=== release 1.8.3 ===

2016-08-19  Sebastian Dröge <slomo@coaxion.net>

	* configure.ac:
	  releasing 1.8.3

2016-08-19 11:57:57 +0300  Sebastian Dröge <sebastian@centricular.com>

	* po/af.po:
	* po/az.po:
	* po/bg.po:
	* po/ca.po:
	* po/cs.po:
	* po/da.po:
	* po/de.po:
	* po/el.po:
	* po/en_GB.po:
	* po/eo.po:
	* po/es.po:
	* po/eu.po:
	* po/fi.po:
	* po/fr.po:
	* po/gl.po:
	* po/hr.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/ro.po:
	* po/ru.po:
	* po/sk.po:
	* po/sl.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:
	  Update .po files

2016-08-17 09:49:04 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: Allow mimetypes with properties as long as they're application/sdp
	  Some servers add properties like charset, e.g.
	  application/sdp; charset=utf8
	  Ideally we should also parse the charset and do conversion of all messages,
	  but that's for a later time.

2016-08-11 16:32:21 -0600  Thomas Bluemel <tbluemel@control4.com>

	* gst/udp/gstmultiudpsink.c:
	  multiudpsink: Initialize bytes_sent field.
	  This fixes endpoints not receiving any data intermittently.
	  https://bugzilla.gnome.org/show_bug.cgi?id=769773

2016-08-10 11:26:17 -0600  Thomas Bluemel <tbluemel@control4.com>

	* gst/rtpmanager/rtpjitterbuffer.c:
	  rtpjitterbuffer: Don't warn for duplicate packets
	  This is a normal scenario and should not be a warning.  This can
	  happen frequently when re-transmits of lost packets are enabled.
	  https://bugzilla.gnome.org/show_bug.cgi?id=762208

2016-08-08 13:49:19 +1000  Jan Schmidt <jan@centricular.com>

	* gst/multifile/gstsplitmuxsink.c:
	  splitmux: Fix typo converting to running time.
	  Use the correct collected timestamp.

2016-08-08 02:53:48 +1000  Jan Schmidt <jan@centricular.com>

	* gst/multifile/gstsplitmuxsink.c:
	* gst/multifile/gstsplitmuxsink.h:
	  Revert "splitmuxsink: Use GstBin async-handling instead of our own."
	  This reverts commit fa008f271a52f82dededc28bd81b020ca7939b47.
	  async-handling in GstBin causes the pipeline to spin at 100%
	  CPU as the top-level pipeline tries to change that state
	  to PLAYING constantly. This is a workaround for a core
	  problem, essentially, but an improvement in this case for now.

2016-08-08 00:56:38 +1000  Jan Schmidt <jan@centricular.com>

	* gst/multifile/gstsplitmuxsink.c:
	  splitmux: Recheck state after unlocking mutex.
	  After dropping the splitmux lock, re-check the state,
	  don't just fall through and sleep unconditionally,
	  as we may have already missed the wakeup.
	  https://bugzilla.gnome.org/show_bug.cgi?id=769514

2016-08-03 03:32:07 +1000  Jan Schmidt <jan@centricular.com>

	* gst/multifile/gstsplitmuxsrc.c:
	  splitmuxsrc: Don't stop and error on EOS flow return
	  Don't immediately halt on EOS flow return from downstream
	  due to out of segment. Let the demuxer handle it and send
	  EOS.

2016-07-25 18:20:03 +1000  Jan Schmidt <jan@centricular.com>

	* gst/multifile/gstsplitmuxsink.c:
	  splitmuxsink: Fix debug statement signedness.
	  The ts variable is a GstClockTime, don't print it
	  as a GstClockTimeDiff.

2016-07-17 22:41:02 +1000  Jan Schmidt <jan@centricular.com>

	* gst/multifile/gstsplitmuxsink.c:
	* gst/multifile/gstsplitmuxsink.h:
	  splitmuxsink: Handle negative running time
	  Use signed clock times for running time everywhere
	  so that we handle negative running times without
	  going haywire, similar to what queue and multiqueue
	  do these days.

2016-07-18 00:12:55 +1000  Jan Schmidt <jan@centricular.com>

	* gst/multifile/gstsplitmuxsink.c:
	  splitmuxsink: Drop lock when sending dummy event
	  When pushing the dummy event into the multiqueue,
	  drop the splitmux lock or else we might deadlock.

2016-07-25 13:34:02 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtp/gstrtpbvpay.c:
	* gst/rtp/gstrtpceltpay.c:
	* gst/rtp/gstrtpg722pay.c:
	* gst/rtp/gstrtph263ppay.c:
	* gst/rtp/gstrtph265pay.c:
	* gst/rtp/gstrtpilbcpay.c:
	  rtp: Filter with the filter caps in the payloader's getcaps

2016-06-30 01:56:41 +1000  Jan Schmidt <thaytan@noraisin.net>

	* gst/rtp/gstrtph264pay.c:
	  rtph264pay: Intersect with filter caps in getcaps function.
	  Always intersect with the filter caps in the getcaps function
	  to make sure we return a subset of what was requested.
	  Other payloaders also have this problem and need fixing
	  in future commits.

2016-06-30 14:40:40 +0200  Thomas Scheuermann <Thomas.Scheuermann@barco.com>

	* ext/jack/gstjackaudioclient.c:
	  jack: don't wait for callbacks if the jack server shut down
	  Otherwise we'll wait forever.
	  https://bugzilla.gnome.org/show_bug.cgi?id=747275

2016-07-07 18:23:07 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/matroska/matroska-mux.c:
	  matroskamux: Error out if we start writing data with some pads not having a codec id yet
	  This can only happen if a) upstream somehow gets around the CAPS event failing
	  or b) there never being any CAPS event.
	  The following code assumes that all pads have a codec-id.
	  https://bugzilla.gnome.org/show_bug.cgi?id=768509

2016-07-04 17:45:40 +0200  Sebastian Dröge <sebastian@centricular.com>

	* tests/check/elements/qtmux.c:
	  qtmux: Use complete AAC caps with codec_data in the tests

2016-07-04 16:58:38 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/audioparsers/gstaacparse.c:
	  aacparse: Reject raw AAC if no codec_data is found in the caps
	  If necessary, a demuxer will have to invent something here but this is only a
	  problem with non-conformant files anyway.

2016-07-04 16:55:32 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Invent AAC codec_data if none is present
	  Without, raw AAC can't be handled and we have some information available in
	  the decoder that most likely allows us to decode the stream in one way or
	  another. This is the same code already used by matroskademux for the same
	  reasons, and ffmpeg/vlc play such files just fine too by guesswork.

2016-07-04 14:54:13 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/gstqtmux.c:
	  qtmux: Reject raw AAC caps without codec_data
	  The resulting file is not going to be playable without guesswork and raw caps
	  should always have codec_data.

2016-07-05 21:11:35 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: Set to PLAYING after a seek again after setting up the segment and everything else
	  There's a small window for a race condition otherwise.

2016-07-01 10:05:00 +0000  Brad Lackey <blackey@gmail.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: Don't disable UDP protocols on redirecting
	  https://bugzilla.gnome.org/show_bug.cgi?id=768232

2016-07-10 21:30:27 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/udp/gstudpsrc.c:
	  udpsrc: Use correct in6_pktinfo struct instead of in_pktinfo
	  Fixes the build on FreeBSD, which does not have the latter.
	  https://bugzilla.gnome.org/show_bug.cgi?id=768623

2016-07-08 12:30:10 -0400  Nicolas Dufresne <nicolas.dufresne@collabora.com>

	* sys/v4l2/gstv4l2.c:
	* sys/v4l2/gstv4l2bufferpool.c:
	* sys/v4l2/gstv4l2deviceprovider.c:
	* sys/v4l2/gstv4l2object.c:
	* sys/v4l2/gstv4l2object.h:
	* sys/v4l2/gstv4l2radio.c:
	* sys/v4l2/gstv4l2sink.c:
	* sys/v4l2/v4l2_calls.c:
	  Revert "v4l2: use opened device caps instead of physical device ones"
	  This reverts commit d6c8cb63859368e7100b0002354ad742e4286eba.
	  Apparently this was not fully tested and cause regressions.

2016-07-02 01:56:07 -0700  Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>

	* sys/v4l2/gstv4l2.c:
	* sys/v4l2/gstv4l2bufferpool.c:
	* sys/v4l2/gstv4l2deviceprovider.c:
	* sys/v4l2/gstv4l2object.c:
	* sys/v4l2/gstv4l2object.h:
	* sys/v4l2/gstv4l2radio.c:
	* sys/v4l2/gstv4l2sink.c:
	* sys/v4l2/v4l2_calls.c:
	  v4l2: use opened device caps instead of physical device ones
	  The same physical device can export multiple devices. In
	  this case, the capabilities field now contains a union of
	  all caps available from all exported V4L2 devices alongside
	  a V4L2_CAP_DEVICE_CAPS flag that should be used to decide
	  what capabilities to consider. In our case, we need the
	  ones from the exported device we are using.
	  https://bugzilla.gnome.org/show_bug.cgi?id=768195

2016-07-01 17:28:17 +0900  Seungha Yang <sh.yang@lge.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Push caps only when it was updated
	  Commit 7873bede3134b15e5066e8d14e54d1f5054d2063 caused new caps
	  event per moof without consideration of duplication.
	  https://bugzilla.gnome.org/show_bug.cgi?id=768268

2016-06-30 15:01:46 +0200  Jonas Holmberg <jonashg@axis.com>

	* gst/rtp/gstrtph265depay.c:
	  rtph265depay: fix invalid memory access
	  10 bytes was allocated for stream_format but size of "byte-stream" is
	  more. Use g_strdup() instead.
	  https://bugzilla.gnome.org/show_bug.cgi?id=753760

2016-06-28 16:44:50 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/udp/gstudpsrc.c:
	  udpsrc: Windows has no ipi_spec_dst in struct in_pktinfo

2016-06-28 15:15:14 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/udp/gstudpsrc.c:
	  udpsrc: #define __APPLE_USE_RFC_3542 to be able to use IPV6_PKTINFO on OSX/iOS

2016-06-28 15:08:04 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/udp/gstudpsrc.c:
	  udpsrc: Move #includes around to a) work around broken glibc header and b) Windows

2016-06-28 14:25:03 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/udp/gstudpsrc.c:
	  udpsrc: Fix compilation on Windows and *BSD/OSX

2016-06-23 20:21:59 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/udp/gstudpsrc.c:
	  udpsrc: Filter out multicast packets that are not for our multicast address
	  https://bugzilla.gnome.org/show_bug.cgi?id=767980

2016-06-28 10:57:27 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: When seeking, consider the current element state or pending state instead of the RTSP state
	  If we consider the RTSP state, what can happen is that it is PLAYING but the
	  element already asynchronously tried to PAUSE and it just did not happen yet.
	  We would then override this setting to PAUSED (while the element actually is
	  in PAUSED) and set the RTSP state to PLAYING again. This would then cause us
	  to produce packets while the sinks are all PAUSED, piling up thousands of
	  packets in the rtpjitterbuffer and other elements and finally failing.

2016-06-17 16:08:08 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/gstqtmux.c:
	  qtmux: The prores variant is stored in the variant field, not format
	  And the caps in the sink pad template already used variant (only).

2016-05-23 10:18:48 +0200  Miguel París Díaz <mparisdiaz@gmail.com>

	* gst/rtpmanager/rtpsession.c:
	  rtpsession: take the lock when changing stats
	  https://bugzilla.gnome.org/show_bug.cgi?id=766025

2016-06-15 11:19:43 +0200  Jürgen Slowack <jurgen.slowack@barco.com>

	* gst/rtp/gstrtph265pay.c:
	  rtph265: fix NAL unit type parsing and SPS/PPS/VPS detection
	  Fixes sps/pps/vps insertion via the config-interval property.
	  https://bugzilla.gnome.org//show_bug.cgi?id=767680

2016-06-10 13:42:01 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/flv/gstflvdemux.c:
	  flvdemux: fix date parsing when there are trailing spaces
	  Fixes parsing of "Thu May 11 15:57:46 2006 ".
	  https://bugzilla.gnome.org/show_bug.cgi?id=767496