=== release 1.1.4 ===

2013-08-28  Sebastian Dröge <sebastian.droege@collabora.co.uk>

	* configure.ac:
	  releasing 1.1.4

2013-08-28 12:32:10 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* po/pt_BR.po:
	  po: update translations

2013-08-27 15:25:16 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/matroska/matroska-mux.c:
	  matroska-mux: remove framerate restriction
	  Remove the framerate restriction on the caps.

2013-08-27 09:38:16 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpsession.c:
	  session: only update next check time when reconsidering
	  Don't update the next RTCP check time in all cases but only when we
	  reconsidered. This avoids delaying sending a full RTCP packet when we
	  are doing early feedback.

2013-08-27 09:37:33 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpsession.c:
	  session: add more debug

2013-08-27 09:34:46 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	* gst/rtpmanager/gstrtpsession.c:
	  jitterbuffer: fix types of the retransmission event

2013-08-27 09:33:03 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: only timeout EXPECTED timers on gap
	  Only timeout the EXPECTED timers when we detect a large seqnum gap.

2013-08-26 13:47:53 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* configure.ac:
	  configure.ac: Don't set BZ2_LIBS if bz2 is not found

2013-08-26 11:50:27 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpsession.c:
	  rtsession: fix locking
	  We need to take the session lock when getting and manipulating the
	  source.

2013-08-26 11:50:13 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpsession.c:
	  rtpsession: add some more debug

2013-08-20 22:12:03 +0200  Mathieu Duponchelle <mathieu.duponchelle@epitech.eu>

	* gst/videomixer/videomixer2.c:
	  videomixer: don't send flush_stop twice.
	  If we get flush start and a seek we need to only send flush_stop once.
	  More info at #706441

2013-08-23 15:56:43 +0100  Tim-Philipp Müller <tim@centricular.net>

	* gst/multipart/multipartdemux.c:
	* gst/multipart/multipartdemux.h:
	  multipartdemux: propagate discont

2013-08-23 15:49:47 +0100  Tim-Philipp Müller <tim@centricular.net>

	* gst/multipart/multipartdemux.c:
	  multipartdemux: remove dynamic sourcpads when going from PAUSED to READY

2013-08-23 15:29:28 +0100  Tim-Philipp Müller <tim@centricular.net>

	* gst/multipart/multipartdemux.c:
	* gst/multipart/multipartdemux.h:
	  multipartdemux: timestamp output buffers based on first input buffer that provided bytes not last
	  https://bugzilla.gnome.org/show_bug.cgi?id=637754

2013-08-23 15:47:25 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtprtxqueue.c:
	* gst/rtpmanager/gstrtprtxqueue.h:
	  rtxqueue: add property to configure queue size

2013-08-23 12:07:55 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* tests/examples/rtp/client-H264-rtx.sh:
	* tests/examples/rtp/server-VTS-H264-rtx.sh:
	  tests: add retransmission example

2013-08-23 11:55:02 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpbin.c:
	* gst/rtpmanager/gstrtpbin.h:
	  rtpbin: proxy jitterbuffer do-retransmission property

2013-08-23 11:17:45 +0200  Michael Olbrich <m.olbrich@pengutronix.de>

	* gst/avi/gstavimux.c:
	  avimux: unmap the correct buffer
	  The audio buffer was mapped so unmap it and not the video buffer
	  https://bugzilla.gnome.org/show_bug.cgi?id=706642

2013-08-18 23:32:22 -0400  Olivier Crête <olivier.crete@collabora.com>

	* ext/pulse/pulsesink.c:
	* ext/pulse/pulsesink.h:
	  pulsesink: Add property to find out the device currently in use
	  https://bugzilla.gnome.org/show_bug.cgi?id=590768

2013-08-18 23:31:15 -0400  Olivier Crête <olivier.crete@collabora.com>

	* ext/pulse/pulsesink.c:
	  pulsesink: De-duplicate code to get the current sink input info
	  https://bugzilla.gnome.org/show_bug.cgi?id=590768

2013-08-18 22:27:37 -0400  Olivier Crête <olivier.crete@collabora.com>

	* ext/pulse/pulsesink.c:
	  pulsesink: Implement changing the device while playing
	  https://bugzilla.gnome.org/show_bug.cgi?id=590768

2013-08-18 23:32:22 -0400  Olivier Crête <olivier.crete@collabora.com>

	* ext/pulse/pulsesrc.c:
	* ext/pulse/pulsesrc.h:
	  pulsesrc: Add property to find out the device currently in use
	  https://bugzilla.gnome.org/show_bug.cgi?id=590768

2013-08-18 23:31:15 -0400  Olivier Crête <olivier.crete@collabora.com>

	* ext/pulse/pulsesrc.c:
	  pulsesrc: De-duplicate code to get the current source output info
	  https://bugzilla.gnome.org/show_bug.cgi?id=590768

2013-08-18 22:27:37 -0400  Olivier Crête <olivier.crete@collabora.com>

	* ext/pulse/pulsesrc.c:
	  pulsesrc: Implement changing the device while playing
	  https://bugzilla.gnome.org/show_bug.cgi?id=590768

2013-08-22 14:55:14 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* configure.ac:
	  configure: Fix bz2 configure check for Windows
	  Due to function decorations on Windows AC_CHECK_LIB can't be used to check for bz2.
	  https://bugzilla.gnome.org/show_bug.cgi?id=465924

2013-02-22 20:57:00 +0900  Akihiro Tsukada <atsukada@users.sourceforge.net>

	* ext/pulse/pulsesink.c:
	* ext/pulse/pulsesink.h:
	* ext/pulse/pulseutil.c:
	* ext/pulse/pulseutil.h:
	  pulsesink: Add support for AAC pass-through
	  https://bugzilla.gnome.org/show_bug.cgi?id=694445

2013-06-24 17:29:37 +0200  Kishore Arepalli <kishore.arepalli@gmail.com>

	* ext/gdk_pixbuf/gstgdkpixbufoverlay.c:
	  gdkpixbufoverlay: crashes if any property changes during playback when location property is not set
	  https://bugzilla.gnome.org/show_bug.cgi?id=702988

2013-08-21 14:54:26 -0400  Olivier Crête <olivier.crete@collabora.com>

	* ext/pulse/pulsesink.c:
	* ext/pulse/pulsesink.h:
	* ext/pulse/pulsesrc.c:
	* ext/pulse/pulseutil.h:
	  pulse: Share static caps definition between src and sink
	  The src was also missing 24-bit sample formats

2013-08-21 16:53:59 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtprtxqueue.c:
	* gst/rtpmanager/gstrtprtxqueue.h:
	  rtx: various improvements
	  Use locking
	  Don't push from the event handler, collected packets in a queue and push from
	  the chain function.
	  Clear queues on shutdown.

2013-08-21 16:50:59 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpsession.c:
	  session: generate events correctly
	  Do correct shifting of the bitmask for lost packets.

2013-08-21 16:47:40 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpmanager.c:
	  rtp: register rtx element better

2013-08-21 16:32:50 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* sys/directsound/gstdirectsoundsink.c:
	  directsoundsink: WAVEFORMATEX is unsigned for 8 bit integers, and signed for others
	  Probably fixes
	  https://bugzilla.gnome.org/show_bug.cgi?id=705477

2013-08-21 13:03:34 +0100  Tim-Philipp Müller <tim@centricular.net>

	* ext/jpeg/gstjpegenc.c:
	  jpegenc: don't ignore return value from _finish_frame()
	  gst_video_encoder_finish_frame() will return FLOW_OK here if
	  there's no output buffer.

2013-08-21 12:56:35 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtpjpegdepay.c:
	  jpegdepay: add some more debug

2013-08-21 12:10:00 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtpgstdepay.c:
	* gst/rtp/gstrtpgstdepay.h:
	  rtpgstdepay: only push events when they changed
	  Keep track of the STREAM_START and TAG events and only push them
	  when they changed.

2013-08-21 10:52:59 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtpgstpay.c:
	  rtpgstpay: taglists should not be merged in 1.0

2013-08-21 10:28:50 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtpgstdepay.c:
	  rtpgstdepay: flush on FLUSH_STOP event

2013-08-21 10:03:52 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtpgstpay.c:
	  rtpgstpay: reset on state change
	  Do full reset on state change to READY

2013-08-21 09:55:20 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtpgstpay.c:
	  rtpgstpay: reset on FLUSH_STOP
	  Clear the adapter and pending buffer list on FLUSH_STOP.

2013-08-21 09:39:30 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtpgstpay.c:
	  rtpgstpay: don't use clock for config interval
	  We can't use the clock to time our config-interval because we are not
	  live (or there might not be a clock or the clock might not be running).
	  Instead just simply take the timestamp diff.

2013-08-21 09:33:04 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtpgstpay.h:
	  rtpgstay: don't use // comments

2013-08-08 11:55:22 -0400  Youness Alaoui <youness.alaoui@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: Fix response argument in handle-request signal

2013-08-08 11:54:41 -0400  Youness Alaoui <youness.alaoui@collabora.co.uk>

	* gst/rtsp/gstrtspsrc.c:
	* gst/rtsp/gstrtspsrc.h:
	  rtspsrc: Add sdes property and proxy it to rtpbin

2013-08-07 09:47:35 -0400  Youness Alaoui <youness.alaoui@collabora.co.uk>

	* gst/rtp/gstrtpgstpay.c:
	* gst/rtp/gstrtpgstpay.h:
	  Send a stream-start whenever we send tags This is to make sure tags are cleared on the client if the stream-start was previously lost, otherwise, the client may end up with a merged taglist of multiple songs

2013-07-25 21:12:05 -0400  Youness Alaoui <youness.alaoui@collabora.co.uk>

	* gst/rtp/gstrtpgstpay.c:
	* gst/rtp/gstrtpgstpay.h:
	  rtpgstpay: Add a config-interval property to resend the caps/tags at a regular interval This is useful in case the packet containing the inlined caps was lost or if new client joins an already running RTP stream and they missed the previous tag events. This also makes the payloader keep a list of merged tags so the retransmitted tag event contains all previously received. A STREAM_START event will flush the list of tags.

2013-07-25 21:10:10 -0400  Youness Alaoui <youness.alaoui@collabora.co.uk>

	* gst/rtp/gstrtpgstpay.c:
	  rtpgstpay: Refactor the setcaps and use new method to send arbitrary caps at any time

2013-07-25 21:03:34 -0400  Youness Alaoui <youness.alaoui@collabora.co.uk>

	* gst/rtp/gstrtpgstpay.c:
	  rtpgstpay: Do not flush events for stream-start and avoid conflict between event and pending inline caps

2013-07-25 20:54:50 -0400  Youness Alaoui <youness.alaoui@collabora.co.uk>

	* gst/rtp/gstrtpgstpay.c:
	* gst/rtp/gstrtpgstpay.h:
	  rtpgstpay: Add a create_from_adapter API and use a list of GstBufferList This is necessary to fix event/caps sending. If we send a STREAM_START packet, it will cause an error because the stream didn't receive its caps and new-segment events, so we must wait for the first buffer before sending the stream-start event buffer. However, the caps will be sent at the same time and so the 'inline caps' will be set for the event. We need to be able to payload individual packets (data, caps or events) and only send them when we call flush.

2013-07-25 17:56:38 -0400  Youness Alaoui <youness.alaoui@collabora.co.uk>

	* gst/rtp/gstrtpgstdepay.c:
	* gst/rtp/gstrtpgstpay.c:
	  rtpgstpay: Add etype=4 for payloading GST_EVENT_STREAM_START

2013-07-25 17:52:16 -0400  Youness Alaoui <youness.alaoui@collabora.co.uk>

	* gst/rtp/gstrtpgstpay.c:
	  rtpgstpay: Fix typo, GST_EVENT_CUSTOM_BOTH has etype of 3

2013-08-20 14:36:59 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: handle EOS
	  When the queue is empty, and we received EOS, pause and push an EOS
	  event downstream.
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=706387

2013-08-20 10:26:15 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: update docs

2013-08-20 10:25:17 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: update all timers
	  Keep looping over all registered timers so that we can mark them lost instead of
	  stopping as soon as we find the timer for the current seqnum.

2013-08-20 08:55:50 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: remove unused variables

2013-08-19 21:10:00 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: reorganize timer handling
	  Restructure handling of incomming packet and the gap with the expected seqnum
	  and register all timers from the _chain function.
	  Convert a timer to a LOST packet timer when the max amount of retransmission
	  requests has been reached.

2013-08-19 21:37:44 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: refactor packet spacing calculation

2013-08-19 21:34:38 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: keep track of last seqnum and dts

2013-08-19 21:29:49 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: small cleanups

2013-08-19 21:21:08 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: reset retransmission timers in add/reschedule
	  Reset the retransmission timers when adding and rescheduling a timer.

2013-08-19 21:12:13 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: rename variables for packet spacing

2013-08-19 14:58:01 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: remove lost timer when we get the packet
	  When we receive a packet, also remove the LOST timer for it.

2013-08-19 14:56:49 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: expected seqnum must increase
	  Only update the expected seqnum when it is bigger than the previous expected
	  seqnum.

2013-08-19 14:55:49 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: add more debug

2013-08-12 16:15:54 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/Makefile.am:
	* gst/rtpmanager/gstrtpmanager.c:
	* gst/rtpmanager/gstrtprtxqueue.c:
	* gst/rtpmanager/gstrtprtxqueue.h:
	  rtxqueue: add retransmission queue element

2013-08-12 14:53:33 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpsession.c:
	  session: add some docs

2013-08-06 16:29:54 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpsession.c:
	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpsession.h:
	  session: handle NACK feedback and generate events
	  Handle and parse the feedback NACK packets and generate a Retransmission
	  event for each NACKed packet

2013-08-19 13:19:42 -0400  Olivier Crête <olivier.crete@collabora.com>

	* sys/v4l2/gstv4l2object.c:
	  v4l2: Add forward declaration for gst_v4l2_object_get_format_list

2012-10-22 17:58:07 -0400  Olivier Crête <olivier.crete@collabora.com>

	* sys/v4l2/gstv4l2object.c:
	* sys/v4l2/gstv4l2object.h:
	* sys/v4l2/gstv4l2sink.c:
	* sys/v4l2/gstv4l2sink.h:
	* sys/v4l2/gstv4l2src.c:
	* sys/v4l2/gstv4l2src.h:
	  v4l2: De-duplicate caps probing between src and sink

2013-08-13 17:32:17 -0400  Olivier Crête <olivier.crete@collabora.com>

	* ext/pulse/Makefile.am:
	* ext/pulse/pulseprobe.c:
	* ext/pulse/pulseprobe.h:
	* ext/pulse/pulsesink.c:
	* ext/pulse/pulsesink.h:
	* ext/pulse/pulsesrc.c:
	* ext/pulse/pulsesrc.h:
	  pulse: Remove unused GstPulseProbe

2013-08-19 12:46:45 -0400  Olivier Crête <olivier.crete@collabora.com>

	* sys/v4l2/gstv4l2tuner.c:
	* sys/v4l2/tuner.c:
	* sys/v4l2/tunerchannel.c:
	* sys/v4l2/tunernorm.c:
	  v4l2: Use G_DEFINE_ macros for added thread safety

2013-08-17 11:28:13 +0200  Thibault Saunier <thibault.saunier@collabora.com>

	* gst/videomixer/videomixer2.c:
	* gst/videomixer/videomixer2.h:
	  videomixer: Do not send flush_stop ourself after a flush_start
	  When we receive a flush_start, we should wait for the next flush_stop
	  and foward it, not create a flush_stop ourself.

2013-08-16 17:10:31 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtp/gstrtph264depay.c:
	  h264depay: init debug category early
	  Init the debug variable when we register the element because it is also used by
	  the payloader element when it calls the add_sps_pps method.

2013-08-16 13:26:28 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* ext/flac/gstflacenc.c:
	  flacenc: Properly set headers via the base class instead of just pushing them downstream
	  Prevents buffers from being send before the caps and segment events.

2013-08-15 10:59:10 +0100  Chris Bass <floobleflam@gmail.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: check denominator isn't zero before scaling duration.
	  When gst_qtdemux_configure_stream sets fps_d, check that n_samples is
	  non-zero before using it as a denominator to scale the stream duration.
	  https://bugzilla.gnome.org/show_bug.cgi?id=706076

2013-08-15 15:08:05 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* ext/jpeg/gstjpegdec.c:
	* ext/jpeg/gstjpegenc.c:
	* ext/libpng/gstpngdec.c:
	* ext/vpx/gstvp8dec.c:
	* ext/vpx/gstvp9dec.c:
	  ext: Use new flush vfunc of video codec base classes and remove reset implementations

2013-08-14 16:19:32 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: forward flush before stopping dataflow
	  First forward the flush event and then stop our loop function.

2013-08-14 13:10:32 +0100  Tim-Philipp Müller <tim@centricular.net>

	* configure.ac:
	  configure: require libsoup >= 2.38
	  Bump libsoup requirement for newer API used, like headers_get_one().
	  2.38 is from early 2012 and is in linen with our GLib requirement.

2013-08-14 11:54:19 +0100  Tim-Philipp Müller <tim@centricular.net>

	* ext/soup/gstsouphttpsrc.c:
	  soup: don't use deprecated soup_message_headers_get() API

2013-08-13 17:44:50 +0200  Edward Hervey <edward@collabora.com>

	* .gitignore:
	  .gitignore: Ignore files from automake test-driver

2013-08-12 15:28:34 -0400  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtp/gstrtph264pay.c:
	* gst/rtp/gstrtph264pay.h:
	  rtph264pay: Use the SPS/PPS handling function from the depayloader
	  Remove duplicated copies
	  https://bugzilla.gnome.org/show_bug.cgi?id=705553

2013-08-12 15:26:08 -0400  Olivier Crête <olivier.crete@collabora.com>

	* gst/rtp/gstrtph264depay.c:
	* gst/rtp/gstrtph264depay.h:
	  rtph264depay: Make the SPS/PPS deduplication function generic
	  Make it not touch any internals of the depayloader
	  https://bugzilla.gnome.org/show_bug.cgi?id=705553

2013-08-13 14:09:20 +0100  Chris Bass <floobleflam@gmail.com>

	* gst/audioparsers/gstaacparse.c:
	  aacparse: allow conversion from raw AAC to ADTS
	  This patch will prepend ADTS headers to raw AAC audio frames, allowing
	  upstream elements to link to decoders that only support AAC in ADTS format.
	  Note that no error correction bits are added to ADTS frames in this code.
	  https://bugzilla.gnome.org/show_bug.cgi?id=615740

2013-08-13 12:44:11 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: Only free GCheckSum after its last usage
	  https://bugzilla.gnome.org/show_bug.cgi?id=705760

2013-08-13 12:02:29 +0200  Andoni Morales Alastruey <ylatuya@gmail.com>

	* ext/soup/gstsouphttpsrc.c:
	  souphttpsrc: fix critical setting a NULL uri redirection

2013-07-13 01:50:56 +0200  Andoni Morales Alastruey <ylatuya@gmail.com>

	* ext/soup/gstsouphttpsrc.c:
	* ext/soup/gstsouphttpsrc.h:
	  souphttpsrc: add redirection to the URI query

2013-07-31 10:42:07 +0200  Matej Knopp <matej.knopp@gmail.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: elst should offset samples instead of buffers
	  The current approach where buffers are offset is not ideal, as during seek
	  and loop current time is compared to sample times.
	  https://bugzilla.gnome.org/show_bug.cgi?id=700264

2013-08-07 19:32:07 +0200  Thibault Saunier <thibault.saunier@collabora.com>

	* gst/videomixer/videomixer2.c:
	* tests/check/elements/videomixer.c:
	  videomixer: Send EOS if buf_end >= segment.stop
	  That means the whole segment is already played, and we are sure we
	  are EOS at that point.
	  Also handle segment seeks, and do not send EOS in that case.

2013-08-04 14:40:38 +0200  Matej Knopp <matej.knopp@gmail.com>

	* gst/avi/gstavidemux.c:
	  avidemux: send proper stream_start event
	  https://bugzilla.gnome.org//show_bug.cgi?id=705449

2013-08-08 11:51:17 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* gst/matroska/ebml-read.c:
	* gst/matroska/matroska-demux.c:
	  matroskademux: Don't print warnings during flushing and stop as soon as possible
	  https://bugzilla.gnome.org//show_bug.cgi?id=705442

2013-08-07 11:14:38 +0100  Tim-Philipp Müller <tim@centricular.net>

	* gst/rtp/gstrtpvp8depay.c:
	  rtpvp8depay: mark key frames and delta frames properly
	  https://bugzilla.gnome.org/show_bug.cgi?id=705550

2013-08-05 23:23:57 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpsession.c:
	  session: add NACK feedback in RTCP

2013-08-05 23:22:16 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpsource.c:
	* gst/rtpmanager/rtpsource.h:
	  source: add methods to register NACK
	  Add a method to register a missing packet for an ssrc along with
	  methods to get the missing packets and clear them.

2013-08-04 23:05:36 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpsession.c:
	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpsession.h:
	  session: handle Retransmission event and schedule NACK
	  Handle the retransmission event from downstream and use it to schedule a NACK
	  request.

2013-08-05 23:20:29 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpsession.c:
	  session: pass data to remove func
	  Pass the data to the remove function because we are going to deref it when there
	  is pli or fir.

2013-08-06 15:28:50 +0200  Thibault Saunier <thibault.saunier@collabora.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Fix compilation

2013-08-06 15:17:44 +0200  Thibault Saunier <thibault.saunier@collabora.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Raw buffer DTS should always be CLOCK_TIME_NONE

2013-08-06 11:58:38 +0200  Thibault Saunier <thibault.saunier@collabora.com>

	* gst/videomixer/videomixer2.c:
	  videomixer: Make sure to send EOS if the buffer end time equals the segment end time
	  Otherwize EOS never gets sent in that particular case.

2013-08-05 08:49:50 +0200  Sjoerd Simons <sjoerd.simons@collabora.co.uk>

	* gst/goom/gstgoom.c:
	  goom: Ensure src caps are writable
	  In some cases the src caps determined by goom weren't writable, causing
	  a bunch of assertion failures and failed caps. Fixed by always
	  explicitely making the caps writable
	  https://bugzilla.gnome.org/show_bug.cgi?id=705475

2013-08-04 23:18:29 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpsession.c:
	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpsession.h:
	  session: use common send_rtcp method
	  Reuse the send_rtcp method that already asks for the current time when
	  requesting a keyframe.

2013-08-04 23:12:50 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpsession.c:
	* gst/rtpmanager/rtpsession.h:
	  session: Don't use ClockTimeDiff for unsigned delays

2013-08-04 16:52:15 +0200  Edward Hervey <edward@collabora.com>

	* gst/isomp4/gstqtmux.c:
	  qtmux: Use buffer PTS if DTS is not set
	  Avoids ending up with completely bogus scaled duration/pts when new
	  buffers have invalid DTS.

2013-08-04 14:32:47 +0100  Tim-Philipp Müller <tim@centricular.net>

	* tests/check/elements/souphttpsrc.c:
	  tests: skip https test if there's no TLS support in soup/glib

2013-08-04 11:20:41 +0100  Tim-Philipp Müller <tim@centricular.net>

	* gst/rtsp/gstrtpdec.c:
	  rtpdec: use generic marshaller

2013-08-04 10:52:33 +0100  Tim-Philipp Müller <tim@centricular.net>

	* Makefile.am:
	* sys/v4l2/.gitignore:
	* sys/v4l2/Makefile.am:
	* sys/v4l2/gstv4l2-marshal.list:
	* sys/v4l2/tuner-marshal.list:
	* sys/v4l2/tuner.c:
	* sys/v4l2/tuner.h:
	* win32/MANIFEST:
	* win32/common/tuner-enumtypes.c:
	* win32/common/tuner-enumtypes.h:
	* win32/common/tuner-marshal.c:
	* win32/common/tuner-marshal.h:
	  v4l2: remove unused enumtypes and use generic marshaller

2013-08-04 10:47:38 +0100  Tim-Philipp Müller <tim@centricular.net>

	* Makefile.am:
	* gst/udp/.gitignore:
	* win32/common/gstudp-enumtypes.c:
	* win32/common/gstudp-enumtypes.h:
	* win32/common/gstudp-marshal.c:
	* win32/common/gstudp-marshal.h:
	  udp: remove unused marshal and enumtypes files

2013-08-04 09:38:19 +0100  Tim-Philipp Müller <tim@centricular.net>

	* Makefile.am:
	* gst/rtpmanager/.gitignore:
	* gst/rtpmanager/Makefile.am:
	* gst/rtpmanager/gstrtpbin-marshal.list:
	* gst/rtpmanager/gstrtpbin.c:
	* gst/rtpmanager/gstrtpjitterbuffer.c:
	* gst/rtpmanager/gstrtpptdemux.c:
	* gst/rtpmanager/gstrtpsession.c:
	* gst/rtpmanager/gstrtpssrcdemux.c:
	* gst/rtpmanager/rtpsession.c:
	* win32/MANIFEST:
	* win32/common/gstrtpbin-marshal.c:
	* win32/common/gstrtpbin-marshal.h:
	  rtpmanager: use generic marshaller

2013-08-04 00:13:07 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: send event in right direction

2013-08-02 17:38:34 -0700  David Schleef <ds@schleef.org>

	* configure.ac:
	* tests/check/Makefile.am:
	  tests: create/remove orc directory at proper time
	  Before automake creates .deps directories, and during distclean.

2013-08-03 00:25:44 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpsession.c:
	  session: add FIR and PLI like other RTCP packets
	  Add the FIR and PLI packets like the other RTCP packet instead of from the
	  on-sending-rtcp default signal handler.

2013-08-02 17:22:55 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: fix property ranges

2013-08-02 16:42:52 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: push retransmission events

2013-08-02 14:12:16 +0200  Lubosz Sarnecki <lubosz@gmail.com>

	* configure.ac:
	  build: add subdir-objects to AM_INIT_AUTOMAKE
	  Fixes warnings with automake 1.14
	  https://bugzilla.gnome.org/show_bug.cgi?id=705350

2013-08-02 14:54:56 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: add support for retransmission retry
	  When we didn't receive a packet after requesting retransmission, retry
	  asking for retransmission for a certain period.

2013-08-02 14:19:54 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: add properties
	  Add properties to control retransmission parameters

2013-08-02 12:44:58 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: use corrected timeout when rescheduling
	  When we recalculate the timeout, use the corrected timeout value depending on
	  the timer type.

2013-08-02 12:43:00 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: update timers after queueing
	  Else we might update the timer needlessly for duplicates.

2013-08-02 12:42:08 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: move method up

2013-08-02 06:28:32 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: small cleanup

2013-08-01 23:26:06 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: unschedule old expected packets
	  When we receive a new packet, unschedule old outstanding packets when their
	  seqnum is too far away.

2013-08-01 23:29:23 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: refactor timer update

2013-08-01 23:24:29 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: update timers when removing
	  Update the timers when we remove a timer.
	  Handle canceled timers, make them unschedule the current timer and
	  trigger the timeout code.

2013-08-01 23:22:02 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: fix typo

2013-08-01 15:40:52 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: improve timeout management
	  If we change the seqnum of an existing timer and we were waiting for
	  that timer, unschedule it. If we change the timeout of an existing timer and we
	  were waiting on it, only unschedule when the new time is smaller.

2013-08-01 15:05:35 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: install timer for expected arrival
	  Install a timer that is triggered when the expected arrival time of a packet
	  expired.

2013-08-01 14:56:00 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: improve unschedule of timers
	  Conflicts:
	  gst/rtpmanager/gstrtpjitterbuffer.c

2013-08-01 12:21:53 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: move code around

2013-08-01 12:07:11 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: estimate inter packet spacing
	  When we see two packets with consecutive seqnums and a different RTP time, use
	  the DTS difference as the inter packet spacing estimate.

2013-08-01 12:01:15 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: keep track of current timeout

2013-08-01 11:49:10 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: cleanup timer handling

2013-08-01 11:40:41 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: reset is only possible with a GAP

2013-08-01 11:29:32 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	* gst/rtpmanager/rtpjitterbuffer.c:
	  jitterbuffer: operate on DTS
	  Make the jitterbuffer schedule the timeouts based on the DTS instead
	  of the PTS. This makes it all smoother with reordered frames and gives
	  the decoder time to reorder the frames in time.

2013-08-01 11:14:12 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: rename timout variable

2013-07-31 17:08:58 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: small cleanup

2013-07-31 16:59:58 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: block output in paused or buffering

2013-07-31 16:59:09 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: store pts in timer
	  Only store the pts in the timer so that we can both do timeouts with timings on
	  the input and output of the jitterbuffer.

2013-07-30 23:14:24 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpjitterbuffer: refactor jitterbuffer
	  Refactor the jitterbuffer code. Make separate function for peeking a buffer,
	  pushing the next buffer, waiting for timeouts and handling the timeouts.
	  The main loop now tries to push as many buffers as it can until it runs out of
	  buffers or when it detects a seqnum discont. Then it will wait for some event to
	  happen before attempting to push more buffers.
	  Make methods to register timeouts in an array. These timeouts are registered
	  when we detect a missing packet, sync for the first packet or when we find an
	  estimation for the end-of-stream.
	  This greatly simplifies and clarifies the code and also makes it possible to
	  register more complicated timeout schemes later.

2013-07-30 18:52:58 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/rtpjitterbuffer.c:
	  rtpjitterbuffer: use NULL to ignore percent
	  If we pass NULL to pop and push we ignore the percent result.

2013-07-30 07:00:19 +0200  Wim Taymans <wim.taymans@collabora.co.uk>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  jitterbuffer: refactor
	  Move eos estimation into separate function

2013-07-30 14:28:19 +0100  Tim-Philipp Müller <tim@centricular.net>

	* gst/flv/gstflvdemux.c:
	  flvdemux: don't leak stream_id string
	  https://bugzilla.gnome.org/show_bug.cgi?id=705142

2013-07-29 19:53:52 +0100  Tim-Philipp Müller <tim@centricular.net>

	* po/LINGUAS:
	* po/da.po:
	* po/de.po:
	* po/el.po:
	* po/gl.po:
	* po/hr.po:
	* po/hu.po:
	* po/ja.po:
	* po/nb.po:
	* po/nl.po:
	* po/pl.po:
	* po/ru.po:
	* po/sl.po:
	* po/tr.po:
	* po/uk.po:
	* po/vi.po:
	* po/zh_CN.po:
	  po: update translations

2013-07-29 19:48:54 +0100  Tim-Philipp Müller <tim@centricular.net>

	* tests/check/elements/.gitignore:
	  tests: ignore new test binaries

2013-07-29 14:47:49 +0200  Sebastian Dröge <slomo@circular-chaos.org>

	* configure.ac:
	  Back to development