=== release 1.6.2 ===

2015-12-14  Sebastian Dröge <slomo@coaxion.net>

	* configure.ac:
	  releasing 1.6.2

2015-12-14 19:12:53 +0100  Sebastian Dröge <sebastian@centricular.com>

	* po/el.po:
	* po/hu.po:
	* po/sv.po:
	* po/uk.po:
	  po: Update translations

2015-12-14 14:15:00 +0100  Thomas Roos <thomas.roos@industronic.de>

	* sys/directsound/gstdirectsoundsink.c:
	  directsoundsink: Fix handling of the mute property
	  - set mute value at startup
	  - correct set and get mute functions
	  https://bugzilla.gnome.org/show_bug.cgi?id=755106

2015-12-01 18:20:23 +0100  Wim Taymans <wtaymans@redhat.com>

	* sys/v4l2/gstv4l2bufferpool.c:
	  v4l2bufferpool: don't block when resurecting a buffer
	  When we are resurecting a buffer, don't block. instead let us copy a
	  buffer.

2015-12-11 11:23:13 +0100  Thomas Roos <thomas.roos@industronic.de>

	* sys/directsound/gstdirectsoundsink.c:
	  directsoundsink: Check the return value of GetStatus() too to decide if there was an error
	  If GetStatus() fails, the status itself won't be very meaningful but we also
	  have to look at its return value. This fixes blocking pipelines when removing
	  sound devices or during other errors, where we wouldn't notice the error and
	  then wait forever.
	  https://bugzilla.gnome.org/show_bug.cgi?id=734098

2015-11-15 17:16:29 -0800  Josep Torra <n770galaxy@gmail.com>

	* gst/rtp/gstrtpgstdepay.c:
	  rtpgstdepay: Properly handle backward compat for event deserialization
	  Actual code is checking for a NULL terminator and a ';' terminator,
	  for backward compat, in a chained way that cause all events being rejected.
	  The proper condition is to reject the events when terminator isn't
	  in ['\0', ';'] set.
	  https://bugzilla.gnome.org/show_bug.cgi?id=758151

2015-11-15 17:11:02 -0800  Josep Torra <n770galaxy@gmail.com>

	* tests/check/elements/rtp-payloading.c:
	  tests: rtp-payloading: Test for handling of custom events in rtpgst
	  Add a simple test that checks proper serialization/deserialization
	  of custom events with rtpgstpay and rtpgstdepay.

2015-12-09 12:24:09 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpjitterbuffer: Fix packet dropping after a big discont
	  We would queue 5 consective packets before considering a reset and a proper
	  discont here. Instead of expecting the next output packet to have the current
	  seqnum (i.e. the fifth), expect it to have the first seqnum. Otherwise we're
	  going to drop all queued up packets.

2015-11-18 16:10:11 +0100  Michael Olbrich <m.olbrich@pengutronix.de>

	* gst/avi/gstavimux.c:
	  avimux: don't crash if we never got audio caps before stopping
	  auds.blockalign is set once the first caps arrive. If
	  gst_avi_mux_stop_file() is called before this happens then auds.blockalign
	  is zero and gst_avi_mux_audsink_set_fields() cause a crash:
	  [...]
	  avipad->parent.hdr.rate = avipad->auds.av_bps / avipad->auds.blockalign;
	  [...]
	  https://bugzilla.gnome.org/show_bug.cgi?id=758912

2015-12-01 00:03:21 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* gst/wavparse/gstwavparse.c:
	  wavparse: flush upstream when seeking in pull mode
	  Makes sure upstream will unblock and return the thread so that
	  seeking can continue
	  https://bugzilla.gnome.org/show_bug.cgi?id=758861

2015-11-18 19:07:53 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Cast to signed integers to prevent unsigned compare between negative and positive numbers
	  This fixes seeking if the first entries in the samples table are negative. The
	  binary search would always fail on this as the array would not be sorted if
	  interpreting the negative numbers as huge positive numbers. This caused us to
	  always output buffers from the beginning after a seek instead of close to the
	  seek position.
	  Also add a case to the comparison function for equality.

2015-11-11 14:10:53 +0900  Vineeth TM <vineeth.tm@samsung.com>

	* gst/multifile/gstsplitmuxpartreader.c:
	  splitmuxpartreader: Fix GCond leak
	  inactive_cond is not being cleared resulting in memory leak.
	  https://bugzilla.gnome.org/show_bug.cgi?id=757924

2015-11-05 18:39:33 +0530  Nirbheek Chauhan <nirbheek@centricular.com>

	* ext/pulse/pulsedeviceprovider.c:
	  pulse: Don't leak caps and structures in the device provider

2015-08-12 13:35:40 +0200  Philippe Normand <philn@igalia.com>

	* gst/isomp4/qtdemux.c:
	* gst/isomp4/qtdemux.h:
	  qtdemux: support for cenc auxiliary info parsing outside of moof box
	  When the cenc aux info index is out of moof boundaries, keep track of
	  it and parse the beginning of the mdat box, before the first sample.
	  https://bugzilla.gnome.org/show_bug.cgi?id=755614

2015-11-03 22:01:07 +0530  Arun Raghavan <git@arunraghavan.net>

	* sys/v4l2/gstv4l2.c:
	  v4l2: Set O_CLOEXEC on the device fd
	  This is needed to make sure that child processes don't inherit the video
	  device fd which can cause problems with some drivers.

2015-10-21 11:16:01 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/flv/gstflvmux.c:
	  flvmux: fix writing of creation time
	  Don't write time as e.g. 11:9:42

2015-09-22 22:59:16 +0200  Havard Graff <havard.graff@gmail.com>

	* gst/flv/gstflvmux.c:
	* tests/check/elements/flvmux.c:
	  flvmux: GST_BUFFER_OFFSETs should be GST_BUFFER_OFFSET_NONE
	  Or else flvdemux don't understand it
	  https://bugzilla.gnome.org/show_bug.cgi?id=754435

2015-09-02 10:44:59 +0200  Havard Graff <havard.graff@gmail.com>

	* gst/flv/gstflvmux.c:
	* tests/check/elements/flvmux.c:
	  flvmux: use time segment and copy timestamps when streamable
	  Add a basic test using speex data to verify timestamping.
	  https://bugzilla.gnome.org/show_bug.cgi?id=754435

=== release 1.6.1 ===

2015-10-30 16:40:55 +0200  Sebastian Dröge <sebastian@centricular.com>

	* ChangeLog:
	* NEWS:
	* RELEASE:
	* configure.ac:
	* docs/plugins/gst-plugins-good-plugins.args:
	* 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-apetag.xml:
	* docs/plugins/inspect/plugin-audiofx.xml:
	* docs/plugins/inspect/plugin-audioparsers.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-dtmf.xml:
	* docs/plugins/inspect/plugin-dv.xml:
	* docs/plugins/inspect/plugin-effectv.xml:
	* docs/plugins/inspect/plugin-equalizer.xml:
	* docs/plugins/inspect/plugin-flac.xml:
	* docs/plugins/inspect/plugin-flv.xml:
	* docs/plugins/inspect/plugin-flxdec.xml:
	* docs/plugins/inspect/plugin-gdkpixbuf.xml:
	* docs/plugins/inspect/plugin-goom.xml:
	* docs/plugins/inspect/plugin-goom2k1.xml:
	* docs/plugins/inspect/plugin-icydemux.xml:
	* docs/plugins/inspect/plugin-id3demux.xml:
	* docs/plugins/inspect/plugin-imagefreeze.xml:
	* docs/plugins/inspect/plugin-interleave.xml:
	* docs/plugins/inspect/plugin-isomp4.xml:
	* docs/plugins/inspect/plugin-jack.xml:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-level.xml:
	* docs/plugins/inspect/plugin-matroska.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-oss4.xml:
	* docs/plugins/inspect/plugin-ossaudio.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* docs/plugins/inspect/plugin-pulseaudio.xml:
	* docs/plugins/inspect/plugin-replaygain.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-rtpmanager.xml:
	* docs/plugins/inspect/plugin-rtsp.xml:
	* docs/plugins/inspect/plugin-shapewipe.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-videobox.xml:
	* docs/plugins/inspect/plugin-videocrop.xml:
	* docs/plugins/inspect/plugin-videofilter.xml:
	* docs/plugins/inspect/plugin-videomixer.xml:
	* docs/plugins/inspect/plugin-vpx.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 1.6.1

2015-10-30 16:27:32 +0200  Sebastian Dröge <sebastian@centricular.com>

	* po/af.po:
	* po/az.po:
	* po/bg.po:
	* po/ca.po:
	* po/da.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/or.po:
	* po/pt_BR.po:
	* po/ro.po:
	* po/sk.po:
	* po/sl.po:
	* po/sq.po:
	* po/sr.po:
	* po/sv.po:
	* po/tr.po:
	* po/uk.po:
	* po/zh_HK.po:
	* po/zh_TW.po:
	  Update .po files

2015-10-30 14:28:04 +0200  Sebastian Dröge <sebastian@centricular.com>

	* po/cs.po:
	* po/de.po:
	* po/nl.po:
	* po/pl.po:
	* po/ru.po:
	* po/vi.po:
	* po/zh_CN.po:
	  po: Update translations

2015-10-28 23:47:30 +0530  Nirbheek Chauhan <nirbheek@centricular.com>

	* sys/v4l2/gstv4l2deviceprovider.c:
	  v4l2: fix double-unref in the v4l2 device provider

2015-10-27 10:48:00 +0100  Nicola Murino <nicola.murino@gmail.com>

	* gst/matroska/matroska-ids.c:
	  matroskamux: don't drop JPEG frames that only have PTS but no DTS set
	  For the MS/VfW codec ids, we want to write DTS timestamps instead
	  of PTS because that's what everyone else seems to do (and it's also
	  how it is in AVI). So for those input formats we use the buffer DTS
	  instead of the PTS. However, if there's no DTS set but only the PTS
	  then just take the PTS instead of dropping the input buffer. This
	  is useful especially for I-frame only codecs like JPEG and huffyuv,
	  but should also be fine as fallback in general.
	  Fixes regression with input JPEG frames that only have PTS set on them.
	  https://bugzilla.gnome.org/show_bug.cgi?id=756967

2015-10-11 12:06:26 +0100  Thibault Saunier <tsaunier@gnome.org>

	* gst/isomp4/gstqtmux.c:
	* gst/isomp4/gstqtmuxmap.c:
	  qtmux: Allow negotiating to S8 as a raw format but stop making it best choice
	  Negotiation to audio/x-raw,format=S8 was not possible because S8 does
	  not have a bit order so we ended up doing `if (!entry.fourcc) goto refuse_caps;`
	  https://bugzilla.gnome.org/show_bug.cgi?id=756387

2015-10-11 09:18:40 +0100  Thibault Saunier <tsaunier@gnome.org>

	* gst/isomp4/gstqtmux.c:
	* gst/isomp4/gstqtmuxmap.c:
	  qtmux: Add prores support
	  https://bugzilla.gnome.org/show_bug.cgi?id=756388

2015-10-24 23:57:38 +0200  George Kiagiadakis <george.kiagiadakis@collabora.com>

	* tests/check/elements/splitmux.c:
	  tests/check/splitmux: test that the release_pad vfunc of splitmuxsink actually releases pads
	  https://bugzilla.gnome.org/show_bug.cgi?id=753622

2015-10-24 23:57:29 +0200  George Kiagiadakis <george.kiagiadakis@collabora.com>

	* gst/multifile/gstsplitmuxsink.c:
	  splitmuxsink: do not destroy the multiqueue & muxer when going to NULL
	  Instead, delay it until all request pads have been released. This is
	  because the release_pad() vfunc requires the multiqueue and muxer to
	  be there in order to release their request pads as well. If those
	  elements are destroyed earlier, release_pad() does not work, no
	  pads are released and some resources are leaked.
	  https://bugzilla.gnome.org/show_bug.cgi?id=753622

2015-10-27 17:32:22 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/audiofx/gstscaletempo.c:
	  scaletempo: F64 support is only available in GIT master at this point

2015-10-24 17:14:07 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/audiofx/gstscaletempo.c:
	* gst/audiofx/gstscaletempo.h:
	  scaletempo: Fix handling of rate < 0
	  We have to reverse all samples in a buffer before processing them to properly
	  have continuous data from one buffer to another. As a result we will have a
	  negative applied rate and a rate of 1.0.
	  Also make sure that input buffers are correctly clipped to the segment,
	  otherwise our calculations are going to go wrong.
	  Also copy over the segment event's sequence number to the output segment while
	  we're at it.
	  https://bugzilla.gnome.org/show_bug.cgi?id=757033

2015-10-26 00:41:28 +1100  Jan Schmidt <jan@centricular.com>

	* tests/files/Makefile.am:
	  check: Dist splitvideo0[012].ogg test files.

2015-10-13 12:42:56 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* gst/rtp/gstrtpj2kpay.c:
	  rtpj2kpay: update fragment offset
	  It was always being set to 0, making the resulting stream broken
	  for the receiver
	  https://bugzilla.gnome.org/show_bug.cgi?id=756422

2015-10-19 15:36:37 +0300  Ryan Hendrickson <ryan.hendrickson@alum.mit.edu>

	* gst/isomp4/gstqtmux.c:
	  qtmux: Don't unconditionally use strnlen()
	  It's not available on older OSX and we can as well use memchr() here.
	  https://bugzilla.gnome.org/show_bug.cgi?id=756154

2015-10-12 10:48:23 +0900  Vineeth TM <vineeth.tm@samsung.com>

	* gst/audioparsers/gstmpegaudioparse.c:
	  mpegaudioparse: Fix buffer memory leak during failures
	  mapped buffer is not being unmapped during failures
	  https://bugzilla.gnome.org/show_bug.cgi?id=756231

2015-10-19 17:38:32 +0900  Vineeth TM <vineeth.tm@samsung.com>

	* gst/auparse/gstauparse.c:
	  auparse: Fix event memory leak
	  Free the event after being handled to prevent memory leak.
	  https://bugzilla.gnome.org/show_bug.cgi?id=756799

2015-10-14 12:03:15 +0200  Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>

	* gst/isomp4/qtdemux.c:
	  qtdemux: fix caps leak
	  If the QtDemuxStream are re-used they may already have caps which used
	  to be leaked.
	  Reproduced using the
	  validate.dash.playback.seek_forward.dash_exMPD_BIP_TC1 validate
	  scenario.
	  https://bugzilla.gnome.org/show_bug.cgi?id=756561

2015-10-14 09:29:50 +0900  Vineeth TM <vineeth.tm@samsung.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Fix taglist memory leak
	  Free the stream and its sub items instead of just the stream
	  https://bugzilla.gnome.org/show_bug.cgi?id=756544

2015-10-07 17:14:57 +0900  Vineeth TM <vineeth.tm@samsung.com>

	* gst/isomp4/gstqtmux.c:
	  qtmux: Fix date memory leak
	  When getting date from taglist, the memory should be freed after
	  using it.
	  https://bugzilla.gnome.org/show_bug.cgi?id=756171

2015-10-05 11:03:38 +0900  Vineeth TM <vineeth.tm@samsung.com>

	* gst/isomp4/gstqtmux.c:
	  qtmux: Fix sample memory leak
	  When getting sample from taglist, the memory should be freed after
	  using it.
	  https://bugzilla.gnome.org/show_bug.cgi?id=756068

2015-10-05 13:10:56 +0900  Vineeth TM <vineeth.tm@samsung.com>

	* gst/cutter/gstcutter.c:
	  cutter: Fix buffer leak
	  Buffer is added to the internal cache, and pushed only when accumulated
	  buffer duration crosses 200 ms. So when the chain ends, the buffer accumulated
	  is not freed. Freeing the cache when the state changes from PAUSED to READY.
	  https://bugzilla.gnome.org/show_bug.cgi?id=754212

2015-04-17 14:25:43 +0200  George Kiagiadakis <george.kiagiadakis@collabora.com>

	* gst/multifile/gstsplitmuxsink.c:
	* gst/multifile/gstsplitmuxsink.h:
	  splitmuxsink: post messages when fragments are being opened and closed
	  This can be useful for applications that need to track the created fragments
	  (to log them in a recording database, for example)
	  https://bugzilla.gnome.org/show_bug.cgi?id=750108

2015-04-29 18:23:28 +0100  Ramiro Polla <ramiro.polla@collabora.co.uk>

	* gst/multifile/gstsplitmuxsink.c:
	* gst/multifile/gstsplitmuxsink.h:
	  splitmuxsink: allow non-video streams to serve as reference
	  In the absence of a video stream, the first stream will be used as
	  reference.
	  https://bugzilla.gnome.org/show_bug.cgi?id=753617

2015-07-22 17:45:12 +0200  George Kiagiadakis <george.kiagiadakis@collabora.com>

	* gst/multifile/gstsplitmuxsink.c:
	  splitmuxsink: initialize mux_start_time properly
	  mux_start_time refers to the running_time of the buffer
	  that goes first in the output file. Normally this time is
	  0, so this variable is initialized to 0 during the state
	  change to PAUSED.
	  However, when dealing with dynamic pipelines and starting
	  a recording while the pipeline has already run for a while,
	  the running_time of the first buffer is > 0 and this causes
	  a problem with detecting the end of the first file(s) when
	  splitting by duration, because the code will later compare
	  the threshold_time with (last buffer running_time - mux_start_time)
	  and will get it wrong until mux_start_time advances enough
	  to make this difference < threshold_time, creating empty files
	  in the meantime.
	  https://bugzilla.gnome.org/show_bug.cgi?id=753624

2015-09-29 11:12:48 +0100  Tim-Philipp Müller <tim@centricular.com>

	* ext/gdk_pixbuf/gstgdkpixbufoverlay.c:
	  gdkpixbufsink: don't leak old pixel buffer when setting a new overlay
	  https://bugzilla.gnome.org/show_bug.cgi?id=755773

2015-09-28 20:25:22 +0100  Tim-Philipp Müller <tim@centricular.com>

	* ext/flac/gstflacenc.c:
	  flacenc: avoid potential string overflow
	  We don't necessarily have full control over the input tags, so
	  it's possible that the ISRC tag contains a longer string than
	  expected, in which case we'd write over the end of the static-size
	  13 byte buffer that is FLAC__StreamMetadata_CueSheet_Track::isrc.
	  Make sure to only copy the ISRC if it's not too long, and make
	  sure the buffer we write to is always NUL-terminated by using
	  g_strlcpy().
	  CID 1324931.

2015-09-28 18:03:51 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/matroska/matroska-demux.c:
	  matroskademux: Remove leftover assertion from 0.10
	  We now allocate memory via GstAllocator and as such can handle arbitrary
	  alignments, not only <= G_MEM_ALIGN.
	  https://bugzilla.gnome.org/show_bug.cgi?id=755708

2015-09-25 10:01:37 +0200  Guillaume Marquebielle <guillaume.marquebielle@parrot.com>

	* gst/audioparsers/gstaacparse.c:
	  aacparse: fix uninitialized variables in LOAS config reading
	  On reading LOAS config, flag v=1 and vA=1 combination can occur, leading to warning
	  "Spec says "TBD"...". Returning TRUE on this case while parameters 'sample_rate' and
	  'channels' are pointing to uninitialized values can end on setting random values as
	  rate and channels on src caps.
	  https://bugzilla.gnome.org/show_bug.cgi?id=755611

=== release 1.6.0 ===

2015-09-25 23:15:55 +0200  Sebastian Dröge <sebastian@centricular.com>

	* ChangeLog:
	* NEWS:
	* RELEASE:
	* configure.ac:
	* docs/plugins/gst-plugins-good-plugins.args:
	* 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-apetag.xml:
	* docs/plugins/inspect/plugin-audiofx.xml:
	* docs/plugins/inspect/plugin-audioparsers.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-dtmf.xml:
	* docs/plugins/inspect/plugin-dv.xml:
	* docs/plugins/inspect/plugin-effectv.xml:
	* docs/plugins/inspect/plugin-equalizer.xml:
	* docs/plugins/inspect/plugin-flac.xml:
	* docs/plugins/inspect/plugin-flv.xml:
	* docs/plugins/inspect/plugin-flxdec.xml:
	* docs/plugins/inspect/plugin-gdkpixbuf.xml:
	* docs/plugins/inspect/plugin-goom.xml:
	* docs/plugins/inspect/plugin-goom2k1.xml:
	* docs/plugins/inspect/plugin-icydemux.xml:
	* docs/plugins/inspect/plugin-id3demux.xml:
	* docs/plugins/inspect/plugin-imagefreeze.xml:
	* docs/plugins/inspect/plugin-interleave.xml:
	* docs/plugins/inspect/plugin-isomp4.xml:
	* docs/plugins/inspect/plugin-jack.xml:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-level.xml:
	* docs/plugins/inspect/plugin-matroska.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-oss4.xml:
	* docs/plugins/inspect/plugin-ossaudio.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* docs/plugins/inspect/plugin-pulseaudio.xml:
	* docs/plugins/inspect/plugin-replaygain.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-rtpmanager.xml:
	* docs/plugins/inspect/plugin-rtsp.xml:
	* docs/plugins/inspect/plugin-shapewipe.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-videobox.xml:
	* docs/plugins/inspect/plugin-videocrop.xml:
	* docs/plugins/inspect/plugin-videofilter.xml:
	* docs/plugins/inspect/plugin-videomixer.xml:
	* docs/plugins/inspect/plugin-vpx.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 1.6.0

2015-09-25 22:57:34 +0200  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

2015-09-25 14:08:09 +0200  Thibault Saunier <tsaunier@gnome.org>

	* gst/smpte/gstsmptealpha.c:
	  smptealpha: Do not set width/height before comparing with old values
	  Otherwise we end up considering the values did not change and we wrongly
	  work with the old video format (which will lead to wrong
	  behaviour/segfaults).
	  https://bugzilla.gnome.org/show_bug.cgi?id=755621

2015-09-23 20:59:00 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/isomp4/qtdemux.c:
	  qtdemux: Accumulate segments for edit lists before activating the next segment
	  eceb2ccc739092d964d78945e19c2ecedbd214e2 broke segment seeks by always
	  accumulating segments manually when activating a segment. This is only
	  needed when handling edit lists, not when activating a segment because of a
	  seek. Do the accumulation when switching edit list segments instead.
	  This fixes segment seeks again, while keeping edit lists playback working.
	  https://bugzilla.gnome.org/show_bug.cgi?id=755471

2015-09-23 17:43:51 +0530  Vikram Fugro <vikram.fugro@gmail.com>

	* gst/spectrum/gstspectrum.c:
	  spectrum: send phase values in the GstMessage for Phase info
	  https://bugzilla.gnome.org/show_bug.cgi?id=755463

2015-09-22 00:46:01 +1000  Jan Schmidt <jan@centricular.com>

	* gst/matroska/matroska-mux.c:
	  matroska-mux: Don't output a warning on MONO multiview mode.

2015-09-19 17:02:18 +0200  Sebastian Rasmussen <sebras@hotmail.com>

	* gst/rtp/gstrtptheoradepay.c:
	  rtptheoradepay: Fix memory leaks
	  The same memory leaks were fixed in identical fashion for
	  vorbisdepay in 06efeff5d979576a252e5dae57f46d6445b1df12 in 2009.
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=755277

2015-09-19 17:04:07 +0200  Sebastian Rasmussen <sebras@hotmail.com>

	* gst/rtp/gstrtptheorapay.c:
	* gst/rtp/gstrtpvorbisdepay.c:
	* gst/rtp/gstrtpvorbispay.c:
	  rtp{vorbis,theora}{pay,depay}: Cosmetic cleanup
	  * use g_list_free_full(), don't iterate elements maually when freeing
	  * call gst_rtp_*_pay_clear_packet(), don't duplicate its code
	  * use gst_buffer_unref() to clarify that it is buffers being released,
	  instead of refering directly to gst_mini_object_unref()
	  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=755277

2015-09-19 18:44:22 +0200  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtp/gstrtptheorapay.c:
	* gst/rtp/gstrtpvorbispay.c:
	  rtp{vorbis,theora}pay: Store headers in the packet buffers lists, not a NULL buffer
	  https://bugzilla.gnome.org/show_bug.cgi?id=755265

=== release 1.5.91 ===

2015-09-18 19:33:13 +0200  Sebastian Dröge <sebastian@centricular.com>

	* ChangeLog:
	* NEWS:
	* RELEASE:
	* configure.ac:
	* docs/plugins/gst-plugins-good-plugins.args:
	* docs/plugins/gst-plugins-good-plugins.signals:
	* 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-apetag.xml:
	* docs/plugins/inspect/plugin-audiofx.xml:
	* docs/plugins/inspect/plugin-audioparsers.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-dtmf.xml:
	* docs/plugins/inspect/plugin-dv.xml:
	* docs/plugins/inspect/plugin-effectv.xml:
	* docs/plugins/inspect/plugin-equalizer.xml:
	* docs/plugins/inspect/plugin-flac.xml:
	* docs/plugins/inspect/plugin-flv.xml:
	* docs/plugins/inspect/plugin-flxdec.xml:
	* docs/plugins/inspect/plugin-gdkpixbuf.xml:
	* docs/plugins/inspect/plugin-goom.xml:
	* docs/plugins/inspect/plugin-goom2k1.xml:
	* docs/plugins/inspect/plugin-icydemux.xml:
	* docs/plugins/inspect/plugin-id3demux.xml:
	* docs/plugins/inspect/plugin-imagefreeze.xml:
	* docs/plugins/inspect/plugin-interleave.xml:
	* docs/plugins/inspect/plugin-isomp4.xml:
	* docs/plugins/inspect/plugin-jack.xml:
	* docs/plugins/inspect/plugin-jpeg.xml:
	* docs/plugins/inspect/plugin-level.xml:
	* docs/plugins/inspect/plugin-matroska.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-oss4.xml:
	* docs/plugins/inspect/plugin-ossaudio.xml:
	* docs/plugins/inspect/plugin-png.xml:
	* docs/plugins/inspect/plugin-pulseaudio.xml:
	* docs/plugins/inspect/plugin-replaygain.xml:
	* docs/plugins/inspect/plugin-rtp.xml:
	* docs/plugins/inspect/plugin-rtpmanager.xml:
	* docs/plugins/inspect/plugin-rtsp.xml:
	* docs/plugins/inspect/plugin-shapewipe.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-videobox.xml:
	* docs/plugins/inspect/plugin-videocrop.xml:
	* docs/plugins/inspect/plugin-videofilter.xml:
	* docs/plugins/inspect/plugin-videomixer.xml:
	* docs/plugins/inspect/plugin-vpx.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 1.5.91

2015-09-18 19:23:57 +0200  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

2015-09-18 11:50:31 +0200  Sebastian Dröge <sebastian@centricular.com>

	* po/zh_CN.po:
	  po: Update translations

2015-09-17 10:50:01 +0900  Eunhae Choi <eunhae1.choi@samsung.com>

	* gst/avi/gstavidemux.c:
	  avidemux: Fix taglist leak
	  gst_tag_list_insert() does not take ownership of the inserted taglist.
	  https://bugzilla.gnome.org/show_bug.cgi?id=755138

2015-09-16 07:05:36 +1000  Jan Schmidt <jan@centricular.com>

	* gst/audioparsers/gstaacparse.c:
	  aacparse: Skip LOAS AAC until a valid config is seen.
	  It's normal when dropping into the middle of a stream to
	  not always have the config available immediately, so skip LOAS
	  until a valid config is seen without either setting invalid
	  caps or erroring out.
	  https://bugzilla.gnome.org/show_bug.cgi?id=751386

2015-09-13 15:41:38 +0200  Mark Nauwelaerts <mnauw@users.sourceforge.net>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpjitterbuffer: reset just a bit more upon flush_stop

2015-09-13 15:40:09 +0200  Mark Nauwelaerts <mnauw@users.sourceforge.net>

	* gst/rtpmanager/gstrtpjitterbuffer.c:
	  rtpjitterbuffer: remove dead struct member

2015-09-11 17:09:28 +0900  Vineeth TM <vineeth.tm@samsung.com>

	* gst/udp/gstmultiudpsink.c:
	  multiudpsink: fix GError memory leak when hostname resolution fails
	  https://bugzilla.gnome.org/show_bug.cgi?id=754869

2015-09-10 15:26:54 -0300  Thiago Santos <thiagoss@osg.samsung.com>

	* gst/matroska/ebml-write.c:
	  matroskamux: drop HEADER flag from output buffers
	  Drop HEADER flag from output buffers if they are not indeed
	  headers.
	  Fixes resending of headers in tcp connection handling
	  https://bugzilla.gnome.org/show_bug.cgi?id=754768

2015-09-10 16:00:50 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/matroska/ebml-write.c:
	  matroskamux: fix matroskamux ! matroskademux
	  Don't carry over DISCONT flags from the input buffers to the
	  output buffer, or the demuxer might reset its state when it
	  receives the first data buffer just after parsing the simple
	  block header, and then expect sane data to follow.
	  Fixes matroskamux ! demux erroring out.
	  https://bugzilla.gnome.org/show_bug.cgi?id=754768
	  https://bugzilla.gnome.org/show_bug.cgi?id=657805

2015-09-09 12:51:40 -0700  Martin Kelly <martin@surround.io>

	* gst/rtsp/README:
	  rtsp: fix small README typo
	  https://bugzilla.gnome.org/show_bug.cgi?id=754807

2015-09-04 19:45:37 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/audioparsers/gstwavpackparse.c:
	  wavpackparse: set both pts and dts so baseparse doesn't make up wrong dts after seeks
	  https://bugzilla.gnome.org/show_bug.cgi?id=752106

2015-09-04 19:34:41 +0100  Tim-Philipp Müller <tim@centricular.com>

	* gst/audioparsers/gstflacparse.c:
	  flacparse: set both pts and dts so baseparse doesn't make up wrong dts after a seek
	  flac contains the sample offset in the frame header, so after a seek
	  without index flacparse will know the exact position we landed on and
	  timestamp buffers accordingly. It only set the pts though, which means
	  the baseparse-set dts which was set to the seek position prevails, and
	  since the seek was based on an estimate, there's likely a discrepancy
	  between where we wanted to land and where we did land, so from here on
	  that dts/pts difference will be maintained, with dts possibly multiple
	  seconds ahead of pts, which is just wrong. The easiest way to fix this
	  is to just set both pts and dts based on the sample offset, but perhaps
	  parsed audio should just not have dts set at all.
	  https://bugzilla.gnome.org/show_bug.cgi?id=752106

2015-09-06 16:33:02 +0100  Tim-Philipp Müller <tim@centricular.com>

	* docs/plugins/gst-plugins-good-plugins.args:
	* docs/plugins/gst-plugins-good-plugins.signals:
	  docs: remove properties and signals that no longer exist
	  https://bugzilla.gnome.org/show_bug.cgi?id=726443

2013-10-11 15:13:00 +0000  George Chriss <gschriss@gmail.com>

	* gst/flv/gstflvmux.c:
	  flvmux: Make the element count in arrays not include end
	  One-line removal of tags_written++
	  This should fix rtmp output to crtmpserver, and hopefully
	  noone is expecting that the element count includes the end
	  element, as different bits of documentation say different
	  things about whether it should or not.
	  https://bugzilla.gnome.org/show_bug.cgi?id=661624

2015-07-30 00:59:15 +1000  Jan Schmidt <jan@centricular.com>

	* gst/flv/gstflvmux.c:
	* gst/flv/gstflvmux.h:
	  flvmux: Store incoming bitrate tags and send in the metadata
	  Apparently the Microsoft Azure RTMP server requires that the
	  videodatarate and audiodatarate metadata be provided, so
	  set those, even if it's to 0. Use the actual input bitrate
	  tags if available.

2015-09-04 00:06:29 +1000  Jan Schmidt <jan@centricular.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: Don't parse key data more than needed.
	  When an auxilliary streams are present in the SDP media,
	  there's no need to re-parse the SDP attributes multiple
	  times.

2015-09-03 20:56:55 +1000  Jan Schmidt <jan@centricular.com>

	* gst/rtsp/gstrtspsrc.c:
	  rtspsrc: Fix SRTP + RTX, auth access, a leak, and an invalid memory access.
	  In parse_keymgmt(), don't mutate the input string that's been passed
	  as const, especially since we might need the original value again if
	  the same key info applies to multiple streams (RTX, for example).
	  When a resource is 404, and we have auth info - retry with the auth
	  info the same as if we had receive unauthorised, in case the resource
	  isn't even visible until credentials are supplied.
	  Fix a memory leak handling Mikey data.
	  When generating a random keystring, don't overrun the 30 byte
	  buffer by generating 32 bytes into it.

2015-09-04 15:18:05 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/udp/gstudpsrc.c:
	  udpsrc: Fix build with GLib < 2.44
	  G_IO_ERROR_CONNECTION_CLOSED was added in 2.44.

2015-09-04 12:01:52 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/udp/gstudpsrc.c:
	  udpsrc: Ignore G_IO_ERROR_CONNECTION_CLOSED when receiving data
	  This happens on Windows if we use the same socket for sending packets,
	  and the remote sends ICMP port/host unreachable messages.
	  https://bugzilla.gnome.org/show_bug.cgi?id=754534

2015-09-02 21:12:41 +0300  Sebastian Dröge <sebastian@centricular.com>

	* gst/rtp/gstrtptheoradepay.c:
	* gst/rtp/gstrtpvorbisdepay.c:
	  rtpvorbis/theoradepay: Fix handling of fragmented packets
	  This was broken in b1089fb520 by not considering the full packet length of a
	  fragmented packet but only the length of the first one.
	  https://bugzilla.gnome.org/show_bug.cgi?id=754417

2015-09-01 15:39:22 -0400  Olivier Crête <olivier.crete@collabora.com>

	* gst/dtmf/gstdtmfsrc.c:
	* gst/dtmf/gstrtpdtmfsrc.c:
	  dtmfsrc: Reply to latency query

2015-08-31 16:42:30 -0400  Olivier Crête <olivier.crete@collabora.com>

	* tests/check/elements/rtpsession.c:
	  tests: Fix rtpsession test failure
	  The time of the first RTCP packet is semi-random, so
	  sometimes it was produced before enough packets from
	  the second SSRC were received. First drop queued RTCP
	  packets, then advance the clock enough to ensure
	  that at least one new RTCP packet is produced.
	  https://bugzilla.gnome.org/show_bug.cgi?id=750731

2015-08-31 13:56:04 +0200  Stefan Sauer <ensonic@users.sf.net>

	* tests/check/elements/level.c:
	  level: improve the test for multi-channel mode
	  Change the test to verify the read-index for multiple messages per buffer.
	  See https://bugzilla.gnome.org/show_bug.cgi?id=754144

2015-08-31 12:46:52 +0200  Jan Alexander Steffens (heftig) <jan.steffens@gmail.com>

	* gst/matroska/matroska-demux.c:
	  matroskademux: Align raw video frames to 32 bytes
	  Outputting unaligned video frames causes videoscale et al to
	  crash when attempting SIMD-accelerated conversion.
	  https://bugzilla.gnome.org/show_bug.cgi?id=736965

2015-08-26 23:16:46 +0200  Stefan Sauer <ensonic@users.sf.net>

	* gst/level/gstlevel.c:
	  level: fix level calculations for mutliple channels
