function in -base.
+	  Rework the jitterbuffer insert code to make it clearer and more
+	  performant by only retrieving the seqnum of the input buffer once and by
+	  adding some G_LIKELY compiler hints.
+	  Improve debugging for duplicate packets.
+	  * gst/rtpmanager/rtpsource.c: (rtp_source_process_rtp):
+	  Fix a comment, we don't do skew correction here..
+
+2008-05-26 10:00:24 +0000  Håvard Graff <havard.graff@tandberg.com>
+
+	  gst/rtpmanager/gstrtpbin.c: Propagate the do-lost and latency properties to the jitterbuffers when they are changed o...
+	  Original commit message from CVS:
+	  Patch by: Håvard Graff <havard dot graff at tandberg dot com>
+	  * gst/rtpmanager/gstrtpbin.c:
+	  (gst_rtp_bin_propagate_property_to_jitterbuffer),
+	  (gst_rtp_bin_set_property):
+	  Propagate the do-lost and latency properties to the jitterbuffers when
+	  they are changed on rtpbin.
+
+2008-05-26 09:57:40 +0000  Wim Taymans <wim.taymans@gmail.com>
+
+	  Don't use _gst_pad().
+	  Original commit message from CVS:
+	  * examples/switch/switcher.c: (switch_timer):
+	  * gst/replaygain/gstrgvolume.c: (gst_rg_volume_init):
+	  * gst/rtpmanager/gstrtpclient.c: (create_stream):
+	  * gst/sdp/gstsdpdemux.c: (gst_sdp_demux_stream_configure_udp),
+	  (gst_sdp_demux_stream_configure_udp_sink):
+	  * tests/check/elements/deinterleave.c: (GST_START_TEST),
+	  (pad_added_setup_data_check_float32_8ch_cb):
+	  * tests/check/elements/rganalysis.c: (send_eos_event),
+	  (send_tag_event):
+	  Don't use _gst_pad().
+
+2008-05-16 19:56:30 +0000  Jan Schmidt <thaytan@mad.scientist.com>
+
+	  docs/Makefile.am: Don't attempt to build plugin docs when they're disabled.
+	  Original commit message from CVS:
+	  * docs/Makefile.am:
+	  Don't attempt to build plugin docs when they're disabled.
+	  * gst/bayer/Makefile.am:
+	  Add libgstvideo to the link.
+	  * gst/rtpmanager/Makefile.am:
+	  Fix link order, and move LIBS things to _LIBS
+
+2008-05-14 21:02:19 +0000  Wim Taymans <wim.taymans@gmail.com>
+
+	  gst/rtpmanager/gstrtpjitterbuffer.c: Simply drop bad RTP packets with a warning instead of just posting an error and ...
+	  Original commit message from CVS:
+	  * gst/rtpmanager/gstrtpjitterbuffer.c:
+	  (gst_rtp_jitter_buffer_chain):
+	  Simply drop bad RTP packets with a warning instead of just posting an
+	  error and stopping. This is a perfectly recoverable event and we don't
+	  force people to use an rtpbin to filter out bad packets first.
+
+2008-05-13 09:06:51 +0000  Wim Taymans <wim.taymans@gmail.com>
+
+	  gst/rtpmanager/gstrtpbin.c: Actually add the do-lost property to the object.
+	  Original commit message from CVS:
+	  * gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_class_init):
+	  Actually add the do-lost property to the object.
+
+2008-05-12 18:43:41 +0000  Wim Taymans <wim.taymans@gmail.com>
+
+	  gst/rtpmanager/gstrtpjitterbuffer.c: Avoid waiting for a negative (huge) duration when the last packet has a lower ti...
+	  Original commit message from CVS:
+	  * gst/rtpmanager/gstrtpjitterbuffer.c:
+	  (gst_rtp_jitter_buffer_loop):
+	  Avoid waiting for a negative (huge) duration when the last packet has a
+	  lower timestamp than the current packet.
+
+2008-05-12 14:28:09 +0000  Peter Kjellerstedt <pkj@axis.com>
+
+	  gst/rtpmanager/gstrtpsession.c: Make sure to unref the rtpsession returned by gst_pad_get_parent() to prevent a memor...
+	  Original commit message from CVS:
+	  * gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_query_send_rtcp_src):
+	  Make sure to unref the rtpsession returned by gst_pad_get_parent() to
+	  prevent a memory leak.
+
+2008-05-12 14:12:08 +0000  Jan Schmidt <thaytan@mad.scientist.com>
+
+	  gst/rtpmanager/gstrtpjitterbuffer.c: Initialise with GST_CLOCK_TIME_NONE to avoid compiler warning.
+	  Original commit message from CVS:
+	  * gst/rtpmanager/gstrtpjitterbuffer.c:
+	  (gst_rtp_jitter_buffer_loop):
+	  Initialise with GST_CLOCK_TIME_NONE to avoid compiler warning.
+
+2008-05-09 07:41:58 +0000  Peter Kjellerstedt <pkj@axis.com>
+
+	  gst/rtpmanager/rtpsource.c: Make sure to unref the caps used by RTPSource to prevent a memory leak.
+	  Original commit message from CVS:
+	  * gst/rtpmanager/rtpsource.c: (rtp_source_finalize):
+	  Make sure to unref the caps used by RTPSource to prevent a memory leak.
+
+2008-05-08 09:43:33 +0000  Olivier Crete <tester@tester.ca>
+
+	  gst/rtpmanager/rtpsession.c: Unlock the session lock when calling one of our callbacks.
+	  Original commit message from CVS:
+	  Patch by: Olivier Crete <tester at tester dot ca>
+	  * gst/rtpmanager/rtpsession.c: (source_clock_rate),
+	  (rtp_session_process_bye), (rtp_session_send_bye_locked):
+	  Unlock the session lock when calling one of our callbacks.
+	  Fixes #532011.
+
+2008-05-08 06:23:39 +0000  Sjoerd Simons <sjoerd@luon.net>
+
+	  gst/rtpmanager/gstrtpsession.c: Send RTP BYE command on EOS. Fixes bug #531955.
+	  Original commit message from CVS:
+	  Patch by: Sjoerd Simons <sjoerd at luon dot net>
+	  * gst/rtpmanager/gstrtpsession.c:
+	  (gst_rtp_session_event_send_rtp_sink):
+	  Send RTP BYE command on EOS. Fixes bug #531955.
+
+2008-04-25 11:32:09 +0000  Wim Taymans <wim.taymans@gmail.com>
+
+	  gst/rtpmanager/gstrtpbin.*: Expose new jitterbuffer property in rtpbin too.
+	  Original commit message from CVS:
+	  * gst/rtpmanager/gstrtpbin.c: (create_stream), (gst_rtp_bin_init),
+	  (gst_rtp_bin_set_property), (gst_rtp_bin_get_property):
+	  * gst/rtpmanager/gstrtpbin.h:
+	  Expose new jitterbuffer property in rtpbin too.
+
+2008-04-25 11:22:13 +0000  Wim Taymans <wim.taymans@gmail.com>
+
+	  gst/rtpmanager/gstrtpjitterbuffer.c: Disable sending out rtp packet lost events by default and make a property to ena...
+	  Original commit message from CVS:
+	  * gst/rtpmanager/gstrtpjitterbuffer.c:
+	  (gst_rtp_jitter_buffer_class_init), (gst_rtp_jitter_buffer_init),
+	  (gst_rtp_jitter_buffer_loop), (gst_rtp_jitter_buffer_set_property),
+	  (gst_rtp_jitter_buffer_get_property):
+	  Disable sending out rtp packet lost events by default and make a
+	  property to enabe it. We will likely enable it by default when the base
+	  depayloaders have a default handler for them so that we don't send these
+	  events all through the pipeline for now.
+
+2008-04-25 09:35:43 +0000  Wim Taymans <wim.taymans@gmail.com>
+
+	  gst/rtpmanager/gstrtpjitterbuffer.c: Remove private version of a function that is in -base now.
+	  Original commit message from CVS:
+	  * gst/rtpmanager/gstrtpjitterbuffer.c:
+	  (gst_rtp_jitter_buffer_init), (gst_rtp_jitter_buffer_flush_stop),
+	  (gst_rtp_jitter_buffer_src_event), (gst_rtp_jitter_buffer_chain),
+	  (gst_rtp_jitter_buffer_loop):
+	  Remove private version of a function that is in -base now.
+	  Add src event handler.
+	  Rework the jitterbuffer pushing loop so that it can quickly react to
+	  lost packets and instruct the depayloader of them. This can then be used
+	  to implement error concealment data.
+
+2008-04-25 08:21:06 +0000  Wim Taymans <wim.taymans@gmail.com>
+
+	  gst/rtpmanager/gstrtpsession.c: Set up some internal links functions for the RTCP and sync pads because the defaults ...
+	  Original commit message from CVS:
+	  * gst/rtpmanager/gstrtpsession.c:
+	  (gst_rtp_session_query_send_rtcp_src), (create_recv_rtcp_sink),
+	  (create_send_rtcp_src):
+	  Set up some internal links functions for the RTCP and sync pads because
+	  the defaults are really not correct.
+	  Implement a query handler for the RTCP src pad, mostly to correctly
+	  report about the latency.
+
+2008-04-25 08:15:58 +0000  Wim Taymans <wim.taymans@gmail.com>
+
+	  gst/rtpmanager/: Also keep track of the first buffer timestamp together with the first
+	  Original commit message from CVS:
+	  * gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_associate),
+	  (gst_rtp_bin_sync_chain):
+	  * gst/rtpmanager/rtpsession.c: (update_arrival_stats),
+	  (rtp_session_process_sr), (rtp_session_on_timeout):
+	  * gst/rtpmanager/rtpsource.c: (rtp_source_init),
+	  (calculate_jitter):
+	  * gst/rtpmanager/rtpsource.h:
+	  * gst/rtpmanager/rtpstats.h:
+	  Also keep track of the first buffer timestamp together with the first
+	  RTP timestamp as they both are needed to construct the timing of
+	  outgoing packets in the jitterbuffer and are therefore also needed to
+	  manage lip-sync. This fixes lip-sync if the first RTP packets arrive
+	  with a wildly different gap.
+
+2008-04-21 08:26:37 +0000  Olivier Crete <tester@tester.ca>
+
+	  gst/rtpmanager/gstrtpbin.c: Ref caps when inserting into the cache.
+	  Original commit message from CVS:
+	  Patch by: Olivier Crete <tester at tester dot ca>
+	  * gst/rtpmanager/gstrtpbin.c: (create_session), (get_pt_map),
+	  (new_ssrc_pad_found):
+	  Ref caps when inserting into the cache.
+	  Don't leak pads.
+	  * gst/rtpmanager/gstrtpjitterbuffer.c:
+	  (gst_rtp_jitter_buffer_get_clock_rate),
+	  (gst_rtp_jitter_buffer_query):
+	  Avoid a caps leak.
+	  Don't leak refcount in query.
+	  * gst/rtpmanager/gstrtpptdemux.c: (gst_rtp_pt_demux_get_caps),
+	  (gst_rtp_pt_demux_chain):
+	  Avoid caps leaks.
+	  * gst/rtpmanager/gstrtpsession.c: (source_get_sdes_structure),
+	  (gst_rtp_session_init), (return_true),
+	  (gst_rtp_session_clear_pt_map), (gst_rtp_session_cache_caps),
+	  (gst_rtp_session_clock_rate):
+	  Ref caps when inserting into the cache.
+	  Fix some more caps leaks. Fixes #528245.
+
+2008-04-17 07:31:44 +0000  Wim Taymans <wim.taymans@gmail.com>
+
+	  gst/rtpmanager/: Unset GValues after g_signal_emitv so that we avoid a refcount leak.
+	  Original commit message from CVS:
+	  * gst/rtpmanager/gstrtpbin.c: (get_pt_map), (free_client),
+	  (gst_rtp_bin_associate), (gst_rtp_bin_get_free_pad_name):
+	  * gst/rtpmanager/gstrtpjitterbuffer.c:
+	  (gst_rtp_jitter_buffer_get_clock_rate):
+	  * gst/rtpmanager/gstrtpptdemux.c: (gst_rtp_pt_demux_get_caps):
+	  * gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_clock_rate):
+	  Unset GValues after g_signal_emitv so that we avoid a refcount leak.
+	  Don't leak a padname.
+	  Don't leak client streams list.
+	  Lock rtpbin when associating streams. Fixes #528245.
+
+2008-04-09 22:27:50 +0000  Peter Kjellerstedt <pkj@axis.com>
+
+	  gst/rtpmanager/: Avoid leaking pads in the RTP manager.
+	  Original commit message from CVS:
+	  * gst/rtpmanager/gstrtpbin.c: (free_session):
+	  * gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_finalize):
+	  Avoid leaking pads in the RTP manager.
+
+2008-03-11 12:40:58 +0000  Olivier Crete <tester@tester.ca>
+
+	  gst/rtpmanager/rtpsession.*: Implement collision and loop detection in rtpmanager.
+	  Original commit message from CVS:
+	  Patch by: Olivier Crete <tester at tester dot ca>
+	  * gst/rtpmanager/rtpsession.c: (find_add_conflicting_addresses),
+	  (check_collision), (obtain_source), (rtp_session_create_new_ssrc),
+	  (rtp_session_create_source), (rtp_session_process_rtp),
+	  (rtp_session_process_sr), (rtp_session_process_rr),
+	  (rtp_session_process_sdes), (rtp_session_process_bye),
+	  (rtp_session_send_bye_locked), (rtp_session_send_bye),
+	  (rtp_session_on_timeout):
+	  * gst/rtpmanager/rtpsession.h:
+	  Implement collision and loop detection in rtpmanager.
+	  Fixes #520626.
+	  * gst/rtpmanager/rtpsource.c: (rtp_source_reset),
+	  (rtp_source_init):
+	  * gst/rtpmanager/rtpsource.h:
+	  Add method to reset stats.
+
+2008-03-11 11:36:03 +0000  Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>
+
+	  gst/rtpmanager/gstrtpsession.c: Avoid a deadlock when joining the RTCP thread in PAUSED because it might be blocked d...
+	  Original commit message from CVS:
+	  Based on patch by: Ole André Vadla Ravnås  <ole.andre.ravnas@tandberg.com>
+	  * gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_init),
+	  (rtcp_thread), (start_rtcp_thread), (stop_rtcp_thread),
+	  (join_rtcp_thread), (gst_rtp_session_change_state):
+	  Avoid a deadlock when joining the RTCP thread in PAUSED because it might
+	  be blocked downstream. Also avoid spawning multiple rtcp threads.
+	  Fixes #520894.
+
+2008-03-11 10:43:32 +0000  Stefan Kost <ensonic@users.sf.net>
+
+	  gst/rtpmanager/rtpjitterbuffer.c: Don't try to reset the clock skew when we have no timestamps.
+	  Original commit message from CVS:
+	  Patch by: Stefan Kost <ensonic@users.sf.net>
+	  * gst/rtpmanager/rtpjitterbuffer.c: (calculate_skew):
+	  Don't try to reset the clock skew when we have no timestamps.
+	  Fixes #519005.
+
+2008-02-20 09:33:25 +0000  Olivier Crete <tester@tester.ca>
+
+	  gst/rtpmanager/gstrtpbin.c: Fix small memory leak, leaking caps. Fixes #bug 517571.
+	  Original commit message from CVS:
+	  Patch by: Olivier Crete <tester at tester dot ca>
+	  * gst/rtpmanager/gstrtpbin.c: (new_ssrc_pad_found):
+	  Fix small memory leak, leaking caps. Fixes #bug 517571.
+
+2008-02-14 16:25:51 +0000  Olivier Crete <tester@tester.ca>
+
+	  gst/rtpmanager/gstrtpbin.c: Ignore streams that did not receive an SR packet when doing synchronisation. Fixes #516160.
+	  Original commit message from CVS:
+	  Patch by: Olivier Crete <tester@tester.ca>
+	  * gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_associate):
+	  Ignore streams that did not receive an SR packet when doing
+	  synchronisation. Fixes #516160.
+
+2008-01-29 18:57:27 +0000  Thijs Vermeir <thijsvermeir@gmail.com>
+
+	  gst/rtpmanager/gstrtpjitterbuffer.c: Try to get the new clock-rate from the buffer caps when we receive a new payload...
+	  Original commit message from CVS:
+	  Patch by: Thijs Vermeir  <thijsvermeir at gmail dot com>
+	  * gst/rtpmanager/gstrtpjitterbuffer.c:
+	  (gst_rtp_jitter_buffer_chain):
+	  Try to get the new clock-rate from the buffer caps when we receive a new
+	  payload type instead of always firing the signal. Fixes #512774.
+
+2008-01-25 16:58:00 +0000  Olivier Crete <tester@tester.ca>
+
+	  gst/rtpmanager/gstrtpbin.c: Also handle lip-sync when the clock-rate is not provided with caps but with a signal.
+	  Original commit message from CVS:
+	  Patch by: Olivier Crete <tester@tester.ca>
+	  * gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_associate),
+	  (create_stream), (payload_type_change), (new_ssrc_pad_found):
+	  Also handle lip-sync when the clock-rate is not provided with caps but
+	  with a signal.
+
+2008-01-25 16:00:52 +0000  Olivier Crete <tester@tester.ca>
+
+	  gst/rtpmanager/: Remove the fixed clock-rate from the jitterbuffer and extend it so that a clock-rate can be provided...
+	  Original commit message from CVS:
+	  Patch by: Olivier Crete <tester@tester.ca>
+	  * gst/rtpmanager/gstrtpjitterbuffer.c:
+	  (gst_jitter_buffer_sink_parse_caps), (gst_rtp_jitter_buffer_chain):
+	  * gst/rtpmanager/rtpjitterbuffer.c: (calculate_skew),
+	  (rtp_jitter_buffer_insert):
+	  * gst/rtpmanager/rtpjitterbuffer.h:
+	  Remove the fixed clock-rate from the jitterbuffer and extend it so that
+	  a clock-rate can be provided with each buffer instead. Fixes #511686.
+
+2008-01-25 15:49:55 +0000  Olivier Crete <tester@tester.ca>
+
+	  gst/rtpmanager/gstrtpjitterbuffer.c: Remove old unused variable.
+	  Original commit message from CVS:
+	  Patch by: Olivier Crete <tester@tester.ca>
+	  * gst/rtpmanager/gstrtpjitterbuffer.c:
+	  (gst_rtp_jitter_buffer_init), (gst_rtp_jitter_buffer_change_state),
+	  (gst_rtp_jitter_buffer_chain), (gst_rtp_jitter_buffer_loop):
+	  Remove old unused variable.
+	  Track pt on input buffers and get the clock-rate when it changes.
+	  Ignore packets with unknown clock-rate. See #511686.
+
+2008-01-25 01:44:27 +0000  Olivier Crete <tester@tester.ca>
+
+	  gst/rtpmanager/rtpsource.c: Fix unref of buffer using the wrong function.  Fixes #511920
+	  Original commit message from CVS:
+	  Patch by: Olivier Crete <tester@tester.ca>
+	  * gst/rtpmanager/rtpsource.c: Fix unref of buffer using the
+	  wrong function.  Fixes #511920
+
+2008-01-11 17:02:30 +0000  Wim Taymans <wim.taymans@gmail.com>
+
+	  gst/rtpmanager/gstrtpsession.c: If we find the caps in the cache, use it to parse the clock-rate instead of returning...
+	  Original commit message from CVS:
+	  * gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_clock_rate):
+	  If we find the caps in the cache, use it to parse the clock-rate instead
+	  of returning an error. Fixes a TODO as found by Youness Alaoui.
+
+2008-01-11 16:45:57 +0000  Youness Alaoui <youness.alaoui@collabora.co.uk>
+
+	  gst/rtpmanager/: Make it possible to use different user_data for each of the callbacks.
+	  Original commit message from CVS:
+	  Patch by: Youness Alaoui <youness dot alaoui at collabora dot co dot uk>
+	  * gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_clock_rate):
+	  * gst/rtpmanager/rtpsession.c: (rtp_session_set_callbacks),
+	  (rtp_session_set_process_rtp_callback),
+	  (rtp_session_set_send_rtp_callback),
+	  (rtp_session_set_send_rtcp_callback),
+	  (rtp_session_set_sync_rtcp_callback),
+	  (rtp_session_set_clock_rate_callback),
+	  (rtp_session_set_reconsider_callback), (source_push_rtp),
+	  (source_clock_rate), (rtp_session_process_bye),
+	  (rtp_session_process_rtcp), (rtp_session_send_bye),
+	  (rtp_session_on_timeout):
+	  * gst/rtpmanager/rtpsession.h:
+	  Make it possible to use different user_data for each of the callbacks.
+	  Fixes #508587.
+
+2008-01-10 20:57:17 +0000  Thijs Vermeir <thijsvermeir@gmail.com>
+
+	  gst/rtpmanager/gstrtpbin.c: Fix documentation for latest patch
+	  Original commit message from CVS:
+	  * gst/rtpmanager/gstrtpbin.c:
+	  Fix documentation for latest patch
+
+2008-01-10 14:34:30 +0000  Thijs Vermeir <thijsvermeir@gmail.com>
+
+	  gst/rtpmanager/gstrtpbin.c: Allow request_new_pad with name NULL (bug #508515)
+	  Original commit message from CVS:
+	  * gst/rtpmanager/gstrtpbin.c:
+	  Allow request_new_pad with name NULL (bug #508515)
+
+2008-01-09 14:39:44 +0000  Wim Taymans <wim.taymans@gmail.com>
+
+	  gst/rtpmanager/gstrtpsession.c: Don't set fixed caps, we can basically do everything the upsteam peer pad can renegot...
+	  Original commit message from CVS:
+	  * gst/rtpmanager/gstrtpsession.c: (create_send_rtp_sink):
+	  Don't set fixed caps, we can basically do everything the upsteam peer
+	  pad can renegotiate to. Fixes #507940.
+
+2008-01-04 18:47:57 +0000  Wim Taymans <wim.taymans@gmail.com>
+
+	  gst/rtpmanager/gstrtpjitterbuffer.c: Don't unref the popped buffer when we don't have ownership.
+	  Original commit message from CVS:
+	  * gst/rtpmanager/gstrtpjitterbuffer.c:
+	  (gst_rtp_jitter_buffer_loop):
+	  Don't unref the popped buffer when we don't have ownership.
+	  Fixes #507020.
+
+2007-12-31 13:12:06 +0000  Wim Taymans <wim.taymans@gmail.com>
+
+	  gst/rtpmanager/gstrtpssrcdemux.c: Don't clean up pads when going to PAUSED.
+	  Original commit message from CVS:
+	  * gst/rtpmanager/gstrtpssrcdemux.c:
+	  (gst_rtp_ssrc_demux_change_state):
+	  Don't clean up pads when going to PAUSED.
+
+2007-12-12 16:59:03 +0000  Wim Taymans <wim.taymans@gmail.com>
+
+	  gst/rtpmanager/: Clean up the dynamic pads when going to READY.
+	  Original commit message from CVS:
+	  * gst/rtpmanager/gstrtpptdemux.c: (gst_rtp_pt_demux_finalize),
+	  (gst_rtp_pt_demux_setup), (gst_rtp_pt_demux_release),
+	  (gst_rtp_pt_demux_change_state):
+	  * gst/rtpmanager/gstrtpssrcdemux.c: (gst_rtp_ssrc_demux_reset),
+	  (gst_rtp_ssrc_demux_dispose), (gst_rtp_ssrc_demux_src_query),
+	  (gst_rtp_ssrc_demux_change_state):
+	  Clean up the dynamic pads when going to READY.
+
+2007-12-12 12:11:53 +0000  Wim Taymans <wim.taymans@gmail.com>
+
+	  gst/rtpmanager/: Fix some leaks.
+	  Original commit message from CVS:
+	  * gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_finalize),
+	  (gst_rtp_bin_set_sdes_string), (gst_rtp_bin_get_sdes_string),
+	  (gst_rtp_bin_handle_message):
+	  * gst/rtpmanager/rtpsession.c: (rtp_session_finalize),
+	  (rtp_session_send_bye):
+	  * gst/rtpmanager/rtpsource.c: (rtp_source_finalize):
+	  Fix some leaks.
+
+2007-12-10 18:36:04 +0000  Wim Taymans <wim.taymans@gmail.com>
+
+	  gst/rtpmanager/: Post a message when the SDES infor changes for a source.
+	  Original commit message from CVS:
+	  * gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_class_init),
+	  (gst_rtp_bin_handle_message):
+	  * gst/rtpmanager/gstrtpsession.c: (source_get_sdes_structure),
+	  (on_ssrc_sdes):
+	  Post a message when the SDES infor changes for a source.
+	  * gst/rtpmanager/rtpsession.c:
+	  * gst/rtpmanager/rtpsource.c:
+	  Update some comments.
+
+2007-12-10 15:34:19 +0000  Wim Taymans <wim.taymans@gmail.com>
+
+	  gst/rtpmanager/: Add signal to notify of an SDES change.
+	  Original commit message from CVS:
+	  * gst/rtpmanager/gstrtpbin.c: (on_ssrc_sdes), (create_session),
+	  (gst_rtp_bin_class_init):
+	  * gst/rtpmanager/gstrtpbin.h:
+	  * gst/rtpmanager/gstrtpclient.c:
+	  * gst/rtpmanager/gstrtpclient.h:
+	  * gst/rtpmanager/gstrtpjitterbuffer.h:
+	  * gst/rtpmanager/gstrtpmanager.c:
+	  * gst/rtpmanager/gstrtpptdemux.c:
+	  * gst/rtpmanager/gstrtpptdemux.h:
+	  * gst/rtpmanager/gstrtpsession.c: (on_ssrc_sdes),
+	  (gst_rtp_session_class_init), (gst_rtp_session_init):
+	  * gst/rtpmanager/gstrtpsession.h:
+	  * gst/rtpmanager/gstrtpssrcdemux.c:
+	  * gst/rtpmanager/gstrtpssrcdemux.h:
+	  * gst/rtpmanager/rtpjitterbuffer.c:
+	  * gst/rtpmanager/rtpjitterbuffer.h:
+	  * gst/rtpmanager/rtpsession.c: (rtp_session_class_init),
+	  (on_ssrc_sdes), (rtp_session_process_sdes):
+	  * gst/rtpmanager/rtpsession.h:
+	  * gst/rtpmanager/rtpsource.c:
+	  * gst/rtpmanager/rtpsource.h:
+	  * gst/rtpmanager/rtpstats.c:
+	  * gst/rtpmanager/rtpstats.h:
+	  Add signal to notify of an SDES change.
+	  Fix object type in the signal callbacks.
+
+2007-12-10 14:03:32 +0000  Wim Taymans <wim.taymans@gmail.com>
+
+	  gst/rtpmanager/gstrtpbin.*: Expose SDES items as properties and configure the session managers with them.
+	  Original commit message from CVS:
+	  * gst/rtpmanager/gstrtpbin.c: (create_session),
+	  (gst_rtp_bin_class_init), (gst_rtp_bin_init), (sdes_type_to_name),
+	  (gst_rtp_bin_set_sdes_string), (gst_rtp_bin_get_sdes_string),
+	  (gst_rtp_bin_set_property), (gst_rtp_bin_get_property):
+	  * gst/rtpmanager/gstrtpbin.h:
+	  Expose SDES items as properties and configure the session managers with
+	  them.
+	  * gst/rtpmanager/rtpsource.c: (rtp_source_class_init),
+	  (rtp_source_set_property):
+	  Fix SSRC property.
+
+2007-12-10 11:08:11 +0000  Wim Taymans <wim.taymans@gmail.com>
+
+	  gst/rtpmanager/: Update comment.
+	  Original commit message from CVS:
+	  * gst/rtpmanager/gstrtpbin.c: (create_session):
+	  * gst/rtpmanager/rtpjitterbuffer.c:
+	  Update comment.
+	  * gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_class_init),
+	  (gst_rtp_session_set_property), (gst_rtp_session_get_property):
+	  Define some GObject properties to set SDES and other configuration.
+	  * gst/rtpmanager/rtpsession.c: (rtp_session_class_init),
+	  (rtp_session_init), (rtp_session_finalize),
+	  (rtp_session_set_property), (rtp_session_get_property),
+	  (on_ssrc_sdes), (rtp_session_set_bandwidth),
+	  (rtp_session_get_bandwidth), (rtp_session_set_rtcp_fraction),
+	  (rtp_session_get_rtcp_fraction), (rtp_session_set_sdes_string),
+	  (rtp_session_get_sdes_string), (obtain_source),
+	  (rtp_session_get_internal_source), (rtp_session_process_sdes),
+	  (rtp_session_send_rtp), (rtp_session_next_timeout), (session_sdes),
+	  (is_rtcp_time):
+	  * gst/rtpmanager/rtpsession.h:
+	  Add signal when new SDES infor has been found for a source.
+	  Create properties for SDES and other info.
+	  Simplify the SDES API.
+	  Add method for getting the internal source object of the session.
+	  * gst/rtpmanager/rtpsource.c: (rtp_source_class_init),
+	  (rtp_source_finalize), (rtp_source_set_property),
+	  (rtp_source_get_property), (rtp_source_set_callbacks),
+	  (rtp_source_get_ssrc), (rtp_source_set_as_csrc),
+	  (rtp_source_is_as_csrc), (rtp_source_is_active),
+	  (rtp_source_is_validated), (rtp_source_is_sender),
+	  (rtp_source_received_bye), (rtp_source_get_bye_reason),
+	  (rtp_source_set_sdes), (rtp_source_set_sdes_string),
+	  (rtp_source_get_sdes), (rtp_source_get_sdes_string),
+	  (rtp_source_get_new_sr), (rtp_source_get_new_rb):
+	  * gst/rtpmanager/rtpsource.h:
+	  Add GObject properties for various things.
+	  Don't leak the bye reason.
+
+2007-11-22 09:08:27 +0000  Wim Taymans <wim.taymans@gmail.com>
+
+	  gst/rtpmanager/gstrtpjitterbuffer.c: jitterbuffer can buffer an unlimited amount of time and thus has no max_latency ...
+	  Original commit message from CVS:
+	  * gst/rtpmanager/gstrtpjitterbuffer.c:
+	  (gst_rtp_jitter_buffer_query):
+	  jitterbuffer can buffer an unlimited amount of time and thus has no
+	  max_latency requirements.
+
+2007-11-02 21:45:38 +0000  Ole André Vadla Ravnås <ole.andre.ravnas@tandberg.com>
+
+	  gst/rtpmanager/gstrtpsession.c: Fix bad function signatures (#492798).
+	  Original commit message from CVS:
+	  Patch by: Ole André Vadla Ravnås  <ole.andre.ravnas@tandberg.com>
+	  * gst/rtpmanager/gstrtpsession.c:
+	  Fix bad function signatures (#492798).
+
+2007-10-09 10:01:39 +0000  Laurent Glayal <spglegle@yahoo.fr>
+
+	  gst/rtpmanager/gstrtpbin.c: Fix memleak. Fixes #484990.
+	  Original commit message from CVS:
+	  Patch by: Laurent Glayal <spglegle at yahoo dot fr>
+	  * gst/rtpmanager/gstrtpbin.c: (create_stream),
+	  (gst_rtp_bin_class_init):
+	  Fix memleak. Fixes #484990.
+
+2007-10-08 17:46:45 +0000  Jan Schmidt <thaytan@mad.scientist.com>
+
+	  gst/: Fix compiler warnings shown by Forte.
+	  Original commit message from CVS:
+	  * gst/librfb/rfbbuffer.c: (rfb_buffer_new_and_alloc):
+	  * gst/librfb/rfbbuffer.h:
+	  * gst/librfb/rfbdecoder.c: (rfb_socket_get_buffer):
+	  * gst/mpegvideoparse/mpegvideoparse.c: (gst_mpegvideoparse_chain):
+	  * gst/nsf/nes6502.c: (nes6502_execute):
+	  * gst/real/gstrealaudiodec.c: (gst_real_audio_dec_setcaps):
+	  * gst/real/gstrealvideodec.c: (open_library):
+	  * gst/real/gstrealvideodec.h:
+	  * gst/rtpmanager/gstrtpsession.c: (create_recv_rtp_sink),
+	  (create_recv_rtcp_sink), (create_send_rtp_sink):
+	  Fix compiler warnings shown by Forte.
+
+2007-10-08 10:39:35 +0000  Wim Taymans <wim.taymans@gmail.com>
+
+	  gst/rtpmanager/gstrtpbin.c: Fix caps refcounting for payload maps.
+	  Original commit message from CVS:
+	  * gst/rtpmanager/gstrtpbin.c: (get_pt_map),
+	  (gst_rtp_bin_clear_pt_map), (gst_rtp_bin_class_init):
+	  Fix caps refcounting for payload maps.
+	  When clearing payload maps, also clear sessions and streams payload
+	  maps.
+	  * gst/rtpmanager/gstrtpptdemux.c: (gst_rtp_pt_demux_get_caps),
+	  (gst_rtp_pt_demux_clear_pt_map), (gst_rtp_pt_demux_chain),
+	  (find_pad_for_pt):
+	  Implement clearing the payload map.
+	  * gst/rtpmanager/gstrtpsession.c:
+	  (gst_rtp_session_event_send_rtp_sink):
+	  Forward flush events instead of leaking them.
+	  * gst/rtpmanager/gstrtpssrcdemux.c:
+	  (gst_rtp_ssrc_demux_rtcp_sink_event):
+	  Correctly refcount events before pushing them.
+
+2007-10-05 17:26:14 +0000  Wim Taymans <wim.taymans@gmail.com>
+
+	  gst/rtpmanager/rtpsession.c: When reconsidering RTCP timeouts, set the next timeout against the last report time inst...
+	  Original commit message from CVS:
+	  * gst/rtpmanager/rtpsession.c: (rtp_session_next_timeout),
+	  When reconsidering RTCP timeouts, set the next timeout against the last
+	  report time instead of the current clock time so that we don't end up
+	  reconsidering forever.
+
+2007-10-05 12:07:37 +0000  Wim Taymans <wim.taymans@gmail.com>
+
+	  gst/rtpmanager/gstrtpjitterbuffer.c: Only peek at the tail element instead of popping it off, which allows us to grea...
+	  Original commit message from CVS:
+	  * gst/rtpmanager/gstrtpjitterbuffer.c:
+	  (gst_rtp_jitter_buffer_chain), (gst_rtp_jitter_buffer_loop):
+	  Only peek at the tail element instead of popping it off, which allows
+	  us to greatly simplify things when the tail element changes.
+	  * gst/rtpmanager/gstrtpsession.c:
+	  (gst_rtp_session_event_recv_rtp_sink):
+	  * gst/rtpmanager/gstrtpssrcdemux.c:
+	  (gst_rtp_ssrc_demux_sink_event):
+	  Forward FLUSH events instead of leaking them.
+	  * gst/rtpmanager/rtpjitterbuffer.c: (rtp_jitter_buffer_reset_skew),
+	  (calculate_skew), (rtp_jitter_buffer_insert):
+	  * gst/rtpmanager/rtpjitterbuffer.h:
+	  Remove the tail-changed callback in favour of a simple boolean when we
+	  insert a buffer in the queue.
+	  Add method to peek the tail of the buffer.
+
+2007-10-02 10:27:45 +0000  Wim Taymans <wim.taymans@gmail.com>
+
+	  gst/rtpmanager/gstrtpjitterbuffer.c: Remove some old unused variables.
+	  Original commit message from CVS:
+	  * gst/rtpmanager/gstrtpjitterbuffer.c:
+	  (gst_rtp_jitter_buffer_flush_start),
+	  (gst_rtp_jitter_buffer_flush_stop),
+	  (gst_rtp_jitter_buffer_change_state), (apply_offset),
+	  (gst_rtp_jitter_buffer_loop):
+	  Remove some old unused variables.
+	  Don't add the latency to the skew corrected timestamp, latency is only
+	  used to sync against the clock.
+	  Improve debugging.
+	  * gst/rtpmanager/rtpjitterbuffer.c: (rtp_jitter_buffer_init),
+	  (rtp_jitter_buffer_reset_skew), (calculate_skew):
+	  * gst/rtpmanager/rtpjitterbuffer.h:
+	  Handle case where server timestamp goes backwards or wildly jumps by
+	  temporarily pausing the skew correction.
+	  Improve debugging.
+
+2007-09-28 14:51:58 +0000  Wim Taymans <wim.taymans@gmail.com>
+
+	  gst/rtpmanager/gstrtpbin.c: Fix crasher in dispose.
+	  Original commit message from CVS:
+	  * gst/rtpmanager/gstrtpbin.c: (free_client):
+	  Fix crasher in dispose.
+	  * gst/rtpmanager/rtpjitterbuffer.c: (calculate_skew):
+	  Handle cases where input buffers have no timestamps so that no clock
+	  skew can be calculated, in this case interpollate timestamps based on
+	  rtp timestamp and assume a 0 clock skew.
+
+2007-09-28 11:17:35 +0000  Wim Taymans <wim.taymans@gmail.com>
+
+	  gst/rtpmanager/gstrtpjitterbuffer.c: Remove jitter correction code, it's now in the lower level object.
+	  Original commit message from CVS:
+	  * gst/rtpmanager/gstrtpjitterbuffer.c: (apply_latency),
+	  (gst_rtp_jitter_buffer_loop), (gst_rtp_jitter_buffer_query):
+	  Remove jitter correction code, it's now in the lower level object.
+	  Use new -core method for doing a peer query.
+	  * gst/rtpmanager/rtpjitterbuffer.c: (rtp_jitter_buffer_init),
+	  (calculate_skew), (rtp_jitter_buffer_insert):
+	  * gst/rtpmanager/rtpjitterbuffer.h:
+	  Move jitter correction to the lowlevel jitterbuffer.
+	  Increase the max window size.
+	  When filling the window, already start estimating the skew using a
+	  parabolic weighting factor so that we have a much better startup
+	  behaviour that gets more accurate with the more samples we have.
+	  Increase the default weighting factor for the steady state to get
+	  smoother timestamps.
+
+2007-09-26 20:08:28 +0000  Wim Taymans <wim.taymans@gmail.com>
+
+	  gst/rtpmanager/gstrtpbin.c: Fix cleanup crasher.
+	  Original commit message from CVS:
+	  * gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_dispose),
+	  (gst_rtp_bin_finalize):
+	  Fix cleanup crasher.
+	  * gst/rtpmanager/rtpjitterbuffer.c: (rtp_jitter_buffer_init),
+	  (calculate_skew):
+	  * gst/rtpmanager/rtpjitterbuffer.h:
+	  Dynamically adjust the skew calculation window so that we calculate it
+	  over a period of around 2 seconds.
+
+2007-09-20 14:34:57 +0000  Wim Taymans <wim.taymans@gmail.com>
+
+	  gst/rtpmanager/: Add notification of active SSRCs to various RTP elements. Fixes #478566.
+	  Original commit message from CVS:
+	  * gst/rtpmanager/gstrtpbin.c: (on_ssrc_active), (create_session),
+	  (gst_rtp_bin_class_init):
+	  * gst/rtpmanager/gstrtpbin.h:
+	  * gst/rtpmanager/gstrtpsession.c: (on_ssrc_active),
+	  (gst_rtp_session_class_init), (gst_rtp_session_init),
+	  (gst_rtp_session_event_send_rtp_sink):
+	  * gst/rtpmanager/gstrtpsession.h:
+	  * gst/rtpmanager/rtpsession.c: (rtp_session_class_init),
+	  (on_ssrc_active), (rtp_session_process_rb):
+	  * gst/rtpmanager/rtpsession.h:
+	  Add notification of active SSRCs to various RTP elements. Fixes #478566.
+
+2007-09-17 02:01:41 +0000  Wim Taymans <wim.taymans@gmail.com>
+
+	  gst/rtpmanager/gstrtpbin.c: Link to the right pads regardless of which one was created first in the ssrc demuxer.
+	  Original commit message from CVS:
+	  * gst/rtpmanager/gstrtpbin.c: (new_ssrc_pad_found):
+	  Link to the right pads regardless of which one was created first in the
+	  ssrc demuxer.
+	  * gst/rtpmanager/gstrtpjitterbuffer.c:
+	  (gst_rtp_jitter_buffer_chain), (gst_rtp_jitter_buffer_loop):
+	  * gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_process_rtp),
+	  (gst_rtp_session_chain_recv_rtp), (gst_rtp_session_chain_send_rtp):
+	  * gst/rtpmanager/rtpsource.c: (calculate_jitter):
+	  Improve debugging.
+	  * gst/rtpmanager/gstrtpssrcdemux.c: (create_demux_pad_for_ssrc),
+	  (gst_rtp_ssrc_demux_init), (gst_rtp_ssrc_demux_finalize),
+	  (gst_rtp_ssrc_demux_sink_event),
+	  (gst_rtp_ssrc_demux_rtcp_sink_event), (gst_rtp_ssrc_demux_chain),
+	  (gst_rtp_ssrc_demux_rtcp_chain),
+	  (gst_rtp_ssrc_demux_internal_links):
+	  * gst/rtpmanager/gstrtpssrcdemux.h:
+	  Fix race in creating the RTP and RTCP pads when a new SSRC is detected.
+
+2007-09-16 19:40:31 +0000  Wim Taymans <wim.taymans@gmail.com>
+
+	  gst/rtpmanager/gstrtpbin.c: Use lock to protect variable.
+	  Original commit message from CVS:
+	  * gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_set_property),
+	  (gst_rtp_bin_get_property):
+	  Use lock to protect variable.
+	  * gst/rtpmanager/gstrtpjitterbuffer.c:
+	  (gst_rtp_jitter_buffer_class_init),
+	  (gst_jitter_buffer_sink_parse_caps), (gst_rtp_jitter_buffer_chain),
+	  (convert_rtptime_to_gsttime), (gst_rtp_jitter_buffer_loop):
+	  Reconstruct GST timestamp from RTP timestamps based on measured clock
+	  skew and sync offset.
+	  * gst/rtpmanager/rtpjitterbuffer.c: (rtp_jitter_buffer_init),
+	  (rtp_jitter_buffer_set_tail_changed),
+	  (rtp_jitter_buffer_set_clock_rate),
+	  (rtp_jitter_buffer_get_clock_rate), (calculate_skew),
+	  (rtp_jitter_buffer_insert), (rtp_jitter_buffer_peek):
+	  * gst/rtpmanager/rtpjitterbuffer.h:
+	  Measure clock skew.
+	  Add callback to be notfied when a new packet was inserted at the tail.
+	  * gst/rtpmanager/rtpsource.c: (rtp_source_init),
+	  (calculate_jitter), (rtp_source_send_rtp):
+	  * gst/rtpmanager/rtpsource.h:
+	  Remove clock skew detection, it's move to the jitterbuffer now.
+
+2007-09-15 18:48:03 +0000  Wim Taymans <wim.taymans@gmail.com>
+
+	  gst/rtpmanager/gstrtpbin.c: Also set NTP base time on new sessions.
+	  Original commit message from CVS:
+	  * gst/rtpmanager/gstrtpbin.c: (create_session):
+	  Also set NTP base time on new sessions.
+	  * gst/rtpmanager/gstrtpjitterbuffer.c:
+	  (gst_rtp_jitter_buffer_loop), (gst_rtp_jitter_buffer_query),
+	  (gst_rtp_jitter_buffer_set_property),
+	  (gst_rtp_jitter_buffer_get_property):
+	  Use the right lock to protect our variables.
+	  Fix some comment.
+	  * gst/rtpmanager/gstrtpsession.c:
+	  (gst_rtp_session_getcaps_send_rtp),
+	  (gst_rtp_session_chain_send_rtp), (create_send_rtp_sink):
+	  Implement getcaps on the sender sinkpad so that payloaders can negotiate
+	  the right SSRC.
+
+2007-09-12 21:23:47 +0000  Wim Taymans <wim.taymans@gmail.com>
+
+	  gst/rtpmanager/: Various leak fixes.
+	  Original commit message from CVS:
+	  * gst/rtpmanager/gstrtpbin.c: (create_session), (free_session),
+	  (get_client), (free_client), (gst_rtp_bin_associate),
+	  (free_stream), (gst_rtp_bin_class_init), (gst_rtp_bin_dispose),
+	  (gst_rtp_bin_finalize):
+	  * gst/rtpmanager/gstrtpjitterbuffer.c:
+	  (gst_rtp_jitter_buffer_class_init),
+	  (gst_rtp_jitter_buffer_finalize):
+	  * gst/rtpmanager/gstrtpptdemux.c: (gst_rtp_pt_demux_release):
+	  * gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_finalize),
+	  (gst_rtp_session_set_property), (gst_rtp_session_chain_recv_rtp),
+	  (gst_rtp_session_chain_send_rtp):
+	  * gst/rtpmanager/gstrtpssrcdemux.c:
+	  (gst_rtp_ssrc_demux_class_init), (gst_rtp_ssrc_demux_dispose):
+	  * gst/rtpmanager/rtpsession.c: (rtp_session_finalize):
+	  * gst/rtpmanager/rtpsession.h:
+	  Various leak fixes.
+
+2007-09-12 18:04:32 +0000  Wim Taymans <wim.taymans@gmail.com>
+
+	  gst/rtpmanager/gstrtpbin.c: Calculate and configure the NTP base time so that we can generate better
+	  Original commit message from CVS:
+	  * gst/rtpmanager/gstrtpbin.c: (calc_ntp_ns_base),
+	  (gst_rtp_bin_change_state), (new_payload_found), (create_send_rtp):
+	  Calculate and configure the NTP base time so that we can generate better
+	  NTP times in SR packets.
+	  Set caps on new ghostpad.
+	  * gst/rtpmanager/gstrtpjitterbuffer.c:
+	  (gst_rtp_jitter_buffer_loop):
+	  Clean debug statement.
+	  * gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_class_init),
+	  (gst_rtp_session_init), (gst_rtp_session_set_property),
+	  (gst_rtp_session_get_property), (get_current_ntp_ns_time),
+	  (rtcp_thread), (gst_rtp_session_event_recv_rtp_sink),
+	  (gst_rtp_session_internal_links), (gst_rtp_session_chain_recv_rtp),
+	  (gst_rtp_session_event_send_rtp_sink),
+	  (gst_rtp_session_chain_send_rtp), (create_recv_rtp_sink),
+	  (create_send_rtp_sink):
+	  * gst/rtpmanager/gstrtpsession.h:
+	  Add ntp-ns-base property to convert running_time to NTP time.
+	  Handle NEWSEGMENT events on send and recv RTP pads so that we can
+	  calculate the running time and thus NTP time of the packets.
+	  Simplify getting the current NTP time using the pipeline clock.
+	  Implement internal links functions.
+	  Use the buffer timestamp to calculate the NTP time instead of the clock.
+	  * gst/rtpmanager/gstrtpssrcdemux.c: (create_demux_pad_for_ssrc),
+	  (gst_rtp_ssrc_demux_init), (gst_rtp_ssrc_demux_sink_event),
+	  (gst_rtp_ssrc_demux_chain), (gst_rtp_ssrc_demux_rtcp_chain),
+	  (gst_rtp_ssrc_demux_internal_links),
+	  (gst_rtp_ssrc_demux_src_query):
+	  * gst/rtpmanager/gstrtpssrcdemux.h:
+	  Implement internal links function.
+	  Calculate the diff between different streams, this might be used later
+	  to get the inter stream latency.
+	  * gst/rtpmanager/rtpsession.c: (rtp_session_send_rtp):
+	  Simple cleanup.
+	  * gst/rtpmanager/rtpsource.c: (rtp_source_init),
+	  (calculate_jitter), (rtp_source_send_rtp), (rtp_source_get_new_sr):
+	  Make the clock skew window a little bigger.
+	  Apply the clock skew to all buffers, not just one with a new timestamp.
+	  Calculate and debug sender clock drift.
+	  Use extended last timestamp to interpollate for SR reports.
+
+2007-09-04 15:23:34 +0000  Tim-Philipp Müller <tim@centricular.net>
+
+	  gst/rtpmanager/gstrtpsession.c: Make compiler happy: fix compilation with -Wall -Werror (#473562).
+	  Original commit message from CVS:
+	  * gst/rtpmanager/gstrtpsession.c:
+	  Make compiler happy: fix compilation with -Wall -Werror
+	  (#473562).
+
+2007-09-03 21:19:34 +0000  Wim Taymans <wim.taymans@gmail.com>
+
+	  gst/rtpmanager/: Updated example pipelines in docs.
+	  Original commit message from CVS:
+	  * gst/rtpmanager/gstrtpbin-marshal.list:
+	  * gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_get_client),
+	  (gst_rtp_bin_associate), (gst_rtp_bin_sync_chain), (create_stream),
+	  (gst_rtp_bin_init), (caps_changed), (new_ssrc_pad_found),
+	  (create_recv_rtp), (create_recv_rtcp), (create_send_rtp):
+	  * gst/rtpmanager/gstrtpbin.h:
+	  Updated example pipelines in docs.
+	  Handle sync_rtcp buffers from the SSRC demuxer to perform lip-sync.
+	  Set the default latency correctly.
+	  Add some more points where we can get caps.
+	  * gst/rtpmanager/gstrtpjitterbuffer.c:
+	  (gst_rtp_jitter_buffer_class_init),
+	  (gst_jitter_buffer_sink_parse_caps), (gst_rtp_jitter_buffer_loop),
+	  (gst_rtp_jitter_buffer_query),
+	  (gst_rtp_jitter_buffer_set_property),
+	  (gst_rtp_jitter_buffer_get_property):
+	  Add ts-offset property to control timestamping.
+	  * gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_class_init),
+	  (gst_rtp_session_init), (gst_rtp_session_set_property),
+	  (gst_rtp_session_get_property), (get_current_ntp_ns_time),
+	  (rtcp_thread), (stop_rtcp_thread), (gst_rtp_session_change_state),
+	  (gst_rtp_session_send_rtcp), (gst_rtp_session_sync_rtcp),
+	  (gst_rtp_session_cache_caps), (gst_rtp_session_clock_rate),
+	  (gst_rtp_session_sink_setcaps), (gst_rtp_session_chain_recv_rtp),
+	  (gst_rtp_session_event_send_rtp_sink),
+	  (gst_rtp_session_chain_send_rtp), (create_recv_rtp_sink),
+	  (create_recv_rtcp_sink), (create_send_rtp_sink),
+	  (create_send_rtcp_src):
+	  Various cleanups.
+	  Feed rtpsession manager with NTP time based on pipeline clock when
+	  handling RTP packets and RTCP timeouts.
+	  Perform all RTCP with the system clock.
+	  Set caps on RTCP outgoing buffers.
+	  * gst/rtpmanager/gstrtpssrcdemux.c: (find_demux_pad_for_ssrc),
+	  (create_demux_pad_for_ssrc), (gst_rtp_ssrc_demux_base_init),
+	  (gst_rtp_ssrc_demux_init), (gst_rtp_ssrc_demux_sink_event),
+	  (gst_rtp_ssrc_demux_rtcp_sink_event), (gst_rtp_ssrc_demux_chain),
+	  (gst_rtp_ssrc_demux_rtcp_chain):
+	  * gst/rtpmanager/gstrtpssrcdemux.h:
+	  Also demux RTCP messages.
+	  * gst/rtpmanager/rtpsession.c: (rtp_session_set_callbacks),
+	  (update_arrival_stats), (rtp_session_process_rtp),
+	  (rtp_session_process_rb), (rtp_session_process_sr),
+	  (rtp_session_process_rr), (rtp_session_process_rtcp),
+	  (rtp_session_send_rtp), (rtp_session_send_bye),
+	  (session_start_rtcp), (session_report_blocks), (session_cleanup),
+	  (rtp_session_on_timeout):
+	  * gst/rtpmanager/rtpsession.h:
+	  Remove the get_time callback, the GStreamer part will feed us with
+	  enough timing information.
+	  Split sync timing and RTCP timing information.
+	  Factor out common RB handling for SR and RR.
+	  Send out SR RTCP packets for lip-sync.
+	  Move SR and RR packet info generation to the source.
+	  * gst/rtpmanager/rtpsource.c: (rtp_source_init),
+	  (rtp_source_update_caps), (get_clock_rate), (calculate_jitter),
+	  (rtp_source_process_rtp), (rtp_source_send_rtp),
+	  (rtp_source_process_sr), (rtp_source_process_rb),
+	  (rtp_source_get_new_sr), (rtp_source_get_new_rb),
+	  (rtp_source_get_last_sr):
+	  * gst/rtpmanager/rtpsource.h:
+	  * gst/rtpmanager/rtpstats.h:
+	  Use caps on incomming buffers to get timing information when they are
+	  there.
+	  Calculate clock scew of the receiver compared to the sender and adjust
+	  the rtp timestamps.
+	  Calculate the round trip in sources.
+	  Do SR and RR calculations in the source.
+
+2007-08-31 15:26:14 +0000  Wim Taymans <wim.taymans@gmail.com>
+
+	  gst/rtpmanager/gstrtpjitterbuffer.c: Use extended timestamp to release buffers from the jitterbuffer so that we can h...
+	  Original commit message from CVS:
+	  * gst/rtpmanager/gstrtpjitterbuffer.c:
+	  (gst_rtp_jitter_buffer_flush_stop),
+	  (gst_rtp_jitter_buffer_change_state), (gst_rtp_jitter_buffer_loop):
+	  Use extended timestamp to release buffers from the jitterbuffer so that
+	  we can handle the rtp wraparound correctly.
+
+2007-08-29 16:56:27 +0000  Wim Taymans <wim.taymans@gmail.com>
+
+	  gst/rtpmanager/gstrtpjitterbuffer.c: Improve Comments.
+	  Original commit message from CVS:
+	  * gst/rtpmanager/gstrtpjitterbuffer.c:
+	  (gst_rtp_jitter_buffer_loop):
+	  Improve Comments.
+	  * gst/rtpmanager/gstrtpsession.c: (stop_rtcp_thread),
+	  (gst_rtp_session_change_state), (gst_rtp_session_parse_caps),
+	  (gst_rtp_session_clock_rate), (gst_rtp_session_sink_setcaps),
+	  (gst_rtp_session_event_send_rtp_sink), (create_recv_rtp_sink),
+	  (create_send_rtp_sink):
+	  Also parse the sink caps for clock-rate instead of only relying on the
+	  result of the signal.
+	  * gst/rtpmanager/rtpsource.c: (rtp_source_send_rtp):
+	  Make sure we fetch the clock rate for payloads we are sending out so
+	  that we can use it for SR reports.
+
+2007-08-29 01:22:43 +0000  Wim Taymans <wim.taymans@gmail.com>
+
+	  gst/rtpmanager/gstrtpsession.*: Distribute synchronisation parameters to the session manager so that it can generate ...
+	  Original commit message from CVS:
+	  * gst/rtpmanager/gstrtpsession.c: (stop_rtcp_thread),
+	  (gst_rtp_session_change_state),
+	  (gst_rtp_session_event_send_rtp_sink):
+	  * gst/rtpmanager/gstrtpsession.h:
+	  Distribute synchronisation parameters to the session manager so that it
+	  can generate correct SR packets for lip-sync.
+	  * gst/rtpmanager/rtpsession.c: (rtp_session_set_base_time),
+	  (rtp_session_set_timestamp_sync), (session_start_rtcp):
+	  * gst/rtpmanager/rtpsession.h:
+	  Add methods for setting sync parameters.
+	  Set correct RTP time in SR packets using the sync params.
+	  * gst/rtpmanager/rtpsource.c: (rtp_source_send_rtp):
+	  * gst/rtpmanager/rtpsource.h:
+	  Record last RTP <-> GST timestamp so that we can use them to convert NTP
+	  to RTP timestamps in SR packets.
+
+2007-08-28 20:30:16 +0000  Wim Taymans <wim.taymans@gmail.com>
+
+	  gst/rtpmanager/gstrtpbin.c: Add some more advanced example pipelines.
+	  Original commit message from CVS:
+	  * gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_clear_pt_map):
+	  Add some more advanced example pipelines.
+	  * gst/rtpmanager/gstrtpsession.c: (rtcp_thread),
+	  (stop_rtcp_thread), (gst_rtp_session_send_rtcp):
+	  Add some debug and FIXME.
+	  Release LOCK when performing session cleanup.
+	  * gst/rtpmanager/rtpsession.c: (session_report_blocks):
+	  Add some debug.
+	  * gst/rtpmanager/rtpsource.c: (calculate_jitter),
+	  (rtp_source_send_rtp):
+	  Make sure we always send RTP packets with the session SSRC.
+
+2007-08-27 21:17:21 +0000  Wim Taymans <wim.taymans@gmail.com>
+
+	  gst/rtpmanager/gstrtpjitterbuffer.c: When synchronizing buffers, take peer latency into account.
+	  Original commit message from CVS:
+	  * gst/rtpmanager/gstrtpjitterbuffer.c:
+	  (gst_rtp_jitter_buffer_change_state), (gst_rtp_jitter_buffer_loop),
+	  (gst_rtp_jitter_buffer_query):
+	  When synchronizing buffers, take peer latency into account.
+	  Don't try to add our latency to invalid peer max latency values.
+
+2007-08-23 21:39:58 +0000  Tim-Philipp Müller <tim@centricular.net>
+
+	  Rename all GstRTPFoo structs to GstRtpFoo so that GST_BOILERPLATE registers a GType that's different than the GstRTPF...
+	  Original commit message from CVS:
+	  * docs/plugins/gst-plugins-bad-plugins-sections.txt:
+	  * docs/plugins/gst-plugins-bad-plugins.hierarchy:
+	  * docs/plugins/gst-plugins-bad-plugins.interfaces:
+	  * docs/plugins/gst-plugins-bad-plugins.signals:
+	  * gst/rtpmanager/gstrtpbin.c:
+	  * gst/rtpmanager/gstrtpbin.h:
+	  * gst/rtpmanager/gstrtpclient.c:
+	  * gst/rtpmanager/gstrtpclient.h:
+	  * gst/rtpmanager/gstrtpjitterbuffer.c:
+	  * gst/rtpmanager/gstrtpjitterbuffer.h:
+	  * gst/rtpmanager/gstrtpptdemux.c:
+	  * gst/rtpmanager/gstrtpptdemux.h:
+	  * gst/rtpmanager/gstrtpsession.c:
+	  * gst/rtpmanager/gstrtpsession.h:
+	  * gst/rtpmanager/gstrtpssrcdemux.c:
+	  * gst/rtpmanager/gstrtpssrcdemux.h:
+	  Rename all GstRTPFoo structs to GstRtpFoo so that GST_BOILERPLATE
+	  registers a GType that's different than the GstRTPFoo types that
+	  farsight registers (luckily GType names are case sensitive). Should
+	  finally fix #430664.
+
+2007-08-21 17:18:29 +0000  Wim Taymans <wim.taymans@gmail.com>
+
+	  gst/rtpmanager/gstrtpjitterbuffer.c: When drop-on-latency is set but we have no latency configured, just push the buf...
+	  Original commit message from CVS:
+	  * gst/rtpmanager/gstrtpjitterbuffer.c:
+	  (gst_rtp_jitter_buffer_chain),
+	  (gst_rtp_jitter_buffer_set_property):
+	  When drop-on-latency is set but we have no latency configured, just push
+	  the buffer as fast as possible.
+	  Fix typo in comment.
+
+2007-08-21 16:04:47 +0000  Wim Taymans <wim.taymans@gmail.com>
+
+	  gst/rtpmanager/rtpjitterbuffer.*: Fix undefined overflow prone ts_diff handling.
+	  Original commit message from CVS:
+	  * gst/rtpmanager/rtpjitterbuffer.c:
+	  (rtp_jitter_buffer_get_ts_diff):
+	  * gst/rtpmanager/rtpjitterbuffer.h:
+	  Fix undefined overflow prone ts_diff handling.
+
+2007-08-16 11:40:16 +0000  Wim Taymans <wim.taymans@gmail.com>
+
+	  gst/rtpmanager/gstrtpjitterbuffer.c: Fix EOS handling.
+	  Original commit message from CVS:
+	  * gst/rtpmanager/gstrtpjitterbuffer.c:
+	  (gst_rtp_jitter_buffer_sink_event), (gst_rtp_jitter_buffer_chain),
+	  (gst_rtp_jitter_buffer_loop):
+	  Fix EOS handling.
+	  Convert some DEBUG into WARNINGs.
+	  Pause task when flushing.
+	  * gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_class_init),
+	  (rtcp_thread), (gst_rtp_session_event_recv_rtcp_sink):
+	  Use system clock for RTCP session management timeouts.
+	  * gst/rtpmanager/rtpsession.c: (on_new_ssrc), (on_ssrc_collision),
+	  (on_ssrc_validated), (on_bye_ssrc), (on_bye_timeout), (on_timeout):
+	  Release the session lock when emiting signals.
+
+2007-08-13 06:16:40 +0000  Stefan Kost <ensonic@users.sourceforge.net>
+
+	  gst/rtpmanager/rtpjitterbuffer.c: Include stdlib.
+	  Original commit message from CVS:
+	  * gst/rtpmanager/rtpjitterbuffer.c:
+	  Include stdlib.
+
+2007-08-10 17:16:53 +0000  Wim Taymans <wim.taymans@gmail.com>
+
+	  gst/rtpmanager/: Remove complicated async queue and replace with more simple jitterbuffer code while also fixing some...
+	  Original commit message from CVS:
+	  * gst/rtpmanager/Makefile.am:
+	  * gst/rtpmanager/async_jitter_queue.c:
+	  * gst/rtpmanager/async_jitter_queue.h:
+	  * gst/rtpmanager/rtpjitterbuffer.c: (rtp_jitter_buffer_class_init),
+	  (rtp_jitter_buffer_init), (rtp_jitter_buffer_finalize),
+	  (rtp_jitter_buffer_new), (compare_seqnum),
+	  (rtp_jitter_buffer_insert), (rtp_jitter_buffer_pop),
+	  (rtp_jitter_buffer_flush), (rtp_jitter_buffer_num_packets),
+	  (rtp_jitter_buffer_get_ts_diff):
+	  * gst/rtpmanager/rtpjitterbuffer.h:
+	  Remove complicated async queue and replace with more simple jitterbuffer
+	  code while also fixing some bugs.
+	  * gst/rtpmanager/gstrtpbin-marshal.list:
+	  * gst/rtpmanager/gstrtpbin.c: (on_new_ssrc), (on_ssrc_collision),
+	  (on_ssrc_validated), (on_bye_ssrc), (on_bye_timeout), (on_timeout),
+	  (create_session), (gst_rtp_bin_class_init), (create_recv_rtp),
+	  (create_send_rtp):
+	  * gst/rtpmanager/gstrtpbin.h:
+	  * gst/rtpmanager/gstrtpjitterbuffer.c:
+	  (gst_rtp_jitter_buffer_init), (gst_rtp_jitter_buffer_dispose),
+	  (gst_jitter_buffer_sink_parse_caps),
+	  (gst_rtp_jitter_buffer_flush_start),
+	  (gst_rtp_jitter_buffer_flush_stop),
+	  (gst_rtp_jitter_buffer_change_state),
+	  (gst_rtp_jitter_buffer_sink_event), (gst_rtp_jitter_buffer_chain),
+	  (gst_rtp_jitter_buffer_loop), (gst_rtp_jitter_buffer_set_property):
+	  * gst/rtpmanager/gstrtpsession.c: (on_new_ssrc),
+	  (on_ssrc_collision), (on_ssrc_validated), (on_bye_ssrc),
+	  (on_bye_timeout), (on_timeout), (gst_rtp_session_class_init),
+	  (gst_rtp_session_init):
+	  * gst/rtpmanager/gstrtpsession.h:
+	  * gst/rtpmanager/rtpsession.c: (on_bye_ssrc), (session_cleanup):
+	  Use new jitterbuffer code.
+	  Expose some new signals in preparation for handling EOS.
+
+2007-07-18 07:35:32 +0000  Stefan Kost <ensonic@users.sourceforge.net>
+
+	  Add stdlib include (free, atoi, exit).
+	  Original commit message from CVS:
+	  * examples/app/appsrc_ex.c:
+	  * examples/switch/switcher.c:
+	  * ext/neon/gstneonhttpsrc.c:
+	  * ext/timidity/gstwildmidi.c:
+	  * ext/x264/gstx264enc.c:
+	  * gst/mve/mveaudioenc.c: (mve_compress_audio):
+	  * gst/rtpmanager/gstrtpclient.c:
+	  * gst/rtpmanager/gstrtpjitterbuffer.c:
+	  * gst/spectrum/demo-audiotest.c:
+	  * gst/spectrum/demo-osssrc.c:
+	  * sys/dvb/gstdvbsrc.c:
+	  Add stdlib include (free, atoi, exit).
+
+2007-06-22 20:23:18 +0000  Jens Granseuer <jensgr@gmx.net>
+
+	  gst/: Build fixes for gcc-2.9x (no mid-block variable declarations etc.).
+	  Original commit message from CVS:
+	  Patch by: Jens Granseuer  <jensgr at gmx net>
+	  * gst/equalizer/gstiirequalizer.c:
+	  * gst/equalizer/gstiirequalizer10bands.c:
+	  * gst/equalizer/gstiirequalizer3bands.c:
+	  * gst/equalizer/gstiirequalizernbands.c:
+	  * gst/rtpmanager/async_jitter_queue.c:
+	  (async_jitter_queue_push_sorted):
+	  * gst/rtpmanager/gstrtpjitterbuffer.c:
+	  (gst_rtp_jitter_buffer_chain):
+	  * gst/switch/gstswitch.c: (gst_switch_chain):
+	  Build fixes for gcc-2.9x (no mid-block variable declarations etc.).
+	  Fixes #450185.
+
+2007-05-28 16:37:47 +0000  Wim Taymans <wim.taymans@gmail.com>
+
+	  Rename elements to avoid conflict with farsight elements with the same name. Fixes #430664.
+	  Original commit message from CVS:
+	  * docs/plugins/gst-plugins-bad-plugins-docs.sgml:
+	  * docs/plugins/gst-plugins-bad-plugins-sections.txt:
+	  * gst/rtpmanager/gstrtpbin.c: (create_session), (create_stream),
+	  (gst_rtp_bin_class_init), (create_recv_rtp), (create_recv_rtcp),
+	  (create_send_rtp), (create_rtcp), (gst_rtp_bin_request_new_pad):
+	  * gst/rtpmanager/gstrtpclient.c: (create_stream),
+	  (gst_rtp_client_request_new_pad):
+	  * gst/rtpmanager/gstrtpjitterbuffer.c:
+	  (gst_rtp_jitter_buffer_clear_pt_map), (gst_rtp_jitter_buffer_loop):
+	  * gst/rtpmanager/gstrtpmanager.c: (plugin_init):
+	  * gst/rtpmanager/gstrtpptdemux.c:
+	  * gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_class_init),
+	  (gst_rtp_session_request_new_pad):
+	  * gst/rtpmanager/gstrtpssrcdemux.c:
+	  Rename elements to avoid conflict with farsight elements with the same
+	  name. Fixes #430664.
+
+2007-05-23 13:08:52 +0000  Wim Taymans <wim.taymans@gmail.com>
+
+	  Document stuff.
+	  Original commit message from CVS:
+	  * docs/plugins/Makefile.am:
+	  * docs/plugins/gst-plugins-bad-plugins-docs.sgml:
+	  * docs/plugins/gst-plugins-bad-plugins-sections.txt:
+	  * gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_class_init):
+	  * gst/rtpmanager/gstrtpbin.h:
+	  * gst/rtpmanager/gstrtpclient.c:
+	  * gst/rtpmanager/gstrtpjitterbuffer.c:
+	  (gst_rtp_jitter_buffer_class_init),
+	  (gst_rtp_jitter_buffer_clear_pt_map), (gst_rtp_jitter_buffer_loop):
+	  * gst/rtpmanager/gstrtpjitterbuffer.h:
+	  * gst/rtpmanager/gstrtpptdemux.c: (gst_rtp_pt_demux_class_init),
+	  (gst_rtp_pt_demux_clear_pt_map):
+	  * gst/rtpmanager/gstrtpptdemux.h:
+	  * gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_class_init),
+	  (rtcp_thread), (gst_rtp_session_clear_pt_map):
+	  * gst/rtpmanager/gstrtpsession.h:
+	  * gst/rtpmanager/gstrtpssrcdemux.c:
+	  (gst_rtp_ssrc_demux_class_init):
+	  Document stuff.
+	  Add clear-pt-map action signal where needed.
+
+2007-05-15 13:29:53 +0000  Wim Taymans <wim.taymans@gmail.com>
+
+	  gst/rtpmanager/gstrtpptdemux.c: We always use fixed caps.
+	  Original commit message from CVS:
+	  * gst/rtpmanager/gstrtpptdemux.c: (gst_rtp_pt_demux_chain):
+	  We always use fixed caps.
+
+2007-05-15 03:45:45 +0000  David Schleef <ds@schleef.org>
+
+	  gst/rtpmanager/gstrtpbin.c: g_hash_table_remove_all() only exists in 2.12.  Work around.
+	  Original commit message from CVS:
+	  * gst/rtpmanager/gstrtpbin.c:
+	  g_hash_table_remove_all() only exists in 2.12.  Work around.
+
+2007-05-14 15:28:36 +0000  Wim Taymans <wim.taymans@gmail.com>
+
+	  gst/rtpmanager/async_jitter_queue.c: Fix leak when flushing.
+	  Original commit message from CVS:
+	  * gst/rtpmanager/async_jitter_queue.c:
+	  (async_jitter_queue_set_flushing_unlocked):
+	  Fix leak when flushing.
+	  * gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_clear_pt_map),
+	  (gst_rtp_bin_class_init):
+	  * gst/rtpmanager/gstrtpbin.h:
+	  Add clear-pt-map signal.
+	  * gst/rtpmanager/gstrtpjitterbuffer.c:
+	  (gst_rtp_jitter_buffer_flush_stop),
+	  (gst_rtp_jitter_buffer_sink_event), (gst_rtp_jitter_buffer_loop):
+	  Init clock-rate to -1 to mark unknow clock rate.
+	  Fix flushing.
+
+2007-05-10 14:02:07 +0000  Stefan Kost <ensonic@users.sourceforge.net>
+
+	  gst/qtdemux/qtdemux.c (gst_qtdemux_move_stream, gst_qtdemux_loop_state_header, gst_qtdemux_activate_segment, gst_qtde...
+	  Original commit message from CVS:
+	  * gst/qtdemux/qtdemux.c (gst_qtdemux_move_stream,
+	  gst_qtdemux_loop_state_header, gst_qtdemux_activate_segment,
+	  gst_qtdemux_prepare_current_sample, gst_qtdemux_combine_flows,
+	  gst_qtdemux_loop_state_movie, gst_qtdemux_loop,
+	  qtdemux_parse_segments, qtdemux_parse_trak):
+	  * gst/rtpmanager/rtpsession.c (rtp_session_get_bandwidth,
+	  rtp_session_get_rtcp_bandwidth, rtp_session_get_cname,
+	  rtp_session_get_name, rtp_session_get_email, rtp_session_get_phone,
+	  rtp_session_get_location, rtp_session_get_tool,
+	  rtp_session_process_bye, session_report_blocks):
+	  * gst/rtpmanager/rtpsource.c (rtp_source_process_rtp,
+	  rtp_source_send_rtp, rtp_source_process_sr, rtp_source_process_rb):
+	  More format arg fixing (spotted by Ali Sabil <ali.sabil@gmail.com>).
+	  * gst/switch/Makefile.am:
+	  Add require libraries(spotted by Ali Sabil <ali.sabil@gmail.com>).
+
+2007-05-10 12:38:49 +0000  Stefan Kost <ensonic@users.sourceforge.net>
+
+	* gst/rtpmanager/async_jitter_queue.c:
+	  gst/rtpmanager/async_jitter_queue.c (tail_buffer_duration, async_jitter_queue_ref, async_jitter_queue_ref_unlocked, a...
+	  Original commit message from CVS:
+	  * gst/rtpmanager/async_jitter_queue.c (tail_buffer_duration,
+	  async_jitter_queue_ref, async_jitter_queue_ref_unlocked,
+	  async_jitter_queue_set_low_threshold,
+	  async_jitter_queue_length_ts_units_unlocked,
+	  async_jitter_queue_unref_and_unlock, async_jitter_queue_unref,
+	  async_jitter_queue_lock, async_jitter_queue_push,
+	  async_jitter_queue_push_unlocked, async_jitter_queue_push_sorted,
+	  async_jitter_queue_pop_intern_unlocked, async_jitter_queue_pop,
+	  async_jitter_queue_pop_unlocked, async_jitter_queue_length_unlocked,
+	  async_jitter_queue_set_flushing_unlocked,
+	  async_jitter_queue_unset_flushing_unlocked):
+	  Format arg fix (spotted by Ali Sabil <ali.sabil@gmail.com>)
+
+2007-05-09 11:24:22 +0000  Wim Taymans <wim.taymans@gmail.com>
+
+	  gst/rtpmanager/gstrtpjitterbuffer.c: Pass queries upstream.
+	  Original commit message from CVS:
+	  * gst/rtpmanager/gstrtpjitterbuffer.c:
+	  (gst_rtp_jitter_buffer_query):
+	  Pass queries upstream.
+
+2007-05-04 12:32:27 +0000  Wim Taymans <wim.taymans@gmail.com>
+
+	  gst/rtpmanager/gstrtpjitterbuffer.c: Add some debug info.
+	  Original commit message from CVS:
+	  * gst/rtpmanager/gstrtpjitterbuffer.c:
+	  (gst_rtp_jitter_buffer_query):
+	  Add some debug info.
+	  * gst/rtpmanager/rtpsession.c: (rtp_session_init),
+	  (rtp_session_send_rtp):
+	  Store real user name in the session.
+
+2007-04-30 13:41:30 +0000  Wim Taymans <wim.taymans@gmail.com>
+
+	  gst/rtpmanager/async_jitter_queue.c: Fix the case where the buffer underruns and does not block.
+	  Original commit message from CVS:
+	  * gst/rtpmanager/async_jitter_queue.c: (signal_waiting_threads),
+	  (async_jitter_queue_pop_intern_unlocked):
+	  Fix the case where the buffer underruns and does not block.
+	  * gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_base_init),
+	  (create_recv_rtcp), (create_send_rtp), (create_rtcp),
+	  (gst_rtp_bin_request_new_pad):
+	  Rename RTCP send pad, like in the session manager.
+	  Allow getting an RTCP pad for receiving even if we don't receive RTP.
+	  fix handling of send_rtp_src pad.
+	  * gst/rtpmanager/gstrtpptdemux.c: (gst_rtp_pt_demux_chain):
+	  When no pt map could be found, fall back to the sinkpad caps.
+	  * gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_process_rtp),
+	  (gst_rtp_session_send_rtp), (create_recv_rtp_sink),
+	  (create_recv_rtcp_sink), (create_send_rtp_sink),
+	  (create_send_rtcp_src):
+	  Fix pad names.
+	  * gst/rtpmanager/rtpsession.c: (source_push_rtp),
+	  (rtp_session_create_source), (rtp_session_process_sr),
+	  (rtp_session_send_rtp), (session_start_rtcp):
+	  * gst/rtpmanager/rtpsession.h:
+	  Unlock session when performing a callback.
+	  Add callbacks for the internal session object.
+	  Fix sending of RTP packets.
+	  first attempt at adding NTP times in the SR packets.
+	  Small debug and doc improvements.
+	  * gst/rtpmanager/rtpsource.c: (rtp_source_send_rtp):
+	  Update stats for SR reports.
+
+2007-04-29 14:46:27 +0000  Wim Taymans <wim.taymans@gmail.com>
+
+	  gst/rtpmanager/gstrtpsession.c: Remove debug.
+	  Original commit message from CVS:
+	  * gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_send_rtcp):
+	  Remove debug.
+	  * gst/rtpmanager/rtpsession.c: (rtp_session_process_sr),
+	  (rtp_session_process_sdes), (calculate_rtcp_interval),
+	  (rtp_session_next_timeout), (session_report_blocks):
+	  * gst/rtpmanager/rtpstats.c: (rtp_stats_calculate_rtcp_interval):
+	  Improve debugging
+	  Fix interval for BYE/RTCP packets.
+
+2007-04-27 15:09:12 +0000  Wim Taymans <wim.taymans@gmail.com>
+
+	  gst/rtpmanager/gstrtpsession.c: Move reconsideration code to the rtpsession object.
+	  Original commit message from CVS:
+	  * gst/rtpmanager/gstrtpsession.c: (rtcp_thread),
+	  (gst_rtp_session_send_rtcp), (gst_rtp_session_reconsider):
+	  Move reconsideration code to the rtpsession object.
+	  Simplify timout handling and add reconsideration.
+	  * gst/rtpmanager/rtpsession.c: (rtp_session_class_init),
+	  (rtp_session_init), (rtp_session_finalize), (on_bye_ssrc),
+	  (on_bye_timeout), (on_timeout), (rtp_session_set_callbacks),
+	  (obtain_source), (rtp_session_create_source),
+	  (update_arrival_stats), (rtp_session_process_rtp),
+	  (rtp_session_process_sr), (rtp_session_process_rr),
+	  (rtp_session_process_bye), (rtp_session_process_rtcp),
+	  (calculate_rtcp_interval), (rtp_session_send_bye),
+	  (rtp_session_next_timeout), (session_start_rtcp),
+	  (session_report_blocks), (session_cleanup), (session_sdes),
+	  (session_bye), (is_rtcp_time), (rtp_session_on_timeout):
+	  * gst/rtpmanager/rtpsession.h:
+	  Handle timeout of inactive sources and senders.
+	  Implement BYE scheduling.
+	  * gst/rtpmanager/rtpsource.c: (calculate_jitter),
+	  (rtp_source_process_sr), (rtp_source_get_last_sr),
+	  (rtp_source_get_last_rb):
+	  * gst/rtpmanager/rtpsource.h:
+	  Add members to check for timeouts.
+	  * gst/rtpmanager/rtpstats.c: (rtp_stats_init_defaults),
+	  (rtp_stats_calculate_rtcp_interval), (rtp_stats_add_rtcp_jitter),
+	  (rtp_stats_calculate_bye_interval):
+	  * gst/rtpmanager/rtpstats.h:
+	  Use RFC algorithm for calculating the reporting interval.
+
+2007-04-25 16:38:03 +0000  Wim Taymans <wim.taymans@gmail.com>
+
+	  gst/rtpmanager/gstrtpsession.c: Implement forward and reverse reconsideration.
+	  Original commit message from CVS:
+	  * gst/rtpmanager/gstrtpsession.c: (rtcp_thread):
+	  Implement forward and reverse reconsideration.
+	  * gst/rtpmanager/rtpsession.c: (rtp_session_get_num_sources),
+	  (rtp_session_get_num_active_sources), (rtp_session_process_sr),
+	  (session_report_blocks):
+	  * gst/rtpmanager/rtpsession.h:
+	  Small cleanups.
+
+2007-04-25 15:48:46 +0000  Wim Taymans <wim.taymans@gmail.com>
+
+	  gst/rtpmanager/gstrtpbin.*: Make default jitterbuffer latency configurable.
+	  Original commit message from CVS:
+	  reviewed by: <delete if not using a buddy>
+	  * gst/rtpmanager/gstrtpbin.c: (create_stream),
+	  (gst_rtp_bin_class_init), (gst_rtp_bin_set_property),
+	  (gst_rtp_bin_get_property):
+	  * gst/rtpmanager/gstrtpbin.h:
+	  Make default jitterbuffer latency configurable.
+	  * gst/rtpmanager/gstrtpjitterbuffer.c:
+	  (gst_rtp_jitter_buffer_class_init),
+	  (gst_jitter_buffer_sink_parse_caps), (gst_rtp_jitter_buffer_loop),
+	  (gst_rtp_jitter_buffer_set_property),
+	  (gst_rtp_jitter_buffer_get_property):
+	  Debuging cleanups.
+
+2007-04-25 13:19:36 +0000  Wim Taymans <wim.taymans@gmail.com>
+
+	  gst/rtpmanager/gstrtpjitterbuffer.c: Report NO_PREROLL when going to PAUSED.
+	  Original commit message from CVS:
+	  * gst/rtpmanager/gstrtpjitterbuffer.c:
+	  (gst_rtp_jitter_buffer_change_state):
+	  Report NO_PREROLL when going to PAUSED.
+	  * gst/rtpmanager/gstrtpsession.c: (rtcp_thread):
+	  Don't send RTCP right before we are shutting down.
+	  * gst/rtpmanager/rtpsession.c: (rtp_session_process_rtp),
+	  (rtp_session_process_sr), (session_report_blocks),
+	  (rtp_session_perform_reporting):
+	  Improve report blocks.
+	  * gst/rtpmanager/rtpsource.c: (calculate_jitter), (init_seq),
+	  (rtp_source_process_rtp), (rtp_source_process_sr),
+	  (rtp_source_process_rb), (rtp_source_get_last_sr),
+	  (rtp_source_get_last_rb):
+	  * gst/rtpmanager/rtpsource.h:
+	  * gst/rtpmanager/rtpstats.h:
+	  Cleanups, add methods to access stats.
+
+2007-04-25 08:30:48 +0000  Wim Taymans <wim.taymans@gmail.com>
+
+	  gst/rtpmanager/gstrtpbin.c: fix for pad name change
+	  Original commit message from CVS:
+	  * gst/rtpmanager/gstrtpbin.c: (create_rtcp):
+	  fix for pad name change
+	  * gst/rtpmanager/gstrtpsession.c: (rtcp_thread),
+	  (gst_rtp_session_send_rtcp), (gst_rtp_session_clock_rate):
+	  Fix for renamed methods.
+	  * gst/rtpmanager/rtpsession.c: (rtp_session_init),
+	  (rtp_session_finalize), (rtp_session_set_cname),
+	  (rtp_session_get_cname), (rtp_session_set_name),
+	  (rtp_session_get_name), (rtp_session_set_email),
+	  (rtp_session_get_email), (rtp_session_set_phone),
+	  (rtp_session_get_phone), (rtp_session_set_location),
+	  (rtp_session_get_location), (rtp_session_set_tool),
+	  (rtp_session_get_tool), (rtp_session_set_note),
+	  (rtp_session_get_note), (source_push_rtp), (obtain_source),
+	  (rtp_session_add_source), (rtp_session_get_source_by_ssrc),
+	  (rtp_session_create_source), (rtp_session_process_rtp),
+	  (rtp_session_process_sr), (rtp_session_process_sdes),
+	  (rtp_session_process_rtcp), (rtp_session_send_rtp),
+	  (rtp_session_get_reporting_interval), (session_report_blocks),
+	  (session_sdes), (rtp_session_perform_reporting):
+	  * gst/rtpmanager/rtpsession.h:
+	  Prepare for implementing SSRC sampling.
+	  Create SSRC for the session.
+	  Add methods to set the SDES entries.
+	  fix accounting of senders/receivers.
+	  Implement SR/RR/SDES RTCP reporting.
+	  * gst/rtpmanager/rtpsource.c: (rtp_source_init), (init_seq),
+	  (rtp_source_process_rtp), (rtp_source_process_sr):
+	  * gst/rtpmanager/rtpsource.h:
+	  Implement extended sequence number.
+	  * gst/rtpmanager/rtpstats.c: (rtp_stats_calculate_rtcp_interval):
+	  * gst/rtpmanager/rtpstats.h:
+	  Rename some fields.
+
+2007-04-21 19:21:49 +0000  Tim-Philipp Müller <tim@centricular.net>
+
+	  gst/rtpmanager/rtpsession.c: Don't use GLib-2.10 API, we only require GLib 2.8 at the moment.
+	  Original commit message from CVS:
+	  * gst/rtpmanager/rtpsession.c: (rtp_session_finalize):
+	  Don't use GLib-2.10 API, we only require GLib 2.8 at the moment.
+
+2007-04-18 18:58:53 +0000  Wim Taymans <wim.taymans@gmail.com>
+
+	  configure.ac: Disable rtpmanager for now because it depends on CVS -base.
+	  Original commit message from CVS:
+	  * configure.ac:
+	  Disable rtpmanager for now because it depends on CVS -base.
+	  * gst/rtpmanager/Makefile.am:
+	  Added new files for session manager.
+	  * gst/rtpmanager/gstrtpjitterbuffer.h:
+	  * gst/rtpmanager/gstrtpbin.c: (create_session), (get_pt_map),
+	  (create_stream), (pt_map_requested), (new_ssrc_pad_found):
+	  Some cleanups.
+	  the session manager can now also request a pt-map.
+	  * gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_base_init),
+	  (gst_rtp_session_class_init), (gst_rtp_session_init),
+	  (gst_rtp_session_finalize), (rtcp_thread), (start_rtcp_thread),
+	  (stop_rtcp_thread), (gst_rtp_session_change_state),
+	  (gst_rtp_session_process_rtp), (gst_rtp_session_send_rtp),
+	  (gst_rtp_session_send_rtcp), (gst_rtp_session_clock_rate),
+	  (gst_rtp_session_get_time), (gst_rtp_session_event_recv_rtp_sink),
+	  (gst_rtp_session_chain_recv_rtp),
+	  (gst_rtp_session_event_recv_rtcp_sink),
+	  (gst_rtp_session_chain_recv_rtcp),
+	  (gst_rtp_session_event_send_rtp_sink),
+	  (gst_rtp_session_chain_send_rtp), (create_send_rtcp_src),
+	  (gst_rtp_session_request_new_pad):
+	  * gst/rtpmanager/gstrtpsession.h:
+	  We can ask for pt-map now too when the session manager needs it.
+	  Hook up to the new session manager, implement the needed callbacks for
+	  pushing data, getting clock time and requesting clock-rates.
+	  Rename rtcp_src to send_rtcp_src to make it clear that this RTCP is to
+	  be send to clients.
+	  Add code to start and stop the thread that will schedule RTCP through
+	  the session manager.
+	  * gst/rtpmanager/rtpsession.c: (rtp_session_class_init),
+	  (rtp_session_init), (rtp_session_finalize),
+	  (rtp_session_set_property), (rtp_session_get_property),
+	  (on_new_ssrc), (on_ssrc_collision), (on_ssrc_validated),
+	  (on_bye_ssrc), (rtp_session_new), (rtp_session_set_callbacks),
+	  (rtp_session_set_bandwidth), (rtp_session_get_bandwidth),
+	  (rtp_session_set_rtcp_bandwidth), (rtp_session_get_rtcp_bandwidth),
+	  (source_push_rtp), (source_clock_rate), (check_collision),
+	  (obtain_source), (rtp_session_add_source),
+	  (rtp_session_get_num_sources),
+	  (rtp_session_get_num_active_sources),
+	  (rtp_session_get_source_by_ssrc),
+	  (rtp_session_get_source_by_cname), (rtp_session_create_source),
+	  (update_arrival_stats), (rtp_session_process_rtp),
+	  (rtp_session_process_sr), (rtp_session_process_rr),
+	  (rtp_session_process_sdes), (rtp_session_process_bye),
+	  (rtp_session_process_app), (rtp_session_process_rtcp),
+	  (rtp_session_send_rtp), (rtp_session_get_rtcp_interval),
+	  (rtp_session_produce_rtcp):
+	  * gst/rtpmanager/rtpsession.h:
+	  The advanced beginnings of the main session manager that handles the
+	  participant database of RTPSources, SSRC probation, SSRC collisions,
+	  parse RTCP to update source stats. etc..
+	  * gst/rtpmanager/rtpsource.c: (rtp_source_class_init),
+	  (rtp_source_init), (rtp_source_finalize), (rtp_source_new),
+	  (rtp_source_set_callbacks), (rtp_source_set_as_csrc),
+	  (rtp_source_set_rtp_from), (rtp_source_set_rtcp_from),
+	  (push_packet), (get_clock_rate), (calculate_jitter),
+	  (rtp_source_process_rtp), (rtp_source_process_bye),
+	  (rtp_source_send_rtp), (rtp_source_process_sr),
+	  (rtp_source_process_rb):
+	  * gst/rtpmanager/rtpsource.h:
+	  Object that encapsulates an SSRC and its state in the database.
+	  Calculates the jitter and transit times of data packets.
+	  * gst/rtpmanager/rtpstats.c: (rtp_stats_init_defaults),
+	  (rtp_stats_calculate_rtcp_interval), (rtp_stats_add_rtcp_jitter):
+	  * gst/rtpmanager/rtpstats.h:
+	  Various stats regarding the session and sources.
+	  Used to calculate the RTCP interval.
+
+2007-04-13 09:20:55 +0000  Wim Taymans <wim.taymans@gmail.com>
+
+	  gst/rtpmanager/: Protect lists and structures with locks.
+	  Original commit message from CVS:
+	  * gst/rtpmanager/gstrtpbin.c: (create_session), (get_pt_map),
+	  (gst_rtp_bin_init), (gst_rtp_bin_finalize), (new_ssrc_pad_found),
+	  (create_recv_rtp), (gst_rtp_bin_request_new_pad):
+	  * gst/rtpmanager/gstrtpbin.h:
+	  * gst/rtpmanager/gstrtpclient.c:
+	  * gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_class_init),
+	  (gst_rtp_session_init), (gst_rtp_session_finalize),
+	  (gst_rtp_session_event_recv_rtp_sink),
+	  (gst_rtp_session_event_recv_rtcp_sink),
+	  (gst_rtp_session_chain_recv_rtcp),
+	  (gst_rtp_session_request_new_pad):
+	  Protect lists and structures with locks.
+	  Return FLOW_OK from RTCP messages for now.
+
+2007-04-12 08:18:32 +0000  Wim Taymans <wim.taymans@gmail.com>
+
+	  gst/rtpmanager/gstrtpbin.c: Emit pt map requests and cache results.
+	  Original commit message from CVS:
+	  * gst/rtpmanager/gstrtpbin.c: (create_session), (get_pt_map),
+	  (create_stream), (gst_rtp_bin_class_init), (pt_map_requested):
+	  Emit pt map requests and cache results.
+	  * gst/rtpmanager/gstrtpjitterbuffer.c:
+	  (gst_rtp_jitter_buffer_class_init),
+	  (gst_jitter_buffer_sink_parse_caps),
+	  (gst_jitter_buffer_sink_setcaps),
+	  (gst_rtp_jitter_buffer_get_clock_rate),
+	  (gst_rtp_jitter_buffer_chain), (gst_rtp_jitter_buffer_loop):
+	  * gst/rtpmanager/gstrtpjitterbuffer.h:
+	  * gst/rtpmanager/gstrtpptdemux.c: (gst_rtp_pt_demux_chain):
+	  Emit request-pt-map signals.
+
+2007-04-11 13:49:54 +0000  Wim Taymans <wim.taymans@gmail.com>
+
+	  gst/rtpmanager/gstrtpbin-marshal.list: Some more custom marshallers.
+	  Original commit message from CVS:
+	  * gst/rtpmanager/gstrtpbin-marshal.list:
+	  Some more custom marshallers.
+	  * gst/rtpmanager/gstrtpbin.c: (create_session), (get_pt_map),
+	  (clock_rate_request), (create_stream), (gst_rtp_bin_class_init),
+	  (pt_map_requested), (new_ssrc_pad_found), (create_recv_rtp):
+	  * gst/rtpmanager/gstrtpbin.h:
+	  Prepare for caching pt maps.
+	  Connect to signals to collect pt maps.
+	  * gst/rtpmanager/gstrtpjitterbuffer.c:
+	  (gst_rtp_jitter_buffer_class_init),
+	  (gst_jitter_buffer_sink_setcaps), (gst_rtp_jitter_buffer_loop):
+	  * gst/rtpmanager/gstrtpjitterbuffer.h:
+	  Add request_clock_rate signal.
+	  Use scale insteat of scale_int because the later does not deal with
+	  negative numbers.
+	  * gst/rtpmanager/gstrtpptdemux.c: (gst_rtp_pt_demux_class_init),
+	  (gst_rtp_pt_demux_chain):
+	  * gst/rtpmanager/gstrtpptdemux.h:
+	  Implement request-pt-map signal.
+
+2007-04-10 09:14:07 +0000  Wim Taymans <wim.taymans@gmail.com>
+
+	  gst/rtpmanager/: Added custom marshallers for signals.
+	  Original commit message from CVS:
+	  * gst/rtpmanager/.cvsignore:
+	  * gst/rtpmanager/Makefile.am:
+	  * gst/rtpmanager/gstrtpbin-marshal.list:
+	  Added custom marshallers for signals.
+	  * gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_class_init):
+	  * gst/rtpmanager/gstrtpbin.h:
+	  Prepare for emiting pt map signals.
+	  * gst/rtpmanager/gstrtpptdemux.c: (gst_rtp_pt_demux_class_init):
+	  * gst/rtpmanager/gstrtpssrcdemux.c:
+	  (gst_rtp_ssrc_demux_class_init):
+	  Fix signals.
+
+2007-04-06 12:28:29 +0000  Wim Taymans <wim.taymans@gmail.com>
+
+	  gst/rtpmanager/gstrtpbin.*: Provide a clock.
+	  Original commit message from CVS:
+	  * gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_class_init),
+	  (gst_rtp_bin_init), (gst_rtp_bin_provide_clock):
+	  * gst/rtpmanager/gstrtpbin.h:
+	  Provide a clock.
+
+2007-04-06 12:07:30 +0000  Wim Taymans <wim.taymans@gmail.com>
+
+	  gst/rtpmanager/gstrtpbin.c: Fix pad template name parsing.
+	  Original commit message from CVS:
+	  * gst/rtpmanager/gstrtpbin.c: (create_rtcp):
+	  Fix pad template name parsing.
+
+2007-04-05 16:10:24 +0000  Wim Taymans <wim.taymans@gmail.com>
+
+	  gst/rtpmanager/gstrtpjitterbuffer.c: Add some debug and comments.
+	  Original commit message from CVS:
+	  * gst/rtpmanager/gstrtpjitterbuffer.c:
+	  (gst_jitter_buffer_sink_setcaps), (gst_rtp_jitter_buffer_chain),
+	  (gst_rtp_jitter_buffer_loop):
+	  Add some debug and comments.
+	  Fix double unref() in error cases.
+
+2007-04-05 13:54:23 +0000  Wim Taymans <wim.taymans@gmail.com>
+
+	  gst/rtpmanager/gstrtpbin.*: Add debugging category.
+	  Original commit message from CVS:
+	  * gst/rtpmanager/gstrtpbin.c: (find_session_by_id),
+	  (create_session), (find_stream_by_ssrc), (create_stream),
+	  (gst_rtp_bin_class_init), (new_payload_found),
+	  (new_ssrc_pad_found), (create_recv_rtp), (create_recv_rtcp),
+	  (create_send_rtp), (create_rtcp):
+	  * gst/rtpmanager/gstrtpbin.h:
+	  Add debugging category.
+	  Added RTPStream to manage stream per SSRC, each with its own
+	  jitterbuffer and ptdemux.
+	  Added SSRCDemux.
+	  Connect to various SSRC and PT signals and create ghostpads, link stuff.
+	  * gst/rtpmanager/gstrtpmanager.c: (plugin_init):
+	  Added rtpbin to elements.
+	  * gst/rtpmanager/gstrtpptdemux.c: (gst_rtp_pt_demux_chain):
+	  Fix caps and forward GstFlowReturn
+	  * gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_class_init),
+	  (gst_rtp_session_event_recv_rtp_sink),
+	  (gst_rtp_session_chain_recv_rtp),
+	  (gst_rtp_session_event_recv_rtcp_sink),
+	  (gst_rtp_session_chain_recv_rtcp),
+	  (gst_rtp_session_event_send_rtp_sink),
+	  (gst_rtp_session_chain_send_rtp), (create_recv_rtp_sink),
+	  (create_recv_rtcp_sink), (create_send_rtp_sink), (create_rtcp_src),
+	  (gst_rtp_session_request_new_pad):
+	  Add debug category.
+	  Add event handling
+	  * gst/rtpmanager/gstrtpssrcdemux.c: (find_rtp_pad_for_ssrc),
+	  (create_rtp_pad_for_ssrc), (gst_rtp_ssrc_demux_class_init),
+	  (gst_rtp_ssrc_demux_init), (gst_rtp_ssrc_demux_chain),
+	  (gst_rtp_ssrc_demux_change_state):
+	  * gst/rtpmanager/gstrtpssrcdemux.h:
+	  Add debug category.
+	  Add new-pt-pad signal.
+
+2007-04-04 10:23:15 +0000  Wim Taymans <wim.taymans@gmail.com>
+
+	  gst/rtpmanager/: Added simple SSRC demuxer.
+	  Original commit message from CVS:
+	  * gst/rtpmanager/Makefile.am:
+	  * gst/rtpmanager/gstrtpmanager.c: (plugin_init):
+	  * gst/rtpmanager/gstrtpssrcdemux.c: (find_pad_for_ssrc),
+	  (create_pad_for_ssrc), (gst_rtp_ssrc_demux_base_init),
+	  (gst_rtp_ssrc_demux_class_init), (gst_rtp_ssrc_demux_init),
+	  (gst_rtp_ssrc_demux_finalize), (gst_rtp_ssrc_demux_sink_event),
+	  (gst_rtp_ssrc_demux_chain), (gst_rtp_ssrc_demux_src_event),
+	  (gst_rtp_ssrc_demux_change_state):
+	  * gst/rtpmanager/gstrtpssrcdemux.h:
+	  Added simple SSRC demuxer.
+
+2007-04-03 11:35:39 +0000  Wim Taymans <wim.taymans@gmail.com>
+
+	  gst/rtpmanager/: Some more ghostpad magic.
+	  Original commit message from CVS:
+	  * gst/rtpmanager/gstrtpbin.c: (find_session_by_id),
+	  (create_session), (gst_rtp_bin_base_init), (create_recv_rtp),
+	  (create_recv_rtcp), (create_send_rtp), (create_rtcp),
+	  (gst_rtp_bin_request_new_pad):
+	  * gst/rtpmanager/gstrtpbin.h:
+	  * gst/rtpmanager/gstrtpclient.c:
+	  Some more ghostpad magic.
+
+2007-04-03 09:51:13 +0000  Wim Taymans <wim.taymans@gmail.com>
+
+	  gst/rtpmanager/Makefile.am: Add .h file so it can be disted properly.
+	  Original commit message from CVS:
+	  * gst/rtpmanager/Makefile.am:
+	  Add .h file so it can be disted properly.
+
+2007-04-03 09:13:17 +0000  Wim Taymans <wim.taymans@gmail.com>
+
+	  Add RTP session management elements. Still in progress.
+	  Original commit message from CVS:
+	  * configure.ac:
+	  * gst/rtpmanager/Makefile.am:
+	  * gst/rtpmanager/async_jitter_queue.c: (async_jitter_queue_new),
+	  (signal_waiting_threads), (async_jitter_queue_ref),
+	  (async_jitter_queue_ref_unlocked),
+	  (async_jitter_queue_set_low_threshold),
+	  (async_jitter_queue_set_high_threshold),
+	  (async_jitter_queue_set_max_queue_length),
+	  (async_jitter_queue_get_g_queue), (calculate_ts_diff),
+	  (async_jitter_queue_length_ts_units_unlocked),
+	  (async_jitter_queue_unref_and_unlock), (async_jitter_queue_unref),
+	  (async_jitter_queue_lock), (async_jitter_queue_unlock),
+	  (async_jitter_queue_push), (async_jitter_queue_push_unlocked),
+	  (async_jitter_queue_push_sorted),
+	  (async_jitter_queue_push_sorted_unlocked),
+	  (async_jitter_queue_insert_after_unlocked),
+	  (async_jitter_queue_pop_intern_unlocked), (async_jitter_queue_pop),
+	  (async_jitter_queue_pop_unlocked), (async_jitter_queue_length),
+	  (async_jitter_queue_length_unlocked),
+	  (async_jitter_queue_set_flushing_unlocked),
+	  (async_jitter_queue_unset_flushing_unlocked),
+	  (async_jitter_queue_set_blocking_unlocked):
+	  * gst/rtpmanager/async_jitter_queue.h:
+	  * gst/rtpmanager/gstrtpbin.c: (gst_rtp_bin_base_init),
+	  (gst_rtp_bin_class_init), (gst_rtp_bin_init),
+	  (gst_rtp_bin_finalize), (gst_rtp_bin_set_property),
+	  (gst_rtp_bin_get_property), (gst_rtp_bin_change_state),
+	  (gst_rtp_bin_request_new_pad), (gst_rtp_bin_release_pad):
+	  * gst/rtpmanager/gstrtpbin.h:
+	  * gst/rtpmanager/gstrtpclient.c: (new_pad), (create_stream),
+	  (free_stream), (find_stream_by_ssrc), (gst_rtp_client_base_init),
+	  (gst_rtp_client_class_init), (gst_rtp_client_init),
+	  (gst_rtp_client_finalize), (gst_rtp_client_set_property),
+	  (gst_rtp_client_get_property), (gst_rtp_client_change_state),
+	  (gst_rtp_client_request_new_pad), (gst_rtp_client_release_pad):
+	  * gst/rtpmanager/gstrtpclient.h:
+	  * gst/rtpmanager/gstrtpjitterbuffer.c:
+	  (gst_rtp_jitter_buffer_base_init),
+	  (gst_rtp_jitter_buffer_class_init), (gst_rtp_jitter_buffer_init),
+	  (gst_rtp_jitter_buffer_dispose), (gst_rtp_jitter_buffer_getcaps),
+	  (gst_jitter_buffer_sink_setcaps), (free_func),
+	  (gst_rtp_jitter_buffer_flush_start),
+	  (gst_rtp_jitter_buffer_flush_stop),
+	  (gst_rtp_jitter_buffer_src_activate_push),
+	  (gst_rtp_jitter_buffer_change_state), (priv_compare_rtp_seq_lt),
+	  (compare_rtp_buffers_seq_num), (gst_rtp_jitter_buffer_sink_event),
+	  (gst_rtp_jitter_buffer_chain), (gst_rtp_jitter_buffer_loop),
+	  (gst_rtp_jitter_buffer_query),
+	  (gst_rtp_jitter_buffer_set_property),
+	  (gst_rtp_jitter_buffer_get_property):
+	  * gst/rtpmanager/gstrtpjitterbuffer.h:
+	  * gst/rtpmanager/gstrtpmanager.c: (plugin_init):
+	  * gst/rtpmanager/gstrtpptdemux.c: (gst_rtp_pt_demux_base_init),
+	  (gst_rtp_pt_demux_class_init), (gst_rtp_pt_demux_init),
+	  (gst_rtp_pt_demux_finalize), (gst_rtp_pt_demux_chain),
+	  (gst_rtp_pt_demux_getcaps), (find_pad_for_pt),
+	  (gst_rtp_pt_demux_setup), (gst_rtp_pt_demux_release),
+	  (gst_rtp_pt_demux_change_state):
+	  * gst/rtpmanager/gstrtpptdemux.h:
+	  * gst/rtpmanager/gstrtpsession.c: (gst_rtp_session_base_init),
+	  (gst_rtp_session_class_init), (gst_rtp_session_init),
+	  (gst_rtp_session_finalize), (gst_rtp_session_set_property),
+	  (gst_rtp_session_get_property), (gst_rtp_session_change_state),
+	  (gst_rtp_session_chain_recv_rtp),
+	  (gst_rtp_session_chain_recv_rtcp),
+	  (gst_rtp_session_chain_send_rtp), (create_recv_rtp_sink),
+	  (create_recv_rtcp_sink), (create_send_rtp_sink), (create_rtcp_src),
+	  (gst_rtp_session_request_new_pad), (gst_rtp_session_release_pad):
+	  * gst/rtpmanager/gstrtpsession.h:
+	  Add RTP session management elements. Still in progress.
+
+2009-08-10 13:30:23 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
+
+	* gst/avi/gstavidemux.c:
+	  avidemux: push mode; cater for chunk padding
+
+2009-08-04 19:45:43 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
+
+	* gst/avi/gstavidemux.c:
+	  avidemux: only use stream's pad after having checked it exists
+
+2009-08-04 13:38:09 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
+
+	* gst/avi/gstavidemux.c:
+	  avidemux: sprinkle some more GST_DEBUG_FUNCPTR
+
+2009-08-04 13:36:36 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
+
+	* gst/avi/gstavidemux.c:
+	  avidemux: post error message if no pads to push EOS event on
+
+2009-08-04 11:39:59 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
+
+	* gst/avi/gstavidemux.c:
+	  avidemux: fix typo in warning message
+
+2009-08-04 11:39:39 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
+
+	* gst/avi/gstavidemux.c:
+	  avidemux: fix some buffer ref handling
+
+2009-08-04 11:37:16 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
+
+	* gst/avi/gstavidemux.c:
+	  avidemux: do not exceed maximum number of supported streams
+
+2009-08-04 11:35:18 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
+
+	* gst/avi/gstavidemux.c:
+	  avidemux: prevent double unref; gst_avi_demux_parse_avih already unrefs
+
+2009-08-04 11:32:27 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
+
+	* gst/avi/gstavidemux.c:
+	  avidemux: verify size of INFO LIST to satisfy subsequent expectations
+
+2009-07-29 15:25:38 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
+
+	* gst/avi/gstavidemux.c:
+	  avidemux: check video stream framerate against avi header frame duration
+	  The former might be bogus in silly cases, and the latter seems to
+	  carry more weight.
+
+2009-08-04 12:16:13 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
+
+	* gst/avi/gstavidemux.c:
+	  avidemux: streamline stream duration calculation
+
+2009-07-03 14:04:13 +0200  Edward Hervey <bilboed@bilboed.com>
+
+	* ext/raw1394/gstdv1394src.c:
+	  dv1394src: Fix element for live usage... which has been broken for 2 years :(
+	  This is a live source, therefore:
+	  * Use GST_FORMAT_TIME as the default format
+	  * set_timestamp to True
+	  * properly implement query latency.
+	  This allows expected live usage like : playbin2 uri=dv://
+
+2009-08-09 09:43:41 +0200  Edward Hervey <bilboed@bilboed.com>
+
+	* ext/raw1394/gstdv1394src.c:
+	  raw1394: Remove unneeded variable
+
+2009-08-09 09:43:29 +0200  Edward Hervey <bilboed@bilboed.com>
+
+	* gst/matroska/matroska-demux.c:
+	  matroska: remove dead assignments
+
+2009-08-09 09:43:00 +0200  Edward Hervey <bilboed@bilboed.com>
+
+	* gst/rtp/gstrtpac3depay.c:
+	* gst/rtp/gstrtpceltdepay.c:
+	* gst/rtp/gstrtpj2kdepay.c:
+	* gst/rtp/gstrtpj2kpay.c:
+	  rtp: Remove dead assignments and resulting unneeded variables.
+
+2009-08-10 09:53:28 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+	* configure.ac:
+	* ext/wavpack/Makefile.am:
+	* ext/wavpack/gstwavpackenc.c:
+	* ext/wavpack/gstwavpackenc.h:
+	* ext/wavpack/md5.c:
+	* ext/wavpack/md5.h:
+	  wavpack: Use GLib GChecksum instead of our own MD5 implementation
+	  This requires GLib 2.16 but that version is already required by core anyway.
+
+2009-08-08 00:47:48 -0300  Thiago Santos <thiagoss@embedded.ufcg.edu.br>
+
+	* gst/matroska/matroska-demux.c:
+	* gst/matroska/matroska-mux.c:
+	* gst/matroska/matroska-mux.h:
+	  matroska: Adds support to muxing/demuxing WMA
+	  Adds support for muxing wma audio family and fixes
+	  demuxing of wma family in matroskademux. matroskademux
+	  was broken because it missed codec_data.
+
+2009-08-06 20:15:17 -0300  Thiago Santos <thiagoss@embedded.ufcg.edu.br>
+
+	* gst/matroska/matroska-mux.c:
+	  matroskamux: adds support for wmv family
+	  Adds support to WMV1, WMV2, WMV3 and other family formats that
+	  are signaled by the 'format' field in the caps (i.e. WVC1).
+	  Partially fixes #576378
+
+2009-08-09 14:19:42 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+	* sys/v4l2/gstv4l2object.c:
+	  v4l2src: if max == min width/height put an int in the probed caps, not an int range
+	  Fixes #560033.
+
+2009-08-09 13:58:07 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+	* sys/osxaudio/gstosxaudiosrc.c:
+	  osxaudiosrc: if max_channels == min_channels, use an int instead of an int range in the caps
+
+2009-08-09 12:52:17 +0200  LoneStar <lone@auvtech.com>
+
+	* gst/id3demux/id3v2frames.c:
+	  id3demux: Try GST_*_TAG_ENCODING and locale encoding if tags are not UTF8
+	  Fixes bug #499242.
+
+2009-08-09 01:29:50 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+	* configure.ac:
+	  configure: bump core/base requirements to latest release
+	  To avoid confusion.
+
+2009-08-09 01:27:01 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+	* tests/check/elements/flvmux.c:
+	  check: fix flvmux unit test on big endian machines
+	  flvmux only accepts raw audio in little endian, but audiotestsrc
+	  produces audio in the native endianness, which makes linking
+	  between audiotestsrc and flvmux fail on big endian machines. Add
+	  an audioconvert element in between the two to fix this.
+
+2009-02-15 18:49:44 +0000  Vincent Penquerc'h <ogg.k.ogg.k@googlemail.com>
+
+	* gst/matroska/matroska-demux.c:
+	* gst/matroska/matroska-ids.h:
+	* gst/matroska/matroska-mux.c:
+	  matroska: add kate subtitle support to matroska muxer and demuxer
+	  See #525743.
+
+2009-08-07 16:51:45 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+	* gst/id3demux/id3v2.3.0.html:
+	  id3demux: add ID3 v2.3 spec as well
+
+2009-08-07 16:42:39 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+	* gst/id3demux/id3v2frames.c:
+	  id3demux: sizes in ID3 v2.3 are unlikely to be sync-safe integers
+	  In ID3 v2.3 compressed frames will have a 4-byte data length indicator
+	  after the frame header to indicate the size of the decompressed data.
+	  This integer is unlikely to be a sync-safe integer for v2.3 tags,
+	  only in v2.4 it's sync-safe.
+
+2009-08-07 16:36:55 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+	* gst/id3demux/id3tags.c:
+	  id3demux: fix typo in debug message
+
+2009-08-07 16:02:23 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+	* gst/id3demux/id3tags.c:
+	* gst/id3demux/id3tags.h:
+	* gst/id3demux/id3v2frames.c:
+	* tests/check/elements/id3demux.c:
+	* tests/files/Makefile.am:
+	* tests/files/id3-588148-unsynced-v24.tag:
+	  id3demux: fix parsing of unsync'ed ID3 v2.4 tags and frames
+	  Reversing the unsynchronisation seems to work slightly differently
+	  for ID3 v2.3 tags and v2.4 tags: v2.3 tags don't have syncsafe frame
+	  sizes in the frame header, so the unsynchronisation is applied to
+	  the whole frame data including all the frame headers. v2.4 frames
+	  have sync-safe sizes, however, so the unsynchronisation only needs
+	  to be applied to the actual frame data, and it seems that's what's
+	  being done as well. So we need to undo the unsynchronisation on a
+	  per-frame basis for v2.4 tags for things to work properly.
+	  Fixes extraction of coverart/images from APIC frames in ID3 v2.4
+	  tags (#588148).
+	  Add unit test for this as well.
+
+2009-08-06 21:24:14 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+	* ext/soup/gstsouphttpsrc.c:
+	  souphttpsrc: Use SOUP_METHOD_GET instead of "GET" string
+	  Fixes bug #590970.
+
+2009-08-06 13:00:59 +0200  Wim Taymans <wim.taymans@collabora.co.uk>
+
+	* ext/pulse/pulsesrc.c:
+	  pulsesrc: set the default slave method to skew
+	  Set the default slave method to the much better skew algorithm. This is the
+	  default in the new base class but we override this here as well for the
+	  upcomming release.
+
+2009-08-06 10:20:34 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+	* ext/pulse/pulsesrc.c:
+	  pulsesrc: fix compilation with --disable-gst-debug
+
+2009-08-03 18:59:32 +0200  Wim Taymans <wim.taymans@collabora.co.uk>
+
+	* gst/rtp/gstrtph264pay.c:
+	* gst/rtp/gstrtph264pay.h:
+	  rtph264pay: use array instead of queue
+
+2009-08-03 18:55:19 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
+
+	* gst/rtp/gstrtph264pay.c:
+	* gst/rtp/gstrtph264pay.h:
+	  rtph264pay: push NALs only after SPS/PPS
+	  parse complete (bytestream) buffer for SPS/PPS before pushing NALs.
+	  Fixes #564501.
+
+2009-08-04 14:44:36 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+	* sys/v4l2/v4l2_calls.h:
+	  v4l2: Directly use GST_PTR_FORMAT for printing caps with the LOG_CAPS macro
+
+2009-08-04 11:17:17 +0200  Edward Hervey <bilboed@bilboed.com>
+
+	* gst/rtp/gstrtpqdmdepay.c:
+	  rtpqdm2depay: Fix debug statement.
+
+2009-08-04 09:32:07 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+	* sys/v4l2/gstv4l2sink.c:
+	* sys/v4l2/v4l2_calls.h:
+	  v4l2: Remove some OMAP specific hacks
+	  They require special build flags and are not useful in general.
+
+2009-08-04 09:22:29 +0200  Rob Clark <rob@ti.com>
+
+	* sys/v4l2/gstv4l2bufferpool.c:
+	* sys/v4l2/gstv4l2bufferpool.h:
+	* sys/v4l2/gstv4l2sink.c:
+	* sys/v4l2/v4l2src_calls.c:
+	  v4l2sink: change where buffers get dequeued
+	  It seems to cause strange occasional high latencies (almost 200ms) when dequeuing buffers from _buffer_alloc().  It is simpler and seems to work much better to dqbuf from the same thread that is queuing the next buffer.
+
+2009-08-04 09:14:20 +0200  Rob Clark <rob@ti.com>
+
+	* sys/v4l2/Makefile.am:
+	* sys/v4l2/gstv4l2.c:
+	* sys/v4l2/gstv4l2bufferpool.c:
+	* sys/v4l2/gstv4l2bufferpool.h:
+	* sys/v4l2/gstv4l2object.c:
+	* sys/v4l2/gstv4l2object.h:
+	* sys/v4l2/gstv4l2sink.c:
+	* sys/v4l2/gstv4l2sink.h:
+	* sys/v4l2/gstv4l2src.c:
+	* sys/v4l2/gstv4l2src.h:
+	* sys/v4l2/v4l2_calls.c:
+	* sys/v4l2/v4l2_calls.h:
+	* sys/v4l2/v4l2src_calls.c:
+	* sys/v4l2/v4l2src_calls.h:
+	  v4l2: Add v4l2sink element
+	  This also does the following changes:
+	  (1) pull the bufferpool code out into gstv4l2bufferpool.c, and make a
+	  bit more generic so it can be used both for v4l2src and v4l2sink
+	  (2) move some of the device probing/configuration/caps stuff into
+	  gstv4l2object.c so it does not have to be duplicated between
+	  v4l2src and v4l2sink
+	  Fixes bug #590280.
+
+2009-08-04 07:07:45 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+	* tests/check/Makefile.am:
+	  flvmux: Enable unit test now that it passes
+
+2009-08-03 21:21:39 +0200  Edward Hervey <bilboed@bilboed.com>
+
+	* gst/rtp/gstrtpqdmdepay.c:
+	* gst/rtp/gstrtpsv3vdepay.c:
+	  rtpqdm2depay,rtpsv3vdepay: Add debugging category.
+
+2009-08-03 21:22:48 +0200  Edward Hervey <bilboed@bilboed.com>
+
+	* gst/rtp/gstrtpqdmdepay.c:
+	* gst/rtp/gstrtpqdmdepay.h:
+	  rtpqdm2depay: Handle gaps in incoming packets.
+	  Whenever we see a gap, we flush the temporary packets (but not the adapter). If we
+	  had some data temporarily stored it will be outputted (the sound will sound a bit
+	  garbled... but that's how it sounds on MacOSX :)
+
+2009-08-03 19:01:07 +0200  Edward Hervey <bilboed@bilboed.com>
+
+	* gst/rtp/gstrtpqdmdepay.c:
+	  rtpqdmdepay: Fix CRC calculation and remove commented code.
+
+2009-08-02 13:42:12 +0200  Edward Hervey <bilboed@bilboed.com>
+
+	* gst/rtp/Makefile.am:
+	* gst/rtp/gstrtp.c:
+	* gst/rtp/gstrtpqdmdepay.c:
+	* gst/rtp/gstrtpqdmdepay.h:
+	  rtp: New QDM2 rtp depayloader.
+	  Reverse-engineered by comparing:
+	  * A rtp hinted file provided by DarwinStreamingServer
+	  * The output procued by DSS for that same file
+	  Also used various streaming sources available on the internet to fine-tune
+	  the code.
+	  The header/codec_data extraction methods are from FFMpeg (LGPL).
+
+2009-08-03 21:24:44 +0200  Edward Hervey <bilboed@bilboed.com>
+
+	* gst/rtp/gstrtpsv3vdepay.c:
+	  rtpsv3vdepay: Properly fill codec_data and cleanup code a bite more.
+
+2009-08-03 19:02:17 +0200  Edward Hervey <bilboed@bilboed.com>
+
+	* gst/rtp/gstrtpsv3vdepay.c:
+	* gst/rtp/gstrtpsv3vdepay.h:
+	  rtpsv3vdepay: Only output buffers once we're configured.
+
+2009-08-03 19:02:00 +0200  Edward Hervey <bilboed@bilboed.com>
+
+	* gst/rtp/gstrtpsv3vdepay.c:
+	  rtpsv3vdepay: Add more encoding-name variants
+
+2009-08-03 20:08:33 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+	* tests/check/elements/flvmux.c:
+	  flvmux: Fix unit test to correctly handle request pads
+	  Request pads are removed by the element instance in PAUSED->READY
+	  so we need to re-request pads for every run and link them again.
+	  Last fix for bug #590447.
+
+2009-08-03 20:08:00 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+	* gst/flv/gstflvmux.c:
+	  flvmux: Fix writing of the index for < 128 buffers
+	  Partially fixes bug #590447.
+
+2009-08-03 20:07:00 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+	* gst/flv/gstflvmux.c:
+	  flvmux: Fix resetting of the element
+	  Reset the have_video/have_audio flags and make sure to
+	  properly release the request pads.
+	  Partially fixes bug #590447.
+
+2009-08-03 18:13:46 +0200  Wim Taymans <wim.taymans@collabora.co.uk>
+
+	* gst/rtsp/gstrtspsrc.c:
+	  rtspsrc: don't add non-utf8 chars to structures
+
+2009-08-03 18:02:31 +0200  Luc Deschenaux <luc.deschenaux at freesurf.ch>
+
+	* gst/rtp/gstrtpjpegdepay.c:
+	* gst/rtp/gstrtpjpegdepay.h:
+	  jpegdepay: use attributes for extra properties
+	  Use some of the SDP attributes when they are present to specify the output
+	  dimension and framerate. This allows us to receive jpeg frames larger than
+	  2040 width/height.
+	  Fixes #564437
+
+2009-08-03 18:01:27 +0200  Wim Taymans <wim.taymans@collabora.co.uk>
+
+	* gst/rtp/README:
+	  RTP docs: update with attributes in caps
+
+2009-08-03 17:21:44 +0200  Luc Deschenaux <luc.deschenaux at freesurf.ch>
+
+	* gst/rtsp/gstrtspsrc.c:
+	  rtspsrc: put all SDP attributes on caps
+	  Put the SDP attributes on the caps too so that they can be used by
+	  depayloaders.
+	  See #564437
+
+2009-08-03 13:32:12 +0200  Jonathan Tellier <jonathan.tellier at gmail.com>
+
+	* ext/pulse/pulsesrc.c:
+	  pulsesrc: initialize the probe with the server
+	  When creating a new probe, pass the server instead of the device string.
+	  fixes #590401
+
+2009-08-02 11:44:03 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+	* gst/udp/gstmultiudpsink.c:
+	  multiudpsink: don't do things with side-effects inside g_return_val_if_fail()
+	  Someone might compile this code with -DG_DISABLE_ASSERT some day.
+
+2009-08-01 21:39:30 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+	* ext/pulse/pulsesink.c:
+	  pulsesink: don't do logic within g_assert() statements
+	  Otherwise that code will just be expanded to nothing when compiled
+	  -DG_DISABLE_ASSERT (PS: why is mainloop_start() called in the init
+	  function and not when changing state to READY?)
+
+2009-08-01 17:07:42 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+	* ext/flac/gstflacdec.c:
+	  flacdec: send newsegment event when operating push-based and unframed
+	  For some reason flac doesn't call our metadata callback when we operate
+	  in push mode with unframed input, but that's where we set up the
+	  newsegment event (since that's where we'd get the duration from the
+	  stream info header), so we didn't send a newsegment event at all in this
+	  case. Hack around this by storing a generic newsegment event for now
+	  which will be used if we don't replace it with a better one that
+	  includes the duration.
+
+2009-08-01 16:48:36 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+	* ext/flac/gstflacdec.c:
+	  flacdec: small cleanups
+	  Remove some callback indirections which are no longer needed because
+	  there's only one decoder object type now. Also remove unused variable.
+
+2009-08-01 15:22:49 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+	* ext/flac/gstflacdec.c:
+	  flacdec: use gst_adapter_copy() to avoid unnecessary buffer merges
+	  gst_adapter_peek() will merge buffers as needed, which we can avoid
+	  here since we're doing a memcpy anyway and then flush the copied
+	  data from the adapter right away.
+
+2009-08-01 00:00:41 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+	* ext/flac/gstflacdec.c:
+	  flacdec: repair some broken indenting
+
+2009-08-01 12:19:41 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+	* tests/check/Makefile.am:
+	* tests/check/elements/.gitignore:
+	* tests/check/elements/flvmux.c:
+	  checks: add basic unit test for flvmux, but disable it for now
+	  Basic unit test for flvmux. Fails miserably, hence disabled for now.
+
+2009-07-31 23:28:12 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+	* tests/check/Makefile.am:
+	* tests/check/elements/.gitignore:
+	* tests/check/elements/flvdemux.c:
+	* tests/files/Makefile.am:
+	* tests/files/pcm16sine.flv:
+	  check: add basic unit test for flvdemux
+	  In particular, test re-use of flvdemux in both pull and push mode
+	  (see #583030).
+
+2009-07-31 20:25:17 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+	* gst/flv/gstflvmux.c:
+	  flvmux: fix invalid write caused by using sizeof("string") as length
+	  sizeof("foo") includes the string's NUL-terminator in the size returned,
+	  but we're writing strings here with an explicit size at the beginning
+	  and no NUL-terminator. In most cases using sizeof("foo") as length in
+	  memcpy is not harmful, but it is where the string goes right at the
+	  end of our buffer to write, since we don't allocate space for that
+	  NUL terminator.
+
+2009-07-27 18:44:45 +0200  Edward Hervey <bilboed@bilboed.com>
+
+	* ext/soup/gstsouphttpsrc.c:
+	  soup: Use "GET" instead of SOUP_METHOD_GET. Fixes build with libsoup-2.7.*
+	  This is due to a quality API change in libsoup 2.7. SOUP_METHOD_* are now
+	  integers and not strings... they could have changed the names.
+
+2009-07-30 17:57:53 +0300  Stefan Kost <ensonic@users.sf.net>
+
+	* ext/jpeg/gstjpegdec.c:
+	* ext/jpeg/gstjpegenc.c:
+	  jpeg: use longer macro names to not clash with some stupid windows defines
+	  libjpeg headers pull some windows system inlcudes (on windows) that contain a
+	  define for DEFAULT_QUALITY.
+
+2009-07-29 14:31:48 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+	* gst/avi/gstavidemux.c:
+	  avidemux: Fix last commit and improve readability
+
+2009-07-24 19:04:31 +0400  Руслан Ижбулатов <lrn1986@gmail.com>
+
+	* gst/avi/gstavidemux.c:
+	  Fixed the fix for TIME->DEFAULT conversion.
+	  Fixes bug #578052 again.
+
+2009-07-29 13:38:03 +0200  Edward Hervey <bilboed@bilboed.com>
+
+	* gst/rtp/gstrtpsv3vdepay.c:
+	  rtpsv3depay: Fix width/height calculation, bring up to marginal rank.
+	  Based on documentation found on http://wiki.multimedia.cx/
+
+2009-07-29 12:13:20 +0200  Wim Taymans <wim.taymans@collabora.co.uk>
+
+	* ext/pulse/pulsesink.c:
+	* ext/pulse/pulsesrc.c:
+	  pulse: conditionally compile newer stuff
+	  configured_sink/source_usec in the timing_info is only since 0.9.11 so
+	  conditionally compile this information.
+	  fixes #590038
+
+2009-07-28 18:29:07 +0200  Wim Taymans <wim.taymans@collabora.co.uk>
+
+	* ext/pulse/pulsesrc.c:
+	* ext/pulse/pulsesrc.h:
+	  pulsesrc: cleanups
+	  Keep track of the paused state of the source and leave the read function when
+	  paused.
+	  don't wait for a latency update when the delay is not yet known but simply
+	  return 0 instead of blocking.
+	  Keep track of the corked state of the stream.
+	  Fix the state changes.
+
+2009-07-28 16:11:18 +0200  Wim Taymans <wim.taymans@collabora.co.uk>
+
+	* ext/pulse/pulsesrc.c:
+	  pulsesrc: set maxlength always to -1
+
+2009-07-28 15:53:57 +0200  Wim Taymans <wim.taymans@collabora.co.uk>
+
+	* ext/pulse/pulsesrc.c:
+	* ext/pulse/pulsesrc.h:
+	  pulsesrc; cleanups, report real latency
+	  Add some more debug info
+	  Avoid some type casts
+	  Report the real latency to the application.
+
+2009-07-28 16:11:36 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
+
+	* ext/jpeg/gstjpegdec.c:
+	  jpegdec: when scanning for 0xff marker ends, ensure desired result
+	  Otherwise, any non 0xff byte at end of data would be mistaken for
+	  a tag byte, and in case of a frame_len 0 tag subsequently lead to an
+	  infinite loop.
+
+2009-07-28 00:30:43 -0300  Thiago Santos <thiagoss@embedded.ufcg.edu.br>
+
+	* gst/avi/gstavimux.c:
+	  avimux: adds support to wma
+
+2009-07-28 00:07:15 -0300  Thiago Santos <thiagoss@embedded.ufcg.edu.br>
+
+	* gst/avi/gstavimux.c:
+	  avimux: adds support to wmv
+
+2009-07-27 21:34:22 -0300  Thiago Santos <thiagoss@embedded.ufcg.edu.br>
+
+	* gst/qtdemux/qtdemux.c:
+	  qtdemux: Downgrade warning message to debug
+
+2009-07-27 11:51:39 +0200  Wim Taymans <wim.taymans@collabora.co.uk>
+
+	* ext/pulse/pulsesink.c:
+	  pulsesink: avoid using ivalid stream indexes
+	  when we get an invalid stream index from pulse because we were just starting,
+	  avoid using it for getting and setting the volume.
+	  Fixes #589365
+
+2009-07-24 19:38:07 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+	* gst/effectv/gstaging.c:
+	* gst/effectv/gstdice.c:
+	* gst/effectv/gstquark.c:
+	* gst/effectv/gstradioac.c:
+	* gst/effectv/gstripple.c:
+	* gst/effectv/gstshagadelic.c:
+	* gst/effectv/gststreak.c:
+	* gst/effectv/gstvertigo.c:
+	* gst/effectv/gstwarp.c:
+	  effectv: Don't allow caps changes for some effectv filters
+	  These filters use information from previous frames to
+	  generate the current frame and a caps change will make
+	  the effect start from the beginning again.
+
+2009-07-24 19:37:09 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+	* gst/effectv/gstwarp.c:
+	* gst/effectv/gstwarp.h:
+	  warptv: Make the sine table global instead of having it in every instance
+
+2009-07-24 10:47:44 +0300  Stefan Kost <ensonic@users.sf.net>
+
+	* ext/jpeg/gstjpegenc.c:
+	  jpeg: make encoder work with libjpeg v7
+	  We have to specify do_fancy_downsampling = FALSE in the encoder with did not exist before.
+
+2009-07-24 00:42:33 +0300  Stefan Kost <ensonic@users.sf.net>
+
+	* common:
+	  Automatic update of common submodule
+	  From fedaaee to 94f95e3
+
+2009-07-23 12:06:27 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+	* ext/flac/gstflacdec.c:
+	  flacdec: Implement SEEKING query
+	  Fixes bug #589423.
+
+2009-07-22 11:16:06 +0100  Colin Guthrie <cguthrie@mandriva.org>
+
+	* ext/pulse/pulsesink.c:
+	  pulsesink: Fix a couple error messages that mentioned incorrect function names.
+	  Fixes #589459.
+
+2009-07-23 11:50:16 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+	* gst/flv/gstflvdemux.c:
+	* gst/flv/gstflvparse.c:
+	  flvdemux: Implement SEEKING query
+	  Also add some more query types to the answer of the query type function.
+	  Fixes bug #589424.
+
+2009-07-21 19:46:55 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+	* ext/flac/gstflacdec.c:
+	* ext/flac/gstflacdec.h:
+	  flacdec: fix intermittent FLAC__STREAM_DECODER_ABORTED errors when seeking
+	  When seeking in a local flac file (ie. operating pull-based), the decoder
+	  would often just error out after the loop function sees a DECODER_ABORTED
+	  status. This, however, is the read callback's way of telling our loop
+	  function that pull_range failed and streaming should stop, in this case
+	  because of the flush-start event that the seek handler pushed upstream
+	  from the seeking thread. Handle this slightly better by storing the last
+	  flow return from pull_range, so the loop function can evaluate it properly
+	  when it encounters a DECODER_ABORTED and take the right action.
+	  Fixes #578612.
+
+2009-07-21 10:07:00 +0300  Stefan Kost <ensonic@users.sf.net>
+
+	* gst/interleave/interleave.c:
+	  interleave: fix indenting and upgrade two debugs to warnings.
+	  Fix newlines in variable decls. Change two debugs to become warnings as they
+	  indicate that things will not work.
+
+2009-07-21 10:04:36 +0300  Stefan Kost <ensonic@users.sf.net>
+
+	* ext/jpeg/gstjpeg.c:
+	* ext/jpeg/gstjpegdec.c:
+	* ext/jpeg/gstjpegenc.c:
+	* ext/jpeg/gstjpegenc.h:
+	  jpeg: code cleanups for encoder
+	  Remove some disabled code in encoder. Try #if 0'ed code and add comments about
+	  why it is disabled. Move idct-method enum to jpeg.c and use in both encoder and
+	  decoder. Add idct-method property to encoder.
+
+2009-07-21 07:50:46 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+	* gst/matroska/matroska-demux.c:
+	  matroskademux: Answer SEEKING queries in the original format
+
+2009-07-21 01:12:44 +0200  Josep Torra <n770galaxy@gmail.com>
+
+	* gst/udp/gstudpnetutils.c:
+	  udputils: initialize struct content with 0.
+	  Fixes some random crashes.
+
+2009-07-20 19:09:19 +0200  Wim Taymans <wim.taymans@collabora.co.uk>
+
+	* ext/pulse/pulsesink.c:
+	  pulsesink: set some values to their defaults
+	  Set the minreq and maxlength buffer attributes to -1 to let puleseaudio select a
+	  sensible value.
+
+2009-07-20 19:04:09 +0200  Wim Taymans <wim.taymans@collabora.co.uk>
+
+	* ext/pulse/pulsesink.c:
+	  pulsesink: don't wait for posted message
+	  We can't wait for the ENTER/LEAVE messages to be be posted because the base
+	  class sometimes calls the start method with the object lock, which would block
+	  the message posting.
+	  Instead, just assume that the message will be posted soon and continue. We'll
+	  have to fix this in the base class.
+
+2009-07-20 18:11:33 +0200  Wim Taymans <wim.taymans@collabora.co.uk>
+
+	* ext/pulse/pulsesink.c:
+	  pulsesink: use relative seeks
+	  Use relative seeks because I was told that absolute seeks don't work.
+
+2009-07-20 16:52:19 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+	* gst/matroska/matroska-demux.c:
+	  matroskademux: Implement SEEKING query
+
+2009-07-20 08:07:13 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+	* ext/cairo/gstcairorender.c:
+	  cairorender: Add support for ARGB/BGRA input
+	  Note that videotestsrc outputs 100% transparent video
+	  which will result in white output from cairorender.
+
+2009-07-17 13:22:57 +0100  Elaine Xiong <Elaine.Xiong@Sun.COM>
+
+	* sys/v4l2/gstv4l2object.h:
+	* sys/v4l2/gstv4l2src.c:
+	* sys/v4l2/v4l2_calls.c:
+	* sys/v4l2/v4l2src_calls.c:
+	  v4l2: Fix v4l2src on OpenSolaris
+	  The v4l2 driver for USB webcams on OpenSolaris does not support select()
+	  calls. Detect when select() fails, and skip polling the device afterward,
+	  which restores the pre 0.10.14 behaviour on OpenSolaris.
+	  Signed-off-by: Jan Schmidt <thaytan@noraisin.net>
+
+2009-07-17 11:22:06 +0100  Jan Schmidt <thaytan@noraisin.net>
+
+	* tests/check/elements/.gitignore:
+	* tests/examples/v4l2/.gitignore:
+	  gitignore: Ignore some new binaries
+
+2009-07-17 13:49:21 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+	* docs/plugins/Makefile.am:
+	* docs/plugins/gst-plugins-good-plugins-docs.sgml:
+	* docs/plugins/gst-plugins-good-plugins-sections.txt:
+	* docs/plugins/gst-plugins-good-plugins.args:
+	* docs/plugins/gst-plugins-good-plugins.hierarchy:
+	* docs/plugins/inspect/plugin-cairo.xml:
+	* ext/cairo/gstcairorender.c:
+	  cairorender: Add to the documentation
+
+2009-07-17 13:42:49 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+	* ext/cairo/gstcairorender.c:
+	  cairorender: Return not-negotiated if we have no caps
+
+2009-07-17 13:41:19 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+	* ext/cairo/gstcairorender.c:
+	* ext/cairo/gstcairorender.h:
+	  cairorender: Fix caps and colorspace handling
+
+2009-07-17 13:30:02 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+	* ext/cairo/gstcairorender.c:
+	  cairorender: Use correct mimetypes for PDF and SVG
+
+2009-07-17 13:24:28 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+	* ext/cairo/gstcairorender.c:
+	  cairorender: Remove pull mode, it only adds complexity but not advantages
+
+2009-07-16 21:55:31 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+	* ext/cairo/gstcairorender.c:
+	  cairorender: Fix caps negotiation and cairo surface creation
+
+2009-07-16 21:42:21 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+	* ext/cairo/gstcairorender.c:
+	  cairorender: Correctly set srccaps
+
+2009-07-16 21:31:43 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+	* ext/cairo/gstcairorender.c:
+	* ext/cairo/gstcairorender.h:
+	  cairorender: Move instance/class struct definitions to the header
+
+2009-07-16 21:30:00 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+	* ext/cairo/gstcairorender.c:
+	* ext/cairo/gstcairorender.h:
+	  cairorender: Add Lutz' copyright to the file header
+
+2009-07-16 21:27:45 +0200  Lutz Mueller <lutz@topfrose.de>
+
+	* ext/cairo/Makefile.am:
+	* ext/cairo/gstcairo.c:
+	* ext/cairo/gstcairorender.c:
+	* ext/cairo/gstcairorender.h:
+	  cairo: Add cairo-based PDF/PS/SVG encoder element
+	  Fixes bug #331420.
+
+2009-07-16 20:44:40 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+	* ext/flac/gstflacenc.c:
+	* ext/flac/gstflacenc.h:
+	  flacenc: Optionally write a PADDING block
+	  The size of the PADDING block is specified by a new
+	  "padding" property.
+	  Fixes bug #588483.
+
+2009-07-16 19:35:44 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+	* ext/soup/gstsouphttpsrc.c:
+	  souphttpsrc: Only assume seekability if the server provides Content-Length
+	  Previously seekability way always assumed until the first seek actually
+	  failed. Now we assume that all servers are not seekable unless they provide
+	  a Content-Length header. If a seek fails after that we continue to
+	  assume no seekability. Fixes bug #585576.
+
+2009-07-16 15:14:43 +0200  Arnout Vandecappelle <arnout@mind.be>
+
+	* ext/soup/gstsouphttpsrc.c:
+	  souphttpsrc: don't try to authenticate if no username/password is set.
+
+2009-07-16 17:10:21 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+	* gst/effectv/gstwarp.c:
+	  effectv: Chain up finalize to the parent class in warptv
+	  Fixes a memory leak.
+
+2009-07-16 12:55:49 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+	* tests/check/Makefile.am:
+	* tests/check/pipelines/effectv.c:
+	  effectv: Add unit test for all effectv elements
+
+2009-07-16 12:17:32 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+	* docs/plugins/Makefile.am:
+	* docs/plugins/gst-plugins-good-plugins-docs.sgml:
+	* docs/plugins/gst-plugins-good-plugins-sections.txt:
+	* docs/plugins/gst-plugins-good-plugins.args:
+	* docs/plugins/gst-plugins-good-plugins.hierarchy:
+	* docs/plugins/inspect/plugin-alaw.xml:
+	* docs/plugins/inspect/plugin-audiofx.xml:
+	* docs/plugins/inspect/plugin-effectv.xml:
+	* docs/plugins/inspect/plugin-mulaw.xml:
+	* docs/plugins/inspect/plugin-videomixer.xml:
+	  effectv: Add new effectv elements to the docs
+
+2009-07-15 14:37:19 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+	* gst/effectv/Makefile.am:
+	* gst/effectv/gsteffectv.c:
+	* gst/effectv/gstripple.c:
+	* gst/effectv/gstripple.h:
+	  effectv: Add rippletv element
+	  This produces a water ripple effect on the video input,
+	  based on motion or a rain drop algorithm.
+	  Kindly relicensed to LGPL2+ by Kentaro Fukuchi <fukuchi@megaui.net>.
+	  Fixes bug #588695.
+
+2009-07-12 15:42:35 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+	* gst/effectv/Makefile.am:
+	* gst/effectv/gsteffectv.c:
+	* gst/effectv/gststreak.c:
+	* gst/effectv/gststreak.h:
+	  effectv: Add streaktv effect filter element
+	  This combines the StreakTV and BaltanTV filters from the
+	  effectv project.
+	  Kindly relicensed to LGPL2+ by Kentaro Fukuchi <fukuchi@megaui.net>.
+	  Fixes bug #588368.
+
+2009-07-12 12:31:15 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+	* gst/effectv/gstaging.c:
+	* gst/effectv/gstedge.c:
+	* gst/effectv/gstop.c:
+	* gst/effectv/gstquark.c:
+	* gst/effectv/gstradioac.c:
+	* gst/effectv/gstrev.c:
+	* gst/effectv/gstshagadelic.c:
+	* gst/effectv/gstvertigo.c:
+	  effectv: Fix processing on big endian architectures
+
+2009-07-12 11:52:04 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+	* gst/effectv/Makefile.am:
+	* gst/effectv/gsteffectv.c:
+	* gst/effectv/gstradioac.c:
+	* gst/effectv/gstradioac.h:
+	  effectv: Add radioactv effect filter
+	  This filter adds a radiation-like motion blur effect
+	  to the video stream.
+	  Kindly relicensed to LGPL2+ by Kentaro Fukuchi <fukuchi@megaui.net>.
+	  Fixes bug #588359.
+
+2009-07-12 11:26:57 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+	* gst/effectv/gstop.c:
+	* gst/effectv/gstop.h:
+	  effectv: Make the optv threshold property an uint
+
+2009-07-12 10:39:02 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+	* gst/effectv/Makefile.am:
+	* gst/effectv/gsteffectv.c:
+	* gst/effectv/gstop.c:
+	* gst/effectv/gstop.h:
+	  effect: Add optv effect filter from the effectv project
+	  This filter binarizes input frames and combines them with various
+	  optical pattern.
+	  Kindly relicensed to LGPL2+ by Kentaro Fukuchi <fukuchi@megaui.net>.
+	  Fixes bug #588349.
+
+2009-07-03 05:11:26 -0400  Olivier Crête <olivier.crete@collabora.co.uk>
+
+	* ext/pulse/pulsesink.c:
+	  pulsesink: Emit stream-status leave message
+	  Fixes #587695
+
+2009-07-03 05:06:45 -0400  Olivier Crête <olivier.crete@collabora.co.uk>
+
+	* ext/pulse/pulsesink.c:
+	* ext/pulse/pulsesink.h:
+	  pulsesink: Emit stream-status enter message
+	  Emit stream-status messages for the pulse thread.
+	  Don't use our own GCond for signaling but simply use the pulse mainloop
+	  mechanisms for synchronisation.
+	  See #587695
+
+2009-07-14 18:15:59 +0200  Wim Taymans <wim.taymans@collabora.co.uk>
+
+	* ext/pulse/pulsesink.c:
+	  pulsesink: debug the latency update values
+
+2009-07-14 16:12:55 +0200  Wim Taymans <wim.taymans@collabora.co.uk>
+
+	* configure.ac:
+	* ext/pulse/pulsesink.c:
+	* ext/pulse/pulseutil.c:
+	  pulsesink: add 24bit sample formats
+	  Add check for pulseaudio 0.9.15 and enable 24bits samples in that case.
+
+2009-07-13 12:23:37 -0400  Olivier Crête <olivier.crete@collabora.co.uk>
+
+	* common:
+	  Automatic update of common submodule
+	  From 5845b63 to fedaaee
+
+2009-07-13 17:53:25 +0200  Marc Leeman <marc.leeman at gmail.com>
+
+	* gst/rtp/gstrtpmpvpay.c:
+	  mpvpay: Rework the timestamping
+	  Rework the timestamping in the mpv payloader so that the timestamps are more
+	  accurate.
+	  Fixes #587680
+
+2009-07-03 08:47:12 +0200  Filippo Argiolas <filippo.argiolas@gmail.com>
+
+	* configure.ac:
+	* tests/examples/Makefile.am:
+	* tests/examples/v4l2/Makefile.am:
+	* tests/examples/v4l2/probe.c:
+	  v4l2src: add a simple test case for device probing
+
+2009-07-03 08:38:43 +0200  Filippo Argiolas <filippo.argiolas@gmail.com>
+
+	* configure.ac:
+	* sys/v4l2/Makefile.am:
+	* sys/v4l2/gstv4l2object.c:
+	  v4l2src: optional support for device probing with gudev
+	  Enumerate v4l2 devices using gudev if available.
+	  Fixes bug #583640.
+
+2009-07-10 19:54:25 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+	* gst/videomixer/videomixer.c:
+	  videomixer: Random cleanup
+
+2009-07-10 19:54:13 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+	* gst/videomixer/videomixer.c:
+	  videomixer: Send queries to the master pad by default instead of all pads
+
+2009-07-10 19:34:41 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+	* gst/videomixer/Makefile.am:
+	* gst/videomixer/blend_rgb.c:
+	* gst/videomixer/videomixer.c:
+	  videomixer: Add RGB, BGR, xRGB, RGBx, xBGR, BGRx support
+
+2009-07-10 17:43:07 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+	* gst/videomixer/videomixer.c:
+	  videomixer: Clean up debugging a bit
+
+2009-07-10 17:25:48 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+	* gst/videomixer/videomixer.c:
+	  videomixer: Remove some redundant checks and error out immediately if not negotiated
+	  Also stop leaking the output buffer in some error cases.
+
+2009-07-10 17:23:03 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+	* gst/videomixer/blend_ayuv.c:
+	* gst/videomixer/blend_bgra.c:
+	* gst/videomixer/blend_i420.c:
+	* gst/videomixer/videomixer.c:
+	* gst/videomixer/videomixer.h:
+	  videomixer: Remove the calculate_frame_size() function and use libgstvideo instead
+
+2009-06-30 15:13:44 +0200  Edward Hervey <bilboed@bilboed.com>
+
+	* gst/videomixer/videomixer.c:
+	  videomixer: Remove unused link/unlink pad methods
+
+2009-06-30 12:43:04 +0200  Edward Hervey <bilboed@bilboed.com>
+
+	* gst/videomixer/blend_i420.c:
+	  videomixer: I420 mode: Add fast path for 0.0 and 1.0 alpha
+	  If the source alpha is 0.0, we take nothing.
+	  If the source alpha is 1.0, we overwrite everything.
+
+2009-06-30 12:40:02 +0200  Edward Hervey <bilboed@bilboed.com>
+
+	* gst/videomixer/blend_i420.c:
+	  videomixer: I420 blending : Fix main algorithm.
+	  When blending a source layer with an alpha of 'a' on top of another
+	  destination layer we take the sum of:
+	  * 'a' percent of the source layer
+	  * (100 - 'a') percent of the destination layer (the remainder)
+
+2009-06-30 12:39:19 +0200  Edward Hervey <bilboed@bilboed.com>
+
+	* gst/videomixer/blend_i420.c:
+	* gst/videomixer/videomixer.c:
+	* gst/videomixer/videomixer.h:
+	* gst/videomixer/videomixerpad.h:
+	  videomixer: Make debugging category global to all the code.
+
+2009-06-29 19:23:41 +0200  Edward Hervey <bilboed@bilboed.com>
+
+	* gst/videomixer/videomixer.c:
+	  videomixer: improve readability of debugging statements.
+
+2009-07-08 13:38:53 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
+
+	* gst/rtsp/gstrtspsrc.c:
+	  rtspsrc: do not leak timeout message
+
+2009-07-09 07:14:23 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+	* gst/avi/gstavidemux.c:
+	  avi: Don't forward NEWSEGMENT events from upstream
+	  New ones are generated later and simply forwarding them can
+	  result in NEWSEGMENT events of different format going downstream.
+	  Fixes bug #587983.
+
+2009-07-08 18:19:45 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+	* gst/videomixer/blend_ayuv.c:
+	* gst/videomixer/blend_i420.c:
+	  videomixer: Make checker pattern lookup table constant
+
+2009-07-08 18:17:48 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+	* gst/videomixer/Makefile.am:
+	* gst/videomixer/blend_bgra.c:
+	* gst/videomixer/videomixer.c:
+	  videomixer: Add support for ARGB
+	  And clean up the caps parsing.
+
+2009-07-08 15:17:41 +0200  Benjamin Gaignard <benjamin@gaignard.net>
+
+	* gst/udp/gstudpnetutils.c:
+	  udp: Initialize pointer to NULL
+	  Otherwise we're calling free() with some random
+	  memory address in error cases.
+	  Fixes bug #587982.
+
+2009-07-07 16:35:24 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
+
+	* gst/qtdemux/qtdemux.c:
+	  qtdemux: sprinkle some more const
+
+2009-07-07 15:57:55 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
+
+	* gst/qtdemux/qtdemux.c:
+	  qtdemux: perform some more (careful) data buffering
+	  Once buffering has started (with an mdat atom), continue buffering
+	  until moov atom is reached, which handles cases with multiple
+	  mdat atoms.  Also keep adapter/offset better in sync with upstream
+	  and fix some debug statements.  Fixes #587426.
+
+2009-07-06 10:40:31 +0200  Philip J�genstedt <philipj@opera.com>
+
+	* gst/avi/gstavidemux.c:
+	  avidemux: Replace deprecated GST_DISABLE_DEBUG with correct macro. Fixes #587826
+
+2009-07-01 13:07:48 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+	* gst/qtdemux/qtdemux.c:
+	  qtdemux: error out instead of dividing by 0
+	  Error out if timescale is 0.
+
+2009-07-01 09:32:42 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+	* gst/qtdemux/qtdemux.c:
+	  Revert "qtdemux: Make sure we don't blacklist streams by wrongly comparing their"
+	  This reverts commit 5503a59a5779b67451d8a271000181790ee76bc7.
+	  Reverting this since it causes regressions with a lot of sample files
+	  I have, all of which worked fine with the last -good release (#586891).
+
+2009-06-30 15:54:47 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+	* gst/qtdemux/qtdemux.c:
+	  qtdemux: comment out unused structure
+
+2009-06-30 13:12:09 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+	* gst/qtdemux/qtdemux.c:
+	  qtdemux: more size checks, and use g_try_new0() instead of g_new0()
+	  Whenever we alloc something based on a user-supplied size, we should
+	  really use g_try_new(), otherwise we can easily be made to abort by
+	  passing a ridiculously large number to us for allocing. Fixes
+	  problems with some fuzzed files.
+
+2009-06-29 18:58:33 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+	* gst/qtdemux/qtdemux.c:
+	  qtdemux: guard against bogus atom sizes and short reads
+	  Check the possibly 64-bit atom size more carefully before casting it
+	  to an int and passing it to gst_pad_pull_range(), otherwise we might
+	  end up pulling 0 bytes, getting an empty buffer as requested and
+	  dereferencing not available data whilst thinking we actually asked
+	  for and got 0x1000000000000 bytes. Similar fix for push mode operation
+	  where neededbytes ends up being 0 bytes, which makes us assert. Fixes
+	  crash with broken or fuzzed file (NB #122378).
+
+2009-06-29 16:52:41 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+	* gst/qtdemux/qtdemux.c:
+	  qtdemux: use 0x prefix when logging numbers in hex
+
+2009-07-01 08:40:40 +0200  Edward Hervey <bilboed@bilboed.com>
+
+	* ext/flac/gstflacdec.c:
+	  flacdec: Don't send empty string tags
+
+2009-06-30 21:35:37 +0400  LRN <lrn1986 at gmail.com>
+
+	* gst/udp/gstmultiudpsink.c:
+	  Don't use sendmsg()-dependent code on Windows
+	  Fixes #585842
+
+2009-06-30 15:59:20 +0200  Wim Taymans <wim.taymans@collabora.co.uk>
+
+	* gst/law/alaw-decode.c:
+	* gst/law/alaw-encode.c:
+	* gst/law/alaw.c:
+	* gst/law/mulaw-decode.c:
+	* gst/law/mulaw-encode.c:
+	* gst/law/mulaw.c:
+	  law: fix caps and negotiation
+	  Fix the caps to include the depth (instead of width twice) in the caps of
+	  audio/x-raw-int.
+	  Fix negotiation to not only copy the rate/channels of the first structure.
+
+2009-06-30 14:48:09 +0300  Stefan Kost <ensonic@users.sf.net>
+
+	* ext/pulse/pulsesink.c:
+	  pulsesink: include "1.0=100%" in volume and change upper limit
+	  Upper volume limmit was 1000. That appear unneceasrily high. It would also cause
+	  sever distortion if accidentialy used. Now its 10 (~ +15db) which is also in
+	  sync with volume and playbin2.
+
+2009-06-29 15:39:43 +0200  Wim Taymans <wim.taymans@collabora.co.uk>
+
+	* ext/pulse/pulsesrc.c:
+	  pulse: some more trivial cleanups
+
+2009-06-29 15:38:49 +0200  Wim Taymans <wim.taymans@collabora.co.uk>
+
+	* ext/pulse/pulsemixer.c:
+	  pulse: trivial cleanups
+
+2009-06-29 15:20:31 +0200  Wim Taymans <wim.taymans@collabora.co.uk>
+
+	* ext/pulse/pulsesink.c:
+	  pulsesink: clear ringbuffer when asked to
+	  Since we map the ringbuffer to the pulseaudio internal ringbuffer, flush the
+	  pulseaudio buffer when we are asked to clear the ringbuffer.
+	  This avoids some leftover audio after a seek.
+
+2009-06-26 15:00:14 +0100  Jan Schmidt <thaytan@noraisin.net>
+
+	* autogen.sh:
+	  autogen.sh: Actually do the 'echo -n' -> printf change.
+
+2009-06-26 14:40:14 +0100  Jan Schmidt <thaytan@noraisin.net>
+
+	* autogen.sh:
+	  autogen.sh: Use printf instead of 'echo -n'. Check for automake-1.1[01]
+	  Check for more automake command variants. Use printf instead of 'echo -n'
+	  for portability
+
+2009-06-26 13:42:09 +0100  Jan Schmidt <thaytan@noraisin.net>
+
+	* common:
+	  Automatic update of common submodule
+	  From f810030 to 5845b63
+
+2009-06-26 13:19:04 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+	* gst/qtdemux/qtdemux.c:
+	  qtdemux: don't process track_num/track_count tags with a 0 value
+	  Number/count values of 0 mean they're not set. Don't put those in the
+	  taglist.
+
+2009-06-25 18:51:12 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+	* sys/waveform/gstwaveformsink.c:
+	  waveformsink: use 'guint8' instead of 'byte' to fix compilation with MSVC8
+	  We need a cast here for pointer arithmetic to work correctly, but some
+	  MSVC versions don't seem to like 'byte', so use guint8 here. Hopefully
+	  fixes #585361.
+
+2009-06-25 19:39:37 +0300  Stefan Kost <ensonic@users.sf.net>
+
+	* sys/v4l2/v4l2_calls.c:
+	  v4l2src: set structs to zero before using them in ioctls
+	  This fixes valgrind warnings.
+
+2009-06-25 13:23:40 +0200  Julien Moutte <julien@fluendo.com>
+
+	* gst/qtdemux/qtdemux.c:
+	  qtdemux: Make sure we don't blacklist streams by wrongly comparing their duration with entire clip duration.
+
+2009-06-25 13:18:14 +0200  Krzysztof Błaszkowski <kb at sysmikro.com.pl>
+
+	* gst/rtsp/gstrtpdec.c:
+	  rtpdec: fix some buffer leaks
+
+2009-06-25 08:11:09 +0200  Edward Hervey <bilboed@bilboed.com>
+
+	* gst/flv/gstflvparse.c:
+	  flvparse: Add missing break in switch/case.
+
+2009-06-25 08:10:38 +0200  Edward Hervey <bilboed@bilboed.com>
+
+	* gst/flv/gstflvdemux.c:
+	  flvdemux: Remove unused variable, hint branch likeliness, add comments.
+
+2009-06-25 08:09:57 +0200  Edward Hervey <bilboed@bilboed.com>
+
+	* gst/avi/gstavidemux.c:
+	  avidemux: Removed unused variable
+
+2009-06-25 07:41:07 +0200  Edward Hervey <bilboed@bilboed.com>
+
+	* gst/qtdemux/qtdemux.c:
+	  qtdemux: Remove dead assignments and unused variables.
+	  Also add branch likeliness macros.
+
+2009-06-25 07:40:26 +0200  Edward Hervey <bilboed@bilboed.com>
+
+	* gst/qtdemux/qtdemux.c:
+	  qtdemux: Fix uninitialized variables. Fixes build on macosx
+
+2009-06-24 17:43:25 +0300  Stefan Kost <ensonic@users.sf.net>
+
+	* ext/soup/gstsouphttpsrc.c:
+	  souphttpsrc: free memory in finalize
+	  finalize is called only once. no need to clear pointers there. dispose is for
+	  unreffing.
+
+2009-06-24 15:14:14 +0100  Jan Schmidt <jan.schmidt@sun.com>
+
+	* common:
+	  Automatic update of common submodule
+	  From 6ab11d1 to f810030
+
+2009-06-08 14:46:48 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+	* gst/avi/gstavidemux.c:
+	  avidemux: short-circuit gst_avi_demux_src_convert() when parsing the index
+	  Don't call gst_avi_demux_src_convert() for each single index entry. Not
+	  only do we already have the pointer to the stream context, we also know
+	  the formats we want to convert from and to already, so we may just as
+	  well use optimised conversion routines that bypass some of the checks
+	  and lookups made in gst_avi_demux_src_convert().
+
+2009-06-17 16:39:36 +0200  Edward Hervey <bilboed@bilboed.com>
+
+	* gst/qtdemux/qtdemux.c:
+	  qtdemux: Another round of G_*LIKELY micro-optimisations.
+
+2009-06-17 16:20:25 +0200  Edward Hervey <bilboed@bilboed.com>
+
+	* gst/qtdemux/qtdemux.c:
+	  qtdemux: Take last sample duration for dummy segment calculation.
+	  This fixes the cases where files without EDL wouldn't output their
+	  last buffer.
+
+2009-06-24 12:36:31 +0200  Edward Hervey <bilboed@bilboed.com>
+
+	* gst/avi/gstavidemux.c:
+	  avidemux: Sprinkle branch likeliness macros over the code.
+
+2009-06-23 16:54:32 +0200  Edward Hervey <bilboed@bilboed.com>
+
+	* ext/raw1394/gstdv1394src.c:
+	* ext/raw1394/gsthdv1394src.c:
+	  raw1394: sprinkle branch likeliness macros accross the code.
+
+2009-06-14 10:36:17 +0200  Edward Hervey <bilboed@bilboed.com>
+
+	* gst/qtdemux/qtdemux.c:
+	  qtdemux: Add GST_MEMDUMP statements for unknown atoms.
+	  This is to help developers track down and implement unhandled atoms faster.
+
+2009-06-23 17:51:32 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+	* gst/deinterlace/gstdeinterlace.c:
+	  deinterlace: Remove the interlaced field from the output caps if deinterlacing is enabled
+
+2009-06-23 17:48:47 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+	* gst/deinterlace/tvtime/greedyh.c:
+	  deinterlace: Copy the correct line from correct place in the history
+
+2009-06-23 16:35:36 +0200  Wim Taymans <wim.taymans@collabora.co.uk>
+
+	* gst/rtsp/gstrtspsrc.c:
+	  rtspsrc: use same protocols after redirect
+	  After a redirect we want to use the same protocols that we were using for the
+	  current url.
+
+2009-06-23 15:35:37 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+	* gst/qtdemux/qtdemux.c:
+	  qtdemux: don't leak cover art
+
+2009-06-23 14:10:10 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+	* gst/udp/gstudpnetutils.c:
+	  udp: fix compiler warning about EAI_ADDRFAMILY getting redefined in some cases
+	  Include the header from where we include all the system headers with the
+	  socket stuff before we try to define EAI_ADDRFAMILY ourselves, otherwise
+	  we define it ourselves and then get a compiler warning if a system header
+	  defines it as well without guarding against it being defined already.
+
+2009-06-23 14:39:56 +0200  Wim Taymans <wim.taymans@collabora.co.uk>
+
+	* gst/matroska/matroska-ids.h:
+	  matroska: and the new headers too
+
+2009-06-23 14:32:43 +0200  Wim Taymans <wim.taymans@collabora.co.uk>
+
+	* gst/matroska/matroska-demux.c:
+	  matroske: fix compiler error
+	  change gpointer to guint8 * for codec_state and codec_priv as some
+	  functions operate on those types and it avoids breaking strict-aliasing
+	  rules.
+
+2009-06-23 12:42:33 +0200  Wim Taymans <wim.taymans@collabora.co.uk>
+
+	* gst/matroska/matroska-demux.c:
+	  matroskademux: avoid leaking buffers
+	  Don't leak buffers when resyncing to a keyframe.
+	  Avoid leaking buffers when exiting the loop on error conditions.
+	  Add some more debug info.
+	  Fixes #585911
+
+2009-06-22 15:56:58 +0300  Stefan Kost <ensonic@users.sf.net>
+
+	* sys/v4l2/gstv4l2src.c:
+	  v4l2: open/close the device in READY
+	  This allows to query the device in READY. Before one need to switch it to PAUSED
+	  and that also starts streaming.
+
+2009-06-20 15:41:44 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+	* gst/qtdemux/qtdemux.c:
+	* gst/qtdemux/qtdemux_dump.c:
+	  qtdemux: use GST_MEMDUMP
+
+2009-06-19 00:16:41 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+	* gst/apetag/Makefile.am:
+	* gst/apetag/gstapedemux.c:
+	  apedemux: add container-format tag
+	  Use pbutils here because the string is translated.
+
+2009-06-19 00:15:51 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+	* gst/id3demux/Makefile.am:
+	* gst/id3demux/gstid3demux.c:
+	  id3demux: add container-format tag
+	  Using pbutils here because the string is translated.
+
+2009-06-18 23:51:52 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+	* ext/dv/gstdvdemux.c:
+	  dvdemux: post container-format tag
+	  Also merge the two almost identical _add_*_pad() functions into one.
+
+2009-06-18 23:43:49 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+	* ext/dv/gstdvdemux.c:
+	  dvdemux: don't screw up first audio buffer
+	  Query the audio format, esp. dvdemux->num_channels, before we use that
+	  variable to allocate the initial buffer. That way we don't accidentally
+	  push a zero-sized buffer as first audio buffer.
+
+2009-06-18 23:38:30 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+	* gst/multipart/multipartdemux.c:
+	  multipartdemux: post container-format tag
+
+2009-06-18 23:37:11 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+	* gst/matroska/matroska-demux.c:
+	  matroska-demux: post container-format tags
+
+2009-06-18 23:36:28 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+	* gst/avi/gstavidemux.c:
+	  avidemux: post container-format tag
+
+2009-06-18 23:35:29 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+	* gst/qtdemux/qtdemux.c:
+	  qtdemux: post container-format tags
+
+2009-06-21 17:13:43 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+	* gst/audiofx/audioamplify.c:
+	  audioamplify: Fix integer overflows on 32 bit architectures
+
+2009-06-21 09:50:54 +0200  Kipp Cannon <kcannon@ligo.caltech.edu>
+
+	* gst/audiofx/audioamplify.c:
+	  audioamplify: Don't declare a loop index static
+	  The previous patch to add support for additional sample formats possibly
+	  introduced a reentrancy bug:  a variable used for a loop index was declared
+	  static.  This patch fixes that, and also adds a "/* *INDENT-ON* */" annotation
+	  following the macro block.  (I don't know what the annotation is for, but the
+	  adder, where I copied this from, has it).
+
+2009-06-19 22:37:27 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+	* gst/audiofx/audioamplify.c:
+	  audioamplify: Fix off-by-one in wrap-positive mode
+
+2009-06-19 22:20:45 +0200  Kipp Cannon <kcannon@ligo.caltech.edu>
+
+	* gst/audiofx/audioamplify.c:
+	* gst/audiofx/audioamplify.h:
+	  audioamplify: Add noclip method and support for more formats
+	  Fixes bug #585828 and #585831.
+
+2009-06-19 21:46:41 +0200  Koop Mast <kwm@freebsd.org>
+
+	* gst/udp/gstudpnetutils.h:
+	  udp: Fix build on FreeBSD
+	  Fixes bug #586397.
+
+2009-06-19 18:12:27 +0200  Ognyan Tonchev <ognyan.tonchev at axis.com>
+
+	* tests/check/elements/rtp-payloading.c:
+	  tests: add unit tests for buffer-list payloaders
+	  See #585559
+
+2009-06-19 18:00:35 +0200  Ognyan Tonchev <ognyan.tonchev at axis.com>
+
+	* gst/rtp/gstrtpmp4vpay.c:
+	* gst/rtp/gstrtpmp4vpay.h:
+	  rtpmp4vpay: add support for buffer-list
+	  See #585559
+
+2009-06-19 17:57:12 +0200  Ognyan Tonchev <ognyan.tonchev at axis.com>
+
+	* gst/rtp/gstrtpjpegpay.c:
+	* gst/rtp/gstrtpjpegpay.h:
+	  rtpjpegpay: add support for buffer-lists
+	  See #585559
+
+2009-06-19 17:53:32 +0200  Ognyan Tonchev <ognyan.tonchev at axis.com>
+
+	* gst/rtp/gstrtph264pay.c:
+	* gst/rtp/gstrtph264pay.h:
+	  rtph264pay: add support for buffer-lists
+	  See #585559
+
+2009-06-18 11:54:22 +0200  Wim Taymans <wim.taymans@collabora.co.uk>
+
+	* gst/udp/gstudpnetutils.c:
+	  udputils: don't free invalid memory
+	  As spotted by benjiG in IRC.
+	  don't free invalid memory when getaddrinfo failed.
+
+2009-06-17 17:48:31 +0200  Wim Taymans <wim.taymans@collabora.co.uk>
+
+	* ext/pulse/pulsesink.c:
+	  pulseink: don't leak device_description
+	  don't leak the device_description.
+	  some cleanups.
+
+2009-06-19 14:44:40 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+	* po/af.po:
+	* po/az.po:
+	* po/bg.po:
+	* po/ca.po:
+	* po/cs.po:
+	* po/da.po:
+	* po/en_GB.po:
+	* po/es.po:
+	* po/eu.po:
+	* po/fi.po:
+	* po/fr.po:
+	* po/hu.po:
+	* po/id.po:
+	* po/it.po:
+	* po/ja.po:
+	* po/lt.po:
+	* po/mt.po:
+	* po/nb.po:
+	* po/nl.po:
+	* po/or.po:
+	* po/pl.po:
+	* po/pt_BR.po:
+	* po/ru.po:
+	* po/sk.po:
+	* po/sq.po:
+	* po/sr.po:
+	* po/sv.po:
+	* po/uk.po:
+	* po/vi.po:
+	* po/zh_CN.po:
+	* po/zh_HK.po:
+	* po/zh_TW.po:
+	  po: update .po files for sunaudiomixer string changes
+
+2009-06-18 16:58:26 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
+
+	* gst/avi/gstavidemux.c:
+	  avidemux: streaming; adjust sizes to cater for padding in chunks
+
+2009-06-17 11:54:53 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
+
+	* gst/avi/gstavidemux.c:
+	  avidemux: streaming mode; handle data chunks grouped in rec lists.
+	  Fixes #567983.
+
+2009-06-10 12:36:50 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
+
+	* gst/qtdemux/qtdemux.c:
+	  qtdemux: map some tags to COMPOSER rather than ARTIST
+
+2009-06-10 12:34:43 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
+
+	* gst/qtdemux/qtdemux.c:
+	  qtdemux: fix some 3GP tag extraction (keywords, genre, location)
+
+2009-06-09 15:36:50 +0200  Mark Nauwelaerts <mark.nauwelaerts@collabora.co.uk>
+
+	* gst/qtdemux/qtdemux.c:
+	* gst/qtdemux/qtdemux_fourcc.h:
+	  qtdemux: extract pixel-aspect-ratio information
+
+2009-06-17 07:14:09 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+	* gst/matroska/matroska-demux.c:
+	  matroskademux: Fix leaking of the Matroska TITLE element
+
+2009-06-16 20:38:42 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+	* docs/plugins/Makefile.am:
+	* docs/plugins/gst-plugins-good-plugins-docs.sgml:
+	* docs/plugins/gst-plugins-good-plugins-sections.txt:
+	* docs/plugins/gst-plugins-good-plugins.args:
+	* docs/plugins/gst-plugins-good-plugins.hierarchy:
+	* docs/plugins/gst-plugins-good-plugins.interfaces:
+	* docs/plugins/gst-plugins-good-plugins.prerequisites:
+	* docs/plugins/inspect/plugin-1394.xml:
+	* docs/plugins/inspect/plugin-aasink.xml:
+	* docs/plugins/inspect/plugin-alaw.xml:
+	* docs/plugins/inspect/plugin-alpha.xml:
+	* docs/plugins/inspect/plugin-alphacolor.xml:
+	* docs/plugins/inspect/plugin-annodex.xml:
+	* docs/plugins/inspect/plugin-apetag.xml:
+	* docs/plugins/inspect/plugin-audiofx.xml:
+	* docs/plugins/inspect/plugin-auparse.xml:
+	* docs/plugins/inspect/plugin-autodetect.xml:
+	* docs/plugins/inspect/plugin-avi.xml:
+	* docs/plugins/inspect/plugin-cacasink.xml:
+	* docs/plugins/inspect/plugin-cairo.xml:
+	* docs/plugins/inspect/plugin-cutter.xml:
+	* docs/plugins/inspect/plugin-debug.xml:
+	* docs/plugins/inspect/plugin-deinterlace.xml:
+	* docs/plugins/inspect/plugin-dv.xml:
+	* docs/plugins/inspect/plugin-efence.xml:
+	* docs/plugins/inspect/plugin-effectv.xml:
+	* docs/plugins/inspect/plugin-equalizer.xml:
+	* docs/plugins/inspect/plugin-esdsink.xml:
+	* docs/plugins/inspect/plugin-flac.xml:
+	* docs/plugins/inspect/plugin-flv.xml:
+	* docs/plugins/inspect/plugin-flxdec.xml:
+	* docs/plugins/inspect/plugin-gamma.xml:
+	* docs/plugins/inspect/plugin-gconfelements.xml:
+	* docs/plugins/inspect/plugin-gdkpixbuf.xml:
+	* docs/plugins/inspect/plugin-goom.xml:
+	* docs/plugins/inspect/plugin-goom2k1.xml:
+	* docs/plugins/inspect/plugin-halelements.xml:
+	* docs/plugins/inspect/plugin-icydemux.xml:
+	* docs/plugins/inspect/plugin-id3demux.xml:
+	* docs/plugins/inspect/plugin-interleave.xml:
+	* docs/plugins/inspect/plugin-jpeg.xml:
+	* docs/plugins/inspect/plugin-level.xml:
+	* docs/plugins/inspect/plugin-matroska.xml:
+	* docs/plugins/inspect/plugin-monoscope.xml:
+	* docs/plugins/inspect/plugin-mulaw.xml:
+	* docs/plugins/inspect/plugin-multifile.xml:
+	* docs/plugins/inspect/plugin-multipart.xml:
+	* docs/plugins/inspect/plugin-navigationtest.xml:
+	* docs/plugins/inspect/plugin-ossaudio.xml:
+	* docs/plugins/inspect/plugin-png.xml:
+	* docs/plugins/inspect/plugin-pulseaudio.xml:
+	* docs/plugins/inspect/plugin-quicktime.xml:
+	* docs/plugins/inspect/plugin-replaygain.xml:
+	* docs/plugins/inspect/plugin-rtp.xml:
+	* docs/plugins/inspect/plugin-rtsp.xml:
+	* docs/plugins/inspect/plugin-shout2send.xml:
+	* docs/plugins/inspect/plugin-smpte.xml:
+	* docs/plugins/inspect/plugin-soup.xml:
+	* docs/plugins/inspect/plugin-spectrum.xml:
+	* docs/plugins/inspect/plugin-speex.xml:
+	* docs/plugins/inspect/plugin-taglib.xml:
+	* docs/plugins/inspect/plugin-udp.xml:
+	* docs/plugins/inspect/plugin-video4linux2.xml:
+	* docs/plugins/inspect/plugin-videobalance.xml:
+	* docs/plugins/inspect/plugin-videobox.xml:
+	* docs/plugins/inspect/plugin-videocrop.xml:
+	* docs/plugins/inspect/plugin-videoflip.xml:
+	* docs/plugins/inspect/plugin-videomixer.xml:
+	* docs/plugins/inspect/plugin-wavenc.xml:
+	* docs/plugins/inspect/plugin-wavpack.xml:
+	* docs/plugins/inspect/plugin-wavparse.xml:
+	* docs/plugins/inspect/plugin-ximagesrc.xml:
+	* docs/plugins/inspect/plugin-y4menc.xml:
+	* gst/effectv/gstaging.c:
+	* gst/effectv/gstaging.h:
+	* gst/effectv/gstdice.c:
+	* gst/effectv/gstdice.h:
+	* gst/effectv/gstedge.c:
+	* gst/effectv/gstedge.h:
+	* gst/effectv/gstquark.c:
+	* gst/effectv/gstquark.h:
+	* gst/effectv/gstrev.c:
+	* gst/effectv/gstrev.h:
+	* gst/effectv/gstshagadelic.c:
+	* gst/effectv/gstshagadelic.h:
+	* gst/effectv/gstvertigo.c:
+	* gst/effectv/gstvertigo.h:
+	* gst/effectv/gstwarp.c:
+	* gst/effectv/gstwarp.h:
+	  effectv: Add basic documentation for the effectv elements
+
+2009-06-16 20:16:13 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+	* gst/effectv/gstaging.c:
+	* gst/effectv/gstdice.c:
+	* gst/effectv/gsteffectv.h:
+	* gst/effectv/gstquark.c:
+	* gst/effectv/gstshagadelic.c:
+	  effectv: Define the fast PRNG function at a central place
+
+2009-06-16 20:13:35 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+	* gst/effectv/Makefile.am:
+	* gst/effectv/gstaging.c:
+	* gst/effectv/gstaging.h:
+	* gst/effectv/gstdice.c:
+	* gst/effectv/gstdice.h:
+	* gst/effectv/gstedge.c:
+	* gst/effectv/gstedge.h:
+	* gst/effectv/gsteffectv.c:
+	* gst/effectv/gsteffectv.h:
+	* gst/effectv/gstquark.c:
+	* gst/effectv/gstquark.h:
+	* gst/effectv/gstrev.c:
+	* gst/effectv/gstrev.h:
+	* gst/effectv/gstshagadelic.c:
+	* gst/effectv/gstshagadelic.h:
+	* gst/effectv/gstvertigo.c:
+	* gst/effectv/gstvertigo.h:
+	* gst/effectv/gstwarp.c:
+	* gst/effectv/gstwarp.h:
+	  effectv: Move type definitions into separate headers
+	  This is needed for the docs later.
+
+2009-06-16 19:41:02 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+	* gst/effectv/gstaging.c:
+	* gst/effectv/gstdice.c:
+	* gst/effectv/gstedge.c:
+	* gst/effectv/gstquark.c:
+	* gst/effectv/gstrev.c:
+	* gst/effectv/gstshagadelic.c:
+	* gst/effectv/gstvertigo.c:
+	* gst/effectv/gstwarp.c:
+	  effectv: Remove get_unit_size implementations
+	  The default on from GstVideoFilter handles this already.
+
+2009-06-16 14:54:34 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+	* configure.ac:
+	  configure: bump core/base requirements to git
+	  Need git core for basesink bufferlist additions; -base requirement
+	  bumped gratuitously.
+
+2009-06-16 15:25:54 +0200  Wim Taymans <wim.taymans@collabora.co.uk>
+
+	* tests/check/elements/udpsink.c:
+	  tests: add some debug, send newsegment
+
+2009-06-16 15:06:50 +0200  Wim Taymans <wim.taymans@collabora.co.uk>
+
+	* gst/udp/gstudpsrc.c:
+	  udpsrc: add debug line for the socket
+
+2009-06-16 15:06:25 +0200  Wim Taymans <wim.taymans@collabora.co.uk>
+
+	* tests/check/pipelines/flacdec.c:
+	  tests: turn g_print into debug
+
+2009-06-16 15:04:15 +0200  Ognyan Tonchev <ognyan@axis.com>
+
+	* gst/udp/gstmultiudpsink.c:
+	* tests/check/Makefile.am:
+	* tests/check/elements/udpsink.c:
+	  multiudpsink: add support for buffer lists
+	  Add support for BufferList and add a unit test.
+	  Fixes #585842
+
+2009-06-16 00:02:42 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+	* ext/soup/gstsouphttpsrc.c:
+	  souphttpsrc: reset session state when stopping
+	  Increases the chances that the element is actually reusable.
+
+2009-06-15 23:49:48 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+	* ext/soup/gstsouphttpsrc.c:
+	  souphttpsrc: log response and request headers and fix some broken indenting
+
+2009-06-15 22:40:35 +0200  Wim Taymans <wim.taymans@collabora.co.uk>
+
+	* gst/rtp/gstrtpmp4gdepay.c:
+	  mp4gdepay: guess constantDuration better
+	  Do a better job at guessing the constantDuration parameter when it is not
+	  present in the caps.
+	  Fixes #585205
+
+2009-06-15 21:09:47 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+	* gst/effectv/gstwarp.c:
+	  warptv: Clean up warptv element and fix some minor bugs and leaks
+
+2009-06-15 20:53:23 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+	* gst/effectv/gstvertigo.c:
+	  vertigotv: Clean up vertigotv element and fix some minor bugs and leaks
+
+2009-06-15 20:38:01 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+	* gst/effectv/gstdice.c:
+	  dicetv: Use guint8 instead of char (which can be signed or unsigned)
+
+2009-06-15 20:36:39 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+	* gst/effectv/gstshagadelic.c:
+	  shagadelictv: Use guint8/gint8 instead of char (which can be signed or unsigned)
+
+2009-06-15 20:31:30 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+	* gst/effectv/gstshagadelic.c:
+	  shagadelictv: Clean up element and free all memory in finalize
+
+2009-06-15 20:21:58 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+	* gst/effectv/gstrev.c:
+	  revtv: Clean up revtv element
+
+2009-06-15 20:07:42 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+	* gst/effectv/gstquark.c:
+	  quarktv: Simplify some code
+
+2009-06-15 20:07:10 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+	* gst/effectv/gstquark.c:
+	  quarktv: Use the input data if a NULL buffer is chosen instead of the value 0
+
+2009-06-15 20:00:43 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+	* gst/effectv/gstquark.c:
+	  quarktv: Fix setting the planes property of quarktv
+	  Setting it to a value<16 would cause crashes before because
+	  current_plane was set to the old number of planes-1. Also
+	  fix calculations for non-2^n planes values.
+
+2009-06-15 17:50:41 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+	* gst/effectv/gstquark.c:
+	  quarktv: Clean up the quarktv element
+
+2009-06-15 17:39:20 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+	* gst/effectv/gsteffectv.c:
+	  effectv: Make elements list constant
+
+2009-06-15 17:37:53 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+	* gst/effectv/gstedge.c:
+	  edgetv: Clean up edgetv element and fix memory leak
+
+2009-06-15 17:21:58 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+	* gst/effectv/gstdice.c:
+	  dicetv: Clean up dicetv element and fix some smaller issues
+	  This fixes a memory leak (the dice map) and a crash when
+	  setting the square-bits property before caps are set.
+
+2009-06-15 17:20:21 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+	* gst/effectv/Makefile.am:
+	* gst/effectv/gstaging.c:
+	  agingtv: Actually use GstController for syncing the properties to timestamps
+
+2009-06-15 17:03:38 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+	* gst/effectv/gstaging.c:
+	  agingtv: Export some more agingtv properties via GObject properties
+
+2009-06-15 15:06:56 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+	* gst/effectv/gstaging.c:
+	  agingtv: General cleanup and updating of copyright
+	  Also make the scratch-lines property exported via a GObject
+	  property and initialize/reset the internal state correctly.
+
+2009-06-15 15:05:58 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+	* gst/effectv/gstaging.c:
+	  agingtv: Store and update state inside the instance struct
+	  This makes the coloraging effect and pits effect visible.
+
+2009-06-15 15:51:32 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+	* ext/pulse/pulsesink.c:
+	  pulsesink: ref custom ring buffer class and type in class_init
+	  Hack around thread-safety issues in GObject and our racy _get_type()
+	  functions (we could easily fix the _get_type() functions, but we still
+	  need to hack around the GObject class races until we require a newer
+	  GLib version, I think).
+
+2009-06-14 19:19:19 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+	* ext/dv/demo-play.c:
+	* tests/old/examples/Makefile.am:
+	* tests/old/examples/level/Makefile.am:
+	* tests/old/examples/level/README:
+	* tests/old/examples/level/demo.c:
+	* tests/old/examples/level/plot.c:
+	* tests/old/examples/switch/.gitignore:
+	* tests/old/examples/switch/Makefile.am:
+	* tests/old/examples/switch/switcher.c:
+	  Remove a few old example apps from the 0.8 days
+	  Some have been replaced by newer ones, others are demoing elements that
+	  don't exist any longer (not in -good anyway), and others have not been
+	  touched in many years and it seem pointless to keep them around.
+	  Removing these files makes sure we don't have any code in our repository
+	  that uses Gtk+ symbols which are to be removed for GNOME3, and as such
+	  will make some script that greps for this kind of stuff give us a clean
+	  bill of code health. Fixes #585757.
+
+2009-06-13 21:02:45 -0400  Olivier Crête <tester@tester.ca>
+
+	* common:
+	* gst/rtp/gstrtpsirenpay.c:
+	  rtpsirenpay: Remove deprecated symbol
+	  Patch by: Luis Menina
+
+2009-06-13 10:43:55 +0200  Marvin Schmidt <marvin_schmidt@gmx.net>
+
+	* tests/check/Makefile.am:
+	  tests: Don't run the flacdec test if the plugin isn't built. Fixes #585630
+
+2009-06-12 16:06:28 +0200  Patrick Radizi <patrick.radizi at axis.com>
+
+	* gst/rtsp/gstrtspsrc.c:
+	* gst/rtsp/gstrtspsrc.h:
+	  rtspsrc: Add RTP blocksize functionality
+	  Add property to make the client suggest a blocksize to the server.
+	  Fixes #585549
+
+2009-06-11 22:30:06 +0200  Wim Taymans <wim.taymans@collabora.co.uk>
+
+	* gst/rtp/README:
+	  rtp: update README, fix some typos, mention gstrtpbin
+
+2009-06-11 19:10:53 +0200  Wim Taymans <wim.taymans@collabora.co.uk>
+
+	* ext/pulse/pulsesink.c:
+	  pulsesink: handle border cases in resampler
+
+2009-06-11 13:32:22 +0100  Jan Schmidt <thaytan@noraisin.net>
+
+	* common:
+	* docs/Makefile.am:
+	* docs/plugins/Makefile.am:
+	* docs/upload.mak:
+	  docs: Bump common. Use upload-doc.mak instead of upload.mak
+	  Remove the local copy of upload.mak in favour of using the shared
+	  upload-doc.make in common/
+
+2009-06-11 11:39:25 +0100  Jan Schmidt <thaytan@noraisin.net>
+
+	* gst/goom/goom_config_param.h:
+	* gst/videomixer/videomixer.c:
+	  docs: Quieten a couple more docs warnings
+
+2009-06-11 11:27:26 +0100  Jan Schmidt <thaytan@noraisin.net>
+
+	* gst/matroska/lzo.c:
+	  docs: Remove gtk-doc comment marker
+	  These comment blocks aren't gtk-doc comments and cause annoying noise in
+	  the docs build.
+
+2009-06-11 10:05:32 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+	* gst/deinterlace/gstdeinterlace.c:
+	* gst/deinterlace/gstdeinterlace.h:
+	  deinterlace: Implement upstream negotation
+
+2009-06-10 21:47:40 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+	* gst/deinterlace/gstdeinterlace.c:
+	  deinterlace: Improve debugging and clean up some code
+
+2009-06-10 14:55:18 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+	* gst/deinterlace/gstdeinterlace.c:
+	  deinterlace: Clip buffers to the current segment if possible
+
+2009-06-10 14:45:06 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+	* gst/deinterlace/gstdeinterlace.c:
+	* gst/deinterlace/gstdeinterlace.h:
+	  deinterlace: Clean up includes and clean up order of instance struct fields
+
+2009-06-10 16:09:56 -0400  Olivier Crête <olivier.crete@collabora.co.uk>
+
+	* gst/rtp/gstrtph263pay.h:
+	  rtph263pay: Default to doing A, B and C modes, not only A
+
+2009-06-10 09:56:11 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+	* gst/deinterlace/gstdeinterlace.c:
+	  deinterlace: Fix QoS calculations
+	  The diff is a signed integer, not an unsigned one of course.
+	  In modes other than GST_DEINTERLACE_ALL every frame has twice the
+	  duration of the field duration.
+
+2009-06-09 14:13:31 -0400  Olivier Crête <olivier.crete@collabora.co.uk>
+
+	* gst/rtp/gstrtpsirenpay.c:
+	  rtpsirenpay: Put the bitrate in the RTP caps
+	  The MS code seems to require the bitrate to interoperate and
+	  draft-ietf-avt-rtp-g7221-00 also has it.
+
+2009-06-09 19:55:36 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+	* gst/deinterlace/gstdeinterlace.c:
+	* gst/deinterlace/gstdeinterlace.h:
+	  deinterlace: Implement basic QoS
+	  This change is based on Tim's QoS implementation
+	  for jpegdec.
+
+2009-06-09 19:29:51 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+	* gst/deinterlace/gstdeinterlace.c:
+	  deinterlace: Directly proxy events/queries to the peer pads
+	  This removes some overhead introduced by the default handlers
+	  that need to iterate over the other pads.
+
+2009-06-09 10:38:52 +0200  Edward Hervey <bilboed@bilboed.com>
+
+	* gst/avi/gstavidemux.c:
+	  avidemux: debug_memdump() unknown tags. Refactor junk parsing code.
+	  This makes life slightly easier when debugging avi files.
+
+2009-06-08 08:21:43 +0200  Edward Hervey <bilboed@bilboed.com>
+
+	* gst/rtp/Makefile.am:
+	  rtp: Don't forget to dist the headers for the CELT (de)payloaders.
+
+2009-06-07 20:54:06 +0200  Wim Taymans <wim.taymans@collabora.co.uk>
+
+	* gst/qtdemux/qtdemux.c:
+	  Revert "Revert "qtdemux: fill timestamp table completely""
+	  This reverts commit 9f022c8a8503c2ce0fa617fdb50e41706dd412f5.
+	  Sorry, I was thinking about the wrong module.
+
+2009-06-07 20:49:50 +0200  Wim Taymans <wim.taymans@collabora.co.uk>
+
+	* gst/qtdemux/qtdemux.c:
+	  Revert "qtdemux: fill timestamp table completely"
+	  This reverts commit 790b050fc5302cae89cddcd23b258093967d05a9.
+	  I forgot we were frozen.
+
+2009-06-07 20:46:45 +0200  Wim Taymans <wim.taymans@collabora.co.uk>
+
+	* gst/qtdemux/qtdemux.c:
+	  qtdemux: fill timestamp table completely
+	  When there are less timestamps that there are samples, fill up the sample table
+	  with the last know timestamp. This situation can happen when the last sample
+	  does not decode and doesn't need a timestamp. We however calculate the total
+	  track length using the last sample timestamp so we need to have something
+	  sensible in there.
+	  Fixes #585056
+
+2009-06-07 13:37:04 +0200  Wim Taymans <wim.taymans@collabora.co.uk>
+
+	* gst/wavparse/gstwavparse.c:
+	  wavparse: handle LIST INFO of 0 size
+	  Handle LIST INFO chunks of 0 size instead of causing errors.
+	  Fixes #584981
+
+2009-06-07 13:24:45 +0200  Wim Taymans <wim.taymans@collabora.co.uk>
+
+	* gst/wavparse/gstwavparse.c:
+	  Revert "wavparse: Remove dead assignments, move variable to where it's needed."
+	  Reverts commit 44256a78f8dd79a91f3bb2ab7c3aa623c097bb8a and use the result in
+	  error reporting so that we can see what's going on.
+
+2009-06-05 18:55:02 +0200  Wim Taymans <wim.taymans@collabora.co.uk>
+
+	* gst/rtp/Makefile.am:
+	* gst/rtp/gstrtp.c:
+	* gst/rtp/gstrtpceltdepay.c:
+	* gst/rtp/gstrtpceltdepay.h:
+	  celtdepay: add CELT depayloader
+
+2009-06-05 15:30:51 +0200  Wim Taymans <wim.taymans@collabora.co.uk>
+
+	* gst/rtp/Makefile.am:
+	* gst/rtp/gstrtp.c:
+	* gst/rtp/gstrtpceltpay.c:
+	* gst/rtp/gstrtpceltpay.h:
+	  rtpceltpay: add CELT RTP payloader
+
+2009-06-05 16:54:48 +0100  Jan Schmidt <jan.schmidt@sun.com>
+
+	* sys/sunaudio/gstsunaudiomixerctrl.c:
+	* sys/sunaudio/gstsunaudiomixeroptions.c:
+	* sys/sunaudio/gstsunaudiomixertrack.c:
+	  sunaudio: Fix switch setting on some devices. Add debug. Fix a FIXME.
+	  Fix the setting of toggle switches on some broken audio drivers which
+	  report that no audio ports are settable by ignoring the mod_port field
+	  there.
+	  Add some debug statements.
+	  Fix a FIXME now that Good relies on a new enough gst-plugins-base.
+
+2009-06-04 12:27:19 +0100  Jan Schmidt <jan.schmidt@sun.com>
+
+	* sys/sunaudio/Makefile.am:
+	* sys/sunaudio/gstsunaudiomixerctrl.c:
+	* sys/sunaudio/gstsunaudiomixerctrl.h:
+	* sys/sunaudio/gstsunaudiomixeroptions.c:
+	* sys/sunaudio/gstsunaudiomixeroptions.h:
+	* sys/sunaudio/gstsunaudiomixertrack.c:
+	* sys/sunaudio/gstsunaudiomixertrack.h:
+	  sunaudio: Support new flags for options and actions
+	  Use new audio mixer flags added in Base 0.10.23 to expose flags and options
+	  on the SunAudio devices.
+	  Fixes: #583593
+	  Patch By: Brian Cameron <brian.cameron@sun.com>
+	  Patch By: Garrett D'Amore <garrett.damore@sun.com>
+
+2009-05-15 11:50:38 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+	* gst/deinterlace/gstdeinterlace.c:
+	* gst/deinterlace/gstdeinterlace.h:
+	  deinterlace: First try to handle DVD still frames correctly
+	  This helps a bit with bug #582740 but still doesn't make it work.
+
+2009-06-04 17:37:03 +0300  Stefan Kost <ensonic@users.sf.net>
+
+	* ext/pulse/pulsesink.c:
+	  pulsesink: only notify if all checks passed
+	  Replace goto done: with return, as those are checks when we don't want to flag a
+	  pending notify.
+
+2009-06-04 15:19:05 +0200  Wim Taymans <wim.taymans@collabora.co.uk>
+
+	* gst/rtsp/gstrtspsrc.c:
+	  rtspsrc: set the right state on rtpbin
+	  We need to set the state of gstrtpbin to the same state as our source elements.
+	  This fixes fallback to TCP again.
+
+2009-06-03 18:23:53 +0300  Stefan Kost <ensonic@users.sf.net>
+
+	* ext/pulse/pulsesink.c:
+	  pulsesink: check pointer before accessing
+	  Move existing check a few lines up, so that we check before accessing fields.
+
+2009-06-03 18:21:12 +0300  Stefan Kost <ensonic@users.sf.net>
+
+	* ext/pulse/pulsesink.c:
+	  pulsesink: rename gst_pulse_sink_get_time to gst_pulsesink_get_time
+	  Rename internal method for consistency.
+
+2009-06-03 18:19:22 +0300  Stefan Kost <ensonic@users.sf.net>
+
+	* ext/pulse/pulsesink.c:
+	  pulsesink: use values from pa_stream_get_buffer_attr()
+	  We were putting the requested values back into ringbuffer spec, instead of
+	  using the queried values.
+
+2009-06-02 19:32:21 +0200  Wim Taymans <wim.taymans@collabora.co.uk>
+
+	* gst/rtp/gstrtpvrawpay.c:
+	  vrawpay: trim output buffers
+	  Remove the leftover unused bytes in the output buffer.
+	  Fixes #584613
+
+2009-06-02 19:30:30 +0200  Wim Taymans <wim.taymans@collabora.co.uk>
+
+	* gst/rtp/gstrtpvrawdepay.c:
+	  vrawdepay: fix parsing of sampling field
+	  commit a12d9a80f225be97b3674b1a0506ac66544dbf49 broke the parsing of the
+	  sampling.
+
+2009-05-27 17:06:34 +0100  Jan Schmidt <thaytan@noraisin.net>
+
+	* ext/libpng/gstpngdec.c:
+	  pngdec: Avoid possible overflow in calculations
+	  A malformed (or simply huge) PNG file can lead to integer overflow in
+	  calculating the size of the output buffer, leading to crashes or buffer
+	  overflows later. Fixes SA35205 security advisory.
+
+2009-06-02 00:48:00 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+	* ext/flac/gstflacenc.c:
+	  flacenc: some more logging - dump header packets
+	  Also, the final fixing up of the headers is expected and not something
+	  we should warn about.
+
+2009-06-02 00:37:15 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+	* ext/flac/gstflacenc.c:
+	  flacenc: never ever pass values >36bits to _set_total_samples_estimate()
+	  Let's be paranoid and make sure we never pass a number that takes up
+	  more than 36 bits to _set_total_samples_estimate(), since libFLAC
+	  expects all the other bits to be zero, and if this is not the case
+	  neighbouring fields in the global stream info header may get messed
+	  up inadvertently, so that flac -d refuses to decode the stream.
+	  See #584455.
+
+2009-06-01 22:33:02 +0200  Thomas Vander Stichele <thomas (at) apestaart (dot) org>
+
+	* ext/flac/gstflacenc.c:
+	  Address bad FLAC sample length encoding of #5844455
+	  Commit df707c666433a78d3878af6f055698d5756226c4
+	  introduced an obvious bug in the sample length calculation,
+	  using the wrong macro for conversion.
+
+2009-06-01 11:58:21 -0700  Brian Cameron <brian.cameron@sun.com>
+
+	* gst/deinterlace/tvtime/mmx.h:
+	  deinterlace: Fix spurious colons in asm code
+	  Fixes #584174.
+	  Signed-off-by: David Schleef <ds@schleef.org>
+
+2009-06-01 00:40:55 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+	* gst/avi/gstavidemux.c:
+	  avidemux: skip JUNK chunks in data section in streaming mode
+	  Skip JUNK tags in streaming mode as well instead of EOSing
+	  prematurely. Fixes #564100.
+
+2009-05-28 14:01:17 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+	* gst/videomixer/blend_bgra.c:
+	* gst/videomixer/blend_i420.c:
+	* gst/videomixer/videomixer.c:
+	  videomixer: Don't use // comments
+
+2009-05-28 13:56:15 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+	* gst/videomixer/blend_bgra.c:
+	  videomixer: Fix background blitting when a color mode is selected with BGRA
+
+2009-05-28 13:54:14 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+	* gst/videomixer/blend_ayuv.c:
+	* gst/videomixer/blend_bgra.c:
+	* gst/videomixer/blend_i420.c:
+	* gst/videomixer/videomixer.c:
+	* gst/videomixer/videomixer.h:
+	  videomixer: Some cleanup and fix the calculation of the frame size in bytes
+
+2009-05-28 13:35:52 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+	* gst/videomixer/blend_i420.c:
+	  videomixer: Fix I420 blending to actually do something
+	  For this we a) implement the checkers filling and b)
+	  actually blend the src/dest by using the src alpha value
+	  from the pad.
+
+2009-05-28 13:14:13 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+	* gst/videomixer/blend_bgra.c:
+	  videomixer: Fix ARGB blending to actually work
+
+2009-05-28 13:04:51 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+	* gst/videomixer/Makefile.am:
+	* gst/videomixer/blend_bgra.c:
+	  videomixer: Blend BGRA ourselves instead of using Cairo
+
+2009-05-28 12:55:16 +0200  Alex Ugarte <alexugarte@gmail.com>
+
+	* gst/videomixer/Makefile.am:
+	* gst/videomixer/blend_ayuv.c:
+	* gst/videomixer/blend_bgra.c:
+	* gst/videomixer/blend_i420.c:
+	* gst/videomixer/videomixer.c:
+	* gst/videomixer/videomixer.h:
+	  videomixer: Add support for blending BGRA and AYUV
+	  Fixes bug #577017.
+
+2009-05-28 12:39:46 +0200  Ghislain 'Aus' Lacroix <aus@songbirdnest.com>
+
+	* gst/equalizer/gstiirequalizer.c:
+	  equalizer: Use floating point arithmetic internally for the int16 mode
+	  By using int32 arithmetic we will introduce distortions as the
+	  IIR filter is very sensitive to rounding errors. Fixes bug #580214.
+
+2009-05-28 10:55:16 +0100  Christian Schaller <christian.schaller@collabora.co.uk>
+
+	* gst-plugins-good.spec.in:
+	  Update spec file with latest plugins
+
+2009-05-26 17:19:08 +0100  Jan Schmidt <thaytan@noraisin.net>
+
+	* common:
+	  Automatic update of common submodule
+	  From 888e0a2 to c572721
+
+2009-05-26 16:20:35 +0300  Stefan Kost <ensonic@users.sf.net>
+
+	* sys/v4l2/gstv4l2src.c:
+	* sys/v4l2/gstv4l2src.h:
+	  v4l2: cleanup and commenting
+	  Remove newlines inserted by gst-indent once. Remove unused var from instance
+	  struct. Add comments. Add another #define for default property value.
+
+2009-05-06 12:43:35 +0300  Stefan Kost <ensonic@users.sf.net>
+
+	* tests/check/Makefile.am:
+	  makefile: idea about makeing more sources/sinks testable again
+
+2009-05-25 16:33:35 +0200  John Keeping <john.keeping at lineone.net>
+
+	* ext/libpng/gstpngdec.c:
+	  pngdec: match g_malloc() with g_free()
+	  Matching g_malloc() with a g_free() is important when a custom allocator is
+	  installed.
+	  Fixes #583803
+
+2009-05-12 18:39:28 +0200  Wim Taymans <wim.taymans@collabora.co.uk>
+
+	* gst/rtp/gstrtpmp4vpay.c:
+	* gst/rtp/gstrtpmp4vpay.h:
+	  rtpmp4vpay: don't look for headers in some cases
+	  In some streams (starting with 00000100) don't look for the headers but push
+	  data as it is.
+	  Fixes #582153
+
+2009-05-13 11:50:22 +0200  Patrick Radizi <patrick.radizi at axis.com>
+
+	* gst/rtsp/gstrtspsrc.c:
+	  rtspsrc: fix memory leak of messages
+	  Free messages correctly.
+	  Fixes #577318
+
+2009-05-24 19:32:17 +0200  Wim Taymans <wim.taymans@collabora.co.uk>
+
+	* gst/rtsp/gstrtspsrc.c:
+	  rtspsrc: make fakesrc silent
+	  Make the fakesrc that is responsible for sending dummy packets silent.
+
+2009-05-24 16:33:42 +0200  Wim Taymans <wim.taymans@collabora.co.uk>
+
+	* gst/rtsp/gstrtspsrc.c:
+	  rtspsrc: don't send teardown before setup
+	  Don't send a TEARDOWN request when we did not manage to successfully setup a
+	  stream.
+
+2009-05-14 14:46:14 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+	* gst/matroska/matroska-demux.c:
+	* gst/matroska/matroska-demux.h:
+	* gst/matroska/matroska-ids.h:
+	  matroskademux: Populate a GstIndex that is set on matroskademux
+
+2009-05-14 10:35:22 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+	* gst/flv/gstflvmux.c:
+	  flvmux: Get the max duration from upstream if there's no duration tag
+
+2009-05-14 10:29:49 +0200  Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+	* gst/flv/gstflvmux.c:
+	* gst/flv/gstflvmux.h:
+	  flvmux: Write an index table to the end of the file
+
+2009-05-22 01:12:35 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+	* autogen.sh:
+	* configure.ac:
+	  autotools: move the -Wno-portability from autogen.sh to configure.ac
+	  If we're lucky it'll get used on automatic rebuilds as well that way.
+
+2009-05-22 01:10:12 +0100  Tim-Philipp Müller <tim.muller@collabora.co.uk>
+
+	* common:
+	* configure.ac:
+	* m4/gst-fionread.m4:
+	  m4: fix 'suspicious cache id' warnings
+	  and update common to pull in a similar fix. Also check in configure
+	  whether the compiler supports do while macros (GLib wants this
+	  defined and it is needed to avoid warnings with some c++ compilers
+	  apparently).
+
+2009-05-22 01:39:33 +0300  Zeeshan Ali (Khattak) <zeeshanak@gnome.org>
+
+	* configure.ac:
+	  souphttpsrc: Bump-up libsoup-2.24 dep to >= 2.26
+	  The helper function soup_message_headers_get_content_type that we now use
+	  was added in 2.26.
+
+2009-05-20 17:57:59 +0300  Zeeshan Ali (Khattak) <zeeshanak@gnome.org>
+
+	* ext/soup/gstsouphttpsrc.c:
+	  souphttpsrc: Set caps for audio/L16 content-type
+	  When "Content-Type" header is "audio/L16", we need to set the caps on the
+	  outgoing buffers so that downstream elements can have means to detect the
+	  stream type and handle it appropriately. Tested with HTTP stream provided
+	  by pulse-audio's http module (git master).
+
+2009-05-20 15:06:25 +0300  Zeeshan Ali (Khattak) <zeeshanak@gnome.org>
+
+	* ext/soup/gstsouphttpsrc.c:
+	* ext/soup/gstsouphttpsrc.h:
+	  souphttpsrc: Rename icy_caps to src_caps
+
+2009-05-21 23:39:13 +0200  Philippe Normand <philippe at fluendo.com>
+
+	* ext/jpeg/gstjpegdec.c:
+	  jpegdec: bump max size to 65535x65535
+	  Remove artificial jpeg image limits.
+	  Fixes #583048.
+
+2009-05-21 21:36:02 +0100  Jan Schmidt <thaytan@noraisin.net>
+
+	* win32/common/config.h:
+	  win32: Update the win32 config.h
+
+2009-05-19 15:12:09 +0100  Jan Schmidt <thaytan@noraisin.net>
+
+	* gst/matroska/matroska-demux.c:
+	* gst/matroska/matroska-ids.h:
+	  matroskademux: Recognise PGS subpicture streams - the bluray format.
+	  Recognise and apply appropriate caps to PGS (Presentation Graphic Stream)
+	  subpicture streams.
+
+2009-05-15 10:42:19 +0100  Jan Schmidt <thaytan@noraisin.net>
+
+	* ext/pulse/pulsesink.c:
+	  pulsesink: Convert an erroneous assertion
+	  Occasionally, we get a change callback for an old stream, triggering
+	  the assertion unnecessarily. Just ignore such callbacks.
+
+2009-05-20 16:14:40 -0400  Olivier Crête <olivier.crete@collabora.co.uk>
+
+	* ext/pulse/pulsesink.c:
+	  pulse: Print a warning on under/overflows
+
+2009-05-20 18:45:45 +0200  Wim Taymans <wim.taymans@collabora.co.uk>
+
+	* gst/qtdemux/qtdemux.c:
+	* gst/qtdemux/qtdemux_fourcc.h:
+	  qtdemux: parse in24 boxes to get endianness
+	  in24 samples are normally big-endian but an enda box can change this to
+	  little-endian. Recurse into the in24 box and find the enda box so that we get
+	  the endianness right.
+	  Fixes #582515
+
+2009-05-20 14:14:31 +0200  Wim Taymans <wim.taymans@collabora.co.uk>
+
+	* gst/multipart/multipartdemux.c:
+	  multipartdemux: add proper padtemplate
+
+2009-05-20 14:02:43 +0200  Wim Taymans <wim.taymans@collabora.co.uk>
+
+	* gst/multipart/multipartdemux.c:
+	  multipartdemux: add more mime types
+	  Add mime-type for Panasonic g726 and add more required caps properties for other
+	  G726 mime-types.
+	  Make mime-types case insensitive.
+	  See #582169
+
+2009-05-20 13:47:52 +0200  Wim Taymans <wim.taymans@collabora.co.uk>
+
+	* gst/multipart/multipartdemux.c:
+	* gst/multipart/multipartdemux.h:
+	  multipartdemux: add flow aggregation
+
+2009-05-20 13:29:02 +0200  Arnout Vandecappelle <arnout@mind.be>
+
+	* gst/multipart/multipartdemux.c:
+	  multipartdemux: allow content to be empty.
+	  gst_adapter_take_buffer doesn't allow buffer to be empty.
+	  Simply skip any part where the content is empty.  Don't
+	  create a pad for it either.
+	  See #582169
+
+2009-05-18 22:19:39 +0200  Wim Taymans <wim.taymans@collabora.co.uk>
+
+	* gst/rtp/gstrtpchannels.h:
+	  rtp: fix channel positions for mono
+
+2009-05-21 21:02:11 +0100  Jan Schmidt <thaytan@noraisin.net>
+
+	* configure.ac:
+	  Back to hacking -> 0.10.15.1
+
 === release 0.10.15 ===
 
-2009-05-20  Jan Schmidt <jan.schmidt@sun.com>
+2009-05-20 22:34:18 +0100  Jan Schmidt <thaytan@noraisin.net>
 
+	* ChangeLog:
+	* NEWS:
+	* RELEASE:
 	* configure.ac:
-	  releasing 0.10.15, "I've been up all night"
+	* docs/plugins/gst-plugins-good-plugins.args:
+	* docs/plugins/gst-plugins-good-plugins.hierarchy:
+	* docs/plugins/gst-plugins-good-plugins.interfaces:
+	* docs/plugins/gst-plugins-good-plugins.prerequisites:
+	* docs/plugins/inspect/plugin-1394.xml:
+	* docs/plugins/inspect/plugin-aasink.xml:
+	* docs/plugins/inspect/plugin-alaw.xml:
+	* docs/plugins/inspect/plugin-alpha.xml:
+	* docs/plugins/inspect/plugin-alphacolor.xml:
+	* docs/plugins/inspect/plugin-annodex.xml:
+	* docs/plugins/inspect/plugin-apetag.xml:
+	* docs/plugins/inspect/plugin-audiofx.xml:
+	* docs/plugins/inspect/plugin-auparse.xml:
+	* docs/plugins/inspect/plugin-autodetect.xml:
+	* docs/plugins/inspect/plugin-avi.xml:
+	* docs/plugins/inspect/plugin-cacasink.xml:
+	* docs/plugins/inspect/plugin-cairo.xml:
+	* docs/plugins/inspect/plugin-cutter.xml:
+	* docs/plugins/inspect/plugin-debug.xml:
+	* docs/plugins/inspect/plugin-deinterlace.xml:
+	* docs/plugins/inspect/plugin-dv.xml:
+	* docs/plugins/inspect/plugin-efence.xml:
+	* docs/plugins/inspect/plugin-effectv.xml:
+	* docs/plugins/inspect/plugin-equalizer.xml:
+	* docs/plugins/inspect/plugin-esdsink.xml:
+	* docs/plugins/inspect/plugin-flac.xml:
+	* docs/plugins/inspect/plugin-flv.xml:
+	* docs/plugins/inspect/plugin-flxdec.xml:
+	* docs/plugins/inspect/plugin-gamma.xml:
+	* docs/plugins/inspect/plugin-gconfelements.xml:
+	* docs/plugins/inspect/plugin-gdkpixbuf.xml:
+	* docs/plugins/inspect/plugin-goom.xml:
+	* docs/plugins/inspect/plugin-goom2k1.xml:
+	* docs/plugins/inspect/plugin-halelements.xml:
+	* docs/plugins/inspect/plugin-icydemux.xml:
+	* docs/plugins/inspect/plugin-id3demux.xml:
+	* docs/plugins/inspect/plugin-interleave.xml:
+	* docs/plugins/inspect/plugin-jpeg.xml:
+	* docs/plugins/inspect/plugin-level.xml:
+	* docs/plugins/inspect/plugin-matroska.xml:
+	* docs/plugins/inspect/plugin-monoscope.xml:
+	* docs/plugins/inspect/plugin-mulaw.xml:
+	* docs/plugins/inspect/plugin-multifile.xml:
+	* docs/plugins/inspect/plugin-multipart.xml:
+	* docs/plugins/inspect/plugin-navigationtest.xml:
+	* docs/plugins/inspect/plugin-ossaudio.xml:
+	* docs/plugins/inspect/plugin-png.xml:
+	* docs/plugins/inspect/plugin-pulseaudio.xml:
+	* docs/plugins/inspect/plugin-quicktime.xml:
+	* docs/plugins/inspect/plugin-replaygain.xml:
+	* docs/plugins/inspect/plugin-rtp.xml:
+	* docs/plugins/inspect/plugin-rtsp.xml:
+	* docs/plugins/inspect/plugin-shout2send.xml:
+	* docs/plugins/inspect/plugin-smpte.xml:
+	* docs/plugins/inspect/plugin-soup.xml:
+	* docs/plugins/inspect/plugin-spectrum.xml:
+	* docs/plugins/inspect/plugin-speex.xml:
+	* docs/plugins/inspect/plugin-taglib.xml:
+	* docs/plugins/inspect/plugin-udp.xml:
+	* docs/plugins/inspect/plugin-video4linux2.xml:
+	* docs/plugins/inspect/plugin-videobalance.xml:
+	* docs/plugins/inspect/plugin-videobox.xml:
+	* docs/plugins/inspect/plugin-videocrop.xml:
+	* docs/plugins/inspect/plugin-videoflip.xml:
+	* docs/plugins/inspect/plugin-videomixer.xml:
+	* docs/plugins/inspect/plugin-wavenc.xml:
+	* docs/plugins/inspect/plugin-wavpack.xml:
+	* docs/plugins/inspect/plugin-wavparse.xml:
+	* docs/plugins/inspect/plugin-ximagesrc.xml:
+	* docs/plugins/inspect/plugin-y4menc.xml:
+	* gst-plugins-good.doap:
+	* win32/common/config.h:
+	  Release 0.10.15
+
+2009-05-20 22:03:21 +0100  Jan Schmidt <thaytan@noraisin.net>
+
+	* po/af.po:
+	* po/az.po:
+	* po/bg.po:
+	* po/ca.po:
+	* po/cs.po:
+	* po/da.po:
+	* po/en_GB.po:
+	* po/es.po:
+	* po/eu.po:
+	* po/fi.po:
+	* po/fr.po:
+	* po/hu.po:
+	* po/id.po:
+	* po/it.po:
+	* po/ja.po:
+	* po/lt.po:
+	* po/mt.po:
+	* po/nb.po:
+	* po/nl.po:
+	* po/or.po:
+	* po/pl.po:
+	* po/pt_BR.po:
+	* po/ru.po:
+	* po/sk.po:
+	* po/sq.po:
+	* po/sr.po:
+	* po/sv.po:
+	* po/uk.po:
+	* po/vi.po:
+	* po/zh_CN.po:
+	* po/zh_HK.po:
+	* po/zh_TW.po:
+	  Update .po files
 
 2009-05-16 02:59:14 +0100  Jan Schmidt <thaytan@noraisin.net>
 
@@ -4102,11 +9816,7 @@
 	* tests/check/Makefile.am:
 	* tests/check/audiotestsrc.flac:
 	* tests/check/pipelines/flacdec.c:
-	  add a test to check that we get all decoded bytes
-	  from a 10-buffer audiotestsrc flac, in the case of:
-	  - a full decode
-	  - a decode of a seek for the full file
-	  - a decode of a seek for a small part, smaller than the first buffer
+	  add a test to check that we get all decoded bytes from a 10-buffer audiotestsrc flac, in the case of:  - a full decode  - a decode of a seek for the full file  - a decode of a seek for a small part, smaller than the first buffer
 	  The test fails because flacdec drops the first outgoing buffer on a seek
 
 2009-03-03 10:06:52 +0100  Thomas Vander Stichele <thomas (at) apestaart (dot) org>
@@ -5216,8 +10926,7 @@
 
 	* ext/pulse/pulsesink.c:
 	* ext/pulse/pulsesink.h:
-	  pulsesink: Issue property change notification in streaming thread,
-	  rather than PA thread.
+	  pulsesink: Issue property change notification in streaming thread, rather than PA thread.
 	  pa_threaded_mainloop_lock() (a.o.) and by extension get_property should
 	  not be done from a PA thread, but the latter may occur as a result of a
 	  property change notification.  Fixes #571204 (though current situation
@@ -5526,8 +11235,7 @@
 	* ext/pulse/pulsesrc.h:
 	* ext/pulse/pulseutil.c:
 	* ext/pulse/pulseutil.h:
-	  Rewrite the pulse plugin, conditionally enabling new behaviour with
-	  newer pulseaudio.
+	  Rewrite the pulse plugin, conditionally enabling new behaviour with newer pulseaudio.
 	  Fixes: #567794
 	  * Hook pulsesink's volume property up with the stream volume -- not the
 	  sink volume in PA.