gstreamer 1.17.90

[Tim-Philipp Müller]

News


GStreamer 1.18 Release Notes

These release notes are a placeholder, please bear with us while we
finish writing up the real thing.

ChangeLog
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(24.7K)

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  sha256sum: 08ec65757c032621f04c3f8e95db8e42e7646fa4512df99368e8ab34c5970f7b

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gst-plugins-good 1.17.90

[Tim-Philipp Müller]

News


GStreamer 1.18 Release Notes

These release notes are a placeholder, please bear with us while we
finish writing up the real thing.

ChangeLog
=
https://download.gnome.org/sources/gst-plugins-good/1.17/gst-plugins-good-1.17.90.changes
  (94)

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  sha256sum: 077194aebf9cc5496f3e872d203955748a5d28cbcf42c0f6291c63dea190bd6f

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gst-plugins-base 1.17.90

[Tim-Philipp Müller]

News


GStreamer 1.18 Release Notes

These release notes are a placeholder, please bear with us while we
finish writing up the real thing.

ChangeLog
=
https://download.gnome.org/sources/gst-plugins-base/1.17/gst-plugins-base-1.17.90.changes
  (36.5K)

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 (2.98M)
  sha256sum: f6f138d5ce6ccaa225f87b108df9f940968bd8d9d6dd1d94ef473695ad87bb65

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gst-plugins-base 1.17.2

[Tim-Philipp Müller]

ChangeLog
=

2020-07-03 00:24:57 +0100  Tim-Philipp Müller 


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gst-plugins-good 1.17.2

[Tim-Philipp Müller]

ChangeLog
=

2020-07-03 00:27:47 +0100  Tim-Philipp Müller 


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gstreamer 1.17.2

[Tim-Philipp Müller]

ChangeLog
=

2020-07-03 00:22:34 +0100  Tim-Philipp Müller 

* ChangeLog:
* NEWS:
* RELEASE:
* gstreamer.doap:
* meson.build:
  Release 1.17.2

2020-07-01 20:20:16 +0300  Sebastian Dröge 

* gst/gststreams.c:
  stream: Don't use GST_FIXME_OBJECT() when generating a random stream 
id if the caller didn't provide one
  That would call into gst_info_describe_stream(), which takes the same
  mutex a second time and then deadlocks.
  Part-of: 
<https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/552>

2020-05-11 17:44:43 +0200  Mathieu Duponchelle 

* gst/gstutils.c:
  gstutils: fix link in parse_bin_from_description_full doc
  Part-of: 
<https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/475>

2020-06-26 00:50:44 +1000  Jan Schmidt 

* gst/gstbin.h:
  gstbin: Fix docs typo
  element-added-deep -> deep-element-added in the GstBin doc header
  Part-of: 
<https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/547>

2020-06-25 02:14:56 +0900  Seungha Yang 

* docs/gst-plugins-doc-cache-generator.py:
  docs: Specify UTF-8 encoding everywhere
  Otherwise some non-English character might be broken depending on
  OS and/or locale.
  Part-of: 
<https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/545>

2020-06-25 01:37:43 +0200  Mathieu Duponchelle 

* hooks/pre-commit.hook:
  hooks: add opt-in cache update hook
  CI now checks that plugin caches are up to date, for example
  when adding a new property to an element.
  This is something pretty easy to forget, and it can be checked
  in a pre-commit hook provided the cache generator generates no
  unnecessart diff on the developer's machine. This is now the
  case for me in core, -base and -good, and eventually all the
  repositories should behave appropriately, at least on my
  development machine.
  The new check in the pre-commit hook is only performed when
  the user is in the development environment (it checks
  $MESON_BUILD_ROOT to determine that), and when the developer
  has opted-in by setting `GST_CACHE_HOOK=enabled`. That is
  because the hook will actually rebuild the cache with ninja,
  and modify it in the source directory, that's not a behaviour
  we want to enable by default.
  Part-of: 
<https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/546>

2020-06-24 22:51:48 +0900  Seungha Yang 

* tools/gst-inspect.c:
  gst-inspect: Use gst_info_strdup_vprintf to print string
  g_vprintf() will write a string binary to stdout directly using 
fwrite().
  So, depending on character in the string, fwrite to stdout can
  print broken one but printf family might not cause the issue.
  Part-of: 
<https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/544>

2020-06-24 07:54:42 +0200  Michael Olbrich 

* gst/gstbus.c:
* tests/check/gst/gstbus.c:
  bus: clear bus->priv->signal_watch immediately when the source is 
removed
  There is a race-condition that can trigger the assertion in
  gst_bus_add_signal_watch_full():
  If gst_bus_add_signal_watch_full() is called immediately after
  gst_bus_remove_signal_watch() then bus->priv->signal_watch may still 
be set
  because gst_bus_source_dispose() or gst_bus_source_finalize() was not 
yet
  called.
  This happens if the corresponding GMainContext has the source queued 
for
  dispatch. In this case, the following dispatch will only unref and 
delete
  the signal_watch because it was already destroyed. Any pending 
messages
  will remain until a new watch is installed.
  So bus->priv->signal_watch can be cleared immediately when the watch 
is
  removed. This avoid the race condition.
  Part-of: 
<https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/543>

2020-06-23 01:54:15 +0200  Mathieu Duponchelle 

* docs/gst-hotdoc-plugins-scanner.c:
  gst-hotdoc-plugins-scanner: serialize interfaces

2020-06-22 23:41:06 +0200  Mathieu Duponchelle 

* docs/gst-hotdoc-plugins-scanner.c:
  gst-hotdoc-plugins-scanner: don't instantiate base classes

2020-06-21 02:00:54 +0200  Mathieu Duponchelle 

* docs/gst-hotdoc-plugins-scanner.c:
  gst-hotdoc-plugins-scanner: serialize parents in hierarchy when needed

2020-06-22 09:11:07 -0400  Thibault Saunier 

* docs/gst-hotdoc-plugins-scanner.c:
* docs/meson.build:
  docs: Document signals and properties only for current type

202

gst-plugins-good 1.17.1

[Tim-Philipp Müller]

News


GSTREAMER 1.18 RELEASE NOTES


THESE RELEASE NOTES ARE A PLACEHOLDER, PLEASE BEAR WITH US WHILE WE
FINISH WRITING UP THE REAL THING.

GStreamer 1.18.0 has not yet been released. It is scheduled for release
in summer 2020 now.

1.17.x is the unstable development series that is currently being
developed in the git master branch and which will eventually result in
1.18, and 1.17.1 is the current development release in that series.

The schedule for the 1.18 development cycle is yet to be confirmed, but
it is expected that feature freeze will be in June/July 2020, followed
by several 1.17 pre-releases and then a new 1.18 stable release in
July/August 2020.

1.18 will be backwards-compatible to the stable 1.16, 1.14, 1.12, 1.10,
1.8, 1.6, 1.4, 1.2 and 1.0 release series.

See https://gstreamer.freedesktop.org/releases/1.18/ for the latest

ChangeLog
=
https://download.gnome.org/sources/gst-plugins-good/1.17/gst-plugins-good-1.17.1.changes
  (93)

Download

https://download.gnome.org/sources/gst-plugins-good/1.17/gst-plugins-good-1.17.1.tar.xz
 (3.04M)
  sha256sum: ea8957460ed38261e5c4bf6a8bf98d946298b756a695b27c9b1209aa7b76b0d7

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gstreamer 1.17.1

[Tim-Philipp Müller]

News


GSTREAMER 1.18 RELEASE NOTES


THESE RELEASE NOTES ARE A PLACEHOLDER, PLEASE BEAR WITH US WHILE WE
FINISH WRITING UP THE REAL THING.

GStreamer 1.18.0 has not yet been released. It is scheduled for release
in summer 2020 now.

1.17.x is the unstable development series that is currently being
developed in the git master branch and which will eventually result in
1.18, and 1.17.1 is the current development release in that series.

The schedule for the 1.18 development cycle is yet to be confirmed, but
it is expected that feature freeze will be in June/July 2020, followed
by several 1.17 pre-releases and then a new 1.18 stable release in
July/August 2020.

1.18 will be backwards-compatible to the stable 1.16, 1.14, 1.12, 1.10,
1.8, 1.6, 1.4, 1.2 and 1.0 release series.

See https://gstreamer.freedesktop.org/releases/1.18/ for the latest

ChangeLog
=
https://download.gnome.org/sources/gstreamer/1.17/gstreamer-1.17.1.changes  (93)

Download

https://download.gnome.org/sources/gstreamer/1.17/gstreamer-1.17.1.tar.xz 
(2.53M)
  sha256sum: f63566152755aab98414feb859ea8b00d3b07cad2486f304a8feec1265e5aa0e

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gst-plugins-base 1.17.1

[Tim-Philipp Müller]

News


GSTREAMER 1.18 RELEASE NOTES


THESE RELEASE NOTES ARE A PLACEHOLDER, PLEASE BEAR WITH US WHILE WE
FINISH WRITING UP THE REAL THING.

GStreamer 1.18.0 has not yet been released. It is scheduled for release
in summer 2020 now.

1.17.x is the unstable development series that is currently being
developed in the git master branch and which will eventually result in
1.18, and 1.17.1 is the current development release in that series.

The schedule for the 1.18 development cycle is yet to be confirmed, but
it is expected that feature freeze will be in June/July 2020, followed
by several 1.17 pre-releases and then a new 1.18 stable release in
July/August 2020.

1.18 will be backwards-compatible to the stable 1.16, 1.14, 1.12, 1.10,
1.8, 1.6, 1.4, 1.2 and 1.0 release series.

See https://gstreamer.freedesktop.org/releases/1.18/ for the latest

ChangeLog
=
https://download.gnome.org/sources/gst-plugins-base/1.17/gst-plugins-base-1.17.1.changes
  (93)

Download

https://download.gnome.org/sources/gst-plugins-base/1.17/gst-plugins-base-1.17.1.tar.xz
 (2.95M)
  sha256sum: c0d8a84e3262e706ebc88831ca7349e5ae221274ef362599734573b3d3c028f3

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gst-plugins-base 1.16.2

[Tim-Philipp Müller]

ChangeLog
=

2019-12-03 11:03:11 +  Tim-Philipp Müller 

* ChangeLog:
* NEWS:
* RELEASE:
* configure.ac:
* gst-plugins-base.doap:
* meson.build:
  Release 1.16.2

2019-12-03 11:03:11 +  Tim-Philipp Müller 

* docs/plugins/inspect/plugin-adder.xml:
* docs/plugins/inspect/plugin-alsa.xml:
* docs/plugins/inspect/plugin-app.xml:
* docs/plugins/inspect/plugin-audioconvert.xml:
* docs/plugins/inspect/plugin-audiomixer.xml:
* docs/plugins/inspect/plugin-audiorate.xml:
* docs/plugins/inspect/plugin-audioresample.xml:
* docs/plugins/inspect/plugin-audiotestsrc.xml:
* docs/plugins/inspect/plugin-cdparanoia.xml:
* docs/plugins/inspect/plugin-compositor.xml:
* docs/plugins/inspect/plugin-encoding.xml:
* docs/plugins/inspect/plugin-gio.xml:
* docs/plugins/inspect/plugin-libvisual.xml:
* docs/plugins/inspect/plugin-ogg.xml:
* docs/plugins/inspect/plugin-opengl.xml:
* docs/plugins/inspect/plugin-opus.xml:
* docs/plugins/inspect/plugin-overlaycomposition.xml:
* docs/plugins/inspect/plugin-pango.xml:
* docs/plugins/inspect/plugin-pbtypes.xml:
* docs/plugins/inspect/plugin-playback.xml:
* docs/plugins/inspect/plugin-rawparse.xml:
* docs/plugins/inspect/plugin-subparse.xml:
* docs/plugins/inspect/plugin-tcp.xml:
* docs/plugins/inspect/plugin-theora.xml:
* docs/plugins/inspect/plugin-typefindfunctions.xml:
* docs/plugins/inspect/plugin-videoconvert.xml:
* docs/plugins/inspect/plugin-videorate.xml:
* docs/plugins/inspect/plugin-videoscale.xml:
* docs/plugins/inspect/plugin-videotestsrc.xml:
* docs/plugins/inspect/plugin-volume.xml:
* docs/plugins/inspect/plugin-vorbis.xml:
* docs/plugins/inspect/plugin-ximagesink.xml:
* docs/plugins/inspect/plugin-xvimagesink.xml:
  Update docs

2019-12-03 11:03:09 +  Tim-Philipp Müller 

* po/hu.po:
  Update translations

2019-10-02 23:21:09 +0300  Sebastian Dröge 

* ext/gl/effects/gstgleffectblur.c:
* ext/gl/gstglbumper.c:
* ext/gl/gstglbumper.h:
* ext/gl/gstglfiltercube.c:
* ext/gl/gstglfiltercube.h:
* ext/gl/gstglfilterglass.c:
* ext/gl/gstglfilterglass.h:
* ext/gl/gstglmosaic.c:
* ext/gl/gstglmosaic.h:
  glfilters: Don't use static variables for storing per-element state

2019-11-08 17:28:44 +0900  Seungha Yang 

* sys/xvimage/xvimagepool.c:
* sys/xvimage/xvimagepool.h:
  xvimagepool: Update size, stride, and offset with allocated XvImage
  Memory layout of XvImage might be different from that of GstVideoInfo.
  If so, the image size, stride, and offset would be wrongly informed.
  Fixes: 
https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/issues/677

2019-11-08 17:33:17 +0900  Seungha Yang 

* sys/xvimage/xvimagepool.c:
  xvimagepool: Fix confusing debug message for padding size

2019-11-08 15:13:59 +0100  Edward Hervey 

* gst-libs/gst/video/video-converter.c:
  video-converter: Fix RGB-XYZ-RGB conversion
  The matrices were in the wrong order.
  Instead of the conversion matrix being
  _ XYZ_TO_RGB_output * RGB_TO_XYZ_input * input_RGB
  It was
  _ RGB_TO_XYZ_input * XYZ_TO_RGB_output * input_RGB

2019-11-07 22:00:03 +0900  Seungha Yang 

* gst/audiorate/gstaudiorate.c:
  audiorate: Update next_offset per rate change
  To support runtime audio samplerate change, re-calculate next target 
offset
  per caps. Calculating the next buffer offset using the previous
  offset seems to be tricky and rounding error prone.
  Fixes: 
https://gitlab.freedesktop.org/gstreamer/gst-plugins-base/issues/693

2019-11-07 21:42:25 +0900  Seungha Yang 

* gst/audiorate/gstaudiorate.c:
* gst/audiorate/gstaudiorate.h:
  Revert "audiorate: accumulate offset by time diff"
  This reverts commit 4fa850e3e6c039000fc7f648de238af6c2278469.
  The commit would break an constant rate audio stream with gap.

2019-08-26 15:03:48 +0200  Jochen Henneberg 

* gst-libs/gst/audio/gstaudioringbuffer.c:
  audioringbuffer: Reset reorder flag before check
  This function might be revisited with different channel position 
mapping
  while audio source goes into play so the reorder flag needs to be 
reset
  before the checks happen.

2019-11-14 12:37:58 +0100  Sebastian Dröge 

* gst-libs/gst/audio/audio-buffer.c:
  audio-buffer: Don't fail to map buffers with zero samples
  Instead initialize the map infos, etc to NULL like gst_buffer_map()
  would be doing on a zero-sized buffer.
  T

gst-plugins-good 1.16.2

[Tim-Philipp Müller]

ChangeLog
=

2019-12-03 11:06:36 +  Tim-Philipp Müller 

* ChangeLog:
* NEWS:
* RELEASE:
* configure.ac:
* gst-plugins-good.doap:
* meson.build:
  Release 1.16.2

2019-12-03 11:06:35 +  Tim-Philipp Müller 

* 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-gtk.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-lame.xml:
* docs/plugins/inspect/plugin-level.xml:
* docs/plugins/inspect/plugin-matroska.xml:
* docs/plugins/inspect/plugin-mpg123.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-qmlgl.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-shout2.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-twolame.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:
  Update docs

2019-12-03 11:06:34 +  Tim-Philipp Müller 

* po/hu.po:
  Update translations

2016-11-30 15:55:01 +0100  Havard Graff 

* gst/rtp/gstrtpL16depay.c:
  rtpL16depay: don't crash if data is not modulo channels*width

2019-11-29 20:46:26 +0900  Seungha Yang 

* ext/vpx/gstvp9dec.c:
  vp9dec: Fix broken 4:4:4 8bits decoding
  VPX_IMG_FMT_I444 pixel format with sRGB colorspace means
  GBR data.
  Fixes: 
https://gitlab.freedesktop.org/gstreamer/gst-plugins-good/issues/651

2019-11-29 14:23:49 +0100  Havard Graff 

* gst/rtpmanager/gstrtpsession.c:
* tests/check/elements/rtpsession.c:
  rtpsession: add locking for clear-pt-map
  ...or it will segfault from time to time...

2019-08-19 07:30:17 +  Amr Mahdi 

* gst/wavparse/gstwavparse.c:
  wavparse: Fix push mode ignoring audio with a size smaller than 
segment buffer
  In push mode (streaming), if the audio size is smaller than segment 
buffer

gstreamer 1.16.2

[Tim-Philipp Müller]

ChangeLog
=

2019-12-03 10:57:30 +  Tim-Philipp Müller 

* ChangeLog:
* NEWS:
* RELEASE:
* configure.ac:
* gstreamer.doap:
* meson.build:
  Release 1.16.2

2019-12-03 10:57:30 +  Tim-Philipp Müller 

* docs/plugins/inspect/plugin-coreelements.xml:
* docs/plugins/inspect/plugin-coretracers.xml:
  Update docs

2019-12-03 10:57:29 +  Tim-Philipp Müller 

* po/hu.po:
  Update translations

2019-12-03 10:40:37 +  Tim-Philipp Müller 

* gst/parse/grammar.y:
* gst/parse/meson.build:
  Revert "gst/parse: define pure-parser depending on bison version"
  This reverts commit 77141834bb5d124fb5781ed3be6230cc66cb42de.
  This breaks the autotools build and it seems too much effort to
  fix that up just to fix a bison warning.

2019-11-27 15:47:32 +0100  Edward Hervey 

* plugins/tracers/gstlatency.c:
  tracers: Don't leak temporary GstStructure
  CID: 1455462

2019-08-26 12:48:28 +0200  Víctor Manuel Jáquez Leal 

* gst/parse/grammar.y.in:
* gst/parse/meson.build:
  gst/parse: define pure-parser depending on bison version
  After release bison 2.5 the declaration %pure-parser was deprecated
  in favor of %define api.pure
  Nonetheless, until bison 3.4, the declaration was treated as backward
  compatibility, but now bison shows a warning:
  warning: deprecated directive, use ‘%define api.pure’
  The patch's approach is to handle both directives according with the
  used bison's version, by string replacement at source configuration
  stage.

2019-10-25 01:41:27 +0300  Sebastian Dröge 

* plugins/elements/gsttee.c:
  tee: First deactivate the pad and then remove it when releasing pads
  This reverts a96002bb28c21b30fb9338a4620ad20504c70aa5, which is not
  necessary anymore. If we release the pad after removing it then none 
of
  the deactivation code will actually be called because the pad has no
  parent anymore, and we require a parent on the pad for deactivation to
  happen.
  This can then, among other things, cause a streaming thread to be 
still
  stuck in a pad probe because the pad was never flushed, and waiting
  there forever because now the pad will actually never be flushed 
anymore.

2019-10-25 01:39:50 +0300  Sebastian Dröge 

* plugins/elements/gsttee.c:
  tee: Check for the removed pad flag also in the slow pushing path
  If a pad is currently being released we don't want to forward the
  FLUSHING flow return but instead consider it as NOT_LINKED. FLUSHING
  would also cause upstream to be FLUSHING.
  This part was missed in a3c4a3201a705eb1934ceeea34d1ca42d4571c07 and
  resulted in a different (and wrong) workaround in
  a96002bb28c21b30fb9338a4620ad20504c70aa5.

2019-10-25 01:39:05 +0300  Sebastian Dröge 

* plugins/elements/gsttee.c:
  tee: Lock mutex before reading the removed flag of the pads
  Otherwise we're not guaranteed to read the very latest value that
  another thread might've written in there when the pad was released, 
and
  could instead work with an old value.

2019-09-30 11:34:51 +0300  Sebastian Dröge 

* gst/gstbin.c:
  bin: Drop need-context messages without source instead of crashing

2019-09-30 11:49:35 +0300  Sebastian Dröge 

* gst/gstbuffer.c:
* gst/gstcaps.c:
  gst: Don't pass miniobjects to GST_DEBUG_OBJECT() and similar macros
  The argument must be at least a GObject according to the 
GstLogFunction
  definition, and while the default C log function handles miniobjects
  just fine this is crashing bindings and user-supplied log functions 
that
  (rightfully) don't expect anything but GObjects.

2019-08-20 01:02:48 +0900  Seungha Yang 

* tools/gst-launch.c:
  gst-launch: Use gst_print* instead of g_print* to fix broken stdout 
on Windows
  Concurrent Windows' colored debug message and g_print will print
  string hard to read. Instead, use gst_print* which serialize
  debug output and the APIs call.

2019-08-20 00:59:15 +0900  Seungha Yang 

* gst/gstinfo.c:
  info: Take lock around all prinf on Windows
  On Windows, concurrent colored gstreamr debug output and usual
  stdout/stderr string will cause broken output on terminal.
  Since it's OS specific behavior, that's hard to completely avoid it
  but we can protect it at least among our printing interfaces side.



Download

https://download.gnome.org/sources/gstreamer/1.16/gstreamer-1.16.2.tar.xz 
(3.17M)
  sha256sum:

gst-plugins-good 1.16.1

[Tim-Philipp Müller]

News


on 23 September 2019.

ChangeLog
=
https://download.gnome.org/sources/gst-plugins-good/1.16/gst-plugins-good-1.16.1.changes
  (13.1K)

Download

https://download.gnome.org/sources/gst-plugins-good/1.16/gst-plugins-good-1.16.1.tar.xz
 (3.71M)
  sha256sum: 9fbabe69018fcec707df0b71150168776040cde6c1a26bb5a82a136755fa8f1f

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gstreamer 1.16.1

[Tim-Philipp Müller]

News


on 23 September 2019.

ChangeLog
=
https://download.gnome.org/sources/gstreamer/1.16/gstreamer-1.16.1.changes  
(14.8K)

Download

https://download.gnome.org/sources/gstreamer/1.16/gstreamer-1.16.1.tar.xz 
(3.17M)
  sha256sum: 02211c3447c4daa55919c5c0f43a82a6fbb51740d57fc3af0639d46f1cf4377d

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gst-plugins-base 1.16.1

[Tim-Philipp Müller]

News


on 23 September 2019.

ChangeLog
=
https://download.gnome.org/sources/gst-plugins-base/1.16/gst-plugins-base-1.16.1.changes
  (22.4K)

Download

https://download.gnome.org/sources/gst-plugins-base/1.16/gst-plugins-base-1.16.1.tar.xz
 (3.75M)
  sha256sum: 5c3cc489933d0597087c9bc6ba251c93693d64554bcc563539a084fa2d5fcb2b

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gst-plugins-base 1.16.0

[Tim-Philipp Müller]

ChangeLog
=

2019-04-19 00:19:55 +0100  Tim-Philipp Müller 

* ChangeLog:
* NEWS:
* README:
* RELEASE:
* configure.ac:
* gst-plugins-base.doap:
* meson.build:
  Release 1.16.0

2019-04-19 00:19:55 +0100  Tim-Philipp Müller 

* docs/plugins/inspect/plugin-adder.xml:
* docs/plugins/inspect/plugin-alsa.xml:
* docs/plugins/inspect/plugin-app.xml:
* docs/plugins/inspect/plugin-audioconvert.xml:
* docs/plugins/inspect/plugin-audiomixer.xml:
* docs/plugins/inspect/plugin-audiorate.xml:
* docs/plugins/inspect/plugin-audioresample.xml:
* docs/plugins/inspect/plugin-audiotestsrc.xml:
* docs/plugins/inspect/plugin-cdparanoia.xml:
* docs/plugins/inspect/plugin-compositor.xml:
* docs/plugins/inspect/plugin-encoding.xml:
* docs/plugins/inspect/plugin-gio.xml:
* docs/plugins/inspect/plugin-libvisual.xml:
* docs/plugins/inspect/plugin-ogg.xml:
* docs/plugins/inspect/plugin-opengl.xml:
* docs/plugins/inspect/plugin-opus.xml:
* docs/plugins/inspect/plugin-overlaycomposition.xml:
* docs/plugins/inspect/plugin-pango.xml:
* docs/plugins/inspect/plugin-pbtypes.xml:
* docs/plugins/inspect/plugin-playback.xml:
* docs/plugins/inspect/plugin-rawparse.xml:
* docs/plugins/inspect/plugin-subparse.xml:
* docs/plugins/inspect/plugin-tcp.xml:
* docs/plugins/inspect/plugin-theora.xml:
* docs/plugins/inspect/plugin-typefindfunctions.xml:
* docs/plugins/inspect/plugin-videoconvert.xml:
* docs/plugins/inspect/plugin-videorate.xml:
* docs/plugins/inspect/plugin-videoscale.xml:
* docs/plugins/inspect/plugin-videotestsrc.xml:
* docs/plugins/inspect/plugin-volume.xml:
* docs/plugins/inspect/plugin-vorbis.xml:
* docs/plugins/inspect/plugin-ximagesink.xml:
* docs/plugins/inspect/plugin-xvimagesink.xml:
  Update docs

2019-04-19 00:19:53 +0100  Tim-Philipp Müller 

* po/zh_CN.po:
  Update translations

2019-01-23 12:50:40 +0100  Michael Tretter 

* gst-libs/gst/gl/gstglupload.c:
  glupload: prevent segfault when updating caps
  When the glupload element renegotiates the caps, set_caps will reset 
the
  method_impl to NULL, but the method will be kept. transform_caps tries
  to use the method_impl to transform the caps, because a method is set,
  but will segfault.



Download

https://download.gnome.org/sources/gst-plugins-base/1.16/gst-plugins-base-1.16.0.tar.xz
 (3.74M)
  sha256sum: 4093aa7b51e28fb24dfd603893fead8d1b7782f088b05ed0f22a21ef176fb5ae

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gst-plugins-good 1.16.0

[Tim-Philipp Müller]

ChangeLog
=

2019-04-19 00:23:16 +0100  Tim-Philipp Müller 

* ChangeLog:
* NEWS:
* README:
* RELEASE:
* configure.ac:
* gst-plugins-good.doap:
* meson.build:
  Release 1.16.0

2019-04-19 00:23:16 +0100  Tim-Philipp Müller 

* 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-gtk.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-lame.xml:
* docs/plugins/inspect/plugin-level.xml:
* docs/plugins/inspect/plugin-matroska.xml:
* docs/plugins/inspect/plugin-mpg123.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-qmlgl.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-shout2.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-twolame.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:
  Update docs

2019-04-19 00:23:14 +0100  Tim-Philipp Müller 

* po/el.po:
* po/zh_CN.po:
  Update translations

2019-04-18 17:14:18 +0200  Benjamin Sigonneau 

* ext/qt/qtplugin.pro:
  qmlglsink: fix compilation with Qt >= 5.5 on Windows
  As of Qt >= 5.5, qmake do not link to opengl32 by default anymore. 
This commit adds opengl32.lib to the .pro
  file so that the plugin can be build using QtCreator on Windows.

2019-04-17 15:48:26 +0530  Nirbheek Chauhan 

* ext/qt/meson.build:
  meson: Build qt plugin in C++11 mode explicitly
  This works implicitly most of the time, but we need to set it
  explicitly for building with Android.

2019-04-16 14:35:06 +0530  Guillaume Desmottes 


* sys/v4l2/gstv4l2transform.c:
* sys/v4l2/gstv4l2videodec.c:
* sys/v4l2/gstv4l2videoenc.c:
  v4l2: fix use after free when handling events
  The sink_event parent function may consume the event so we shouldn&

gstreamer 1.16.0

[Tim-Philipp Müller]

ChangeLog
=

2019-04-19 00:15:21 +0100  Tim-Philipp Müller 

* ChangeLog:
* NEWS:
* README:
* RELEASE:
* configure.ac:
* gstreamer.doap:
* meson.build:
  Release 1.16.0

2019-04-19 00:15:21 +0100  Tim-Philipp Müller 

* docs/plugins/inspect/plugin-coreelements.xml:
* docs/plugins/inspect/plugin-coretracers.xml:
  Update docs

2019-04-19 00:15:19 +0100  Tim-Philipp Müller 

* po/zh_CN.po:
  Update translations

2019-04-18 10:13:51 +0200  Olivier Crête 

* libs/gst/base/gstbasesrc.c:
  basesrc: Downgrade EOS warning
  In the case of pushfilesrc, this is the expected behaviour, so let's
  downgrade the warning to a debug message to avoid confusing users.

2019-04-17 20:46:58 +0200  Mathieu Duponchelle 

* tests/check/libs/basesrc.c:
  tests: basesrc: unref gst_bus_timed_pop_filtered return

2019-04-16 13:29:00 +0200  Mathieu Duponchelle 

* gst/gst.c:
* gst/gst_private.h:
* gst/gstinfo.c:
  gstinfo: clean up function pointer names hashtable
  And add strduped function pointer names to the global quark
  table, so that they don't get reported as lost by valgrind.
  This allows us to use GST_DEBUG when running tests under
  valgrind.

2019-04-16 23:50:15 +0100  Tim-Philipp Müller 

* libs/gst/base/gstaggregator.c:
  aggregator: fix doc chunk for new buffer-consumed signal
  Fixes 'Warning: GstBase: incorrect number of parameters in
  comment block, parameter annotations will be ignored.' from
  g-ir-scanner.

2019-04-12 09:23:52 -0400  Julian Bouzas 

* plugins/tracers/gstlatency.c:
  tracer: latency: Don't compare element name as element Id is already 
unique

2019-04-12 08:38:03 -0400  Julian Bouzas 

* plugins/tracers/gstlatency.c:
  tracer: latency: Remove redundant if conditions

2019-04-12 08:34:49 -0400  Julian Bouzas 

* plugins/tracers/gstlatency.c:
  tracer: latency: Make GST_DEBUG logs consistent

2019-04-12 08:28:22 -0400  Julian Bouzas 

* plugins/tracers/gstlatency.c:
  tracer: latency: Fix bug when dropping sub-latency probe event
  Fixes #373

2019-04-12 16:37:18 +0200  Philipp Zabel 

* gst/gstevent.c:
  event: fix seek event creation
  Creating seek events segfaults on 32-bit ARM since commit 2fa15d53717c
  ('event: add new seek parameter, "trickmode-interval"'), which missed
  casting the trickmode-interval initializer in the variable argument 
list
  to guint64.

2019-04-11 15:32:51 -0400  Xavier Claessens 

* tools/gst-inspect.c:
  gst-inspect: Do not print warning if 'less' is missing



Download

https://download.gnome.org/sources/gstreamer/1.16/gstreamer-1.16.0.tar.xz 
(3.16M)
  sha256sum: 0e8e2f7118be437cba879353970cf83c2acced825ecb9275ba05d9186ef07c00

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gstreamer 1.15.90

[Tim-Philipp Müller]

News


GSTREAMER 1.16 RELEASE NOTES


GStreamer 1.16 has not been released yet. It is scheduled for release in
April 2019.

1.15.x is the unstable development version that is being developed in
the git master branch and which will eventually result in 1.16.

1.16 will be backwards-compatible to the stable 1.14, 1.12, 1.10, 1.8,
1.6, 1.4, 1.2 and 1.0 release series.

See https://gstreamer.freedesktop.org/releases/1.16/ for the latest

ChangeLog
=
https://download.gnome.org/sources/gstreamer/1.15/gstreamer-1.15.90.changes  
(94)

Download

https://download.gnome.org/sources/gstreamer/1.15/gstreamer-1.15.90.tar.xz 
(3.16M)
  sha256sum: f71fe61f6b90b5a1ded99d5e5a47b90f2855dbcad115863d4356a9d330f6bbb6

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gst-plugins-base 1.15.90

[Tim-Philipp Müller]

News


GSTREAMER 1.16 RELEASE NOTES


GStreamer 1.16 has not been released yet. It is scheduled for release in
April 2019.

1.15.x is the unstable development version that is being developed in
the git master branch and which will eventually result in 1.16.

1.16 will be backwards-compatible to the stable 1.14, 1.12, 1.10, 1.8,
1.6, 1.4, 1.2 and 1.0 release series.

See https://gstreamer.freedesktop.org/releases/1.16/ for the latest

ChangeLog
=
https://download.gnome.org/sources/gst-plugins-base/1.15/gst-plugins-base-1.15.90.changes
  (94)

Download

https://download.gnome.org/sources/gst-plugins-base/1.15/gst-plugins-base-1.15.90.tar.xz
 (3.74M)
  sha256sum: c87fad68f8d5313b77b3473ce5433dcd0d931327635012b226ce2a109129a7a3

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gst-plugins-good 1.15.90

[Tim-Philipp Müller]

News


GSTREAMER 1.16 RELEASE NOTES


GStreamer 1.16 has not been released yet. It is scheduled for release in
April 2019.

1.15.x is the unstable development version that is being developed in
the git master branch and which will eventually result in 1.16.

1.16 will be backwards-compatible to the stable 1.14, 1.12, 1.10, 1.8,
1.6, 1.4, 1.2 and 1.0 release series.

See https://gstreamer.freedesktop.org/releases/1.16/ for the latest

ChangeLog
=
https://download.gnome.org/sources/gst-plugins-good/1.15/gst-plugins-good-1.15.90.changes
  (94)

Download

https://download.gnome.org/sources/gst-plugins-good/1.15/gst-plugins-good-1.15.90.tar.xz
 (3.70M)
  sha256sum: 23330ddba08177f6b279b38e0d88c10a5af5ca4b5550e5fd70a7f0d9db6b086a

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gst-plugins-good 1.14.4

[Tim-Philipp Müller]

News


The fourth 1.14 bug-fix release (1.14.4) was released on 2 October 2018.

This release only contains bugfixes and it should be safe to update from
1.14.x.

Highlighted bugfixes in 1.14.4

-   glviewconvert: wait and set the gl sync meta on buffers
-   glviewconvert: Copy composition meta from the primary buffer to both
outputs
-   glcolorconvert: Don’t copy overlay composition meta over to NULL
outbufs
-   matroskademux: add functionality needed for MSE use case fixing
youtube playback in epiphany/webkit-gtk
-   msdk: fix build on windows
-   opusenc: fix another crash on 32-bit x86 on windows (alignment issue
in SSE optimisations)
-   osxaudio: add support for parsing more channel layouts
-   tagdemux: Use upstream GST_EVENT_STREAM_START (and stream-id) if
present
-   vorbisdec: fix header handling regression: init decoder immediately
once we have headers
-   wasapisink: recover from low buffer levels in shared mode
-   fix GstSegment unit test which would fail on some 32-bit x86 CPUs

Contributors to 1.14.4

Alicia Boya García, Christoph Reiter, Edward Hervey, Jan Schmidt,
Matthew Waters, Nicola Murino, Nicolas Dufresne, Sebastian Dröge,
Tim-Philipp Müller, Wangfei, and many others. Thank you all.

List of bugs fixed in 1.14.4

For a full list of bugfixes see Bugzilla. Note that this is not the full
list of changes. For the full list of changes please refer to the GIT
logs or ChangeLogs of the particular modules.

ChangeLog
=
https://download.gnome.org/sources/gst-plugins-good/1.14/gst-plugins-good-1.14.4.changes
  (6.39K)

Download

https://download.gnome.org/sources/gst-plugins-good/1.14/gst-plugins-good-1.14.4.tar.xz
 (3.62M)
  sha256sum: 5f8b553260cb0aac56890053d8511db1528d53cae10f0287cfce2cb2acc70979

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gstreamer 1.14.4

[Tim-Philipp Müller]

News


The fourth 1.14 bug-fix release (1.14.4) was released on 2 October 2018.

This release only contains bugfixes and it should be safe to update from
1.14.x.

Highlighted bugfixes in 1.14.4

-   glviewconvert: wait and set the gl sync meta on buffers
-   glviewconvert: Copy composition meta from the primary buffer to both
outputs
-   glcolorconvert: Don’t copy overlay composition meta over to NULL
outbufs
-   matroskademux: add functionality needed for MSE use case fixing
youtube playback in epiphany/webkit-gtk
-   msdk: fix build on windows
-   opusenc: fix another crash on 32-bit x86 on windows (alignment issue
in SSE optimisations)
-   osxaudio: add support for parsing more channel layouts
-   tagdemux: Use upstream GST_EVENT_STREAM_START (and stream-id) if
present
-   vorbisdec: fix header handling regression: init decoder immediately
once we have headers
-   wasapisink: recover from low buffer levels in shared mode
-   fix GstSegment unit test which would fail on some 32-bit x86 CPUs

Contributors to 1.14.4

Alicia Boya García, Christoph Reiter, Edward Hervey, Jan Schmidt,
Matthew Waters, Nicola Murino, Nicolas Dufresne, Sebastian Dröge,
Tim-Philipp Müller, Wangfei, and many others. Thank you all.

List of bugs fixed in 1.14.4

For a full list of bugfixes see Bugzilla. Note that this is not the full
list of changes. For the full list of changes please refer to the GIT
logs or ChangeLogs of the particular modules.

ChangeLog
=
https://download.gnome.org/sources/gstreamer/1.14/gstreamer-1.14.4.changes  
(807)

Download

https://download.gnome.org/sources/gstreamer/1.14/gstreamer-1.14.4.tar.xz 
(3.11M)
  sha256sum: f94f6696c5f05a3b3a9183e39c5f5c0b779f75a04c0efa497e7920afa985ffc7

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gst-plugins-base 1.14.4

[Tim-Philipp Müller]

News


The fourth 1.14 bug-fix release (1.14.4) was released on 2 October 2018.

This release only contains bugfixes and it should be safe to update from
1.14.x.

Highlighted bugfixes in 1.14.4

-   glviewconvert: wait and set the gl sync meta on buffers
-   glviewconvert: Copy composition meta from the primary buffer to both
outputs
-   glcolorconvert: Don’t copy overlay composition meta over to NULL
outbufs
-   matroskademux: add functionality needed for MSE use case fixing
youtube playback in epiphany/webkit-gtk
-   msdk: fix build on windows
-   opusenc: fix another crash on 32-bit x86 on windows (alignment issue
in SSE optimisations)
-   osxaudio: add support for parsing more channel layouts
-   tagdemux: Use upstream GST_EVENT_STREAM_START (and stream-id) if
present
-   vorbisdec: fix header handling regression: init decoder immediately
once we have headers
-   wasapisink: recover from low buffer levels in shared mode
-   fix GstSegment unit test which would fail on some 32-bit x86 CPUs

Contributors to 1.14.4

Alicia Boya García, Christoph Reiter, Edward Hervey, Jan Schmidt,
Matthew Waters, Nicola Murino, Nicolas Dufresne, Sebastian Dröge,
Tim-Philipp Müller, Wangfei, and many others. Thank you all.

List of bugs fixed in 1.14.4

For a full list of bugfixes see Bugzilla. Note that this is not the full
list of changes. For the full list of changes please refer to the GIT
logs or ChangeLogs of the particular modules.

ChangeLog
=
https://download.gnome.org/sources/gst-plugins-base/1.14/gst-plugins-base-1.14.4.changes
  (3.32K)

Download

https://download.gnome.org/sources/gst-plugins-base/1.14/gst-plugins-base-1.14.4.tar.xz
 (3.53M)
  sha256sum: ca6139490e48863e7706d870ff4e8ac9f417b56f3b9e4b3ce490c13b09a77461

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gst-plugins-good 1.14.3

[Tim-Philipp Müller]

ChangeLog
=

2018-09-16 16:20:32 +0100  Tim-Philipp Müller 

* ChangeLog:
* NEWS:
* RELEASE:
* configure.ac:
* gst-plugins-good.doap:
* meson.build:
  Release 1.14.3

2018-09-16 16:20:32 +0100  Tim-Philipp Müller 

* 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-gtk.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-lame.xml:
* docs/plugins/inspect/plugin-level.xml:
* docs/plugins/inspect/plugin-matroska.xml:
* docs/plugins/inspect/plugin-mpg123.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-qmlgl.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-shout2.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-twolame.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:
  Update docs

2018-08-15 02:10:25 +1000  Jan Schmidt 

* gst/multifile/gstsplitmuxsink.c:
* gst/multifile/gstsplitmuxsink.h:
  splitmuxsink: Fix reference counting loop
  The stream context was holding a reference to the
  internal queue and pads, with pad probes that were
  in turn holding references to the stream context.
  This lead to a leak if the request pads weren't explicitly
  released.
  https://bugzilla.gnome.org/show_bug.cgi?id=796893

2018-07-17 00:03:19 +1000  Jan Schmidt 

* gst/multifile/gstsplitmuxsink.c:
  splitmux: Improve handling of repeated timestamps
  When handling input with timestamps that repeat, sometimes
  splitmuxsink would get confused and ignore a keyframe.
  The logic in question is a holdover from before the cmd queue
  moved the file cutting to the multiqueue output side and made
  it deterministic, so it's no longer needed on the input
  here.
 

gst-plugins-base 1.14.3

[Tim-Philipp Müller]

ChangeLog
=

2018-09-16 16:17:17 +0100  Tim-Philipp Müller 

* ChangeLog:
* NEWS:
* RELEASE:
* configure.ac:
* gst-plugins-base.doap:
* meson.build:
  Release 1.14.3

2018-09-16 16:17:17 +0100  Tim-Philipp Müller 

* docs/plugins/inspect/plugin-adder.xml:
* docs/plugins/inspect/plugin-alsa.xml:
* docs/plugins/inspect/plugin-app.xml:
* docs/plugins/inspect/plugin-audioconvert.xml:
* docs/plugins/inspect/plugin-audiomixer.xml:
* docs/plugins/inspect/plugin-audiorate.xml:
* docs/plugins/inspect/plugin-audioresample.xml:
* docs/plugins/inspect/plugin-audiotestsrc.xml:
* docs/plugins/inspect/plugin-cdparanoia.xml:
* docs/plugins/inspect/plugin-encoding.xml:
* docs/plugins/inspect/plugin-gio.xml:
* docs/plugins/inspect/plugin-libvisual.xml:
* docs/plugins/inspect/plugin-ogg.xml:
* docs/plugins/inspect/plugin-opengl.xml:
* docs/plugins/inspect/plugin-opus.xml:
* docs/plugins/inspect/plugin-pango.xml:
* docs/plugins/inspect/plugin-pbtypes.xml:
* docs/plugins/inspect/plugin-playback.xml:
* docs/plugins/inspect/plugin-rawparse.xml:
* docs/plugins/inspect/plugin-subparse.xml:
* docs/plugins/inspect/plugin-tcp.xml:
* docs/plugins/inspect/plugin-theora.xml:
* docs/plugins/inspect/plugin-typefindfunctions.xml:
* docs/plugins/inspect/plugin-videoconvert.xml:
* docs/plugins/inspect/plugin-videorate.xml:
* docs/plugins/inspect/plugin-videoscale.xml:
* docs/plugins/inspect/plugin-videotestsrc.xml:
* docs/plugins/inspect/plugin-volume.xml:
* docs/plugins/inspect/plugin-vorbis.xml:
* docs/plugins/inspect/plugin-ximagesink.xml:
* docs/plugins/inspect/plugin-xvimagesink.xml:
  Update docs

2018-09-13 20:34:08 -0300  Marcos Kintschner 

* ext/opus/gstopusenc.c:
  opusenc: fix segmentation fault at x86 version
  The argument 0x0 is interpreted by the x86 compiler as a 32-bit int, 
but
  it is consumed as a 64-bit uint causing a segmentation fault. We need 
to
  explicit cast it to guint64 in order for the va_list to be built 
correctly.
  https://bugzilla.gnome.org/show_bug.cgi?id=797092

2018-09-11 00:41:59 +0530  Nirbheek Chauhan 

* gst-libs/gst/audio/gstaudiosink.c:
* gst-libs/gst/audio/gstaudiosrc.c:
* gst-libs/gst/audio/gstaudioutilsprivate.c:
* gst-libs/gst/audio/gstaudioutilsprivate.h:
  gstaudiosrc/sink: Set audio ringbuffer thread priority
  On Windows, the ringbuffer thread function must have the "Pro Audio"
  priority set, otherwise it sometimes doesn't get scheduled for
  200-300ms, which will immediately cause an underrun unless you set
  a very high latency-time and buffer-time.
  This has no compile-time deps since it tries to load avrt.dll at
  runtime to set the thread priority.

2018-08-16 19:37:33 +0300  Sebastian Dröge 

* ext/vorbis/gstvorbisdec.c:
  vorbisdec: Always handle in-band header packets once the first 
non-header packet arrives
  And clean up any old pending headers if we receive a new 
identification
  header, or if we receive a new set of headers via caps.
  Otherwise it might happen that we receive one or more header but not
  all, and then afterwards all headers again, and libvorbis does not 
like
  getting headers passed multiple times and would error out.
  It only makes sense to pass the very latest headers to the decoder at
  the time we can actually make use of them.
  https://bugzilla.gnome.org/show_bug.cgi?id=796980

2017-09-21 14:03:28 +0900  Yuji Kuwabara 

* gst-libs/gst/gl/dispmanx/gstglwindow_dispmanx_egl.c:
  gl/dispmanx: fix removing foreign window handle
  gst_gl_window_dispmanx_egl_set_window_handle() removes native window 
handle
  (dispmanx element), regardless it was foreign window handle
  (set via gst_video_overlay_set_window_handle()) or not.
  This problem prevents glimagesink reusable.
  (PAUSED -> READY -> PAUSED does not work)
  This patch corrects it comparing the native window handle with 
foreign window
  handle. This behavior is same as gst_gl_window_dispmanx_egl_close().
  https://bugzilla.gnome.org/show_bug.cgi?id=785199

2018-08-27 11:07:47 +0300  Sebastian Dröge 

* ext/ogg/gstoggstream.c:
  oggdemux: Ensure that no pad values are set when setting up the mapper
  Otherwise we might have arbitrary values set that are used later and 
can
  cause undefined behaviour, as found by ossfuzz.

2018-08-26 01:52:41 +0200  Tim-Philipp Müller 

* ext/gl/gstglcolorscale.c:
  glcolorscale:

gstreamer 1.14.3

[Tim-Philipp Müller]

ChangeLog
=

2018-09-16 16:13:01 +0100  Tim-Philipp Müller 

* ChangeLog:
* NEWS:
* RELEASE:
* configure.ac:
* gstreamer.doap:
* meson.build:
  Release 1.14.3

2018-09-16 16:13:00 +0100  Tim-Philipp Müller 

* docs/plugins/inspect/plugin-coreelements.xml:
* docs/plugins/inspect/plugin-coretracers.xml:
  Update docs

2018-08-27 22:32:01 +1000  Jan Schmidt 

* tests/check/gst/gstsegment.c:
  gstsegment: Add check for gst_segment_offset_running_time()
  Add a check for gst_segment_offset_running_time() that values
  are taken directly from the segment base if possible.

2018-08-23 22:34:47 +1000  Jan Schmidt 

* gst/gstsegment.c:
* tests/check/gst/gstsegment.c:
  gstsegment: Handle positions before the segment properly
  Fixes for gst_segment_position_from_running_time_full() when
  converting running_times that precede the segment start (or
  stop in a negative rate segment)
  The return value was incorrectly negated in those cases.
  Add some more unit test checks for those cases, and especially
  for segments with offsets.

2018-09-08 13:05:13 +0100  Philippe Normand 

* gst/gstbin.c:
  bin: Fix use-after-free issue in gst_bin_add()
  gst_element_post_message() takes ownership of the message so we need 
to increase
  its refcount until we no longer require access to its data 
(context_type).
  https://bugzilla.gnome.org/show_bug.cgi?id=797099

2018-09-03 12:06:35 +0100  Philippe Normand 

* gst/gstutils.c:
  utils: Set default values for position and duration query results
  https://bugzilla.gnome.org/show_bug.cgi?id=797066

2018-08-30 17:44:07 +0100  Philippe Normand 

* libs/gst/base/gstbaseparse.c:
  baseparse: avg_bitrate calculation critical warning fix
  The avg_bitrate is an unsigned int, so the gst_util_uin64_scale() 
function can't
  be used for it, as it expects signed integers for the fraction parts 
arguments.
  https://bugzilla.gnome.org/show_bug.cgi?id=797054

2018-08-31 12:15:16 +0300  Sebastian Dröge 

* plugins/elements/gstinputselector.c:
  input-selector: Bring latency handling in sync with GstPad code

2018-08-31 12:12:13 +0300  Sebastian Dröge 

* gst/gstpad.c:
  Revert "pad: Don't drop LATENCY queries with default implementation"
  This reverts commit 794944f779f954375fc74a3fffcc2067bba6a3e5.
  Accumulating non-live latency values generally makes no sense and 
often
  gives invalid results with min>max

2018-08-31 12:12:09 +0300  Sebastian Dröge 

* gst/gstpad.c:
  Revert "pad: Accumulate live/non-live latency values separately"
  This reverts commit f5783e1cacb09867d81ba089b229faa7dd0edd0c.

2018-08-31 11:47:03 +0300  Sebastian Dröge 

* plugins/elements/gstinputselector.c:
  input-selector: Apply GstPad default latency handler fixes here too

2018-08-31 11:41:47 +0300  Sebastian Dröge 

* gst/gstpad.c:
  pad: Accumulate live/non-live latency values separately
  And only ever use the non-live values if all pads are non-live,
  otherwise only use the results of all live pads.
  It's unclear what one would use the values for in the non-live case, 
but
  by this we at least pass them through correctly then.
  This is a follow-up for 794944f779f954375fc74a3fffcc2067bba6a3e5, 
which
  causes wrong latency calculations if the first pad is non-live but a
  later pad is actually live. In that case the live values would be
  accumulated together with the values of the non-live first pad,
  generally causing wrong min/max latencies to be calculated.

2018-08-02 10:55:40 +0300  Sebastian Dröge 

* libs/gst/base/gsttypefindhelper.c:
  typefindhelper: Mark gst_type_find_helper_get_range_full() as Since 
1.14.3

2018-07-30 18:51:35 +0300  Sebastian Dröge 

* gst/gstpad.c:
  pad: Ensure that the pad is blocked for IDLE probes if they are 
called from the streaming thread too
  IDLE probes that are directly called when being added will increase /
  decrease the "number of IDLE probes running" counter around the call,
  but when running from the streaming thread this won't happen.
  This has the effect that when running from a streaming thread it is
  possible to push serialized events or data out of the pad without
  problems, but otherwise it would deadlock because serialized data 
would
  wait for the IDLE probe to finish first (it is blocking after all!).
  With this change it will now always consistently deadlock instead of
  just every once in a while, which should m

gstreamer 1.14.2

[Tim-Philipp Müller]

ChangeLog
=

2018-07-20 00:46:55 +0100  Tim-Philipp Müller 

* ChangeLog:
* NEWS:
* RELEASE:
* configure.ac:
* gstreamer.doap:
* meson.build:
  Release 1.14.2

2018-07-20 00:46:55 +0100  Tim-Philipp Müller 

* docs/plugins/inspect/plugin-coreelements.xml:
* docs/plugins/inspect/plugin-coretracers.xml:
  Update docs

2018-07-18 21:13:57 -0400  Thibault Saunier 

* libs/gst/base/gstbasetransform.c:
  basetransform: Do not check if NULL is an emtpy caps
  gst_base_transform_transform_caps can return NULL in various 
conditions
  thus we should not treat its result as valid caps.
  In all other places NULL is properly handled.

2018-06-18 16:29:18 +0200  Edward Hervey 

* plugins/elements/gstconcat.c:
  concat: Properly forward the SEGMENT seqnum

2018-05-21 09:14:37 +0200  Mark Nauwelaerts 

* gst/gstmeta.c:
* gst/gstprotection.c:
  gst: add some GIR array annotations

2018-05-20 13:53:31 +0200  Christoph Reiter 

* libs/gst/controller/controller_mkenum.py:
  meson: Fix detection of glib-mkenums under MSYS2
  Under MSYS2 glib-mkenums is an executable and has a .exe extension and
  the path does not end with "glib-mkenums".
  Make the script compare the path without the file extension instead.
  https://bugzilla.gnome.org/show_bug.cgi?id=796273



Download

https://download.gnome.org/sources/gstreamer/1.14/gstreamer-1.14.2.tar.xz 
(3.10M)
  sha256sum: 4bd6127299f3f29379046bbd58a526e6353b569e0e72f7b4df2ae70df6882e09

___
release-team@gnome.org
https://mail.gnome.org/mailman/listinfo/release-team
Release-team lurker? Do NOT participate in discussions.

gst-plugins-base 1.14.2

[Tim-Philipp Müller]

ChangeLog
=

2018-07-20 00:51:42 +0100  Tim-Philipp Müller 

* ChangeLog:
* NEWS:
* RELEASE:
* configure.ac:
* gst-plugins-base.doap:
* meson.build:
  Release 1.14.2

2018-07-20 00:51:42 +0100  Tim-Philipp Müller 

* docs/plugins/gst-plugins-base-plugins.args:
* docs/plugins/inspect/plugin-adder.xml:
* docs/plugins/inspect/plugin-alsa.xml:
* docs/plugins/inspect/plugin-app.xml:
* docs/plugins/inspect/plugin-audioconvert.xml:
* docs/plugins/inspect/plugin-audiomixer.xml:
* docs/plugins/inspect/plugin-audiorate.xml:
* docs/plugins/inspect/plugin-audioresample.xml:
* docs/plugins/inspect/plugin-audiotestsrc.xml:
* docs/plugins/inspect/plugin-cdparanoia.xml:
* docs/plugins/inspect/plugin-encoding.xml:
* docs/plugins/inspect/plugin-gio.xml:
* docs/plugins/inspect/plugin-libvisual.xml:
* docs/plugins/inspect/plugin-ogg.xml:
* docs/plugins/inspect/plugin-opengl.xml:
* docs/plugins/inspect/plugin-opus.xml:
* docs/plugins/inspect/plugin-pango.xml:
* docs/plugins/inspect/plugin-pbtypes.xml:
* docs/plugins/inspect/plugin-playback.xml:
* docs/plugins/inspect/plugin-rawparse.xml:
* docs/plugins/inspect/plugin-subparse.xml:
* docs/plugins/inspect/plugin-tcp.xml:
* docs/plugins/inspect/plugin-theora.xml:
* docs/plugins/inspect/plugin-typefindfunctions.xml:
* docs/plugins/inspect/plugin-videoconvert.xml:
* docs/plugins/inspect/plugin-videorate.xml:
* docs/plugins/inspect/plugin-videoscale.xml:
* docs/plugins/inspect/plugin-videotestsrc.xml:
* docs/plugins/inspect/plugin-volume.xml:
* docs/plugins/inspect/plugin-vorbis.xml:
* docs/plugins/inspect/plugin-ximagesink.xml:
* docs/plugins/inspect/plugin-xvimagesink.xml:
  Update docs

2018-07-18 19:53:26 +0900  Seungha Yang 

* sys/xvimage/xvimage.c:
* sys/xvimage/xvimagesink.c:
  xvimage: Fix symbol redefine build error
  https://bugzilla.gnome.org/show_bug.cgi?id=796827

2018-02-27 23:22:24 +1100  Jan Schmidt 

* ext/gl/gstglcolorbalance.c:
  glcolorbalance: Support OES textures for input/passthrough
  glcolorbalance is in the default GL path inside glimagesink,
  so has been causing an possibly-unnecessary extra texture copy
  on Android for a while now. If we're just doing passthrough,
  we can support OES directly. If not, they'll be transformed
  to 2D textures and colourbalanced.

2018-07-18 14:56:28 +0100  Tim-Philipp Müller 

* gst-libs/gst/gl/meson.build:
  meson: gl: fix backported patch

2018-07-16 16:01:34 +0200  Iñigo Huguet 

* gst-libs/gst/gl/egl/gsteglimage.c:
* gst-libs/gst/gl/gstglconfig.h.meson:
* gst-libs/gst/gl/meson.build:
* m4/gst-gl.m4:
  gl/build: fixed failed compilation due to missing EGLuint64KHR typedef
  [Matthew Waters]: add meson differences
  https://bugzilla.gnome.org/show_bug.cgi?id=796820

2018-07-06 10:51:37 +0300  Sebastian Dröge 

* gst/playback/gstparsebin.c:
  parsebin: Don't try to continue autoplugging a parser if we got raw 
caps
  Otherwise we'll fail with an unhelpful message that no decoder for the
  raw caps was found if a container contains raw media.

2018-06-28 20:13:04 +0800  wangzq 

* gst-libs/gst/audio/gstaudiobasesrc.c:
  audiobasesrc: Round down segsize to an integer number of samples
  https://bugzilla.gnome.org/show_bug.cgi?id=796704

2018-06-13 15:29:46 +0300  Vivia Nikolaidou 

* tools/gst-discoverer.c:
  discoverer: Don't crash when running with -v if channel-mask==0 and 
>7 channels
  For e.g. 16-channel audio, if the channel mask is 0 (which it usually
  is), gst_audio_channel_positions_from_mask would get confused,
  ultimately leading into a crash.
  https://bugzilla.gnome.org/show_bug.cgi?id=796578

2018-06-04 22:32:03 +1000  Matthew Waters 

* ext/gl/gstgldownloadelement.c:
  gldownloadelement: fix build with msvc
  msvc doesn't like #ifdef inside macro expansion

2018-05-30 11:39:40 +0200  Edward Hervey 

* gst/subparse/gstsubparse.c:
  subparse: Don't read beyond array
  If num_open_tags is 0, we shouldn't try to get the "last" open tag
  since there isn't any.

2018-05-30 10:56:12 +0200  Edward Hervey 

* ext/ogg/gstoggstream.c:
  ogg: Avoid undefined granule shift
  A granule is a 64bit signed integer, shifting by 63 or more is
  undefined and most likely an indication that the stream is
  corrupted or invalid.
  Detected by oss-fuzz

2014-04-18 10:37:57 +0200

gst-plugins-good 1.14.2

[Tim-Philipp Müller]

ChangeLog
=

2018-07-20 00:54:57 +0100  Tim-Philipp Müller 

* ChangeLog:
* NEWS:
* RELEASE:
* configure.ac:
* gst-plugins-good.doap:
* meson.build:
  Release 1.14.2

2018-07-20 00:54:56 +0100  Tim-Philipp Müller 

* 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-gtk.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-lame.xml:
* docs/plugins/inspect/plugin-level.xml:
* docs/plugins/inspect/plugin-matroska.xml:
* docs/plugins/inspect/plugin-mpg123.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-qmlgl.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-shout2.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-twolame.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:
  Update docs

2018-07-17 08:23:54 +0200  Iñigo Huguet 

* ext/qt/gstqtgl.h:
  qmlgl: Fix conflicting declaration of type GLsync for non-android
  https://bugzilla.gnome.org/show_bug.cgi?id=796821

2018-06-20 10:03:59 +0200  Edward Hervey 

* ext/soup/gstsouphttpsrc.c:
  souphttpsrc: Protect input stream with lock
  This was the last remaining place where modifying/unreffing the
  input stream was not protected by the lock
  https://bugzilla.gnome.org/show_bug.cgi?id=796639

2018-06-18 12:13:48 +0300  Sebastian Dröge 

* gst/multifile/gstsplitmuxsrc.c:
  splitmuxsrc: Make sure events are writable before setting their seqnum

2018-07-13 16:51:24 -0400  Nicolas Dufresne 

* sys/v4l2/gstv4l2bufferpool.c:
  v4l2bufferpool: Validate that capture buffers were queued
  When the pool is started, we allocate and release buffer, expecting
  the pool release-buffer handler to queue them. Though, as we rely
  on release function

gst-plugins-good 1.12.5

[Tim-Philipp Müller]

ChangeLog
=

2018-03-28 14:22:42 +0100  Tim-Philipp Müller 

* ChangeLog:
* NEWS:
* RELEASE:
* configure.ac:
* gst-plugins-good.doap:
* meson.build:
  Release 1.12.5

2018-03-28 14:22:42 +0100  Tim-Philipp Müller 

* docs/plugins/gst-plugins-good-plugins.args:
* docs/plugins/gst-plugins-good-plugins.hierarchy:
* 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-shout2.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:
  Update docs

2018-02-28 19:19:10 +0200  Sebastian Dröge 

* gst/matroska/matroska-mux.c:
  matroskamux: Clip maximum cluster duration to the maximum possible 
value
  Only up to timescale * G_MAXINT16 is possible as cluster duration, 
which
  is already higher than our default value. Using higher values would
  cause overflows and broken files.
  Based on the investigation by Nicola Murino 
  https://bugzilla.gnome.org/show_bug.cgi?id=792775

2018-02-28 19:21:53 +0200  Sebastian Dröge 

* gst/matroska/matroska-mux.c:
  matroskamux: Only mark new clusters as keyframe if they start on a 
keyframe or we're muxing only audio
  Based on a patch by Nicola Murino 
  https://bugzilla.gnome.org/show_bug.cgi?id=792775

2018-03-13 16:03:50 +0200  Sebastian Dröge 

* gst/rtsp/gstrtspsrc.c:
  Revert "rtspsrc: Ignore medias marked as sendonly"
  This reverts commit 86a56cc48c521d4fbd4c73c903a58787313458d4.

2018-03-13 16:03:02 +0200  Sebastian Dröge 

* gst/rtsp/gstrtspsrc.c:
  Revert "rtspsrc: Fix up sendonly/recvonly attribute handling"
  This reverts commit 326e9549e378bcc

gst-plugins-base 1.12.5

[Tim-Philipp Müller]

ChangeLog
=

2018-03-28 14:19:50 +0100  Tim-Philipp Müller 

* ChangeLog:
* NEWS:
* RELEASE:
* configure.ac:
* gst-plugins-base.doap:
* meson.build:
  Release 1.12.5

2018-03-28 14:19:50 +0100  Tim-Philipp Müller 

* docs/plugins/inspect/plugin-adder.xml:
* docs/plugins/inspect/plugin-alsa.xml:
* docs/plugins/inspect/plugin-app.xml:
* docs/plugins/inspect/plugin-audioconvert.xml:
* docs/plugins/inspect/plugin-audiorate.xml:
* docs/plugins/inspect/plugin-audioresample.xml:
* docs/plugins/inspect/plugin-audiotestsrc.xml:
* docs/plugins/inspect/plugin-cdparanoia.xml:
* docs/plugins/inspect/plugin-encoding.xml:
* docs/plugins/inspect/plugin-gio.xml:
* docs/plugins/inspect/plugin-libvisual.xml:
* docs/plugins/inspect/plugin-ogg.xml:
* docs/plugins/inspect/plugin-opus.xml:
* docs/plugins/inspect/plugin-pango.xml:
* docs/plugins/inspect/plugin-pbtypes.xml:
* docs/plugins/inspect/plugin-playback.xml:
* docs/plugins/inspect/plugin-rawparse.xml:
* docs/plugins/inspect/plugin-subparse.xml:
* docs/plugins/inspect/plugin-tcp.xml:
* docs/plugins/inspect/plugin-theora.xml:
* docs/plugins/inspect/plugin-typefindfunctions.xml:
* docs/plugins/inspect/plugin-videoconvert.xml:
* docs/plugins/inspect/plugin-videorate.xml:
* docs/plugins/inspect/plugin-videoscale.xml:
* docs/plugins/inspect/plugin-videotestsrc.xml:
* docs/plugins/inspect/plugin-volume.xml:
* docs/plugins/inspect/plugin-vorbis.xml:
* docs/plugins/inspect/plugin-ximagesink.xml:
* docs/plugins/inspect/plugin-xvimagesink.xml:
  Update docs

2018-03-17 06:33:38 +0100  Edward Hervey 

* ext/ogg/gstoggstream.c:
  oggstream: protect against out-of-bounds read
  We need at least 17 bytes of data for a valid flac header
  oss-fuzz #6974

2018-02-28 23:12:39 -0500  Nicolas Dufresne 

* gst-libs/gst/video/gstvideodecoder.c:
  videodecoder: Reset QoS time after pushing segment
  This fixes playbin gapless playback. An ancient QoS time was used and
  would lead to all frames being dropped.
  https://bugzilla.gnome.org/show_bug.cgi?id=668995

2018-02-12 16:26:01 +0100  Edward Hervey 

* gst-libs/gst/tag/id3v2.c:
  id3v2: re-fix handling of v2.4 extended headers
  The various id3v2 specs handle the extended header sizes differently
  (because hey, it wouldn't be fun otherwise).
  http://id3.org/id3v2.3.0 states:
  "Where the 'Extended header size', currently 6 or 10 bytes, excludes
  itself."
  http://id3.org/id3v2.4.0-structure states:
  Extended header size   4 * %0xxx
  Number of flag bytes   $01
  Extended Flags $xx
  Where the 'Extended header size' is the size of the whole extended
  header, stored as a 32 bit synchsafe integer. An extended header can
  thus never have a size of fewer than six bytes.
  So in id3v2.4.0 it's the *whole* extended header size (a-la ISOBMFF
  atom), whereas in id3v2.3.0 it's the extended header size *excluding*
  those 4 initial bytes.
  And for other versions, god knows..
  Fixes regression introduced in commit da607005.
  https://bugzilla.gnome.org/show_bug.cgi?id=792983

2018-02-15 01:14:52 +0100  Mathieu Duponchelle 

* gst-libs/gst/audio/gstaudiopack.orc:
  gstaudiopack.orc: pack_u32be_swap: actually swap
  Fixes:
  gst-launch-1.0 audiotestsrc ! audio/x-raw, format=U32BE ! \
  audioconvert ! autoaudiosink

2018-02-14 14:37:52 -0500  Nicolas Dufresne 

* docs/libs/gst-plugins-base-libs-docs.sgml:
  doc: Add per version newly added API indexes

2018-02-14 14:11:47 -0500  Nicolas Dufresne 

* gst-libs/gst/allocators/gstfdmemory.h:
* gst-libs/gst/video/video-color.h:
  doc: Remove extra . after Since marker

2018-02-14 14:16:14 -0500  Nicolas Dufresne 

* gst-libs/gst/allocators/gstdmabuf.h:
  doc: Fix since marker in dmabuf to match a stable release

2018-02-14 14:39:32 -0500  Nicolas Dufresne 

* gst/playback/gsturidecodebin.c:
  doc: Remove obsolete Since 0.10.X marks

2018-02-13 08:36:30 +0100  Edward Hervey 

* ext/vorbis/gstvorbisdec.c:
* ext/vorbis/gstvorbisdec.h:
  vorbisdec: Improve "new headers while initialized" handling
  If new headers arrive after we are initialized, we need to make
  sure that they are indeed valid.
  A vorbis bitstream always begins with three header packets and must
  be in order.
  Also some streams have unframed (invalid?) header

gstreamer 1.12.5

[Tim-Philipp Müller]

ChangeLog
=

2018-03-28 14:16:40 +0100  Tim-Philipp Müller 

* ChangeLog:
* NEWS:
* RELEASE:
* configure.ac:
* gstreamer.doap:
* meson.build:
  Release 1.12.5

2018-03-28 14:16:40 +0100  Tim-Philipp Müller 

* docs/plugins/inspect/plugin-coreelements.xml:
* docs/plugins/inspect/plugin-coretracers.xml:
  Update docs

2017-11-20 17:01:04 +0100  Edward Hervey 

* gst/gstregistrychunks.c:
  registrychunks: Make sure we use aligned memory
  This is in the same vein as for all other features. Some systems
  might not allow unaligned read.

2018-02-14 19:12:06 +  Tim-Philipp Müller 

* docs/libs/gstreamer-libs-docs.sgml:
  docs: add index for new symbols in 1.12

2018-02-14 19:37:35 +  Tim-Philipp Müller 

* docs/libs/gstreamer-libs-sections.txt:
  docs: add flow combiner ref/unref to docs
  So new-in-1.12 index actually has some entries.

2017-10-10 15:53:38 +0200  fengalin 

* libs/gst/base/gstflowcombiner.c:
  flowcombiner: Fix version for ref and unref functions
  The functions were introduced in version 1.12.1, GstFlowCombiner was
  introduced in 1.4.
  https://bugzilla.gnome.org/show_bug.cgi?id=788778

2018-01-20 15:30:53 +  Tim-Philipp Müller 

* gst/gstchildproxy.c:
  childproxy: gracefully handle methods being NULL
  Do this for all method invoke functions for consistency.
  https://bugzilla.gnome.org/show_bug.cgi?id=750154

2017-08-01 10:43:32 +0200  Michele Dionisio 

* libs/gst/net/gstptpclock.c:
  ptp: fix build failure with #undef USE_MEASUREMENT_FILTERING
  "Label ‘out’ used but not defined", since it's also used by
  the USE_MEDIAN_PRE_FILTERING branch.
  https://bugzilla.gnome.org/show_bug.cgi?id=785631

2018-01-11 17:10:45 +0000  Tim-Philipp Müller 

* gst/gstplugin.c:
  plugin: plugin_load() must return a ref even if it was loaded already
  Fix refcounting issue when plugin was loaded already.
  gst_plugin_load() is supposed to return a ref, so it
  must always return a ref.
  This also fixes the gstplugin unit test on windows where
  fork is not available and where test_load_coreelements()
  would unref a plugin ref it didn't get and then mess up
  the internal registry plugin list state for the next test,
  in case where the test registry does not exist yet.

2018-01-11 11:36:53 +  Tim-Philipp Müller 

* gst/gstinfo.c:
  info: reset default threshold to LEVEL_DEFAULT not 0
  in set_threshold_from_string().

2017-12-26 18:08:31 +0100  Sebastian Dröge 

* gst/gststreams.h:
  streams: GstStreamType/GstStream are available since 1.10
  Annotate them as such.

2017-12-21 13:47:52 +0200  Sebastian Dröge 

* plugins/elements/gstdownloadbuffer.c:
  downloadbuffer: Don't hold the mutex while posint the 
download-complete message
  Something might handle it from a sync message handler and call back 
into
  downloadbuffer, causing a deadlock.

2017-12-20 18:56:23 +0200  Sebastian Dröge 

* gst/gstsystemclock.c:
  systemclock: set_default() clock parameter can be NULL

2017-12-20 18:11:48 +0200  Sebastian Dröge 

* gst/gstelement.c:
  element: Annotate set_clock() clock parameter with allow-none

2017-12-20 18:09:28 +0200  Sebastian Dröge 

* gst/gstelement.c:
  element: Annotate set_bus() bus parameter as allow-none
  It's possible to replace the bus with NULL/None

2017-12-17 14:18:38 +0200  Sebastian Dröge 

* gst/gstplugin.c:
  plugin: Annotate add_dependency() arguments as NULL-terminated arrays

2017-12-06 13:40:46 +0200  Sebastian Dröge 

* gst/gstevent.c:
* gst/gstmessage.c:
* gst/gstquery.c:
  event/query/message: Annotate get_structure() return value as nullable

2017-12-03 14:48:22 +0200  Sebastian Dröge 

* gst/gstdebugutils.c:
* gst/gstplugin.c:
* gst/gstregistry.c:
  gst: Annotate various strings as type filename if they represent a 
path/filename

2018-01-16 10:17:58 +0100  Edward Hervey 

* gst/gstpad.c:
  gstpad: Avoid stream-dead-lock on deactivation
  The following case can happen when two thread try to activate and
  deactivate a pad at the same time:
  T1: starts to deactivate, calls pre_activate(), sets in_activation
  to TRUE and carries on
  T2: starts to activate, calls pre_activate(), in_activation is TRUE
  so it waits on the GCond
  T1: calls post_activate(), tries to acquire the streaming lock ..
  but can't because T2 is currently holding it
  With this patch, the deadlock

gst-plugins-base 1.14.0

[Tim-Philipp Müller]

News


See https://gstreamer.freedesktop.org/releases/1.14/ for the latest
version of this document.

_Last updated: Monday 19 March 2018, 12:00 UTC (log)_


Introduction

The GStreamer team is proud to announce a new major feature release in
the stable 1.x API series of your favourite cross-platform multimedia
framework!

As always, this release is again packed with new features, bug fixes and
other improvements.


ChangeLog
=
https://download.gnome.org/sources/gst-plugins-base/1.14/gst-plugins-base-1.14.0.changes
  (2.65K)

Download

https://download.gnome.org/sources/gst-plugins-base/1.14/gst-plugins-base-1.14.0.tar.xz
 (3.52M)
  sha256sum: 7e904660ff56e02b036cf7fdfb77a50a540828ca9d2614d69ba931772e5b6940

___
release-team@gnome.org
https://mail.gnome.org/mailman/listinfo/release-team
Release-team lurker? Do NOT participate in discussions.

gst-plugins-good 1.14.0

[Tim-Philipp Müller]

News


See https://gstreamer.freedesktop.org/releases/1.14/ for the latest
version of this document.

_Last updated: Monday 19 March 2018, 12:00 UTC (log)_


Introduction

The GStreamer team is proud to announce a new major feature release in
the stable 1.x API series of your favourite cross-platform multimedia
framework!

As always, this release is again packed with new features, bug fixes and
other improvements.


ChangeLog
=
https://download.gnome.org/sources/gst-plugins-good/1.14/gst-plugins-good-1.14.0.changes
  (4.93K)

Download

https://download.gnome.org/sources/gst-plugins-good/1.14/gst-plugins-good-1.14.0.tar.xz
 (3.59M)
  sha256sum: 6afa35747d528d3ab4ed8f5eac13f7235d7d28100d6a24dd78f81ec7c0d04688

___
release-team@gnome.org
https://mail.gnome.org/mailman/listinfo/release-team
Release-team lurker? Do NOT participate in discussions.

gstreamer 1.14.0

[Tim-Philipp Müller]

News


See https://gstreamer.freedesktop.org/releases/1.14/ for the latest
version of this document.

_Last updated: Monday 19 March 2018, 12:00 UTC (log)_


Introduction

The GStreamer team is proud to announce a new major feature release in
the stable 1.x API series of your favourite cross-platform multimedia
framework!

As always, this release is again packed with new features, bug fixes and
other improvements.


ChangeLog
=
https://download.gnome.org/sources/gstreamer/1.14/gstreamer-1.14.0.changes  
(383)

Download

https://download.gnome.org/sources/gstreamer/1.14/gstreamer-1.14.0.tar.xz 
(3.10M)
  sha256sum: fc361367f0d4b780a868a8833f9f30b9c9f4ac9faea4e6b251db8b4b0398466e

___
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Release-team lurker? Do NOT participate in discussions.

gst-plugins-good 1.13.91

[Tim-Philipp Müller]

ChangeLog
=

2018-03-13 19:16:42 +  Tim-Philipp Müller 

* NEWS:
* RELEASE:
* configure.ac:
* gst-plugins-good.doap:
* meson.build:
  Release 1.13.91

2018-03-13 19:16:42 +  Tim-Philipp Müller 

* 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-gtk.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-lame.xml:
* docs/plugins/inspect/plugin-level.xml:
* docs/plugins/inspect/plugin-matroska.xml:
* docs/plugins/inspect/plugin-mpg123.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-qmlgl.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-shout2.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-twolame.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:
  Update docs

2018-03-12 13:21:08 +  Tim-Philipp Müller 

* gst/rtpmanager/gstrtpbin.c:
  docs: rtpbin: add some Since markers for new properties

2018-03-10 18:57:38 +0530  Nirbheek Chauhan 

* sys/directsound/meson.build:
  meson: Add deviceprovider changes to directsoundsink
  These were missed when they were added to Makefile.am

2018-03-08 10:12:16 +0100  Michael Tretter 

* configure.ac:
  configure.ac: enable largefile support if possible
  https://bugzilla.gnome.org/show_bug.cgi?id=793103

2018-03-07 14:16:02 -0500  Nicolas Dufresne 

* sys/v4l2/gstv4l2object.c:
* sys/v4l2/gstv4l2object.h:
  v4l2: Fix support for 32bit mmap
  https://bugzilla.gnome.org/show_bug.cgi?id=793103



Download

https://download.gnome.org/sources/gst-plugins-good/1.13/gst-plugins-good-1.13.91.tar.xz
 (3.59M)
  sha256sum: 9586e4c4bcf0faf0374655c43bd3de292450ca8dac2029f5a456829d260fddaf

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gst-plugins-base 1.13.91

[Tim-Philipp Müller]

ChangeLog
=

2018-03-13 19:13:18 +  Tim-Philipp Müller 

* NEWS:
* RELEASE:
* configure.ac:
* gst-plugins-base.doap:
* meson.build:
  Release 1.13.91

2018-03-13 19:13:18 +  Tim-Philipp Müller 

* docs/plugins/inspect/plugin-adder.xml:
* docs/plugins/inspect/plugin-alsa.xml:
* docs/plugins/inspect/plugin-app.xml:
* docs/plugins/inspect/plugin-audioconvert.xml:
* docs/plugins/inspect/plugin-audiomixer.xml:
* docs/plugins/inspect/plugin-audiorate.xml:
* docs/plugins/inspect/plugin-audioresample.xml:
* docs/plugins/inspect/plugin-audiotestsrc.xml:
* docs/plugins/inspect/plugin-cdparanoia.xml:
* docs/plugins/inspect/plugin-encoding.xml:
* docs/plugins/inspect/plugin-gio.xml:
* docs/plugins/inspect/plugin-libvisual.xml:
* docs/plugins/inspect/plugin-ogg.xml:
* docs/plugins/inspect/plugin-opengl.xml:
* docs/plugins/inspect/plugin-opus.xml:
* docs/plugins/inspect/plugin-pango.xml:
* docs/plugins/inspect/plugin-pbtypes.xml:
* docs/plugins/inspect/plugin-playback.xml:
* docs/plugins/inspect/plugin-rawparse.xml:
* docs/plugins/inspect/plugin-subparse.xml:
* docs/plugins/inspect/plugin-tcp.xml:
* docs/plugins/inspect/plugin-theora.xml:
* docs/plugins/inspect/plugin-typefindfunctions.xml:
* docs/plugins/inspect/plugin-videoconvert.xml:
* docs/plugins/inspect/plugin-videorate.xml:
* docs/plugins/inspect/plugin-videoscale.xml:
* docs/plugins/inspect/plugin-videotestsrc.xml:
* docs/plugins/inspect/plugin-volume.xml:
* docs/plugins/inspect/plugin-vorbis.xml:
* docs/plugins/inspect/plugin-ximagesink.xml:
* docs/plugins/inspect/plugin-xvimagesink.xml:
  Update docs

2018-03-13 13:49:42 +  Tim-Philipp Müller 

* gst-libs/gst/audio/meson.build:
  meson: install new audio-prelude.h

2018-03-13 11:56:59 +  Tim-Philipp Müller 

* docs/libs/Makefile.am:
* docs/libs/meson.build:
  docs: update api decorators to ignore

2018-03-13 11:48:31 +  Tim-Philipp Müller 

* gst-libs/gst/video/Makefile.am:
* gst-libs/gst/video/colorbalance.h:
* gst-libs/gst/video/colorbalancechannel.h:
* gst-libs/gst/video/gstvideoaffinetransformationmeta.h:
* gst-libs/gst/video/gstvideodecoder.h:
* gst-libs/gst/video/gstvideoencoder.h:
* gst-libs/gst/video/gstvideofilter.h:
* gst-libs/gst/video/gstvideometa.h:
* gst-libs/gst/video/gstvideopool.h:
* gst-libs/gst/video/gstvideosink.h:
* gst-libs/gst/video/gstvideotimecode.h:
* gst-libs/gst/video/gstvideoutils.h:
* gst-libs/gst/video/meson.build:
* gst-libs/gst/video/navigation.h:
* gst-libs/gst/video/video-blend.h:
* gst-libs/gst/video/video-chroma.h:
* gst-libs/gst/video/video-color.h:
* gst-libs/gst/video/video-converter.h:
* gst-libs/gst/video/video-dither.h:
* gst-libs/gst/video/video-event.h:
* gst-libs/gst/video/video-format.h:
* gst-libs/gst/video/video-frame.h:
* gst-libs/gst/video/video-info.h:
* gst-libs/gst/video/video-multiview.h:
* gst-libs/gst/video/video-overlay-composition.h:
* gst-libs/gst/video/video-prelude.h:
* gst-libs/gst/video/video-resampler.h:
* gst-libs/gst/video/video-scaler.h:
* gst-libs/gst/video/video-tile.h:
* gst-libs/gst/video/video.h:
* gst-libs/gst/video/video_mkenum.py:
* gst-libs/gst/video/videodirection.h:
* gst-libs/gst/video/videoorientation.h:
* gst-libs/gst/video/videooverlay.h:
  video: GST_EXPORT -> GST_VIDEO_API
  We need different export decorators for the different libs.
  For now no actual change though, just rename before the release,
  and add prelude headers to define the new decorator to GST_EXPORT.

2018-03-13 11:32:02 +0000  Tim-Philipp Müller 

* gst-libs/gst/tag/Makefile.am:
* gst-libs/gst/tag/gsttagdemux.h:
* gst-libs/gst/tag/gsttagmux.h:
* gst-libs/gst/tag/id3v2.h:
* gst-libs/gst/tag/meson.build:
* gst-libs/gst/tag/tag-prelude.h:
* gst-libs/gst/tag/tag.h:
* gst-libs/gst/tag/tag_mkenum.py:
* gst-libs/gst/tag/xmpwriter.h:
  tag: GST_EXPORT -> GST_TAG_API
  We need different export decorators for the different libs.
  For now no actual change though, just rename before the release,
  and add prelude headers to define the new decorator to GST_EXPORT.

2018-03-13 11:27:11 +0000  Tim-Philipp Müller 

* gst-libs/gst/sdp/Makefile.am:
* gst-libs/gst/sdp/gstmikey.h:
* gst-libs/gst/sdp/gstsdpmessage.h:
* gst-libs/gst/sdp/meson.build:
* gst-libs/gst/s

gstreamer 1.13.91

[Tim-Philipp Müller]

ChangeLog
=

2018-03-13 19:08:54 +  Tim-Philipp Müller 

* NEWS:
* RELEASE:
* configure.ac:
* gstreamer.doap:
* meson.build:
  Release 1.13.91

2018-03-13 19:08:54 +  Tim-Philipp Müller 

* docs/plugins/inspect/plugin-coreelements.xml:
* docs/plugins/inspect/plugin-coretracers.xml:
  Update docs

2018-03-13 11:54:42 +  Tim-Philipp Müller 

* docs/gst/meson.build:
* docs/libs/meson.build:
  meson: docs: update api decorators to ignore

2018-03-12 23:12:13 +  Tim-Philipp Müller 

* docs/libs/Makefile.am:
  docs: fixup for new libs API export decorators

2018-03-12 23:03:26 +  Tim-Philipp Müller 

* libs/gst/net/Makefile.am:
* libs/gst/net/gstnet.h:
* libs/gst/net/gstnetaddressmeta.h:
* libs/gst/net/gstnetclientclock.h:
* libs/gst/net/gstnetcontrolmessagemeta.h:
* libs/gst/net/gstnettimepacket.h:
* libs/gst/net/gstnettimeprovider.h:
* libs/gst/net/gstptpclock.h:
* libs/gst/net/meson.build:
* libs/gst/net/net-prelude.h:
* libs/gst/net/net.h:
  net: GST_EXPORT -> GST_NET_API
  We need different export decorators for the different libs.
  For now no actual change though, just rename before the release,
  and add prelude headers to define the new decorator to GST_EXPORT.

2018-03-12 23:03:26 +0000  Tim-Philipp Müller 

* common:
* libs/gst/controller/Makefile.am:
* libs/gst/controller/controller-prelude.h:
* libs/gst/controller/controller.h:
* libs/gst/controller/controller_mkenum.py:
* libs/gst/controller/gstargbcontrolbinding.h:
* libs/gst/controller/gstdirectcontrolbinding.h:
* libs/gst/controller/gstinterpolationcontrolsource.h:
* libs/gst/controller/gstlfocontrolsource.h:
* libs/gst/controller/gstproxycontrolbinding.h:
* libs/gst/controller/gsttimedvaluecontrolsource.h:
* libs/gst/controller/gsttriggercontrolsource.h:
* libs/gst/controller/meson.build:
  controller: GST_EXPORT -> GST_CONTROLLER_API
  We need different export decorators for the different libs.
  For now no actual change though, just rename before the release,
  and add prelude headers to define the new decorator to GST_EXPORT.

2018-03-12 23:03:26 +0000  Tim-Philipp Müller 

* libs/gst/check/Makefile.am:
* libs/gst/check/check-prelude.h:
* libs/gst/check/check.h:
* libs/gst/check/gstbufferstraw.h:
* libs/gst/check/gstcheck.h:
* libs/gst/check/gstconsistencychecker.h:
* libs/gst/check/gstharness.h:
* libs/gst/check/gsttestclock.h:
* libs/gst/check/meson.build:
  check: GST_EXPORT -> GST_CHECK_API
  We need different export decorators for the different libs.
  For now no actual change though, just rename before the release,
  and add prelude headers to define the new decorator to GST_EXPORT.

2018-03-12 23:03:26 +0000  Tim-Philipp Müller 

* libs/gst/base/Makefile.am:
* libs/gst/base/base-prelude.h:
* libs/gst/base/base.h:
* libs/gst/base/gstadapter.h:
* libs/gst/base/gstaggregator.h:
* libs/gst/base/gstbaseparse.h:
* libs/gst/base/gstbasesink.h:
* libs/gst/base/gstbasesrc.h:
* libs/gst/base/gstbasetransform.h:
* libs/gst/base/gstbitreader.h:
* libs/gst/base/gstbytereader.h:
* libs/gst/base/gstbytewriter.h:
* libs/gst/base/gstcollectpads.h:
* libs/gst/base/gstdataqueue.h:
* libs/gst/base/gstflowcombiner.h:
* libs/gst/base/gstindex.h:
* libs/gst/base/gstpushsrc.h:
* libs/gst/base/gstqueuearray.h:
* libs/gst/base/gsttypefindhelper.h:
* libs/gst/base/meson.build:
  base: GST_EXPORT -> GST_BASE_API
  We need different export decorators for the different libs.
  For now no actual change though, just rename before the release,
  and add prelude headers to define the new decorator to GST_EXPORT.

2018-03-12 23:03:26 +  Tim-Philipp Müller 

* docs/gst/Makefile.am:
* gst/Makefile.am:
* gst/gst.h:
* gst/gst_private.h:
* gst/gstallocator.h:
* gst/gstatomicqueue.h:
* gst/gstbin.h:
* gst/gstbuffer.h:
* gst/gstbufferlist.h:
* gst/gstbufferpool.h:
* gst/gstbus.h:
* gst/gstcaps.h:
* gst/gstcapsfeatures.h:
* gst/gstchildproxy.h:
* gst/gstclock.h:
* gst/gstconfig.h.in:
* gst/gstcontext.h:
* gst/gstcontrolbinding.h:
* gst/gstcontrolsource.h:
* gst/gstdatetime.h:
* gst/gstdebugutils.h:
* gst/gstdevice.h:
* gst/gstdevicemonitor.h:
* gst/gstdeviceprovider.h:
 

gst-plugins-good 1.13.90

[Tim-Philipp Müller]

News


GSTREAMER 1.14 RELEASE NOTES


ChangeLog
=
https://download.gnome.org/sources/gst-plugins-good/1.13/gst-plugins-good-1.13.90.changes
  (181K)

Download

https://download.gnome.org/sources/gst-plugins-good/1.13/gst-plugins-good-1.13.90.tar.xz
 (3.58M)
  sha256sum: f116f8d078c4494f391b382cdcf15a1433bd896e3546c01d104c522ad400c170

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gstreamer 1.13.90

[Tim-Philipp Müller]

News


GSTREAMER 1.14 RELEASE NOTES


ChangeLog
=
https://download.gnome.org/sources/gstreamer/1.13/gstreamer-1.13.90.changes  
(174K)

Download

https://download.gnome.org/sources/gstreamer/1.13/gstreamer-1.13.90.tar.xz 
(3.08M)
  sha256sum: 4ffbd96a3ea1844212757be5609bbd16667f83c21db79a91f9da589c833258b0

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gst-plugins-base 1.13.90

[Tim-Philipp Müller]

News


GSTREAMER 1.14 RELEASE NOTES


ChangeLog
=
https://download.gnome.org/sources/gst-plugins-base/1.13/gst-plugins-base-1.13.90.changes
  (116)

Download

https://download.gnome.org/sources/gst-plugins-base/1.13/gst-plugins-base-1.13.90.tar.xz
 (3.50M)
  sha256sum: 6dfac30152385489bf52e161efd7dc6c581913975ca2c0fd1582edcc2670904b

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gstreamer 1.13.1

[Tim-Philipp Müller]

ChangeLog
=

=== release 1.12.0 ===

2017-05-04  Sebastian Dröge 


Download

https://download.gnome.org/sources/gstreamer/1.13/gstreamer-1.13.1.tar.xz 
(3.05M)
  sha256sum: 7d2dc896d533cf89f08ce41e3ae3914b1ef9b463ae16c3ef689367e114495804

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gst-plugins-good 1.13.1

[Tim-Philipp Müller]

ChangeLog
=

=== release 1.12.0 ===

2017-05-04  Sebastian Dröge 


Download

https://download.gnome.org/sources/gst-plugins-good/1.13/gst-plugins-good-1.13.1.tar.xz
 (3.42M)
  sha256sum: 136cc29d032e3f15035aa5e905cc85cd7882b40a68bf3f7cc9cb5e8cb8b14154

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gst-plugins-base 1.13.1

[Tim-Philipp Müller]

ChangeLog
=

=== release 1.12.0 ===

2017-05-04  Sebastian Dröge 


Download

https://download.gnome.org/sources/gst-plugins-base/1.13/gst-plugins-base-1.13.1.tar.xz
 (3.35M)
  sha256sum: 51cd6cb976d5f6b1cd46b800352cf11fca7af3d971e23fe38e1773505b29e21d

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gstreamer 1.12.4

[Tim-Philipp Müller]

ChangeLog
=

2017-12-07  Sebastian Dröge 

* configure.ac:
  releasing 1.12.4

2017-12-07 18:56:12 +0200  Sebastian Dröge 

* po/nl.po:
  po: Update translations

2017-12-06 17:07:29 +0100  Edward Hervey 

* gst/gstpad.c:
  gstpad: Handle GST_PAD_PROBE_HANDLED on sticky event push
  When actually pushing an event, if we get GST_FLOW_CUSTOM_SUCCESS_1
  (which is the conversion of GST_PAD_PROBE_HANDLED return value),
  don't consider the stick event push as ignored, but as handled

2017-09-04 12:20:43 +0200  Miguel París 

* tests/check/gst/gstpad.c:
  pad: add test to check handled and drop probes
  https://bugzilla.gnome.org/show_bug.cgi?id=787243

2017-09-04 14:33:29 +0200  Edward Hervey 

* gst/gstpad.c:
  pad: Don't call remaining probes after they return DROPPED|HANDLED
  If multiple probes are set on a pad and one probe returns either
  GST_PAD_PROBE_HANDLED or GST_PAD_PROBE_DROPPED we need to stop
  calling the remaining probes.
  https://bugzilla.gnome.org/show_bug.cgi?id=787243

2017-10-02 17:59:17 +0200  Edward Hervey 

* gst/gstbin.c:
* tests/check/gst/gstbin.c:
  bin: iterate_sorted: Ensure sources are always returned last
  For linked elements, the resulting gst_bin_iterate_sorted() will
  properly return elements from sink to sources.
  If we have some elements that are not linked, we *still* want to
  ensure that we return:
  * In priority any sinks
  * Last of all any sources
  * And in between any element which is neither source nor sink
  For this to work, when looking for the next candidate element,
  not only check the degree order, but if there are two candidates
  with the same degree order, prefer the non-source one.
  Amongst other things, this fixes the case where we activating a
  bin containing unlinked sources and other elements. Without this
  we could end up activating sources (which might start adding pads
  to be linked) before other (to which those new source element pads
  might be linked) are not activated
  https://bugzilla.gnome.org/show_bug.cgi?id=788434

2017-12-06 13:36:30 +0200  Sebastian Dröge 

* gst/gstquery.c:
  query: Add an empty structure in writable_structure() if there is 
none yet
  This is consistent with how it works for GstEvent already.

2017-12-02 12:29:20 +  Tim-Philipp Müller 

* gst/gstinfo.c:
  info: always check match patterns for new debug categories
  Not only if a match pattern was set originally via GST_DEBUG.
  Patterns might be set programmatically as well after all.

2017-12-02 12:22:47 +  Tim-Philipp Müller 

* gst/gstinfo.c:
  info: fix performance issue with registering categories after 
gst_init()
  When registering a new debug category after gst_init(), simply check
  the existing patterns against that new category.
  No need to iterate over all categories and recheck them all against
  the existing patterns.
  Also, no need to re-parse the existing pattern string set via 
GST_DEBUG
  and add the same set of match patterns all over again to the existing
  list of match patterns every time we register a new debug category.
  Combined with iterating all debug categories on a change this would
  make adding debug categories after gst_init() very very very slow.

2017-11-25 13:07:12 +0100  Edward Hervey 

* gst/gstinfo.c:
  gstinfo: Use free instead of g_free
  Because

2017-11-25 12:44:11 +0100  Edward Hervey 

* gst/gstinfo.c:
  gstinfo: Don't leak array of strings
  The array provided by backtrace_symbols needs to be freed.

2017-11-24 12:05:26 +0100  Edward Hervey 

* gst/gstutils.c:
  gstutils: Fix linear regression comparision
  The check for dropping precision was wrong when sxx and syy were 
negative.
  if they are negative then "G_MAXINT64 - val" would always overflow
  The check was meant to use G_MININT64 (like in the loop contained just
  after).

2017-11-24 09:40:07 +0100  Tim-Philipp Müller 

* gst/gstghostpad.c:
  ghostpad: access internal pad with lock held

2017-03-30 09:17:08 +0200  Havard Graff 

* gst/gstghostpad.c:
* tests/check/gst/gstghostpad.c:
  ghostpad: fix race-condition while tearing down
  An upstream query will take a ref on the internal proxypad, and can
  hence end up owning the last reference to that pad, causing a crash.

2017-11-23 15:28:39 +0100  Edward Hervey 

* plugins/elements/gstqueue.c:
  queue: Only calculate level if we have valid levels

gst-plugins-base 1.12.4

[Tim-Philipp Müller]

ChangeLog
=

2017-12-07  Sebastian Dröge 

* configure.ac:
  releasing 1.12.4

2017-12-07 11:03:20 +0100  Edward Hervey 

* gst/typefind/gsttypefindfunctions.c:
  typefind: Fix ico out-of-bound read
  The furthest we go to verify the data is reading a guint32 at offset
  18, therefore make sure we can read as much.

2017-11-14 23:07:26 +0900  Seungha Yang 

* sys/xvimage/xvimageallocator.c:
  xvimageallocator: Fix build warning error
  Fix unused variable build error if HAVE_XSHM is undefined
  https://bugzilla.gnome.org/show_bug.cgi?id=790329

2017-12-06 14:22:49 +0100  Edward Hervey 

* ext/ogg/gstogmparse.c:
  ogmparse: Make sure we set valid string on caps
  the fourcc might not contain printable characters, use the convenience
  macro to turn it into a printable fourcc, like it's done in
  gstogmparse

2017-06-06 10:39:15 +0200  Havard Graff 

* gst-libs/gst/audio/gstaudiodecoder.c:
  audiodecoder: fix buffer leak in error code path

2017-12-01 10:14:30 -0300  Thibault Saunier 

* gst-libs/gst/audio/audio-converter.h:
  audio: Add missing G_BEGIN/END_DECLS

2017-11-25 12:45:03 +0100  Edward Hervey 

* gst/playback/gsturisourcebin.c:
  urisourcebin: Don't leak mutexes

2017-11-21 10:15:02 +0100  Edward Hervey 

* gst/videotestsrc/gstvideotestsrc.c:
  videotestsrc: Avoid overflow calculation
  n_frames could end up being quite big (potentially up to G_MAXINT64). 
Which
  would result in overflowing 64bits when multiplying it by GST_SECOND.
  Instead move GST_SECOND to the num argument

2017-11-06 14:28:52 +0100  Edward Hervey 

* ext/ogg/gstoggstream.c:
  oggstream: Ensure enough bytes for fishead header parsing

2017-11-05 12:16:13 +0100  Edward Hervey 

* ext/ogg/gstoggstream.c:
  oggstream: Use proper type for sample calculation
  If we are going to return a (potentially) 64bit integer, don't use
  a 32bit one for calculation, otherwise we could end up exceeding
  the maximum size of a 32bit int.

2017-11-05 12:15:33 +0100  Edward Hervey 

* ext/ogg/gstoggstream.c:
  oggstream: More fixes for invalid granuleshift
  Don't use granuleshift if it wasn't set

2017-11-04 16:56:10 +0100  Edward Hervey 

* gst-libs/gst/rtp/gstrtcpbuffer.c:
  rtcpbuffer: fix left shift override
  Needs to be cast to the target type

2017-11-04 12:19:46 +0100  Edward Hervey 

* gst/videotestsrc/videotestsrc.c:
  videotestsrc: Fix undefined left shift
  Cast value to target type

2017-11-04 12:18:39 +0100  Edward Hervey 

* gst-libs/gst/video/video-converter.c:
  video-converter: Fix undefined left shift
  Cast value to target type

2017-11-04 11:29:52 +0100  Edward Hervey 

* ext/ogg/gstoggstream.c:
  oggstream: Fix default granuleshift usage
  For stream mappers that don't set a specific granuleshift, it will
  have the default value of -1.
  Protect the code for that and return the granule value as-is

2017-11-04 11:28:47 +0100  Edward Hervey 

* ext/ogg/gstoggdemux.c:
  oggdemux: Protect against invalid granule positions
  Only valid values are -1, 0 or positive values. Anything else is
  most likely corrupted data streams

2017-11-03 17:20:44 +0100  Mathieu Duponchelle 

* gst-libs/gst/pbutils/codec-utils.c:
  codecutils: improve input validation in opus header parsing
  Invalid input files do not warrant assertions. Instead
  output error messages and let the error bubble up.

2017-11-02 15:14:49 +0100  Edward Hervey 

* gst/typefind/gsttypefindfunctions.c:
  typefind: Fix out-of-bound read in PNM typefinder

2017-11-02 10:40:37 +0100  Edward Hervey 

* gst-libs/gst/tag/gstvorbistag.c:
  vorbistag: Fix previous comment
  We already NULL-ended the string, don't use the bogus cur_size

2017-11-02 09:05:47 +  Tim-Philipp Müller 

* gst/typefind/gsttypefindfunctions.c:
  typefindfunctions: fix off-by-one in webvtt typefinder
  We're also checking the byte after the WEBVTT magic.

2017-11-02 09:19:21 +0100  Edward Hervey 

* gst-libs/gst/riff/riff-media.c:
  riff-media: Handle strf_data being NULL
  Instead of trying to get the size of a NULL buffer :)

2017-11-02 08:18:26 +0100  Edward Hervey 

* gst-libs/gst/tag/gstvorbistag.c:
  vorbistag: Check whether tag name is valid
  In the same way we check before whether the content of the tag
  is UTF-8

2017-11-01 18:26:20 +0100  Edward Hervey 

* ext/ogg/gstoggdemux.c:
  oggdemux: Only track time for initialized streams
  in p

gst-plugins-good 1.12.4

[Tim-Philipp Müller]

ChangeLog
=

2017-12-07  Sebastian Dröge 

* configure.ac:
  releasing 1.12.4

2017-12-07 19:03:58 +0200  Sebastian Dröge 

* po/el.po:
* po/nl.po:
  po: Update translations

2017-12-07 11:15:19 +  Tim-Philipp Müller 

* gst/audioparsers/gstflacparse.c:
* gst/audioparsers/gstflacparse.h:
  Revert "flacparse: fix header rewriting being ignored"
  This caused broken metadata and also looks a bit dodgy.
  Revert until we can figure out a solution that works for
  all cases and doesn't break anything.
  This reverts commit adeee44b07a173b9ab4253216caba8f66dd43abb.
  https://bugzilla.gnome.org/show_bug.cgi?id=727802
  https://bugzilla.gnome.org/show_bug.cgi?id=785558

2017-08-25 11:57:26 +0200  Haakon Sporsheim 

* gst/rtpmanager/rtpsession.c:
  rtpsession: Handle zero length feedback packets
  https://bugzilla.gnome.org/show_bug.cgi?id=791074

2017-07-10 15:19:34 +0200  Florian Zwoch 

* gst/isomp4/qtdemux.c:
  qtdemux: fix debug log for 'hvcC' codec_data
  https://bugzilla.gnome.org/show_bug.cgi?id=784749

2017-11-23 07:59:07 +0100  Edward Hervey 

* gst/replaygain/rganalysis.c:
  rganalysis: Fix left shift of signed values
  left shifting signed values is undefined.
  Instead of doing "x << offs" which is undefined, do the equivalent
  "x * (1 << offs)" which is well defined

2017-11-24 10:36:01 +0200  Sebastian Dröge 

* gst/audioparsers/gstflacparse.c:
  flacparse: Request at least the full header size when parsing headers
  Otherwise baseparse will incrementally send us bigger buffers until 
the
  full header size is reached, which is not only pointless but also 
means
  that baseparse will reallocate and copy into a bigger buffer for every
  input buffers. In pull mode that's done in 64kb increments, in push 
mode
  usually in much smaller increments, causing a lot of overhead for
  example when parsing high-quality coverart.

2017-10-22 18:26:12 +0800  Jun Xie 

* gst/isomp4/qtdemux.c:
  qtdemux: reset reused QtDemuxStream while parsing a new 'trak'
  if QtDemuxStream is reused, then we need to reset it.
  https://bugzilla.gnome.org/show_bug.cgi?id=788759

2017-11-18 02:01:58 +1100  Jan Schmidt 

* gst/multifile/gstsplitmuxpartreader.c:
  splitmuxsrc: Don't return FALSE from event handling.
  Returning FALSE because we drop an event means that
  internal sources like qtdemux might throw an error
  and break the whole pipeline. The only time it can
  happen is either flushing or shutdown, and those
  will be handled anyway.

2017-03-13 18:14:12 +0900  paul.kim 

* ext/soup/gstsouphttpsrc.c:
  souphttpsrc: Remove range header when seek to 0
  This fixes the previous range header is remained if seek to 0 is
  attempted.
  https://bugzilla.gnome.org/show_bug.cgi?id=779957

2017-06-20 16:28:35 +0200  Jan Alexander Steffens (heftig) 


* tests/files/test-cert.pem:
  tests: souphttpsrc: Update test-cert.pem
  Recent GnuTLS disregards the Common Name and only looks at the Subject
  Alternative Name extension. Since our test-cert has no SAN extension,
  validation fails.
  Generate a new certificate with SAN. In addition to 127.0.0.1, for 
good
  measure make it valid for localhost and ::1, too.
  https://bugzilla.gnome.org/show_bug.cgi?id=784005

2017-06-20 16:39:36 +0200  Jan Alexander Steffens (heftig) 


* tests/check/elements/souphttpsrc.c:
  tests: souphttpsrc: Avoid deprecated ssl-ca-file property
  SoupSession's ssl-ca-file property is deprecated. Use the recommended
  tls-database property.
  This is a bit more complex as it requires creating a GTlsFileDatabase
  object for an absolute (!) path to the CA certificates file.
  https://bugzilla.gnome.org/show_bug.cgi?id=784005

2017-06-20 16:37:55 +0200  Jan Alexander Steffens (heftig) 


* tests/check/elements/souphttpsrc.c:
  tests: souphttpsrc: Avoid deprecated server ssl properties
  The ssl-cert-file and ssl-key-file properties are deprecated. Use the
  soup_server_set_ssl_cert_file function to load the files.
  https://bugzilla.gnome.org/show_bug.cgi?id=784005

2017-06-20 16:34:41 +0200  Jan Alexander Steffens (heftig) 


* tests/check/elements/souphttpsrc.c:
  tests: souphttpsrc: Make ssl_cert/key_file static
  Just a bit of cleanup.
  https://bugzilla.gnome.org/show_bug.cgi?id=784005

2017-11-08 16:34:01 +0100  Edward Hervey 

* ext/soup/gstsouphttpsrc.c:
  soupht

gst-plugins-good 1.6.4

[Tim-Philipp Müller]

ChangeLog
=

2016-04-14  Tim-Philipp Müller 

* configure.ac:
  releasing 1.6.4

2016-04-12 09:41:00 +  Paolo Pettinato 

* gst/rtpmanager/gstrtpmux.c:
  rtpmux: Forward sticky events on buffer lists too, not only on buffers
  https://bugzilla.gnome.org/show_bug.cgi?id=764933

2016-03-08 13:03:55 +0100  Alban Bedel 

* sys/v4l2/gstv4l2object.c:
  v4l2object: fix capture with bayer formats other than bggr
  gst_v4l2_object_get_caps_info() always return V4L2_PIX_FMT_SBGGR8
  for all bayer formats. This is obviously broken if the device use
  another ordering. Fix this by properly reading the format parameter.
  https://bugzilla.gnome.org/show_bug.cgi?id=763318

2016-03-04 15:09:45 +0100  Josep Torra 

* sys/v4l2/gstv4l2object.c:
  v4l2: fix colorimetry for NV12
  Replicate V4L2_MAP_QUANTIZATION_DEFAULT macro behavior.
  At #v4l it was described that documentation might be wrong and that
  we should trust this macro instead.
  https://bugzilla.gnome.org/show_bug.cgi?id=762529

2016-01-21 17:41:55 -0500  Nicolas Dufresne 

* sys/v4l2/gstv4l2object.c:
  v4l2: Re-add colorimetry field for RGB formats
  This time, check if it's an RGB format and sets the transformation
  matrix to identity. The rest of the colorimetry information is
  meaningfull and shall be kept.
  https://bugzilla.gnome.org/show_bug.cgi?id=759624

2016-01-22 10:03:50 +0100  Wim Taymans 

* sys/v4l2/gstv4l2object.c:
  v4l2: fix sRGB colorspace definition
  V4l2 can also use the sRGB colorspace for YUV formats and thus needs a
  default matrix.

2015-11-17 12:39:05 +0200  Sebastian Dröge 

* gst/udp/gstudpsrc.c:
* gst/udp/gstudpsrc.h:
  udpsrc: Add "loop" property for enabling/disabling multicast loopback
  On POSIX, IP_MULTICAST_LOOP is a setting for the sender socket. On 
Windows it
  is a setting for the receiver socket. As such we will need it on 
udpsrc too to
  allow filtering out our own multicast packets.

2016-02-10 19:56:59 +0530  Nirbheek Chauhan 

* sys/v4l2/gstv4l2deviceprovider.c:
  v4l2: Don't leak v4l2 objects and props on probe errors

2016-03-15 03:25:26 +0530  Nirbheek Chauhan 

* gst/rtp/gstrtpjpegdepay.c:
  rtpjpegdepay: Don't send invalid frames downstream after packet loss 
or a DISCONT
  After clearing the adapter due to a DISCONT, as might happen when 
some packet(s)
  have been lost, the depayloader was pushing data into the adapter 
(which had no
  header due to the clear), creating a headerless frame out of it, and 
sending it
  downstream. The downstream decoder would then usually ignore it; 
unless there
  were lots of DISCONTs from the jitterbuffer in which case the decoder 
would reach
  its max_errors limit and throw an element error. Now we just discard 
that data.
  It is probaby not worth trying to salvage this data because 
non-progressive
  jpeg does not degrade gracefully and makes the video unwatchable even 
with
  low packet loss such as 3-5%.

2016-01-31 11:08:38 +1100  Sebastian Dröge 

* gst/rtp/gstrtpjpegpay.c:
  rtpjpegpay: Allow different quantization tables for components 2 and 3
  RFC 2435 mentions in section 4.1 that U/V use table number 1, but 
this seems
  just like an example. Some encoders are not following that and there 
seems to
  be no reason to reject their streams.
  https://bugzilla.gnome.org/show_bug.cgi?id=761345

2016-02-29 23:40:03 -0300  Thiago Santos 

* gst/multifile/gstsplitmuxsink.c:
* tests/check/elements/splitmux.c:
  splitmuxsink: only try to create internal sink if it doesn't exist
  This allows splitmuxsink to be reused after being put to NULL.
  Test included
  https://bugzilla.gnome.org/show_bug.cgi?id=762893

2016-03-04 14:07:19 +0200  Sebastian Dröge 

* gst/udp/gstudpsrc.c:
  udpsrc: Fix multicast group joining with provided sockets on Windows
  On Windows the socket will be bound to ANY instead of the multicast 
group,
  as binding to a multicast group does not work. Which would mean that 
we
  override src->addr to become ANY and won't automatically join a 
multicast
  group anymore on Windows.
  On Linux we would automatically join a multicast group, keep it 
consistent.
  https://bugzilla.gnome.org/show_bug.cgi?id=763093

2015-11-04 14:51:19 +0900  Jihae Yi 

* gst/rtsp/gstrtspsrc.c:
  rtspsrc: avoid potentially overflowing expression
  https://bugzilla.gnome.org/show_bug.cgi?id=757569

2016-03-08 17:37:17 +0100  Thomas Roos 

* sys/directsound/gstdirectsoundsin

gst-plugins-base 1.6.4

[Tim-Philipp Müller]

ChangeLog
=

2016-04-14  Tim-Philipp Müller 

* configure.ac:
  releasing 1.6.4

2016-04-11 11:28:09 +0200  Fabrice Bellet 

* gst-libs/gst/audio/gstaudiosink.c:
* gst-libs/gst/audio/gstaudiosrc.c:
  audio: Fix a race with the audioringbuffer thread
  There is a small window of time where the audio ringbuffer thread
  can access the parent thread variable, before it's initialized
  by the parent thread. The patch replaces this variable use by
  g_thread_self().
  https://bugzilla.gnome.org/show_bug.cgi?id=764865

2016-04-02 10:09:07 +0100  Tim-Philipp Müller 

* gst-libs/gst/pbutils/gstdiscoverer-types.c:
  discoverer: copy over result and seekable fields when copying a 
discoverer info
  The function gst_discoverer_info_copy doesn't copy the data members 
seekable
  and result of the source GstDiscovererInfo.
  In the case of copying a GstDiscovererInfo for later use, the 
seekbale will be
  undefined, which in practice usually will be false, even though the 
seekable of
  the original GstDiscovererInfo is true.
  https://bugzilla.gnome.org/show_bug.cgi?id=762710

2016-03-24 14:59:48 +1100  Jan Schmidt 

* gst/playback/gstdecodebin2.c:
  decodebin2: Hold new buffering_post lock while posting msgs
  There's a small window between decodebin choosing a buffering level
  to post and another thread choosing a different buffering level
  where things can race. Close that window by holding a new lock
  that's only for posting buffering messages - like what was done
  in multiqueue.
  https://bugzilla.gnome.org/show_bug.cgi?id=764020

2016-01-28 16:26:47 +0100  Tom Deseyn 

* gst/tcp/gstmultisocketsink.c:
  multisocketsink: handle client close correctly and EWOULDBLOCK
  Fixes 100% cpu usage when client disconnects. Commit 6db2ee56
  would just make multisocketsink ignore reads of 0 bytes without
  removing the client, so we'd get woken up over and over again
  for the client.
  Fix the original issue differently by handling the non-fatal error 
code.
  https://bugzilla.gnome.org/show_bug.cgi?id=761257
  https://bugzilla.gnome.org/show_bug.cgi?id=743834

2016-03-02 18:47:23 +0200  Sebastian Dröge 

* gst/encoding/gstencodebin.c:
  encodebin: Make dispose() function safe to be called multiple times

2016-01-22 18:26:01 -0800  Reynaldo H. Verdejo Pinochet 


* gst/typefind/gsttypefindfunctions.c:
  typefind: strengthen check for valid H.263 picture layer
  Avoids some false positives leading to miss identification:
  * Prevent picture start code emulation for the first 2 bytes read
  * Add check for valid "picture coding type" and "PB-frames mode" 
combination
  Additionally, change name on confusingly named TR var to what
  it is, the layer's PTYPE.
  https://bugzilla.gnome.org/show_bug.cgi?id=693263

2016-02-23 09:36:50 +0100  Edward Hervey 

* gst/playback/gstplaysink.c:
  Revert "playsink: Properly mark pending blocked pads"
  This reverts commit 3bc67006d4075f5bcbaa7de68230c8b0f551d44a.
  The issue that the patch fixes is only noticeable when using 
decodebin3,
  which isn't yet in master.

2016-02-11 19:47:04 +0100  Wim Taymans 

* gst/audioresample/resample.c:
  resample: avoid overflows
  Avoid overflow in rate calculation. This can cause the resampler to
  start on the wrong phase after a rate change.
  Avoid overflow in cubic fraction calculation. This can cause noise 
when
  dealing with higher samplerates.

2016-02-11 18:01:40 +0100  Wim Taymans 

* gst/audioresample/resample_sse.h:
  resample: fix double interpolation sse code
  We were only reading 2 filter taps and we need to read 4 to do cubic
  interpolation.



Download

https://download.gnome.org/sources/gst-plugins-base/1.6/gst-plugins-base-1.6.4.tar.xz
 (2.70M)
  sha256sum: 7a193e2a66b0d7411160ef2a373184c8aa3cdeaa576fa270be346716220d9606

___
release-team@gnome.org
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Release-team lurker? Do NOT participate in discussions.

gstreamer 1.6.4

[Tim-Philipp Müller]

ChangeLog
=

2016-04-14  Tim-Philipp Müller 

* configure.ac:
  releasing 1.6.4

2016-03-19 12:55:09 +0100  Aurélien Zanelli 

* gst/gstutils.c:
  utils: add 'transfer full' annotation to gst_pad_peer_query_caps
  https://bugzilla.gnome.org/show_bug.cgi?id=763912

2016-03-19 12:39:18 +0100  Aurélien Zanelli 

* gst/gstpad.c:
  pad: add 'transfer full' and 'nullable' annotations to 
gst_pad_get_current_caps
  and also change the description accordingly since function returns an
  incremented caps object or NULL if there is no caps set.
  https://bugzilla.gnome.org/show_bug.cgi?id=763912

2016-03-18 16:02:43 -0400  Ben Iofel 

* gst/gstutils.c:
  utils: fix gir annotation for gst_element_query_convert()
  https://bugzilla.gnome.org/show_bug.cgi?id=763895

2016-03-18 03:08:39 +1100  Jan Schmidt 

* plugins/elements/gstmultiqueue.c:
  multiqueue: Fix not-linked pad handling at EOS
  Ensure that not-linked pads will drain out at EOS by
  correctly detecting the EOS condition based on the EOS
  pad flag (which indicates we actually pushed an EOS),
  and make sure that not-linked pads are woken when doing
  EOS processing on linked pads.
  https://bugzilla.gnome.org/show_bug.cgi?id=763770

2016-01-20 09:57:00 +  Tim-Philipp Müller 

* plugins/elements/gstmultiqueue.c:
  multiqueue: two small fixes for when an existing pad is requested
  Unlock when returning NULL from gst_single_queue_new(), and don't
  crash with debug logging enabled if NULL is returned.
  Spotted by Steven Hoving.

2015-12-09 17:40:02 +0100  Edward Hervey 

* plugins/elements/gstmultiqueue.c:
  multiqueue: Don't use random segment.position from upstream
  segment.position is meant for internal usage only, but the various
  GST_EVENT_SEGMENT creationg/parsing functions won't clear that field.
  Use the appropriate segment boundary as an initial value instead

2016-03-15 16:37:33 +0100  Romain Picard 

* plugins/elements/gsttypefindelement.c:
  typefind: Allow caps query in "have-type" signal handlers
  If an application calls gst_pad_query_caps from its "have-type" 
signal handler,
  then the query fails because typefind->caps has not been set yet.
  This patch sets typefind->caps in the object method handler, before 
the signal
  handlers are called.
  https://bugzilla.gnome.org/show_bug.cgi?id=763491

2016-03-11 14:17:13 +0200  Sebastian Dröge 

* plugins/elements/gsttypefindelement.c:
  typefind: Store caps on the pad before emitting have-type but send it 
downstream only in the default signal handler
  https://bugzilla.gnome.org/show_bug.cgi?id=763491

2016-03-13 10:33:53 +0200  Sebastian Dröge 

* libs/gst/base/gstbaseparse.c:
  baseparse: Recheck after pre_push_frame() if there are tags pending
  Many parsers are storing tags only in pre_push_frame(), if we 
wouldn't check
  afterwards we would push buffers before those tags and a lot of code 
assumes that
  tags are available before preroll.
  https://bugzilla.gnome.org/show_bug.cgi?id=763553

2016-02-04 10:07:22 +  Tim-Philipp Müller 

* libs/gst/base/gstbaseparse.c:
  baseparse: fix stray discont flag set on outgoing buffers in push mode
  We have no guarantees about what flags are set on buffers we take
  out of the GstAdapter. If we push out multiple buffers from the
  first input buffer (which will have discont set), only the first
  buffer we push out should be flagged as discont, not all of the
  buffers produced from that first initial input buffer.
  Fixes issue where the first few mp3 frames/seconds of data in push
  mode were skipped or garbled in some cases, and the discont flags
  would also trip up decoders which were getting drained/flushed for
  every buffer. This was a regression introduced in 1.6 apparently.

2016-02-02 16:35:34 +0100  Thibault Saunier 

* libs/gst/controller/gstdirectcontrolbinding.c:
  controller: Do not unset uninitiallized GValue
  In case the property was not interpollable we might never initialize
  the GValue, we should thus never unset it.



Download

https://download.gnome.org/sources/gstreamer/1.6/gstreamer-1.6.4.tar.xz (3.44M)
  sha256sum: 4ccba88a286b13d6f2d8c1180f78a13dcd49f2fc3cb2b3b3f502b3a23f7c01b5

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Release-team lurker? Do NOT participate in discussions.

gst-plugins-base 1.6.3

[Tim-Philipp Müller]

ChangeLog
=

2016-01-20  Sebastian Dröge 

* configure.ac:
  releasing 1.6.3

2016-01-20 10:19:34 +0100  Wim Taymans 

* gst-libs/gst/video/video-info.c:
  video-info: enfore RGB matrix for RGB formats
  In gst_video_info_to_caps(), make sure we end up with an RGB matrix 
for
  RGB formats and warn when the GstVideoInfo colorimetry is wrong.
  In gst_video_info_from_caps(), fix the GstVideoInfo with an RGB matrix
  for RGB formats and warn about inconsistent caps.
  See https://bugzilla.gnome.org/show_bug.cgi?id=759624

2016-01-20 10:02:20 +0100  Wim Taymans 

* gst-libs/gst/video/video-converter.c:
  video-converter: ignore matrix for RGB formats
  For RGB formats, the matrix in the colorimetry (conversion from YUV to
  RGB) is irrelevant and we should ignore it and assume the identity
  transform for everything we do.
  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=759624

2016-01-19 23:26:57 +0100  Thibault Saunier 

* gst-libs/gst/video/gstvideoencoder.h:
  videoencoder: Deprecate GST_VIDEO_ENCODER_FLOW_DROPPED
  It was never actually supported or used
  https://bugzilla.gnome.org/show_bug.cgi?id=760666

2016-01-19 23:22:35 +0100  Thibault Saunier 

* gst-libs/gst/video/gstvideoencoder.c:
  Revert "videoencoder: Release video frame when ->handle return ERROR 
or DROPPED"
  This reverts commit 63517d0ed348784cce4ab4b295c2c0f1b78baa81.
  It was wrong ref counting wise and we decided to deprecated DROPPED
  return value
  https://bugzilla.gnome.org/show_bug.cgi?id=760666

2016-01-09 04:35:23 +0100  Mathieu Duponchelle 


* gst/playback/gststreamsynchronizer.c:
  streamsynchronizer: Ignore flushing streams [..]
  [..] when resetting group start time. In GES, we are usually connected
  to the streamsynchronizer on one audio and one video pad.
  When seeking the timeline, both nlecompositions often output their 
flush_start
  before any of them has output its flush_stop.
  The current code, when receiving the first flush stop was using the
  running time of the start of the second composition, which could
  be pretty much anything, and means nothing at that point.
  This patch is thread-safe, as STREAM_SYNCHRONIZER_LOCK is taken
  both when setting flushing and when checking it.
  https://bugzilla.gnome.org/show_bug.cgi?id=750013

2016-01-15 13:35:22 +  Thibault Saunier 

* gst-libs/gst/video/gstvideoencoder.c:
  videoencoder: Release video frame when ->handle return ERROR or 
DROPPED
  https://bugzilla.gnome.org/show_bug.cgi?id=760666

2016-01-15 09:50:29 +0100  Edward Hervey 

* gst/playback/gstplaysink.c:
  playsink: Properly mark pending blocked pads
  When blocking input pads, we also need to properly set the appropriate
  pending flag.
  Without this, when switching stream types after initial configuration
  (like going from Audio+Video to Audio+Video+Sub) playsink would never
  wait for *all* input streams to be blocked (it would just wait for the
  new input pad (text in this case) to be blocked).
  Since the reconfiguration might introduce unlinking/relinking of 
elements,
  we need to ensure that *ALL* input streams are blocked.
  Failure to do so would result in having some input streams pushing 
data
  to inactive elements (returning GST_FLOW_FLUSHING) or unlinked pads
  (returning GST_FLOW_NOT_LINKED).
  A later optimization could involve only blocking the input pads that
  might be involved in reconfiguration. But better be safe than sorry 
for
  now :)

2016-01-06 15:49:59 +0100  Aurélien Zanelli 

* gst-libs/gst/video/gstvideopool.c:
  videopool: store videoinfo after choosing the biggest buffer size
  Otherwise, pool could be negotiated with a size which will be 
different
  from the one used in allocation which is the GstVideoInfo.
  https://bugzilla.gnome.org/show_bug.cgi?id=760222

2015-12-28 12:28:26 +0200  Sebastian Dröge 

* gst/typefind/gsttypefindfunctions.c:
  typefindfunctions: Make sure that enough data is available in 
AAC/ADTS typefinder
  We would otherwise read beyond the array bounds and crash every now 
and then.
  This was introduced with 5640ba17c8db80976b7718904e4024dcfe9ee1a0.
  https://bugzilla.gnome.org/show_bug.cgi?id=759910

2015-12-25 11:34:10 +0100  Sebastian Dröge 

* gst/playback/gstplaysink.c:
  playsink: Don't leak audio/video filters due to floating references 
weirdness
  The filters' floating references are sinked during set_property() 
already,
  which means that GstBin takes a new reference w

gst-plugins-good 1.6.3

[Tim-Philipp Müller]

ChangeLog
=

2016-01-20  Sebastian Dröge 

* configure.ac:
  releasing 1.6.3

2016-01-20 13:41:33 +0200  Sebastian Dröge 

* ext/vpx/gstvp8enc.c:
  vp8enc: Ensure that we always have valid frame user data before using 
it
  Otherwise we're going to dereference NULL pointers.

2016-01-20 10:02:48 +0200  Sebastian Dröge 

* ext/vpx/gstvp8dec.c:
* ext/vpx/gstvp9dec.c:
  vpxdec: Unref frame in all code paths of handle_frame()
  https://bugzilla.gnome.org/show_bug.cgi?id=760666

2016-01-19 22:49:20 +0100  Thibault Saunier 

* ext/vpx/gstvp8enc.c:
* ext/vpx/gstvp9enc.c:
  vpxenc: Unref frame on ERROR
  All code paths for handle_frame() must somehow take ownership of the 
frame, be
  it by actually unreffing, forwarding the frame elsewhere or storing 
it for
  later.
  http://bugzilla.gnome.org/show_bug.cgi?id=760666

2016-01-20 10:08:50 +0200  Sebastian Dröge 

* ext/vpx/gstvp9enc.c:
  vp9enc: Return FLOW_ERROR when an error accures
  FALSE would mean FLOW_OK
  https://bugzilla.gnome.org/show_bug.cgi?id=760666

2016-01-19 15:14:59 -0500  Nicolas Dufresne 

* sys/v4l2/gstv4l2object.c:
  v4l2object: Don't set colorimetry for non YUV formats
  Setting colormetry in caps for RGB have no meaning, but worst it
  confuses the converters downstream.
  https://bugzilla.gnome.org/show_bug.cgi?id=759624

2016-01-19 14:57:03 +0200  Sebastian Dröge 

* gst/wavparse/gstwavparse.c:
  wavparse: Don't play anything after the end of the data chunk even 
when seeking
  Especially in push mode we would completely ignore the size of the 
data chunk
  when not stop position is given for the seek. Instead make sure that 
the end
  offset is at most the end of the data chunk if known.
  Without this we would output anything after the data chunk, possibly 
causing
  loud noises if the media file is followed by an INFO chunk or an ID3 
tag.

2016-01-19 14:55:57 +0200  Sebastian Dröge 

* gst/wavparse/gstwavparse.c:
  wavparse: Don't do calculations with -1 offsets when handling SEGMENT 
events
  We use that to signal "infinity", taking the difference between that 
and some
  other value is not going to give us any useful result for the end 
offsets of
  segments.

2016-01-12 14:01:21 -0800  Aleix Conchillo Flaqué 

* gst/rtsp/gstrtspsrc.c:
  rtspsrc: handle rtcp/srtcp caps properly when using interleaved data
  We check the stream profile and use the proper RTCP caps:
  application/x-srtcp if we are using a secure profile and
  application/x-rtcp otherwise.
  https://bugzilla.gnome.org/show_bug.cgi?id=760556

2016-01-15 03:56:49 +0530  Nirbheek Chauhan 

* sys/osxaudio/gstosxaudioringbuffer.c:
* sys/osxaudio/gstosxcoreaudiohal.c:
  osxaudio: Fix error handling when selecting/opening devices
  Post an element error when the CoreAudio device cannot be selected or 
opened.
  Also ensure that we post a GST_ERROR with more detail.

2016-01-13 23:40:20 +0100  Sebastian Dröge 

* gst/wavparse/gstwavparse.c:
  wavparse: When flushing on EOS, don't process more data than the 
"data" size
  Even if we have more data queued up when flushing than the size of 
the data
  chunk, don't process and output it. If the data size is known, this 
likely
  contains another chunk (e.g. an INFO chunk) or things like ID3 tags. 
Just
  outputting them as if they were data is going to cause unexpected 
behaviour
  and unpleasant audio noises.

2016-01-11 16:29:55 +  Tim Sheridan 

* gst/audioparsers/gstsbcparse.c:
  sbcparse: Fix frame length calculation
  SBC frame length calculation wasn't being rounded up to the nearest 
byte
  (as specified in the A2DP 1.0 specification, section 12.9). This could
  cause 'stereo' and 'joint stereo' mode SBC streams to have incorrectly
  calculated frame lengths.
  Incorrect frame length calculation causes frame coalescing to fail, as
  subsequent frames in the stream aren't found in the expected 
locations.
  https://bugzilla.gnome.org/show_bug.cgi?id=742446

2016-01-15 11:36:35 +  Thibault Saunier 

* ext/vpx/gstvp8enc.c:
  vp8enc: Return FLOW_ERROR when an error accures
  FALSE would mean FLOW_OK
  https://bugzilla.gnome.org/show_bug.cgi?id=760666

2016-01-07 20:27:29 +  Tim-Philipp Müller 

* gst/rtp/gstrtpvrawdepay.c:
  rtpvrawdepay: fix major memory leak and performance issue
  We call gst_rtp_buffer_get_payload() which creates a sub-buffer
  

gstreamer 1.6.3

[Tim-Philipp Müller]

ChangeLog
=

2016-01-20  Sebastian Dröge 

* configure.ac:
  releasing 1.6.3

2016-01-16 16:01:38 +  Tim-Philipp Müller 

* gst/gstmessage.c:
  message: add function guard to gst_message_set_buffering_stats()
  https://bugzilla.gnome.org/show_bug.cgi?id=760704

2016-01-15 00:25:05 +0530  Nirbheek Chauhan 

* gst/gstdevicemonitor.c:
  GstDeviceMonitor: Don't remove unmatched class filters
  If no providers for a particular class could be found, then removing 
unmatched
  filters would cause all devices to be returned instead which is not 
at all what
  the user intended. We still return 0 for unmatched filters.

2016-01-13 21:32:20 +  Florin Apostol 

* libs/gst/net/gstnetclientclock.c:
  netclientclock: Fix GError memory leak in handling NTP response
  Error was not released if gst_ntp_packet_receive failed.
  https://bugzilla.gnome.org/show_bug.cgi?id=760598

2015-09-26 18:16:07 +0800  Ting-Wei Lan 

* gst/gstutils.h:
  gstutils: Fix build with clang -Werror=cast-align
  https://bugzilla.gnome.org/show_bug.cgi?id=755657

2016-01-06 19:50:21 +0000  Tim-Philipp Müller 

* plugins/elements/gstqueue2.c:
  queue2: fix fill level arithmetic overflow with large values
  Based on patch by: Aleksander Wabik 
  https://bugzilla.gnome.org/show_bug.cgi?id=755971

2015-12-29 14:32:47 +0200  Sebastian Dröge 

* gst/gstcaps.c:
  caps: Add (transfer full) annotation to simplify() and subtract() 
return value
  https://bugzilla.gnome.org/show_bug.cgi?id=759948

2015-12-29 11:06:39 +0100  Aurélien Zanelli 

* gst/gstcaps.c:
  caps: add 'transfer full' annotation to caps returned by 
interserction functions
  To make clear caller is responsible to unref them.
  https://bugzilla.gnome.org/show_bug.cgi?id=759948



Download

https://download.gnome.org/sources/gstreamer/1.6/gstreamer-1.6.3.tar.xz (3.39M)
  sha256sum: 22f9568d67b87cf700a111f381144bd37cb93790a77e4e331db01fe854a37f24

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gst-plugins-good 1.5.2

[Tim-Philipp Müller]

ChangeLog
=

2015-06-24  Sebastian Dröge 

* configure.ac:
  releasing 1.5.2

2015-06-24 22:56:12 +0200  Sebastian Dröge 

* 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

2015-06-24 11:15:00 +0200  Sebastian Dröge 

* po/nl.po:
  po: Update translations

2015-06-23 18:42:59 -0400  Nicolas Dufresne 

* tests/check/elements/qtmux.c:
  qtmux: Correctly test each segments
  In presence of gaps, qtdemux will emit multiple segments. The
  second segment start should match the CTTS.
  https://bugzilla.gnome.org/show_bug.cgi?id=751361

2015-06-23 17:54:31 -0400  Nicolas Dufresne 

* gst/isomp4/gstqtmux.c:
* gst/isomp4/gstqtmux.h:
  qtmux: Correctly calculate the elst media start
  The media start has nothing to do with the shift we have applied
  but with the value of the first PTS. This is defined as:
  Dt(0) = 0
  Ct(0) = Dt(0) + CTTS(0)
  So the media start is always the first CTTS.
  https://bugzilla.gnome.org/show_bug.cgi?id=751361

2015-06-23 11:49:32 -0300  Thiago Santos 

* gst/isomp4/qtdemux.c:
  qtdemux: accumulate previous edts entries into segment.base
  Allows playing edts editted files with proper synchronization of
  streams. This patch fixes the regression introduced by
  bf95f93c0189aa04f18e264b86b6527e431c5d53 that was added to fix
  segment seeks handling.
  Having the accumulated_base separated from the main segment.base
  allows handling both segment seeks and edts editted files.
  https://bugzilla.gnome.org/show_bug.cgi?id=751361

2015-06-23 00:56:16 -0300  Thiago Santos 

* gst/isomp4/qtdemux.c:
  qtdemux: improve some debug messages
  Those messages are about the stream, use the pad as the
  debug object to make it clear from the logs
  https://bugzilla.gnome.org/show_bug.cgi?id=751361

2015-06-22 22:22:09 -0300  Thiago Santos 

* gst/isomp4/gstqtmux.c:
  qtmux: store last_dts of the first buffer
  Buffers need not to start at running-time 0 so the last_dts needs
  to be the value of the first buffer's dts as it is used to compute
  the duration of the buffers. If it was left at 0 the first buffer
  would have a larger duration when it shouldn't
  https://bugzilla.gnome.org/show_bug.cgi?id=751361

2015-06-23 17:11:57 +0900  Vineeth TM 

* gst/audioparsers/gstflacparse.c:
  flacparse: fix possible memory leak
  when buffer is stored to seektable, and stop gets called due to
  corrupt flac file, then the seektable is not being released
  https://bugzilla.gnome.org/show_bug.cgi?id=751364

2015-06-23 16:28:40 +1000  Jan Schmidt 

* gst/multifile/gstsplitmuxsink.c:
* gst/multifile/gstsplitmuxsink.h:
  Revert "splitmuxsink: Mask async-start/done while switching files."
  This reverts commit d61e5393f110ed482815d77807245d78b52eff46.
  Causes failures muxing larger GOP sizes for some reason. Reverting
  while I figure it out

2015-06-18 23:22:06 +1000  Jan Schmidt 

* gst/multifile/gstsplitmuxpartreader.c:
* gst/multifile/gstsplitmuxsrc.c:
  splitmuxsrc: Fix startup and shutdown races.
  Fix 2 startup races when things happen too quickly, and 1
  at shutdown by holding a ref to the pads in use until the
  loop functions exit.
  Handle errors activating file parts and publish them on
  the bus.
  https://bugzilla.gnome.org/show_bug.cgi?id=750747

2015-06-18 09:26:13 +1000  Jan Schmidt 

* gst/multifile/gstsplitmuxsink.c:
* gst/multifile/gstsplitmuxsink.h:
  splitmuxsink: Mask async-start/done while switching files.
  Sometimes, extra async-start/done from the internal sink
  while the element is still starting up can cause splitmuxsink
  to stall in PAUSED state when it has been set to PLAYING
  by the app. Drop the child's async-start/done messages while
  switching

gstreamer 1.5.2

[Tim-Philipp Müller]

ments/gstconcat.c:
* plugins/elements/gstconcat.h:
  concat: Add adjust-base property
  This disables the segment.base adjustments, which is useful if 
downstream
  takes care of base adjustments already (example: a combination of 
concat
  and streamsynchronizer)
  https://bugzilla.gnome.org/show_bug.cgi?id=751047

2015-06-22 14:04:45 +0200  Sebastian Dröge 

* libs/gst/base/gstbasesink.c:
  basesink: Unset the last buffer list if we only got a buffer
  Also remember any preroll buffer list.

2015-06-22 13:33:29 +0200  Sebastian Dröge 

* docs/gst/gstreamer-sections.txt:
* win32/common/libgstreamer.def:
  sample: Add new API to the docs

2015-06-22 20:02:55 +0900  Hyunjun 

* libs/gst/base/gstbasesink.c:
  basesink: enable to get last sample including buffer list if needed
  In case of a buffer list rendering, last-sample is not updated.
  It needs to be updated and enable to get buffer list from last-sample.
  https://bugzilla.gnome.org/show_bug.cgi?id=751026

2015-06-22 19:35:40 +0900  Hyunjun 

* gst/gstsample.c:
* gst/gstsample.h:
  sample: add gst_sample_set/get_buffer_list apis
  Allowed to set/get buffer list to sample if needed
  https://bugzilla.gnome.org/show_bug.cgi?id=751026

2015-06-19 10:52:10 +0100  Tim-Philipp Müller 

* test.py:
  test.py: remove accidentally committed file

2015-06-18 11:51:48 -0400  Nicolas Dufresne 

* gst/gstbuffer.c:
* gst/gstelementfactory.h:
* gst/gstsegment.h:
* gst/gstsystemclock.h:
* libs/gst/base/gstbasetransform.h:
  doc: Unify Since mark for attribute and enum
  As this show up as prose in the doc, simply make it consistent
  and "arguable" nicer to read.

2015-06-18 11:48:50 -0400  Nicolas Dufresne 

* gst/gstbuffer.h:
  doc: Remove parenthesis around Since:
  This confuse the parser, hence it does not endup in the doc and the
  index properly.

2015-05-06 16:44:48 +1000  Jan Schmidt 

* docs/gst/gstreamer-sections.txt:
* gst/gstbuffer.c:
* gst/gstbuffer.h:
* win32/common/libgstreamer.def:
  Add GstParentBufferMeta
  A core meta which helps implement the old concept
  of sub-buffering in some situations, by making it
  possible for a buffer to keep a ref on a different
  parent buffer. The parent buffer is unreffed when
  the Meta is freed.
  This meta is used to ensure that a buffer whose
  memory is being shared to a child buffer isn't freed
  and returned to a buffer pool until the memory
  is.
  https://bugzilla.gnome.org/show_bug.cgi?id=750039

2015-06-16 18:08:24 -0400  Nicolas Dufresne 

* gst/Makefile.am:
* libs/gst/base/Makefile.am:
* libs/gst/check/Makefile.am:
* libs/gst/controller/Makefile.am:
* libs/gst/net/Makefile.am:
* test.py:
  gi: Use INTROSPECTION_INIT for --add-init-section
  This new define was added to common. The new init section fixed
  compilation warning found in the init line that was spread across
  all files.

2015-06-16 17:46:44 -0400  Nicolas Dufresne 

* common:
  Automatic update of common submodule
  From 6015d26 to f74b2df

2015-06-15 10:06:00 -0400  Nicolas Dufresne 

* gst/gstclock.h:
  clock: Fix _STIME_FORMAT macros
  This macro didn't work well as it relied on the sign on the last
  divided number (number of days). This value is most of the time
  zero, and zero is considered positive in printf. Instead, deal with
  the sign manually, and resuse the original macros for the rest. This
  actually simplify the macro a lot.

2015-06-14 20:48:29 +0100  Tim-Philipp Müller 

* plugins/elements/gsttypefindelement.c:
  typefindelement: reset segment only once streaming has stopped
  Fixes the occasional criticals in the discoverer unit test.
  https://bugzilla.gnome.org/show_bug.cgi?id=745073
  https://bugzilla.gnome.org/show_bug.cgi?id=750823

2015-06-14 11:23:22 -0400  Nicolas Dufresne 

* docs/libs/gstreamer-libs-sections.txt:
  doc: Add more missing symbols in lib-sections.txt
  These where causing broken links.

2015-06-14 11:22:54 -0400  Nicolas Dufresne 

* libs/gst/net/gstnetaddressmeta.c:
  doc: Fix reference to unknown type GstNetAddress

2015-06-14 11:22:36 -0400  Nicolas Dufresne 

* docs/libs/gstreamer-libs-sections.txt:
* libs/gst/controller/gsttimedvaluecontrolsource.h:
  doc: Include and fix GstControlPoint

2015-06-14 11:21:26 -0400  Nicolas Dufresne 

* docs/libs/gstreamer-libs-docs.sgml:
  doc: Add GstNetCont

gst-plugins-base 1.5.2

[Tim-Philipp Müller]

n to NULL when deactivating chain
  The chain is about to be invalidated so we shouldn't keep it around.
  Prevent a double free crash when the demuxer is being finalized.
  https://bugzilla.gnome.org/show_bug.cgi?id=751000

2015-06-19 19:51:25 +0900  Vineeth T M 

* tools/gst-play.c:
  tools: gst-play: fix seeking issue
  For positive seeking segment.stop value will be -1,
  when we change rate to -1, then the stop value will be udpated
  with the current position. And then again if we change rate to 1,
  the segment.stop value does not get updated and remains as position
  where we last changed rate to -1. Hence playback stops at that point.
  In case of positive rates, call gst_element_new_seek with correct 
values
  https://bugzilla.gnome.org/show_bug.cgi?id=751213

2015-06-18 21:02:48 -0400  Nicolas Dufresne 

* gst-libs/gst/rtp/gstrtphdrext.c:
  doc: Fix gsttrtphdrext section name

2015-06-18 18:23:45 -0400  Nicolas Dufresne 

* win32/common/libgstvideo.def:
  video: Add missing new symbol to win32 def file
  Fixes make distcheck

2015-06-19 02:19:12 +1000  Jan Schmidt 

* docs/libs/gst-plugins-base-libs-sections.txt:
  Add gst_video_multiview_guess_half_aspect() to the docs

2015-06-15 16:04:55 +1000  Jan Schmidt 

* gst-libs/gst/video/video-multiview.c:
* gst-libs/gst/video/video-multiview.h:
  multiview: Add gst_video_multiview_guess_half_aspect()
  Add a utility function that, given a video size and a
  packed stereoscopic mode, attempts to guess if the video
  is packed at half resolution per view or not, since
  very few videos provide the information.

2015-06-17 17:09:46 +0200  Wim Taymans 

* gst-libs/gst/video/video-converter.c:
  video-converter: make sure we draw enough border for YUY2 formats
  Round width up to 2 so that we draw all border pixels for YUY2 formats

2015-06-17 16:43:03 +0200  Wim Taymans 

* gst-libs/gst/video/video-scaler.c:
  video-scaler: fix scaling of odd width for YUY2 formats
  We need to scale groups of 4 bytes for YUY2 formats so round up to 4.
  It's possible that there is no Y byte for the last pixel so make sure
  we clamp correctly.

2015-06-17 10:02:08 +0200  Thibault Saunier 

* gst-libs/gst/pbutils/gstdiscoverer-types.c:
  discoverer: Fix a wrong naming in the documentation
  gst_discoverer_stream_get_missing_elements_installer_details does not
  exist, one should use 
gst_discoverer_info_get_missing_elements_installer_details

2015-06-16 18:04:57 -0400  Nicolas Dufresne 

* gst-libs/gst/app/Makefile.am:
* gst-libs/gst/audio/Makefile.am:
* gst-libs/gst/pbutils/Makefile.am:
* gst-libs/gst/riff/Makefile.am:
* gst-libs/gst/rtp/Makefile.am:
* gst-libs/gst/rtsp/Makefile.am:
* gst-libs/gst/tag/Makefile.am:
* gst-libs/gst/video/Makefile.am:
  gi: Use INTROSPECTION_INIT for --add-init-section
  This new define was added to common. The new init section fixed
  compilation warning found in the init line that was spread across
  all files.

2015-06-16 17:47:24 -0400  Nicolas Dufresne 

* common:
  Automatic update of common submodule
  From 6015d26 to f74b2df

2015-06-16 22:32:49 +0100  Tim-Philipp Müller 

* tools/gst-play.c:
  tools: gst-play: error out instead of crashing if there's no playbin 
element

2015-06-16 16:08:39 -0400  Nicolas Dufresne 

* gst-libs/gst/video/video-chroma.c:
* gst-libs/gst/video/video-converter.c:
* gst-libs/gst/video/video-dither.c:
* gst-libs/gst/video/video-scaler.c:
  gi: Skip Scaler, Chroma, Conveter, Dither constructor
  Please box these types before removing the skip mark.

2015-06-16 16:07:27 -0400  Nicolas Dufresne 

* gst-libs/gst/video/gstvideometa.c:
* gst-libs/gst/video/video-overlay-composition.c:
  gi: Add (transfer none) for various video meta
  These method chains gst_buffer_add_meta() which is also transfer
  none.

2015-06-16 15:50:13 -0400  Nicolas Dufresne 

* gst-libs/gst/rtsp/gstrtspconnection.c:
  gi: Fix warnings in GstRtsp
  * The custom GSource is not boxed (skip for now)
  * The comment block has wrong name for _read_socket()

2015-06-16 15:16:33 -0400  Nicolas Dufresne 

* gst-libs/gst/riff/Makefile.am:
  gi: Don't produce gir and typlib for GstRiff
  The API does not follow the type naming convention. Re-enable
  only if one take the time to box and rename (see (rename-to SYMBOL)
  annotation) all types.

2015-06-16 14:36:44 -0400  Nicolas Dufresne 

* gst-libs/gst/fft/gstfftf32.c:
* gst

gst-plugins-good 1.5.1

[Tim-Philipp Müller]

ChangeLog
=

2015-06-07  Sebastian Dröge 


Download

https://download.gnome.org/sources/gst-plugins-good/1.5/gst-plugins-good-1.5.1.tar.xz
 (2.97M)
  sha256sum: 8dd487797b8e3f32bed5cefca09b2a1083ba00fbd69b945b5e4411f3ef00d1bc

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gst-plugins-base 1.5.1

[Tim-Philipp Müller]

ChangeLog
=

2015-06-07  Sebastian Dröge 


Download

https://download.gnome.org/sources/gst-plugins-base/1.5/gst-plugins-base-1.5.1.tar.xz
 (2.66M)
  sha256sum: b1355ee5300ed64126cc69d7f84750a4e00027d05dff89391fed3657d2b27e9a

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gstreamer 1.5.1

[Tim-Philipp Müller]

ChangeLog
=

2015-06-07  Sebastian Dröge 


Download

https://download.gnome.org/sources/gstreamer/1.5/gstreamer-1.5.1.tar.xz (3.32M)
  sha256sum: e4f1503c798ba6f0e076f1e44afecd65df00d1bb55b203755717f3e7b38e53d5

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gst-plugins-good 1.4.5

[Tim-Philipp Müller]

ChangeLog
=

2014-12-18  Sebastian Dröge 

* configure.ac:
  releasing 1.4.5

2014-12-18 12:06:04 +0100  Sebastian Dröge 

* 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:
  po: Update translations

2014-12-08 21:26:18 +0100  Patrick Radizi 

* gst/rtp/gstrtph264pay.c:
  rtph264pay: Fixes buffer leak when using SPS/PPS
  Fixes a buffer leak that would occurr if the pipeline was shutdown
  while a SPS/PPS header was being created.
  https://bugzilla.gnome.org/show_bug.cgi?id=741271

2014-11-21 11:54:18 +0100  Thibault Saunier 

* gst/deinterlace/gstdeinterlace.c:
  Deinterlace: in query_caps return only supported formats if filter is 
interlaced
  In some cases the currently set GstVideoInfo is not interlaced, but
  upstream caps are interlaced and the info is passed in the filter,
  we should take that info into account and make sure that we do not
  consider that case as a "pass through" case.
  https://bugzilla.gnome.org/show_bug.cgi?id=741407

2014-12-03 11:12:55 +0100  Thibault Saunier 

* gst/rtpmanager/gstrtpjitterbuffer.c:
  rtpjitterbuffer: Use an empty iterator in iterate_internal_link when 
no links
  We used to setup an iterator with 1 GValue set with a NULL object
  pointer which is not the normal way to do that. Instead we should make
  sure that the first call to gst_iterator_next returns 
GST_ITERATOR_DONE.

2014-12-03 17:26:56 +0100  Thibault Saunier 

* gst/rtpmanager/gstrtpsession.c:
  rtpsession: Use an empty iterator in iterate_internal_link when no 
links
  And not a NULL Iterator, so it is consistent with the way it usually
  works and avoid user to need a different code paths to handle that.

2014-12-09 16:38:38 +0100  Thibault Saunier 

* ext/vpx/gstvp8enc.c:
* ext/vpx/gstvp9enc.c:
  vpXenc: CLOCK_TIME_NONE is not a valid min_latency value
  We should just use 0 if we do not have the information

2014-12-10 11:35:29 +0100  Sebastian Dröge 

* gst/audioparsers/gstaacparse.c:
  aacparse: Also only unref caps if they're not NULL

2014-12-10 11:34:42 +0100  Sebastian Dröge 

* gst/audioparsers/gstaacparse.c:
  aacparse: gst_pad_get_allowed_caps() will return NULL if there is no 
peer

2014-12-16 13:25:01 +0100  Wim Taymans 

* sys/osxvideo/osxvideosink.m:
  osxvideosink: clear rectangle structures before use

2014-12-09 15:09:56 -0500  Nicolas Dufresne 

* sys/v4l2/gstv4l2object.c:
  v4l2object: Always set format
  Right now we try to be clever by detecting if device format have
  changed or not, and skip setting format in this case. This is valid
  behaviour with V4L2, but it's also very error prone. The rational
  for not setting these all the time is for speed, though I can't
  measure any noticeable gain on any HW I own. Also, until recently,
  we where doing get/set on the format for each format we where
  probing, making it near to impossible that the format would match.
  This also fixes bug where we where skipping frame-rate setting if
  format didn't change.
  https://bugzilla.gnome.org/show_bug.cgi?id=740636

2014-11-02 23:33:23 +  Tim-Philipp Müller 

* sys/v4l2/gstv4l2object.c:
  v4l2src: fix a couple of minor leaks

2014-09-29 11:49:45 +0200  Aurélien Zanelli 

* sys/v4l2/gstv4l2object.c:
  v4l2object: set colorspace for output devices
  When the v4l2 device is an output device, the application shall set 
the
  colorspace. So map GStreamer colorimetry info to V4L2 colorspace and 
set
  on set_format. In case we have no colorimetry information, we try to
  guess it according to pixel format and video size.
  https://bugzilla.gnome.org/show_bug.cgi?id=737579

2014-12-11 10:16:06 +0100  Patrick Radizi 

* gst/rtp/gstrtph264pay.c:
  rtph264pay: fix potential crash when shutting down
  A race condition in the s

gstreamer 1.4.5

[Tim-Philipp Müller]

ChangeLog
=

2014-12-18  Sebastian Dröge 

* configure.ac:
  releasing 1.4.5

2014-12-16 12:21:59 +0100  Wim Taymans 

* gst/gstbufferpool.c:
  bufferpool: log reason for discarded buffers
  PERFORMANCE log the reason why a buffer could not be recycled in the
  bufferpool.

2014-10-07 16:44:45 +0200  Edward Hervey 

* libs/gst/base/gstbasesink.c:
  basesink: clamp reported position based on direction
  When using a negative rate (rate being segment.rate * 
segment.applied_rate),
  we will end up reporting decreasing positions, therefore adjust the 
clamping
  against last reported value accordingly.
  Fixes positions getting properly reported with applied_rate < 0.0
  https://bugzilla.gnome.org/show_bug.cgi?id=738092

2014-11-28 14:15:30 -0300  Thiago Santos 

* libs/gst/base/gstbaseparse.c:
  baseparse: update the duration variable before emitting the bus
  Otherwise the application might still get the old value if it asks
  between the message and the real update.

2014-11-28 16:25:02 +0100  Edward Hervey 

* gst/gstelement.c:
  element: Fix doc and default implementation of send_event
  The documentation states that gst_element_send_event is to "send an 
event
  to an element".
  Therefore we *send* upstream events to a source pad and downstream 
events
  to a sink pad

2014-11-28 11:16:00 +0100  Edward Hervey 

* gst/gstelement.c:
  element: Figure default send_event direction handling
  If we get a downstream event we want to send it to a random SINK pad
  (and vice-versa).

2014-11-23 05:45:24 -0300  Thiago Santos 

* plugins/elements/gstqueue2.c:
  queue2: percentage is relative to high-percent
  When comparing percentage values, compare with 0-100 scale as it
  has already been made relative to 0-high_percent, otherwise we mark
  the queue as not buffering and report a 50% to the user. This leads to
  a buffering stall as the user assumes the queue is still buffering but
  it thinks it isn't.
  https://bugzilla.gnome.org/show_bug.cgi?id=736969

2014-11-23 05:42:51 -0300  Thiago Santos 

* plugins/elements/gstmultiqueue.c:
  multiqueue: percentage is an absolute value
  multiqueue's queues stored percent value is the percentage from 0
  to 100 (max-size-*) and should be compared with the requested limit
  (high_percentage) set by the user and not with 100% to check if
  buffering should stop. Otherwise we are only stopping buffering when 
the
  queue gets completely full.

2014-11-20 13:34:32 +0100  Wim Taymans 

* gst/gststructure.c:
* tests/check/gst/gststructure.c:
  structure: don't overread input when searching for "
  When searching for the string terminator don't read past the ending
  0-byte when escaping characters.
  Add unit test for various escaping cases.

2014-11-03 17:46:57 +  Vincent Penquerc'h 


* gst/gstpad.c:
  pad: fail dropped queries
  Previously, dropping a query from a pad probe would deem the
  query succeeded, and the caller might then assume the query's
  results are valid, and thus dereference an invalid object
  such as a GstCaps.
  We now assume dropped queries did not succeed. Dropped events
  and buffers are still deemed a success.

2014-11-12 11:30:51 +0100  Haakon Sporsheim 

* gst/gsttask.c:
* tests/check/gst/gsttask.c:
  task: Fix pause/stop race condition
  If a task thread is calling pause on it self and the
  controlling/"main" thread stops the task, it could end in a race
  where gst_task_func loops and then checks for paused after the
  controlling thread just changed the task state to stopped.
  Hence the task would actually call func again even though it was
  both paused and stopped.
  https://bugzilla.gnome.org/show_bug.cgi?id=740001



Download

https://download.gnome.org/sources/gstreamer/1.4/gstreamer-1.4.5.tar.xz (3.17M)
  sha256sum: 40801aa7f979024526258a0e94707ba42b8ab6f7d2206e56adbc4433155cb0ae

___
release-team@gnome.org
https://mail.gnome.org/mailman/listinfo/release-team
Release-team lurker? Do NOT participate in discussions.

gst-plugins-base 1.4.5

[Tim-Philipp Müller]

ChangeLog
=

2014-12-18  Sebastian Dröge 

* configure.ac:
  releasing 1.4.5

2014-12-09 22:47:31 -0300  Thiago Santos 

* gst-libs/gst/audio/gstaudiodecoder.c:
  audiodecoder: do not use fixed caps on source pad
  decoders can change the caps on their source pads, so they don't
  use fixed caps. Having fixed caps can cause renegotiation issues.

2014-12-09 22:46:42 -0300  Thiago Santos 

* gst-libs/gst/video/gstvideodecoder.c:
  videodecoder: do not use fixed caps on source pad
  decoders can change the caps on their source pads, so they don't
  use fixed caps. Having fixed caps can cause renegotiation issues.

2014-12-16 15:03:55 +0100  Sebastian Dröge 

* gst-libs/gst/video/gstvideosink.c:
* tests/check/libs/video.c:
  Revert "video: Fix non-default usage of gst_video_sink_center_rect"
  This reverts commit 899461d722e45f591eeddf33c405677170d63de4.
  There seems to be a lot of code out there that does not properly 
initialize
  the rectangles and then causes undefined behaviour. Including our 
video sinks.
  Let's keep this out of 1.4, fix everything everywhere and keep it in 
1.6

2014-12-16 12:57:55 +0100  Wim Taymans 

* sys/ximage/ximagesink.c:
* sys/xvimage/xvimagesink.c:
  ximagesink: clear src and dest rectangles
  Now that the center function also takes into account the x and y
  coordinates of the dest rectangle, better clear all the fields before
  using them.

2014-12-16 12:10:53 +0100  Song Bing 

* gst-libs/gst/video/gstvideopool.c:
* sys/ximage/ximagepool.c:
* sys/xvimage/xvimagepool.c:
  videopool: update buffer size after video alignment
  Update the new buffer size after alignment in the pool configuration
  before calling the parent set_config. This ensures that the parent 
knows
  about the buffer size that we will allocate and makes the size check
  work in the release_buffer method.
  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=741420

2014-12-15 14:10:17 +0100  Edward Hervey 

* gst-libs/gst/video/gstvideosink.c:
* tests/check/libs/video.c:
  video: Fix non-default usage of gst_video_sink_center_rect
  Make sure we take into account non-0 x/y destination rectangles

2014-12-15 09:45:43 +0100  Sebastian Dröge 

* gst/playback/gstdecodebin2.c:
  Revert "decodebin: Only emit the drain signal for the main decode 
chain, not any subchains"
  This reverts commit a391dfe17f1a325f60e1d51a6d40c1a68eb196de.
  It breaks gapless playback: 
https://bugzilla.gnome.org/show_bug.cgi?id=740045

2014-11-28 13:29:37 +  Tim-Philipp Müller 

* tests/check/Makefile.am:
  tests: don't run orc/* tests under valgrind
  They just seem to blow up for some reason that needs investigating.

2014-12-12 14:56:36 +  Tim-Philipp Müller 

* tests/check/libs/audiodecoder.c:
  tests: audiodecoder: fix broken refcounting in unit test
  The set_format vfunc does not pass ownership of the caps
  to the decoder, so we mustn't unref the caps there.
  gst_event_new_caps() does not take ownership of the caps
  passed, so we must unref the caps afterwards.
  Fixes leaks when running test in valgrind in 1.4 branch.

2014-12-11 13:45:38 +0100  Thibault Saunier 

* gst/playback/gstplaybin2.c:
  playbin: Do not mix up stream type when getting stream combiner 
element
  We were always returning the video stream combiner whatever stream 
type
  combiner was wanted.

2014-12-10 13:23:23 -0300  Thiago Santos 

* gst/playback/gstplaybin2.c:
  playbin2: always unref the combiner sinkpad when removing the srcpad
  Create a function to do the pad cleanup of the GstSourceCombine struct
  and use it to not forget to also cleanup the sink pad and fix a memory
  leak.
  https://bugzilla.gnome.org/show_bug.cgi?id=741198

2014-12-11 01:53:15 +1100  Jan Schmidt 

* gst-libs/gst/video/gstvideodecoder.h:
  videodecoder: Add GST_VIDEO_DECODER_CAST macro
  It's used in some macros already, so let's make it exist.

2014-11-25 13:31:48 +0100  Göran Jönsson 

* gst-libs/gst/rtsp/gstrtspconnection.c:
  rtspconnection: No remove child if destroyed.
  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=740730

2014-11-28 15:06:27 +0100  Mathieu Duponchelle 


* gst-libs/gst/audio/gstaudiodecoder.c:
* tests/check/libs/audiodecoder.c:
  audiodecoder: Push pending events before sending EOS.
  Segments are added to the pending events, and pushing a segment
  is mandatory before sending EOS.
  + Adds a test.
  

gst-plugins-base 1.1.90

[Tim-Philipp Müller]

ChangeLog
=

2013-09-19  Sebastian Dröge 

* configure.ac:
  releasing 1.1.90

2013-09-18 20:42:55 -0400  Olivier Crête 

* gst-libs/gst/video/gstvideodecoder.c:
  videodecoder: If there is no PTS at all, assume it starts from the 
segment start
  This is to make the multifilesrc ! pngdec case work
  https://bugzilla.gnome.org/show_bug.cgi?id=688043

2013-09-19 09:44:47 +0200  Sebastian Dröge 

* 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/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: Update translations

2013-09-18 22:05:36 +0200  Sebastian Dröge 

* gst/playback/gstplaybin2.c:
  playbin: Implement context caching for sinks that are not in playsink 
yet

2013-09-18 18:21:54 +0200  Sebastian Dröge 

* gst/playback/gstplaybin2.c:
  playbin: No need to set the GstContext on the sink before activating 
it
  This is all handled by the GstBin code now.

2013-09-04 20:21:54 -0400  Olivier Crête 

* gst-libs/gst/allocators/gstdmabuf.c:
* gst-libs/gst/allocators/gstdmabuf.h:
  dmabuf: Make it not a singleton
  Makes it easier to track how many users there are
  Also make it possible to create a dmabuf struct on systems without 
mmap,
  it just won't be possible to map it.
  https://bugzilla.gnome.org/show_bug.cgi?id=707793

2013-09-13 16:01:42 +0200  Wim Taymans 

* gst-libs/gst/rtp/gstrtpbuffer.c:
  rtpbuffer: check for valid payload type
  The payload type can't be between 72 and 76 because with the marker 
bit set,
  this could be mistaken for an RTCP packet then. We do a relaxed check 
and
  only refuse 72-76 when the marker bit is set. The effect is that when
  we try to map an RTCP packet as an RTP packet, we will certainly fail.

2013-09-13 09:17:38 +0100  Tim-Philipp Müller 

* configure.ac:
  configure: rely solely on pkg-config to find libogg and libvorbis
  And get rid of AS_SCRUB_INCLUDES
  https://bugzilla.gnome.org/show_bug.cgi?id=707658

2013-09-12 12:23:35 +0200  Wim Taymans 

* gst/videoscale/vs_4tap.c:
  videoscale: fix 4tap for RGB15 and RGB16
  Fix component ordering, it's wrong in both the scanline and merge
  function so it cancels eachother out and isn't really a except for
  loss of precision of the green component.
  Fix calculation of the filter weight

2013-09-10 17:02:44 +0200  Wim Taymans 

* gst/videoscale/vs_scanline.c:
  videoscale: optimize merge for RGB15 and RGB16

2013-09-10 16:55:25 +0200  Wim Taymans 

* gst/videoscale/vs_4tap.c:
  videoscale: remove redundant MAX
  The checks above make it inpossible for the value to be smaller than
  what we check against with the MAX call.

2013-09-12 09:42:36 +0200  Sebastian Dröge 

* gst-libs/gst/audio/gstaudiodecoder.c:
* gst-libs/gst/audio/gstaudioencoder.c:
  audioencoder/decoder: Mark pads as requiring reconfiguration again if 
negotiation fails
  Otherwise we might end up in non-optimal configuration, especially
  when a flush happened during reconfiguration.

2013-09-12 09:35:00 +0200  Sebastian Dröge 

* gst-libs/gst/video/gstvideodecoder.c:
* gst-libs/gst/video/gstvideoencoder.c:
  videodecoder/videoencoder: Mark pads as requiring reconfiguration 
again if negotiation fails
  Otherwise we might end up in non-optimal configuration, especially
  when a flush happened during reconfiguration.

2013-09-10 21:44:33 +0200  Matej Knopp 

* gst-libs/gst/pbutils/descriptions.c:
  pbutils: Add description for TechSmith Screen Capture 2
  https://bugzilla.gnome.org/show_bug.cgi?id=707878

2013-09-10 21:44:21 +0200  Matej Knopp 

* gst-libs/gst/riff/riff-media.c:
  riff: Add support for TechSmith Screen Capture 2
  https://bugzilla.gnome.org/show_bug.cgi?id=707878

2013-09-06 15:36:12 -0300  Thiago Santos 

* ext/ogg/gstoggdemux.c:
  oggdemux: check for full eos after a pad goes eos in push mode
  After a pad is on EOS, verify if all pads are EOS and return
  upstre

gstreamer 1.1.90

[Tim-Philipp Müller]

öge 

* gst/gstmeta.h:
  meta: Deprecate GST_META_TAG_MEMORY
  The GQuarks are not exported by any public API

2013-08-22 00:02:28 +0200  Mathieu Duponchelle 

* docs/gst/gstreamer-sections.txt:
* gst/gstmeta.h:
* win32/common/libgstreamer.def:
  meta: Add a #define for memory metadata

2013-08-22 00:01:44 +0200  Mathieu Duponchelle 

* gst/gstmeta.c:
* libs/gst/base/gstbasetransform.c:
  basetransform: implement a default transform_meta. If a metadata has 
no dependency as shown by the tags, copy it.

2013-08-22 21:32:36 +0200  Mathieu Duponchelle 

* gst/gstmeta.c:
* gst/gstmeta.h:
  meta: API: Add gst_meta_api_type_get_tags() to get all meta tags.

2013-09-09 14:21:56 +0200  Sebastian Dröge 

* tests/check/elements/capsfilter.c:
  tests/capsfilter: Fix memory leak and compare caps directly instead 
of strcmp()

2013-09-06 23:03:54 +0200  Sebastian Rasmussen 

* tests/check/elements/capsfilter.c:
  tests/capsfilter: Test caps-related queries and property

2013-09-06 15:09:46 -0300  Gustavo Noronha Silva 

* plugins/elements/gstqueue2.c:
  Update the buffering state before stalling for more data
  In some cases the wait for more data was happening without updating
  the buffering state, meaning the API user would not be able to notice
  it should pause the pipeline and update UI to indicate that is the
  case, the video would likely stutter instead.
  https://bugzilla.gnome.org/show_bug.cgi?id=707648

2013-09-04 15:28:10 -0300  Thiago Santos 

* libs/gst/base/gstbasesrc.c:
  basesrc: preserve seqnum on segments after seeks
  The seqnum of the segment after a seek should be the same of
  the seek event. Downstream elements might rely on seqnums to
  identify events related to a seek.
  This is particularly important when a demuxer maps a TIME seek
  into a BYTES seek for upstream and it needs to identify the
  corresponding segment event and map it back into TIME to push
  downstream, possibly using the values from the original seek
  event.
  https://bugzilla.gnome.org/show_bug.cgi?id=707530

2013-09-05 14:14:42 +0200  Zaheer Abbas Merali 

* libs/gst/base/gstcollectpads.c:
  collectpads: Don't unref NULL GstCollectData
  If a pad is removed while a collectpads element (say adder) is in a 
chain
  function waiting to be collected, there is a possibility that an 
unref happens
  on a NULL pointer.
  https://bugzilla.gnome.org/show_bug.cgi?id=707536

2013-09-04 17:11:20 +0200  Christian Fredrik Kalager Schaller 


* gstreamer.spec.in:
  Remove PyXML from spec file, it is not longer needed

2013-09-04 14:40:57 +0200  Sebastian Dröge 

* plugins/elements/gsttypefindelement.c:
  typefind: Add missing break after handling the GAP event
  Thanks to Edward Hervey for noticing.

2013-09-04 09:18:55 +0100  Tim-Philipp Müller 

* scripts/gst-plot-timeline.py:
* tools/Makefile.am:
* tools/gst-plot-timeline.py:
  tools: move gst-plot-timeline.py into scripts directory
  So it's not in PATH in an uninstalled setup (thwarting
  gst-play autocompletion).

2013-09-03 23:59:05 +0200  Matej Knopp 

* plugins/elements/gstmultiqueue.c:
  multiqueue: Don't reduce single queue visible size below its current 
level
  If the multiqueue has automatically grown chances are good that
  we will cause the pipeline to starve if the maximum level is reduced
  below that automatically grown size.
  https://bugzilla.gnome.org/show_bug.cgi?id=707156

2013-09-02 13:53:51 +0200  Sebastian Dröge 

* plugins/elements/gstoutputselector.c:
  outputselector: Don't adjust segment->start to the current time when 
switching pads
  This does not make any sense at all and breaks timestamp->running_time
  calculations in unpredictable ways.
  https://bugzilla.gnome.org/show_bug.cgi?id=707130

2013-08-29 23:18:31 +0200  Mathieu Duponchelle 

* plugins/elements/gstcapsfilter.c:
  capsfilter: Delete link directly in pending_events.
  When removing a segment event.
  https://bugzilla.gnome.org/show_bug.cgi?id=707088

2013-08-29 11:07:38 +0100  Tim-Philipp Müller 

* libs/gst/base/gstbasesink.c:
  basesink: demote log message, don't spam INFO level when handling 
buffer lists

2013-08-28 13:26:28 +0200  Sebastian Dröge 

* configure.ac:
  Back to development



Download

http://download.gnome.org/sources/gstreamer/1.1/gstreamer-1.1.90.tar.xz (3.04M)
  sha256sum: b38e9cb70e4d20f9b17d2421da639eafe17dd917f7a32aaac3807ade3c165813

__

gst-plugins-good 1.1.90

[Tim-Philipp Müller]

ChangeLog
=

2013-09-19  Sebastian Dröge 

* configure.ac:
  releasing 1.1.90

2013-09-19 09:45:18 +0200  Sebastian Dröge 

* po/cs.po:
* po/nl.po:
* po/pl.po:
* po/uk.po:
* po/vi.po:
  po: Update translations

2013-09-11 14:27:02 -0400  Olivier Crête 

* sys/v4l2/gstv4l2bufferpool.c:
  v4l2bufferpool: dmabuf is not a singleton anymore
  https://bugzilla.gnome.org/show_bug.cgi?id=707793

2013-09-16 13:53:45 -0300  Thiago Santos 

* ext/soup/gstsouphttpsrc.c:
  souphttpsrc: do not do http requests in READY
  HEAD requests to discover if the server is seekable shouldn't be done 
in
  READY as it might lock the main thread that is doing the state change.
  https://bugzilla.gnome.org/show_bug.cgi?id=705371

2013-09-18 16:32:28 +0200  Wim Taymans 

* gst/rtpmanager/gstrtpjitterbuffer.c:
  rtpjitterbuffer: reevaluate the current timer after timeout
  When we trigger the timeout logic of a timer, reevaluate it because 
it is
  possible that it still has the lowest timeout.

2013-09-18 16:31:26 +0200  Wim Taymans 

* gst/rtpmanager/gstrtpjitterbuffer.c:
  rtpjitterbuffer: don't update time when unscheduled
  Don't try to estimate the current time when we got unscheduled.

2013-09-18 16:29:37 +0200  Wim Taymans 

* gst/rtpmanager/gstrtpjitterbuffer.c:
  rtpjitterbuffer: init packet spacing on first buffer
  Already init the packet spacing variables on the first buffer so that 
we can
  calculate the spacing on the second buffer already.

2013-09-18 15:08:45 +0200  Wim Taymans 

* tests/check/elements/rtpjitterbuffer.c:
  tests: fix comments

2013-09-18 14:57:09 +0200  Wim Taymans 

* gst/rtpmanager/gstrtpjitterbuffer.c:
  rtpjitterbuffer: push the lost event from the timer thread
  Instead of pushing the lost event from the chain function, schedule a 
timeout
  that will push the lost event from the timer thread. This avoid 
blocking the
  upstream thread while we push and sync the event.

2013-09-18 14:23:55 +0200  Wim Taymans 

* tests/check/elements/rtpjitterbuffer.c:
  rtpjitterbuffer: add another test
  The test is modified slightly because the late lost packets are only
  generated now when a large gap is received.

2013-09-18 14:12:47 +0200  Wim Taymans 

* gst/rtpmanager/gstrtpjitterbuffer.c:
* tests/check/elements/rtpjitterbuffer.c:
  rtpjitterbuffer: round gap duration to multiple of duration
  Make sure the gap duration in the lost event is a multiple of the 
packet
  duration.
  Enable another test.

2013-09-18 12:29:38 +0200  Wim Taymans 

* gst/rtpmanager/gstrtpjitterbuffer.c:
* tests/check/Makefile.am:
* tests/check/elements/rtpjitterbuffer.c:
  rtpjitterbuffer: keep track of duration
  Keep track of the estimated duration of missing packets and use it in 
the lost
  event.
  Enable another unit test

2013-09-18 11:59:28 +0200  Wim Taymans 

* gst/rtpmanager/gstrtpjitterbuffer.c:
* tests/check/elements/rtpjitterbuffer.c:
  rtpjitterbuffer: handle large gaps with one lost event
  When we have a large number of missing packets, generate one lost 
event for all
  the packets that have no chance of being pushed out in time.
  Fix and activate unit test for large gaps.

2013-09-18 11:56:38 +0200  Wim Taymans 

* gst/rtpmanager/gstrtpjitterbuffer.c:
  rtpjitterbuffer: refactor lost event sending
  Also make sure we only increment the expected seqnum and last
  output timestamp.

2013-09-17 23:21:09 +0200  Wim Taymans 

* gst/rtpmanager/gstrtpjitterbuffer.c:
  jitterbuffer: refactor timeout triggers

2013-09-17 23:03:45 +0200  Wim Taymans 

* gst/rtpmanager/gstrtpjitterbuffer.c:
  jitterbuffer: simplify the timeout code
  Keep track of the current time in the timeout loop.
  Loop over all timers and trigger all the expired ones, we can do this 
in the
  same loop that selects the new best timer.

2013-09-17 23:01:17 +0200  Wim Taymans 

* gst/rtpmanager/gstrtpjitterbuffer.c:
  jitterbuffer: rearrange timer update code
  Also update the timers when retransmission is disabled. We need to
  do this because when we added LOST timers when we detected missing 
packets and
  we need to remove those timers when the packet finally arrives.

2013-09-17 22:02:04 +0100  Tim-Philipp Müller 

* gst/videomixer/Makefile.am:
  videomixer: link to libm for maths stuff
  Fixes undefined references to rint and pow on ubuntu
  build bot.

2013-09-17 15:19:42 +0200  Wim Taymans 

*

gst-plugins-good 1.1.4

[Tim-Philipp Müller]

ChangeLog
=

2013-08-28  Sebastian Dröge 

* configure.ac:
  releasing 1.1.4

2013-08-28 12:32:10 +0200  Sebastian Dröge 

* po/pt_BR.po:
  po: update translations

2013-08-27 15:25:16 +0200  Wim Taymans 

* 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 

* 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 

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

2013-08-27 09:34:46 +0200  Wim Taymans 

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

2013-08-27 09:33:03 +0200  Wim Taymans 

* 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 

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

2013-08-26 11:50:27 +0200  Wim Taymans 

* 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 

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

2013-08-20 22:12:03 +0200  Mathieu Duponchelle 

* 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 

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

2013-08-23 15:49:47 +0100  Tim-Philipp Müller 

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

2013-08-23 15:29:28 +0100  Tim-Philipp Müller 

* 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 

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

2013-08-23 12:07:55 +0200  Wim Taymans 

* 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 

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

2013-08-23 11:17:45 +0200  Michael Olbrich 

* 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 

* 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 

* 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 

* 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 

* 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 

* 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 

* 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 

* configure.ac:
  configure: Fix bz2 configure check for Windows
  Due to function decorations on Windows AC_CHECK_LIB can't be us

gstreamer 1.1.4

[Tim-Philipp Müller]

ChangeLog
=

2013-08-28  Sebastian Dröge 

* configure.ac:
  releasing 1.1.4

2013-08-28 12:30:00 +0200  Sebastian Dröge 

* po/af.po:
* po/az.po:
* po/be.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/nb.po:
* po/nl.po:
* po/pl.po:
* po/pt_BR.po:
* po/ro.po:
* po/ru.po:
* po/rw.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_TW.po:
  po: update translations

2013-08-27 09:31:22 +0200  Alessandro Decina 

* plugins/elements/gstfilesink.c:
  filesink: please gcc (avoid a warn_unused_result warning)

2013-08-27 07:51:35 +0200  Alessandro Decina 

* plugins/elements/gstfilesink.c:
* tests/check/elements/filesink.c:
  filesink: flush (discard data) on FLUSH_STOP
  Reset the write position to 0 and truncate the file on FLUSH_STOP.

2013-08-27 07:05:11 +0200  Alessandro Decina 

* tests/check/elements/filesink.c:
  tests: filesink: small refactoring

2013-08-26 13:19:10 +0100  Tim-Philipp Müller 

* tools/gst-launch.c:
  tools: gst-launch: don't print properties being reset when shutting 
down
  It's just noise.

2013-08-22 19:01:32 +0200  Edward Hervey 

* libs/gst/base/gstbasetransform.c:
  basetransform: Don't push out identical caps
  This avoids triggering plenty of extra code/methods/overhead 
downstream when
  we can just quickly check whenever we want to set caps whether they 
are
  identical or not
  https://bugzilla.gnome.org/show_bug.cgi?id=706600

2013-08-21 12:21:43 +0100  Tim-Philipp Müller 

* gst/gstsample.c:
  docs: flesh out gst_sample_get_buffer() a little
  https://bugzilla.gnome.org/show_bug.cgi?id=706478

2013-08-20 23:59:29 -0700  Kerrick Staley 

* gst/parse/grammar.y:
  parse: make grammar.y work with Bison 3
  YYLEX_PARAM is no longer supported in Bison 3.
  https://bugzilla.gnome.org/show_bug.cgi?id=706462

2013-08-20 17:15:41 +0900  Wonchul Lee 

* gst/gstsample.h:
  sample: Add gst_sample_copy()
  https://bugzilla.gnome.org/show_bug.cgi?id=706454

2013-08-19 14:55:22 -0400  Olivier Crête 

* gst/gstbuffer.c:
* tests/check/gst/gstbuffer.c:
  buffer: Fix gst_buffer_memcmp() where the buffer is smaller than size
  Also add unit tests for gst_buffer_memcmp
  https://bugzilla.gnome.org/show_bug.cgi?id=706162

2013-08-20 17:06:49 +0100  Tim-Philipp Müller 

* gst/gstutils.c:
  docs: flesh out gst_element_query_{duration,position} docs a bit

2013-08-14 16:18:59 +0100  Matthieu Bouron 

* gst/gsttaglist.c:
* gst/gsttaglist.h:
  taglist: handle publisher and interpreted-by tags
  https://bugzilla.gnome.org/show_bug.cgi?id=705999

2013-08-20 13:58:24 +0200  Sebastian Dröge 

* gst/gstpluginloader.c:
  pluginloader: Don't call memcpy() with NULL src and 0 length

2013-08-20 10:16:41 +0200  Sebastian Dröge 

* plugins/elements/gstqueue.c:
  queue: Properly unlock the sinkpad streaming thread when deactivating 
the pad
  https://bugzilla.gnome.org/show_bug.cgi?id=705835

2013-08-20 10:16:05 +0200  Sebastian Dröge 

* plugins/elements/gstqueue2.c:
  queue2: Properly unlock the sinkpad streaming thread when 
deactivating the pad
  https://bugzilla.gnome.org/show_bug.cgi?id=706360

2013-08-19 16:38:50 +0200  Sebastian Dröge 

* plugins/elements/gstmultiqueue.c:
  multiqueue: Clean up after the streaming thread has stopped
  https://bugzilla.gnome.org/show_bug.cgi?id=705835

2013-08-19 16:38:40 +0200  Sebastian Dröge 

* plugins/elements/gstqueue2.c:
  queue2: Clean up after the streaming thread has stopped
  https://bugzilla.gnome.org/show_bug.cgi?id=705835

2013-08-19 16:38:16 +0200  Sebastian Dröge 

* plugins/elements/gstqueue.c:
  queue: Clean up after the streaming thread has stopped
  https://bugzilla.gnome.org/show_bug.cgi?id=705835

2013-07-01 14:04:46 -0600  Brendan Long 

* gst/gstparse.h:
* gst/gstutils.c:
* gst/parse/grammar.y:
  parse: Add GST_FLAG_NO_SINGLE_ELEMENT_BINS
  This makes gst_parse_bin_from_description() return an element instead 
of
  a bin if there's only one element

gst-plugins-base 1.1.4

[Tim-Philipp Müller]

ChangeLog
=

2013-08-28  Sebastian Dröge 

* configure.ac:
  releasing 1.1.4

2013-08-28 12:31:23 +0200  Sebastian Dröge 

* 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/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: update translations

2013-08-27 15:03:54 +0200  Andoni Morales Alastruey 

* gst-libs/gst/video/gstvideoencoder.c:
  videoencoder: fix forwarding of GstForceKeyUnit events
  Use the frame id to match the output forced keyframe with
  the event that forced it.
  https://bugzilla.gnome.org/show_bug.cgi?id=706885

2013-08-26 11:44:06 +0100  Tim-Philipp Müller 

* ext/vorbis/gstvorbisenc.c:
* ext/vorbis/gstvorbisenc.h:
  vorbisenc: remove unused variables

2013-08-26 11:47:40 +0200  Wim Taymans 

* gst-libs/gst/rtp/gstrtcpbuffer.c:
  rtcpbuffer: do additional packet checks
  Check the packet size and avoid crashing on malformed packets.
  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=655727

2013-08-26 11:46:11 +0200  Wim Taymans 

* gst-libs/gst/rtp/gstrtcpbuffer.c:
  rtcpbuffer: improve bye parsing
  It is an error to ask for a non-existing BYE SSRC, the caller should
  check the SSRC count first.

2013-08-23 18:06:36 +0200  Michael Olbrich 

* gst-libs/gst/allocators/gstdmabuf.c:
  dmabuf: fix mmap counting
  A successful gst_dmabuf_mem_map must always increment the mmap count.
  Otherwise the first gst_dmabuf_mem_unmap will unmap the memory and all
  other user will access unmapped memory.
  https://bugzilla.gnome.org/show_bug.cgi?id=706680

2013-08-26 08:08:32 +0200  Alessandro Decina 

* ext/vorbis/gstvorbisenc.c:
  vorbisenc: implement flushing

2013-08-25 10:25:43 +0200  Sebastian Dröge 

* gst-libs/gst/video/gstvideodecoder.c:
* gst-libs/gst/video/gstvideoencoder.c:
  videoencoder: Make sure to return TRUE if the same caps are set again

2013-08-23 19:47:57 +0100  Tim-Philipp Müller 

* gst/audioconvert/gstaudioconvert.c:
  audioconvert: improve fixate_format function readability even more
  Do the flags comparisons only once and re-use the result.

2013-08-23 19:41:32 +0100  Tim-Philipp Müller 

* gst/audioconvert/gstaudioconvert.c:
  audioconvert: simplify fixate_format function some more
  If we have no output format yet, any format will do. The
  !out_info condition existed in every path, so just split
  it our for clarity. KISS.

2013-08-23 19:05:41 +0100  Tim-Philipp Müller 

* gst/audioconvert/gstaudioconvert.c:
  audioconvert: make fixate function more readable
  Use some variables to replace accessor macros to make code
  a little bit mor readable.

2013-08-23 18:52:44 +0100  Tim-Philipp Müller 

* gst/audioconvert/gstaudioconvert.c:
  audioconvert: remove unnecessary deep nesting in fixate function
  Makes it easier to read and removes two levels of indentation.

2013-08-23 19:20:03 +0200  Sebastian Dröge 

* gst-libs/gst/video/gstvideoencoder.c:
  videoencoder: Only set the caps when they actually changed

2013-08-23 19:17:16 +0200  Sebastian Dröge 

* gst-libs/gst/audio/gstaudioencoder.c:
  audioencoder: Simplify pushing of pending events during negotiation
  And also don't send the same caps twice.

2013-08-23 19:10:48 +0200  Sebastian Dröge 

* gst-libs/gst/audio/gstaudiodecoder.c:
  audiodecoder: Fix last commit and simplify code a lot

2013-08-23 18:51:59 +0200  Sebastian Dröge 

* gst/audioconvert/gstaudioconvert.c:
  audioconvert: If we have to lose precision, try to lose as less 
precision as possible
  https://bugzilla.gnome.org/show_bug.cgi?id=706624

2013-08-23 16:59:30 +0200  Edward Hervey 

* gst-libs/gst/audio/gstaudiodecoder.c:
  audiodecoder: Fix previous commit
  (sorry)

2013-08-23 15:22:43 +0200  Edward Hervey 

* gst-libs/gst/video/gstvideodecoder.c:
  videocoder: Don't push out identical caps
  This avoids triggering plenty of extra code/methods/overhead 
downstream when
  w

gst-plugins-good 1.0.10

[Tim-Philipp Müller]

News


Changes since 1.0.9:

  * qtdemux: add variant field to H.263 caps, fixing H.263 playback
  * flvdemux: memory leak fix
  * goom: fix issue with caps writability

Bugs fixed since 1.0.9:

  * 705142 : flvdemux: leaks stream_id string
  * 705475 : goom: negotiation fails when sink has ANY caps
  * 705477 : directsoundsink: Glitches when playing S8
  * 705550 : rtpvp8depay: mark key frames and delta frames properly
  * 706422 : jpegenc is sending wrong buffer on specific input/resolution

ChangeLog
=
http://download.gnome.org/sources/gst-plugins-good/1.0/gst-plugins-good-1.0.10.changes
  (1.56K)

Download

http://download.gnome.org/sources/gst-plugins-good/1.0/gst-plugins-good-1.0.10.tar.xz
 (2.61M)
  sha256sum: c60342d6080ac6f794c5c2427dfbdee9140a2f67b82e7e945e286a2d416428ae

___
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Release-team lurker? Do NOT participate in discussions.

gstreamer 1.0.10

[Tim-Philipp Müller]

News


Changes since 1.0.9:

  * parse: make gst-launch syntax parser work with Bison 3
  * memory: only copy the requested part instead of the complete source 
memory for default system memory
  * queue2: update buffering when changing capacity 
  * g-i: add bindings-friendly buffer.extract() function
  * g-i: miscellaneous gobject-introspection annotation fixes (queries, 
pipeline)

Bugs fixed since 1.0.9:

  * 705630 : directcontrolbinding: Wrong behaviour of control bindings with 
properties that span +-INT_MAX
  * 705678 : sysmem allocator copies too much
  * 705751 : gst_pipeline_use_clock: clock should allow-none
  * 706462 : parse: fix build with Bison 3

ChangeLog
=
http://download.gnome.org/sources/gstreamer/1.0/gstreamer-1.0.10.changes  
(3.67K)

Download

http://download.gnome.org/sources/gstreamer/1.0/gstreamer-1.0.10.tar.xz (3.00M)
  sha256sum: 8e0aa9f41370586171a2616326fbc508bc4b61ffc4d55b2a8c4c3459d0cc1130

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gst-plugins-base 1.0.10

[Tim-Philipp Müller]

News


Changes since 1.0.9:

  * rtpbasedepayload: mark discontinuities after packet loss properly
  * audioconvert: if we have to lose precision, try to lose as little 
precision as possible
  * gio: fix very inefficient data cache handling

Bugs fixed since 1.0.9:

  * 655727 : rtp: segfault in gst_rtcp_packet_get_rb()
  * 705415 : gio: incredibly inefficient when playing mp3s from smb servers
  * 706624 : audioconvert: Converts S32LE to S8 although S16LE would be 
possible too

ChangeLog
=
http://download.gnome.org/sources/gst-plugins-base/1.0/gst-plugins-base-1.0.10.changes
  (1.72K)

Download

http://download.gnome.org/sources/gst-plugins-base/1.0/gst-plugins-base-1.0.10.tar.xz
 (2.26M)
  sha256sum: 9754e83b2a3ff61266112915bfc84c895c4d145519ad9a3381edceb216bc97e8

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gst-plugins-good 1.0.9

[Tim-Philipp Müller]

News


Changes since 1.0.8:

  * udpsink, multiudpsink: unbreak IPv4 fallback on systems without
IPv6 support
  * deinterlace: fix changing 'mode' property on-the-fly to
enable/disable deinterlacing
  * rtp: fix autoplugging of depayloaders if there's only one of
the payload number and the encoding-name
  * pulse: fixes for alaw/mulaw
  * matroskademux, avidemux: fix some buffer leaks
  * flacenc: fix image tag handling

Bugs fixed since 1.0.8:

  * 641599 : deinterlace: changing " mode " property on-the-fly doesn't work
  * 695981 : Segmentation fault on gst_v4l2_buffer_pool_process after 
sending " start-capture " signal to camerabin2 just after setting the pipeline 
state to PLAYING
  * 704162 : aacparse: Too verbose when parsing LOAS stream
  * 704533 : regression: rtppcmudpay no longer autoplugged for rtsp stream 
with mulaw audio
  * 704553 : (dyn|multi)udpsink: broken on !IPv6 systems
  * 705018 : flacenc: segfault when trying to encode from alac decoded file

ChangeLog
=
http://download.gnome.org/sources/gst-plugins-good/1.0/gst-plugins-good-1.0.9.changes
  (9.81K)

Download

http://download.gnome.org/sources/gst-plugins-good/1.0/gst-plugins-good-1.0.9.tar.xz
 (2.60M)
  sha256sum: cfa2e617a76f93e9ddd4ae1109297e93fb4a06b152042b996231234a72c5a5ff

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gstreamer 1.0.9

[Tim-Philipp Müller]

News


Changes since 1.0.8:

  * basesink: when we asynchronously go from READY to PLAYING, also
call the state change function so that subclasses can update
their state for PLAYING, esp. audio sinks added to the pipeline
dynamically
  * bindings: gobject-introspection annotation fixes
  
Bugs fixed since 1.0.8:
  
  * 702282 : basesink: makes element go to PLAYING without 
PAUSED_TO_PLAYING transition

ChangeLog
=
http://download.gnome.org/sources/gstreamer/1.0/gstreamer-1.0.9.changes  (1.61K)

Download

http://download.gnome.org/sources/gstreamer/1.0/gstreamer-1.0.9.tar.xz (2.99M)
  sha256sum: 17a2c60a82baa461ef685ad3de187edb9c03a2c7e07513daba58a5a32efacaa3

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gst-plugins-base 1.0.9

[Tim-Philipp Müller]

News


Changes since 1.0.8:

  * subparse: re-enable sami parser
  * tagdemux: fix ACCURATE seeks in push mode
  * multisocketsink, multifdsink: fix handling of partial writes and 
WOULD_BLOCK errors, and other fixes
  * audiodecoder: fix input caps leak

Bugs fixed since 1.0.8:

  * 693056 : subparse: re-enable sami support
  * 704291 : riff: PNG AVI file does not play
  * 704301 : tagdemux: Breaks seeking forward fails if ACCURATE seek and 
push sources
  * 704926 : audiodecoder: leaks input caps

ChangeLog
=
http://download.gnome.org/sources/gst-plugins-base/1.0/gst-plugins-base-1.0.9.changes
  (4.42K)

Download

http://download.gnome.org/sources/gst-plugins-base/1.0/gst-plugins-base-1.0.9.tar.xz
 (2.26M)
  sha256sum: 963e3b83d651661f495ca2e44ccd2d5c61e986e9d7706246e568276689a372ea

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Release-team lurker? Do NOT participate in discussions.

gstreamer 1.1.2

[Tim-Philipp Müller]

ChangeLog
=

2013-07-11  Sebastian Dröge 

* configure.ac:
  releasing 1.1.2

2013-07-10 15:52:10 +0200  Sebastian Dröge 

* gst/gstbin.c:
  bin: Always forward clock-lost message if we're not a top-level bin
  This makes sure that no bin misses the clock-lost messages, 
independent
  of the state, and could return an old, non-working clock from
  gst_bin_provide_clock_func().
  https://bugzilla.gnome.org/show_bug.cgi?id=701997

2013-07-10 14:30:31 +0200  Sebastian Dröge 

* plugins/elements/gstinputselector.c:
  inputselector: Keep previous active sinkpad around until we're done 
with it
  Otherwise we'll send a new segment event downstream for each buffer.

2013-07-08 15:26:38 +0100  Tim-Philipp Müller 

* gst/gstallocator.c:
  allocator: fix type of gst_memory_alignment to match declaration
  Fixes compiler warnings such as
  gstallocator.c:61:8: error: conflicting types for 
'gst_memory_alignment'
  ../gst/gstallocator.h:52:18: note: previous declaration of 
'gst_memory_alignment' was here

2013-07-05 21:36:27 +0200  Piotr Drąg 

* po/POTFILES.in:
  po: update POTFILES.in
  https://bugzilla.gnome.org/show_bug.cgi?id=703682

2013-07-04 20:39:26 -0400  Thibault Saunier 

* libs/gst/base/gstbasesrc.c:
  basesrc: Do not lock a mutex that does not exist
  The GST_LIVE_LOCK is on GstBaseSrc, not on its source pad.

2013-07-03 21:23:44 +0200  Wim Taymans 

* libs/gst/base/gstbaseparse.c:
  baseparse: reset PTS after seek
  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702778

2013-07-03 13:03:49 -0400  Nicolas Dufresne 

* gst/gstallocator.c:
* gst/gstevent.c:
* gst/gstghostpad.c:
* gst/gstinfo.h:
* gst/gstmessage.c:
* gst/gstminiobject.c:
* gst/gstpad.c:
* gst/gstplugin.c:
* gst/gsttaglist.c:
* gst/gsttypefind.c:
* gst/gstutils.c:
* libs/gst/base/gstcollectpads.c:
* libs/gst/base/gsttypefindhelper.c:
* libs/gst/base/gsttypefindhelper.h:
  Add few missing allow-none annotation

2013-07-03 09:27:13 +0100  Tim-Philipp Müller 

* scripts/gst-uninstalled:
  gst-uninstalled: add new -bad mpegts lib
  And remove signalprocessor/video libs from -bad which have gone
  away or were merged into -base.

2013-07-01 20:35:21 -0400  Olivier Crête 

* plugins/elements/gstfunnel.c:
* plugins/elements/gstfunnel.h:
* tests/check/elements/funnel.c:
  funnel: Re-push all sticky events when buffers come from a different 
pad
  Don't special case segment/caps, just push all sticky events when 
they are
  received on the currently active pad or when the active pad changes.

2013-07-01 20:21:10 -0400  Olivier Crête 

* plugins/elements/gstfunnel.c:
  funnel: Use default pad function for upstream event/queries
  The default functions in 1.x already do the right thing

2013-07-01 20:18:58 -0400  Olivier Crête 

* tests/check/elements/funnel.c:
  tests: Remove funnel pad_alloc test

2013-07-01 20:07:03 -0400  Olivier Crête 

* libs/gst/check/gstcheck.h:
  check: Change stream_id parameter name to match GtkDoc

2013-07-01 11:10:00 +0200  Jonas Holmberg 

* docs/libs/gstreamer-libs-sections.txt:
* libs/gst/check/Makefile.am:
* libs/gst/check/gstcheck.c:
* libs/gst/check/gstcheck.h:
* tests/check/elements/funnel.c:
  check: Added gst_check_setup_events_with_stream_id()
  Added a new function gst_check_setup_events_with_stream_id(), since
  gst_check_setup_events() does not work with multiple pads.
  https://bugzilla.gnome.org/show_bug.cgi?id=703377

2013-06-30 18:39:03 +0200  Sebastian Dröge 

* gst/gstpad.c:
  pad: Don't consider flushing pads as needing reconfiguration
  Renegotiation and reconfiguration will fail because all queries
  and events won't be accepted by the pad if it's flushing. In the
  best case this just causes unneeded work and spurious warnings in
  the debug logs, in the worst case it causes elements to fail 
completely.

2013-06-24 23:25:51 +0200  Wim Taymans 

* plugins/elements/gstqueue2.c:
  queue2: only block serialized query when it's safe
  We must be certain that we don't cause a deadlock when blocking the 
serialized
  queries. One such deadlock can happen when we are buffering and 
downstream is
  blocked in preroll and a serialized query arrives. Downstream will 
not unblock
  (and allow our query to execute) until we complete buffering and 
buffering will
  not complete until we can answer the query..
  

gst-plugins-good 1.1.2

[Tim-Philipp Müller]

57079

2013-06-26 14:58:53 +0200  Wim Taymans 

* gst/rtsp/gstrtspsrc.c:
* gst/rtsp/gstrtspsrc.h:
  rtspsrc: avoid some flushes

2013-06-26 14:41:00 +0200  Wim Taymans 

* gst/rtsp/gstrtspsrc.c:
  rtspsrc: handle data message when waiting for reply
  When we are waiting for a server reply, handle data messages instead 
of
  ignoring them.

2013-06-26 14:27:34 +0200  Wim Taymans 

* gst/rtsp/gstrtspsrc.c:
  rtspsrc: handle data messages in separate method
  Refactor and make a method to handle a data message.

2013-06-25 20:36:18 +0200  Wim Taymans 

* gst/rtsp/gstrtspsrc.c:
  rtspsrc: add some more docs to handle-request signal
  See https://bugzilla.gnome.org/show_bug.cgi?id=702705

2013-06-10 17:20:30 -0400  Youness Alaoui 

* gst/rtsp/gstrtspsrc.c:
  Send a clock_provide message on the bus when we get a netclock

2013-06-10 17:20:14 -0400  Youness Alaoui 

* gst/rtsp/gstrtspsrc.c:
* gst/rtsp/gstrtspsrc.h:
  rtspsrc: Expose use-pipeline-clock property

2013-06-24 17:11:35 +0200  Wim Taymans 

* gst/udp/gstmultiudpsink.c:
  udpsink: bind to the given interface
  Actually call BINDTODEVICE to bind to the interface as given by the
  property.
  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702819

2013-06-22 10:59:17 +0200  Sebastian Dröge 

* ext/vpx/gstvp8dec.c:
  vp8dec: Error out gracefully if we get an unsupported color format
  In theory we can only get I420 though, just to be on the safe side.

2013-06-22 10:57:41 +0200  Sebastian Dröge 

* ext/vpx/gstvp9dec.c:
* ext/vpx/gstvp9enc.c:
  vp9: Add support for YV12, Y42B and Y444 color formats
  The encoder does not work with Y42B and Y444 yet it seems.

2013-06-22 10:26:18 +0200  Sebastian Dröge 

* ext/vpx/gstvp9dec.c:
  vp9dec: Update default postproc settings from vp9_dx_iface.c

2013-06-21 13:11:32 +0200  Sebastian Dröge 

* gst/matroska/matroska-demux.c:
* gst/matroska/matroska-ids.h:
* gst/matroska/matroska-mux.c:
* gst/matroska/webm-mux.c:
  matroska: Add initial VP9 support

2013-06-21 13:07:30 +0200  Sebastian Dröge 

* configure.ac:
* ext/vpx/Makefile.am:
* ext/vpx/gstvp9dec.c:
* ext/vpx/gstvp9dec.h:
* ext/vpx/gstvp9enc.c:
* ext/vpx/gstvp9enc.h:
* ext/vpx/plugin.c:
  vpx: Add initial, experimental VP9 support

2013-06-21 10:32:30 +0200  Youness Alaoui 

* gst/rtsp/gstrtspsrc.c:
  rtsp: go back into the loop after doing pause
  After we do a pause request, go back to loop mode so that we can 
listen
  for server messages again.
  See https://bugzilla.gnome.org/show_bug.cgi?id=702705

2013-06-20 23:16:17 -0400  Olivier Crête 

* gst/rtpmanager/gstrtpptdemux.c:
  rtpptdemux: Wait after the caps to forward the other events
  First forward the stream-start, then the caps, then the rest

2013-06-21 00:42:02 +0100  Tim-Philipp Müller 

* sys/ximage/gstximagesrc.c:
  ximagesrc: clear dts on buffer acquired from pool
  When setting timestamps on outgoing buffers, clear the
  dts explicitly, otherwise it may end up being set to a
  bogus value from last time it was used. Avoids every
  second or so buffer's dts being set to 0. Not that it
  should matter for raw video.

2013-06-20 15:35:11 +0200  Wim Taymans 

* sys/v4l2/gstv4l2.c:
  v4l2: don't redefine the PERFORMANCE debug variable
  It is already defined in core.
  fixes https://bugzilla.gnome.org/show_bug.cgi?id=702732

2013-06-20 14:43:47 +0200  Wim Taymans 

* gst/rtsp/gstrtspsrc.c:
  rtspsrc: fix race in state change to paused
  When we go to paused, we first flush the connection and then send the 
pause
  command. As a result of the flushing, the scheduled paused command 
can get
  lost. Wait until the connection is completely flushed and the rtsp 
task is
  waiting before issuing the paused or playing request.
  Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702705

2013-06-20 11:31:22 +0200  Wim Taymans 

* gst/isomp4/qtdemux.c:
  qtdemux: handle SEGMENT query

2013-06-19 12:37:31 +0200  Sjoerd Simons 

* sys/v4l2/gstv4l2src.c:
  v4l2: Optimize negotiation by removing the query filter
  As cameras tend to have a quite specific set of capabilities (specific
  framerates for each resolution), getting the peer caps filtered by our
  probed caps can cause a big increase in the caps size which slows down
  things quire a bit.
  As for negotiation v4l2 iterates through the caps of the peer to find 
the
  first intersection with the probed caps, getting th

gst-plugins-base 1.1.2

[Tim-Philipp Müller]

ChangeLog
=

2013-07-11  Sebastian Dröge 

* configure.ac:
  releasing 1.1.2

2013-07-10 17:16:14 +0200  Sebastian Dröge 

* gst/playback/gstplaybin2.c:
  playbin: Only give sinks a new bus if they have no parent yet
  Otherwise we will remove the bus that would proxy messages to playsink
  and never set it again. If the sink is already in playsink, all 
failures
  are fatal anyway as it's either a sink that worked before or one that
  was set by the user.
  https://bugzilla.gnome.org/show_bug.cgi?id=701997

2013-07-10 13:22:04 +0200  Sebastian Dröge 

* gst/playback/gstplaybin2.c:
  playbin: Store a/v/t sinks locally too, not just in playsink

2013-07-10 13:21:29 +0200  Sebastian Dröge 

* gst/playback/gstplaysink.c:
  playsink: ref_sink() any sinks that are set on playsink
  Otherwise the behaviour of the properties is inconsistent.

2013-07-10 13:20:34 +0200  Sebastian Dröge 

* tests/check/elements/playbin.c:
  playbin: Fix assumptions in the unit test
  Unused sinks are still set to READY now during autoplugging
  to check their caps. Also playsink owns a ref to the sinks too.

2013-07-10 13:00:21 +0200  Sebastian Dröge 

* gst/playback/gststreamsynchronizer.c:
  streamsynchronizer: Non-TIME segment streams are not waiting 
automatically
  This was leftover code from porting to 1.0 and fixes the playbin
  unit test.
  https://bugzilla.gnome.org/show_bug.cgi?id=701943

2013-07-09 23:04:49 +0200  Branko Subasic 

* win32/common/libgstrtp.def:
  win32: add missing rtp buffer methods

2013-07-09 14:55:57 +0200  Sebastian Dröge 

* gst/playback/gstplaybin2.c:
* gst/playback/gstplaysink.c:
  playbin: Change sink ownership handling to be a bit more sane
  playbin will now only activate the sinks in a single place and
  will never change the states of any sinks that are owned by
  playsink.
  Also handle text-sinks the same way as audio/video sinks inside
  playbin.

2013-07-05 21:55:26 +0200  Piotr Drąg 

* po/POTFILES.in:
  po: update POTFILES.in
  https://bugzilla.gnome.org/show_bug.cgi?id=703684

2013-07-04 17:09:00 +0300  Sreerenj Balachandran 


* gst-libs/gst/video/colorbalance.c:
  colorbalance: Fix the typo in base_init().

2013-07-04 12:54:59 -0400  Thibault Saunier 

* gst/adder/gstadder.c:
  adder: Do not send flush_start event with the stream lock taken
  FLUSH_START is not serialized, so the lock should not be taken when
  sending it.

2013-07-05 00:47:08 +0100  Marcin Lewandowski 

* gst-libs/gst/tag/id3v2frames.c:
  tag: ignore malformed ID3v2 TDAT frames
  Just skip them, don't cause criticals.
  https://bugzilla.gnome.org/show_bug.cgi?id=703283

2013-07-03 09:44:32 +0100  Tim-Philipp Müller 

* gst/audioresample/speex_resampler_int.c:
  audioresample: make explicit that neon is disabled and why
  https://bugzilla.gnome.org/show_bug.cgi?id=703477

2013-07-02 18:20:39 +0200  Carlos Rafael Giani 

* gst/audioresample/speex_resampler_int.c:
  audioresample: disable 16-bit integer NEON support
  it seems to be broken (produces no audio), plus the performance gain
  is small
  Signed-off-by: Carlos Rafael Giani 

2013-07-02 14:25:28 +0200  Sebastian Dröge 

* gst/playback/gstplaybin2.c:
  playbin: If we had a previous autoplugged sink, try to reuse it
  https://bugzilla.gnome.org/show_bug.cgi?id=701997

2013-07-02 14:18:20 +0200  Sebastian Dröge 

* gst/playback/gstplaysink.c:
  playsink: If we switch sinks, make sure that the old sink is set to 
NULL

2013-07-02 14:02:57 +0200  Sebastian Dröge 

* gst/playback/gstplaybin2.c:
  playbin: Don't change the state of sinks that we passed to playsink 
already

2013-07-02 14:01:52 +0200  Sebastian Dröge 

* gst/playback/gstplaysink.c:
  playsink: Consider new audio/video sinks when reconfiguring

2013-07-02 12:27:03 +0200  Sebastian Dröge 

* gst/playback/gstplaybin2.c:
  playbin: Improve debug output regarding sink selection

2013-07-01 12:52:43 -0600  Brendan Long 

* gst/playback/gstplaybin2.c:
  playbin: Post an error message if a stream combiner doesn't return a 
request pad.

2013-07-01 13:45:25 +0200  Sebastian Dröge 

* gst/playback/gstplaybin2.c:
  playbin: Only intersect to check if a sink can handle raw caps
  Doing a subset check requires fixed caps, which we might not have 
here.
  https://bugs.webkit.org/show_bug.cgi?id=116042

2013-07-01 10:39:02 +0100  Vincent Penquerc'h 


* gst-libs/gst/pbutils/descriptions.c:
* gst-libs/gst/

gst-plugins-base 1.1.3

[Tim-Philipp Müller]

ChangeLog
=

2013-07-29  Sebastian Dröge 

* configure.ac:
  releasing 1.1.3

2013-07-29 12:11:38 +0200  Sebastian Dröge 

* ext/ogg/gstoggdemux.c:
  oggdemux: Don't swap start/stop for negative rates in the SEGMENT 
query

2013-07-29 11:40:57 +0200  Sebastian Dröge 

* gst-libs/gst/tag/gsttagdemux.c:
  tagdemux: Properly implement seeking if tagdemux is driving the 
pipeline
  https://bugzilla.gnome.org/show_bug.cgi?id=705062

2013-07-29 10:46:01 +0200  Sebastian Dröge 

* ext/ogg/gstoggdemux.c:
  oggdemux: Implement SEGMENT query

2013-07-28 23:38:06 +0200  Matej Knopp 

* gst/videorate/gstvideorate.c:
  videorate: ignore GAP event
  videorate automatically fills gaps with the previous frames.
  https://bugzilla.gnome.org/show_bug.cgi?id=705048

2013-07-28 23:38:17 +0200  Matej Knopp 

* gst/audiorate/gstaudiorate.c:
  audiorate: ignore GAP event
  audiorate automatically fills gaps with silence.
  https://bugzilla.gnome.org/show_bug.cgi?id=705048

2013-07-28 13:52:28 +0200  Sebastian Dröge 

* gst/tcp/gstmultisocketsink.c:
  multisocketsink: Fix handling of partial writes and WOULD_BLOCK errors

2013-07-28 13:23:41 +0200  Sebastian Dröge 

* gst/tcp/gstmultifdsink.c:
  multifdsink: Update current time after every write
  Each write will update the last_activity_time and otherwise we would
  compare against a too old current time and immediately timeout because
  current time is smaller than last activity time (overflow).

2013-07-28 13:20:48 +0200  Sebastian Dröge 

* gst/tcp/gstmultihandlesink.c:
  multihandlesink: Update current time after every write
  Each write will update the last_activity_time and otherwise we would
  compare against a too old current time and immediately timeout because
  current time is smaller than last activity time (overflow).

2013-07-27 12:16:15 +0200  Edward Hervey 

* gst-libs/gst/pbutils/descriptions.c:
  pbutils: Add description for teletext

2013-07-26 14:28:41 +0200  Matej Knopp 

* gst-libs/gst/audio/gstaudiodecoder.c:
  audiodecoder: do not leak input caps
  https://bugzilla.gnome.org/show_bug.cgi?id=704926

2012-11-02 10:04:16 +0100  Edward Hervey 

* gst/videotestsrc/gstvideotestsrc.c:
  videotestsrc: Initialize GstVideoInfo in ::start()
  Otherwise we end up with bogus values and races

2012-11-02 10:03:38 +0100  Edward Hervey 

* gst/videotestsrc/gstvideotestsrc.c:
  videotestsrc: Don't error on not-negotiated
  Base classes will handle re-negotiation if needed, but emitting an
  error message will make applications stop.

2012-09-12 12:41:01 +0200  Edward Hervey 

* ext/libvisual/gstaudiovisualizer.c:
* ext/pango/gstbasetextoverlay.c:
* gst/encoding/gststreamsplitter.c:
* gst/playback/gststreamsynchronizer.c:
  ext/gst: Ensure default query/event handlers are used
  And in some cases, just remove our implementation that does nothing

2013-07-26 11:02:32 +0200  Sebastian Dröge 

* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/app/gstappsrc.c:
* gst-libs/gst/app/gstappsrc.h:
  appsrc: Also provide function API for current-level-bytes and 
integrate into the docs

2013-07-26 15:00:44 +0900  Changbok Chea 

* gst-libs/gst/app/gstappsrc.c:
* win32/common/libgstapp.def:
  appsrc: Add "current-level-bytes" property
  https://bugzilla.gnome.org/show_bug.cgi?id=704774

2013-07-25 20:03:50 +0100  Vincent Penquerc'h 


* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/pbutils/codec-utils.c:
* gst-libs/gst/pbutils/codec-utils.h:
* win32/common/libgstpbutils.def:
  codec-utils: add a gst_codec_utils_aac_get_index_from_sample_rate
  It maps a sample rate to a well known AAC sample rate index.

2013-07-26 10:22:32 +0200  Sebastian Dröge 

* gst-libs/gst/video/gstvideodecoder.c:
* gst-libs/gst/video/gstvideoencoder.c:
  videodecoder/encoder: Call reset() always between start() and stop() 
and never outside

2013-07-25 14:25:08 +0200  Sebastian Dröge 

* gst-libs/gst/video/gstvideodecoder.c:
* gst-libs/gst/video/gstvideoencoder.c:
  videoencoder/decoder: Call reset() before start() too

2013-07-25 11:56:07 +0100  Tim-Philipp Müller 

* gst/typefind/gsttypefindfunctions.c:
  typefinding: don't detect mp3 based on just a few bits
  Remove dodgy code that detects mp3 with as little as
  a valid frame sync at the beginning. This was only used
  in some unit tests in -good where there were only a few
  bytes after the id3 tag. We now requir

gstreamer 1.1.3

[Tim-Philipp Müller]

ChangeLog
=

2013-07-29  Sebastian Dröge 

* configure.ac:
  releasing 1.1.3

2013-07-29 12:10:45 +0200  Sebastian Dröge 

* libs/gst/base/gstbaseparse.c:
* libs/gst/base/gstbasesink.c:
* libs/gst/base/gstbasesrc.c:
  base: Fix handling of SEGMENT query
  The values should be in stream-time, and start/stop should not
  be swapped for negative rates.

2013-07-29 11:05:09 +0200  Sebastian Dröge 

* plugins/elements/gsttypefindelement.c:
  typefind: Only advance offset by the number of bytes we actually read
  There might be a short read at EOS.

2013-07-29 10:48:30 +0200  Sebastian Dröge 

* libs/gst/base/gstbaseparse.c:
  baseparse: Implement SEGMENT query

2013-07-26 18:36:04 +0100  Tim-Philipp Müller 

* gst/gstbuffer.c:
  buffer: fix Since: marker for new gst_buffer_extract_dup()

2013-07-26 12:19:32 +0200  Wim Taymans 

* gst/gstclock.c:
  clock: debug the clock return values

2013-07-25 12:20:14 -0300  Thiago Santos 

* libs/gst/base/gstbaseparse.c:
  baseparse: fix seqnum handling for seeks
  Use the same seqnum as the seek for flushes/segments that are
  caused by the seek. Also do the same for segment events
  Fixes #676242

2013-07-24 10:29:30 -0700  David Schleef 

* gst/gstinfo.c:
  info: parse debug levels > 9

2013-07-24 16:57:46 +0200  Sebastian Dröge 

* gst/gstvalue.c:
  value: Fix copy&paste mistakes in the bitmask function docs

2013-07-24 11:21:27 +0200  Sebastian Dröge 

* libs/gst/base/gstbasesink.c:
  basesink: Don't shadow variables that are set inside our scope and 
then used outside our scope
  Fixes uninitialized use of these variables.

2013-07-24 10:30:25 +0200  Sebastian Dröge 

* tests/check/gst/struct_arm.h:
* tests/check/gst/struct_hppa.h:
* tests/check/gst/struct_i386.h:
* tests/check/gst/struct_i386w.h:
* tests/check/gst/struct_ppc32.h:
* tests/check/gst/struct_ppc64.h:
* tests/check/gst/struct_sparc.h:
* tests/check/gst/struct_x86_64.h:
  tests: Remove other interface structs from the ABI tests too

2010-10-15 13:16:59 -0300  Thiago Santos 

* tests/check/gst/struct_arm.h:
* tests/check/gst/struct_hppa.h:
* tests/check/gst/struct_i386.h:
* tests/check/gst/struct_i386w.h:
* tests/check/gst/struct_ppc32.h:
* tests/check/gst/struct_ppc64.h:
* tests/check/gst/struct_sparc.h:
* tests/check/gst/struct_x86_64.h:
  tests: Remove GstTagSetter from ABI checks
  Interfaces can have new members added without breaking ABI, so
  remove it from the check.
  https://bugzilla.gnome.org/show_bug.cgi?id=623799

2013-07-23 15:39:53 -0400  Thibault Saunier 

* libs/gst/check/libcheck/check_print.c:
  libcheck: Escape strings in the generated xml files
  This is copy pasted from upstream libcheck

2013-07-23 18:53:44 +0200  Sebastian Dröge 

* libs/gst/base/gstbasesink.c:
  basesink: Print some debug output if a stream-start event without 
group-id arrives
  Ideally all elements would implement handling of that to get proper
  stream-start message handling and other things.

2013-07-22 18:03:01 +0200  Arnaud Vrac 

* plugins/elements/gstinputselector.c:
  input-selector: Fix missing pad activation notification
  A new active pad might not be notified in some cases, which results
  in the current track number not being set in playbin.
  The active-pad notification is only sent in the chain and sink_event
  functions, and only when the buffer or event that triggered the active
  pad selection is from the newly activated pad. So in the other case
  the notification will never be sent.
  https://bugzilla.gnome.org/show_bug.cgi?id=704691

2013-07-22 17:25:47 +0100  Tim-Philipp Müller 

* gst/gstvalue.c:
  value: handle deserialisation of nonexistant enum value more 
gracefully

2013-07-22 14:12:18 +0200  Sebastian Dröge 

* plugins/elements/gstinputselector.c:
* plugins/elements/gstinputselector.h:
  inputselector: Don't push new stream-start events on stream change 
unless they all have group ids
  https://bugzilla.gnome.org/show_bug.cgi?id=704408

2013-07-22 12:06:29 +0200  Sebastian Dröge 

* plugins/elements/gsttypefindelement.c:
  typefind: Use new group-id in stream-start event

2013-07-22 12:06:08 +0200  Sebastian Dröge 

* libs/gst/base/gstbaseparse.c:
* libs/gst/base/gstbasesink.c:
* libs/gst/base/gstbasesrc.c:
  base: Use new group-id field in stream-start event and message

2013-07-22 11:42:18 +0200  Sebastian Dröge 

* gst/gstbin.c:
 

gst-plugins-good 1.1.3

[Tim-Philipp Müller]

 set when the session wants to send BYE.

2013-07-25 16:24:04 +0200  Wim Taymans 

* gst/rtpmanager/rtpsession.c:
* gst/rtpmanager/rtpsession.h:
* gst/rtpmanager/rtpsource.c:
* gst/rtpmanager/rtpsource.h:
  session: configure sdes with structure only
  Remove code to configure the SDES with methods and types, only
  allow configuration with GstStructure

2013-07-25 15:56:39 +0200  Wim Taymans 

* gst/rtpmanager/rtpsession.c:
  session: refactor add and find source
  Make functions to find and add a source to the hashtable.

2013-07-25 15:43:11 +0200  Wim Taymans 

* gst/rtpmanager/gstrtpsession.c:
* gst/rtpmanager/rtpsession.c:
* gst/rtpmanager/rtpsession.h:
  session: remove source from sync_rtcp
  We don't need to know the sender source of the session in the
  callback, the SR packet is for all participants in the session.

2013-07-24 14:18:14 +0200  Wim Taymans 

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

2013-07-15 17:11:45 +0100  Vincent Penquerc'h 


* gst/audioparsers/Makefile.am:
* gst/audioparsers/gstaacparse.c:
* gst/audioparsers/gstaacparse.h:
  aacparse: allow conversion from ADTS to raw AAC
  Some muxers (eg, qtmux) only support raw AAC, so this allows linking
  an encoder that outputs ADTS only to those muxers.
  The conversion is simple (omit the first 7 or 9 bytes of the frame),
  but has to be done in pre_push instead of handle_frame as 1.0 does
  not seem to allow skipping bytes there as 0.10 used to.
  Other conversions are not supported (yet).

2013-07-15 17:15:44 +0100  Vincent Penquerc'h 


* gst/audioparsers/gstaacparse.c:
  aacparse: fix object_type parsing off-by-one in ADTS frame
  According to http://wiki.multimedia.cx/index.php?title=ADTS,
  the value stored in ADTS headers is one less than the object
  type of the AAC stream.
  A look at ffmpeg shows it also adds 1 to the value read off
  the ADTS header.
  Note that this might break other things that happen to have
  an inverse off by one to match the existing code.

2013-07-25 11:13:01 -0300  Thiago Santos 

* gst/avi/gstavidemux.c:
  avidemux: fix seqnum handling for seeks
  Use the same seqnum as the seek for flushes/segments that are
  caused by the seek. Also do the same for segment events
  Fixes #676242

2013-07-25 01:39:58 -0300  Thiago Santos 

* gst/matroska/matroska-demux.c:
* gst/matroska/matroska-demux.h:
  matroskademux: fix seqnum handling for seeks
  Use the same seqnum as the seek for flushes/segments that are
  caused by the seek. Also do the same for segment events
  Fixes #676242

2013-07-25 01:11:31 -0300  Thiago Santos 

* gst/isomp4/qtdemux.c:
  qtdemux: correctly handle seqnum for seeks and segments
  Use the same seqnum on messages and events for derived events.
  Fixed for flushes / stream-start / segment after a seek, and segment
  after a segment.
  Fixes #676242

2013-07-12 20:01:42 +0200  Arnaud Vrac 

* ext/soup/gstsouphttpsrc.c:
  souphttpsrc: always ignore HEAD errors
  https://bugzilla.gnome.org/show_bug.cgi?id=704241

2013-07-25 14:26:07 +0200  Sebastian Dröge 

* ext/jpeg/gstjpegenc.c:
  jpegenc: Clean up reset/start/stop handling

2013-07-25 14:13:10 +0200  Sebastian Dröge 

* ext/jpeg/gstjpegdec.c:
* ext/jpeg/gstjpegdec.h:
  jpegdec: Use base class error handling function instead of 
replicating it here

2013-07-25 14:12:56 +0200  Sebastian Dröge 

* ext/jpeg/gstjpegdec.c:
  jpegdec: Clean up handling of reset/start/stop

2013-07-25 10:41:22 +0100  Tim-Philipp Müller 

* tests/files/id3-407349-1.tag:
* tests/files/id3-407349-2.tag:
* tests/files/id3-447000-wcop.tag:
  tests: fix test ID3 tags up not to rely on dodgy typefinding code
  Change 0xff 0xfb 'mp3' marker to 'fLaC' marker, so we can fix
  the typefinder.
  https://bugzilla.gnome.org/show_bug.cgi?id=681368

2013-07-25 08:22:45 +0200  Alessandro Decina 

* sys/osxaudio/gstosxaudiosink.c:
  osxaudiosink: intersect the probed caps with the filter passed to 
get_caps()

2013-07-24 14:17:45 +0200  Wim Taymans 

* gst/rtpmanager/gstrtpbin.c:
  bin: fix compilation

2013-07-24 12:42:31 +0200  Wim Taymans 

* gst/rtp/gstrtpvrawdepay.c:
  vrawdepay: fix UYVP format

2013-07-24 12:41:58 +0200  Wim Taymans 

* gst/rtp/gstrtpvrawpay.c:
  vrawpay: fix UYVP format

2013-07-24 12:41:44 +0200  Wim Taymans 

* gst/rtp/gstrtpvrawpay.c:
  vra

gst-plugins-good 1.1.1

[Tim-Philipp Müller]

ChangeLog
=

2013-06-05  Sebastian Dröge 


Download

http://download.gnome.org/sources/gst-plugins-good/1.1/gst-plugins-good-1.1.1.tar.xz
 (2.67M)
  sha256sum: 06dc153369a9a960236fd471ab11f597fe1a4871dcfa5dbdcdba6708e482c70e

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gstreamer 1.1.1

[Tim-Philipp Müller]

ChangeLog
=

2013-06-05  Sebastian Dröge 


Download

http://download.gnome.org/sources/gstreamer/1.1/gstreamer-1.1.1.tar.xz (3.02M)
  sha256sum: 74892b1a877cd84592d8179878e270937ecd5dcc3c09a88d777cff5992e5b45f

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gst-plugins-base 1.1.1

[Tim-Philipp Müller]

ChangeLog
=

2013-06-05  Sebastian Dröge 


Download

http://download.gnome.org/sources/gst-plugins-base/1.1/gst-plugins-base-1.1.1.tar.xz
 (2.36M)
  sha256sum: f974e3125d75df5785347b610be5eb9b6128bfc25ef3f0ce8f64f9c51ea74e72

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gstreamer 1.0.8

[Tim-Philipp Müller]

News


Changes since 1.0.7:

  * basesink: improve position reporting without clock
  * fix caps leak in typefind/decodebin/playbin
  * gobject-introspection fixes for bindings

Bugs fixed since 1.0.7:

  * 693365 : gst_structure_is_subset false positive
  * 702617 : buffer: Wrong size/index handling when merging memory
  * 702778 : REGRESSION : Backward seeking doesn't work with mp3 files.
  * 703562 : Missing few allow-none annotation

ChangeLog
=
http://download.gnome.org/sources/gstreamer/1.0/gstreamer-1.0.8.changes  (3.25K)

Download

http://download.gnome.org/sources/gstreamer/1.0/gstreamer-1.0.8.tar.xz (2.99M)
  sha256sum: ff70f45509566b88e35986971ace5e89cb6cb232e9ca249f84502abceef1762d

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gst-plugins-good 1.0.8

[Tim-Philipp Müller]

News


Changes since 1.0.7:

  * pngenc: fix massive memory leak
  * pngdec: parse incoming data into frames before decoding
  * osvideo: many osxvideosink fixes
  * udpsink, multiudpsink, dynudpsink: bind socket before using it, fixes 
sending udp on windows

Bugs fixed since 1.0.7:
 
  * 682110 : qtdemux: discont flag set on multiple buffers in push mode
  * 692400 : udpsrc: fix socket options not getting set on windows, 
resulting in packet drop in high bitrate movie
  * 693727 : rtpvrawpay/depay negotiation broken
  * 699260 : pngenc: unmap source frame when done
  * 699303 : matroskademux: stream-format=raw missing from aac caps
  * 699314 : rtph264pay: CRITICAL **: gst_adapter_map: assertion `size > 0' 
failed
  * 700047 : pngdec: make decoding work without png parser
  * 700382 : qtdemux: handle 96kHz/24 bits ALAC audio
  * 700514 : rtpmp4apay: clear config buffer before using it
  * 700878 : udpsink: Not sending anything on Windows 
  * 701586 : rtspsrc memleak
  * 702167 : matroskademux: missing mutex unlock leading to wrong 
STREAM_LOCK count
  * 702457 : rtpmp2tdepay: unable to depay MPEG2-TS RTP streams from 
GStreamer 0.10
  * 702705 : rtspsrc does not pause properly (race condition)
  * 702732 : v4l2: Does not link statically on linux
  * 703076 : flvdemux: Add flvversion 1 to flash-video caps
  * 703100 : osxvideosink: deadlock on re-use
  * 703171 : rtph264pay: segfault because of double buffer unmap on error
  * 703729 : osxvideosink doesn't display video anymore if once set to READY
  * 691419 : osxvideosink: doesn't close internal window in case of 
window-id assignment

ChangeLog
=
http://download.gnome.org/sources/gst-plugins-good/1.0/gst-plugins-good-1.0.8.changes
  (10.6K)

Download

http://download.gnome.org/sources/gst-plugins-good/1.0/gst-plugins-good-1.0.8.tar.xz
 (2.60M)
  sha256sum: 97831570ccdd8e15557f18858b965e54433d572d27fdabebb8b710cee276cfad

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gst-plugins-base 1.0.8

[Tim-Philipp Müller]

News


Changes since 1.0.7:

  * tag: ignore malformed ID3v2 TDAT frames

Bugs fixed since 1.0.7:

  * 636245 : riff: for ADPCM codecs the average bitrate should be 
calculated instead of trusting the format header
  * 688803 : playbin: converters don't work? not-negotiated error with 
non-1/1 PAR and ximagesink
  * 690420 : decodebin: Race between GstBin and decodebin trying to change 
states of child elements
  * 698896 : liveadder: What is the unit for " latency " ?
  * 699923 : typefind: fix detection of HLS playlists with alternative 
renditions
  * 701976 : flvdemux: Forwards CAPS events from upstream
  * 703128 : videotestsrc leaks memory
  * 703283 : id3: gst_date_time_new: assertion `(month > 0 &  & month < = 
12) || month == -1' failed with malformed TDAT frames
  * 699794 : uridecodebin: Leaking queue2 elements in playbin gapless mode

ChangeLog
=
http://download.gnome.org/sources/gst-plugins-base/1.0/gst-plugins-base-1.0.8.changes
  (5.17K)

Download

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 (2.26M)
  sha256sum: b55c9deea00acf789f82845c088b7c7c17b3ecef24a94831a819071b3dd8ef0d

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gst-plugins-good 1.0.7

[Tim-Philipp Müller]

News


Changes since 1.0.6:

  * osxaudio plugin ported to 1.0

Bugs fixed since 1.0.6:

  * 695709 : vp8enc plugin has invalid parameter error when framerate is " 
0/1 " 
  * 697103 : osxaudio: port to 1.0
  * 677560 : rtpjpegdepay: crash in copy_into_unchecked
  * 685209 : gst_v4l2_object_get_nearest_size VIDIOC_G_FMT bug
  * 693005 : capssetter: do not return a copy of sink caps upstream
  * 693727 : rtpvrawpay/depay negotiation broken
  * 696052 : qtmux expose mulaw
  * 696651 : v4l2loopback:gst_buffer_get_meta: assertion `buffer != NULL' 
failed. " 
  * 697113 : matroskademux: add TrueHD audio codec support
  * 697303 : osxvideosink: compiler warning: implicit declaration of 
function 'pthread_self'
  * 697984 : souphttpsrc: direct stream dump results in broken audio with 
internet radio
  * 698224 : [osxaudio] caps negotiation fails due to invalid channel-mask

ChangeLog
=
http://download.gnome.org/sources/gst-plugins-good/1.0/gst-plugins-good-1.0.7.changes
  (10.4K)

Download

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 (2.57M)
  sha256sum: a016a3b377c86658627aef902b9204108209100b2e88fcc3b695c392af1b4035

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gst-plugins-base 1.0.7

[Tim-Philipp Müller]

News


Changes since 1.0.6:

  * streamsynchronizer is now a public element, useful in HLS pipelines for 
example

Bugs fixed since 1.0.6:

  * 682171 : videoscale: assertion failed: (envelope ((x - (xi - 1)) * ex) 
== 0)
  * 688240 : streamsynchronizer: make element public
  * 696540 : Ring buffer released after parsing caps when setting caps
  * 696899 : oggdemux: non flushing segment seek makes buffers late
  * 697092 : rtspconnection: The " x-server-ip-address " header with wrong 
IP address is added to the GET response
  * 697162 : SDP library is not parsing correctly
  * 697277 : SDP library does not include timing line if there is not 
timing information
  * 697820 : playbin: Enabling visualizations during playback stops the 
pipeline
  * 697824 : videometa: gst_buffer_add_meta() can return NULL

ChangeLog
=
http://download.gnome.org/sources/gst-plugins-base/1.0/gst-plugins-base-1.0.7.changes
  (8.69K)

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  sha256sum: 014805e50b696bc06c3862ea656df079fc0b5fef0c10f16e9f085f290545677a

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gstreamer 1.0.7

[Tim-Philipp Müller]

News


Changes since 1.0.6:

  * Add better support for static plugins

Bugs fixed since 1.0.6:

  * 667305 : build: Add --enable-static-plugins configure argument
  * 696357 : Basesrc may send not-neg error when flushed
  * 696435 : gst_static_caps_get(GST_STATIC_CAPS_NONE) doesn't work properly
  * 697791 : gst-inspect: shows action signal return pointers as pointers

ChangeLog
=
http://download.gnome.org/sources/gstreamer/1.0/gstreamer-1.0.7.changes  (4.09K)

Download

http://download.gnome.org/sources/gstreamer/1.0/gstreamer-1.0.7.tar.xz (2.95M)
  sha256sum: 68cada7ee24ede23e15dc81ccde11898eed1a7a3c6a2d81a8c31596fccb1b5ce

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gst-plugins-good 1.0.6

[Tim-Philipp Müller]

News


Changes since 1.0.5:

  * auparse: fix caps leak
  * avidemux: push mode: handle some more 0-size buffer cases
  * deinterlace: fix infinite loop on EOS with non-default methods or fields
  * dvdemux: don't return FALSE when dropping sink events, fixes flow errors
  * level: send a last message on EOS 
  * mp4mux: in faststart mode, don't output up to 4 kB of garbage at the end
  * osxvideosink: Fix crash in osxvideosink with external window output
  * osxvideosink: Make GstGLView propagate input events to its parent view
  * osxvideosink: Make GstNavigation key input events in osxvideosink 
compatible with x(v)imagesink ones
  * pulsesink: don't error out if pa_stream_proplist_update() with new tags 
fails
  * qtdemux: fix potential crash on short MOOV atom
  * qtdemux: fix sample leak when processing private qt tags
  * qtdemux: push mode: only parse moov 1 once
  * qtdemux: skip disabled tracks
  * qtmux: set stream language code from tag
  * rtph264pay: Don't use upstream caps with peer_query_caps()
  * rtpmp4gdepay: streamtype is not put by all RTSP server, not make it 
optional
  * rtpptdemux: forward sticky events and then set caps
  * rtpsession: Fix wrong code organisation in case of collision
  * rtspsrc: flush connection when stopping
  * rtspsrc: only EOS when our source sends BYE
  * rtspsrc: save the stream SSRC
  * v4l2: don't check stride for encoded formats
  * v4l2: fix compilation against newer kernel headers as on FC19
  * videomixer2: avoid caps leak
  * videomixer: fix eos timestamp check
  * ximagesrc: Set the pixel aspect ratio correctly in the caps
  * build: allow calling autogen.sh from out-of-tree
  * build: fix build with automake 1.13

Bugs fixed since 1.0.5:

  * 628790 : qtdemux: OSD displays persistent chapters subtitles
  * 656068 : pulsesink: errors out if pa_stream_proplist_update() fails 
with old pulseaudio versions
  * 675453 : rtspsrc: only EOS if BYE came from SSRC of the stream
  * 678429 : souphttpsrc: should return GST_FLOW_ERROR from create function 
in case of HTTP error 404 not found
  * 684924 : rtspsrc: flush connection when going to null state
  * 684944 : avidemux: " unhandled buffer size " error for cyuv.avi in push 
mode
  * 688935 : rtpjitterbuffer not always correctly warns about resetting skew
  * 689809 : osxvideosink: inconsistent keyboard navigation events
  * 691484 : osxvideosink: crash when displaying on Qt widget
  * 691570 : [isomp4/qtdemux] lots of critical warnings on this sample file
  * 691580 : ximagesrc sets bad pixel-aspect-ratio caps
  * 691832 : osxvideosink doesn't propagate input events to output window
  * 692309 : allow autogen.sh to be call out-of-tree
  * 692786 : CAPS event is forwarded before STREAM_START in rtpptdemux 
element
  * 692935 : videomixer: hangs on eos
  * 692950 :  " rtpjitterbuffer " properties can't be changed after 
pipeline is started
  * 693055 : level: 32-bit formats don't work because of typo in format 
list: S32LEF32LE
  * 693173 : deinterlace method=greedyl fields=top does not complete
  * 693307 : videomixer2 leaks caps reference
  * 693373 : 1.0 branch is still using AM_CONFIG_HEADER
  * 694010 : qtdemux: crash with partial mp4 / moov atom
  * 694184 : Wrong code organisation in case of collision in rtpsession
  * 694275 : auparse: src_caps are leaked
  * 695629 : rtph264pay: caps negotiation fails if upstream caps is not NULL
  * 695643 : dvdemux: returns wrong value when it is not processing the 
sink events
  * 695644 : deinterlace: going to infinite loop on EOS with 
method=scalerbob fields=bff
  * 696355 : qtdemux: sample leak in tag handling
  * 696358 : qtmux: set language code on streams

ChangeLog
=
http://download.gnome.org/sources/gst-plugins-good/1.0/gst-plugins-good-1.0.6.changes
  (8.82K)

Download

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  sha256sum: 67f7690a9826d9a6ab28b9af2536a6f3e833ee412bd59dd603c48fb3c6823e0d

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gstreamer 1.0.6

[Tim-Philipp Müller]

News


Changes since 1.0.5:

  * bin: reset GValue from iterator after usage, fixing leak
  * pipeline: fix seeking on pipeline with NULL clock
  * message: accept NULL error argument in 
gst_message_parse_{error,warning,info}
  * basesrc: handle renegotiation correctly
  * basesink: handle sync of EOS after item without duration
  * basetransform: avoid warnings from NULL outcaps passed to 
transform_size in shutdown race
  * adapter, basesrc, typefind: handle gst_buffer_map failure
  * queuearray: fix gst_queue_array_drop_element() 
  * queuearray: fix gst_queue_array_find()
  * queue: remove query from queue if queue is flushing
  * g-i: make sure enumeration types and flags are introspected and 
available to bindings
  * g-i: add allow-none annotation for gst_pipeline_new()'s name property

Bugs fixed since 1.0.5:

  * 684804 : basesink: fails to post EOS message with RTSP stream
  * 691185 : g-i: GStreamer enums and flags not usable - " expected 
enumeration type GstFormat, but got Format instead " warning when setting 
GstFormat property on GStreamer appsrc element
  * 691985 : gst_pad_check_reconfigure: only remove flag if set.
  * 691986 : basesrc: set NEED_RECONFIGURE flag if negotiate fails
  * 692691 : queue: dangling pointer when doing allocation query on a 
flushing queue
  * 692868 : basesink: expands GAP event without duration to last to 
segment end
  * 693065 : pipeline: criticals when seeking on pipeline with NULL clock
  * 693587 : Adding offset to GstSegment deserialize/serialize to fix jump 
on uninitialized value
  * 693704 : gst_message_parse_{warning,error} annotations wrongly claim 
allow-none for GError parameter
  * 693996 : basetransform: audioresample shutdown unit test fails with 
'gst_audio_info_from_caps: caps != NULL'  in transform_size

ChangeLog
=
http://download.gnome.org/sources/gstreamer/1.0/gstreamer-1.0.6.changes  (10.4K)

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gst-plugins-base 1.0.6

[Tim-Philipp Müller]

ChangeLog
=

2013-03-22  Tim-Philipp Müller 

* configure.ac:
  releasing 1.0.6

2013-02-14 01:35:45 +0100  Arnaud Vrac 

* gst/subparse/gstsubparse.c:
* gst/subparse/gstsubparse.h:
  subparse: fix off by one offset calculation
  Do not use the buffer start offset when it is invalid, otherwise a
  discontinuity is detected on the next buffer, and the subtitle parser
  reset and some subtitle lines are not shown.
  Also remove unused next_offset field.
  https://bugzilla.gnome.org/show_bug.cgi?id=693981

2013-03-22 09:35:21 +  Tim-Philipp Müller 

* gst/typefind/gsttypefindfunctions.c:
  typefinding: fix y4m caps
  We need to specify the y4mversion field now.
  https://bugzilla.gnome.org/show_bug.cgi?id=696282

2013-03-21 15:37:31 +0100  Sebastian Dröge 

* gst/playback/gstsubtitleoverlay.c:
  subtitleoverlay: Return ANY caps for the GET_CAPS query
  subtitleoverlay handles any caps, not just the ones
  for which a subtitle parser/renderer exist. It will
  just ignore any unsupported streams instead of causing
  an error.
  https://bugzilla.gnome.org/show_bug.cgi?id=688476

2013-03-20 19:34:06 +0100  Jan Schole 

* gst/videoscale/gstvideoscale.c:
  videoscale: Correct DAR and border calculations
  changed: gst_video_scale_set_info in gst/videoscale/gstvideoscale.c
  DAR on sink side now calculated with PAR on sink side
  ratio of output width/height now calculated with inverse PAR
  additional condition that borders are 0:0 for passthrough mode
  https://bugzilla.gnome.org/show_bug.cgi?id=696019

2013-03-21 11:42:43 +0100  Sebastian Dröge 

* gst/playback/gstplaysinkconvertbin.c:
  playsinkconvertbin: Use gst_caps_is_subset() for handling the 
ACCEPT_CAPS query

2013-03-21 13:33:44 +0100  Sebastian Dröge 

* gst/playback/gstplaysinkconvertbin.c:
  playsinkconvertbin: Make sure to return all allowed caps in the 
GET_CAPS query
  Add all the caps that we can convert to to the filter caps,
  otherwise downstream might just return EMPTY caps because
  it doesn't handle the filter caps but we could still convert
  to these caps, causing us to return EMPTY caps although
  conversion would be possible.
  https://bugzilla.gnome.org/show_bug.cgi?id=688803

2013-03-21 13:29:06 +0100  Sebastian Dröge 

* sys/ximage/ximagesink.c:
  ximagesink: Don't access structures of EMPTY caps
  If the intersection between our caps and the filter caps is
  empty, just immediately return EMPTY caps instead of trying
  to access the (non-existant) structures.

2013-03-11 21:55:28 -0600  Greg Rutz 

* ext/libvisual/gstaudiovisualizer.c:
  libvisual: fix improper video frame clear operation
  The current code is memsetting the GstVideoFrame.data address to 0s 
(which
  causes a segfault). This member is actually an array of data buffers 
(one for
  each plane).  This fix iterates over each data plane to clear them 
all.
  https://bugzilla.gnome.org/show_bug.cgi?id=695655

2013-02-13 21:08:48 +0100  Stefan Sauer 

* gst/adder/gstadder.c:
* tests/check/elements/adder.c:
  adder: fix setting caps via the "caps" property
  We were setting the query-func on the sink-pad, which got overwritten 
when
  adding the new pad to collect pads. Instead register our query-func 
with the
  collect pads object. This fixes filter caps. Add a test for it.

2013-02-12 17:22:23 +  Vincent Penquerc'h 


* gst/encoding/gstencodebin.c:
  encodebin: sync muxer state with parent bin
  Other elements were synced, but not the muxer. Makes sure the
  muxer is in the right state even when encodebin gets added to
  the pipeline dynamically after the pipeline has been started.

2013-02-07 21:51:06 +  Tim-Philipp Müller 

* configure.ac:
  build: fix build with automake 1.13
  AM_CONFIG_HEADER -> AC_CONFIG_HEADERS

2013-02-08 00:05:24 +1100  Jan Schmidt 

* ext/pango/gstbasetextoverlay.c:
  pango: 3rd time's the charm. Fix attribute list handling.
  Really really fix attribute list handling by taking a
  copy of the original attributes that pango_attr_list_filter
  can mutate, but keep the original around intact to restore
  later.

2013-02-07 23:45:26 +1100  Jan Schmidt 

* ext/pango/gstbasetextoverlay.c:
  pango: Don't modify the original attributes list.
  Take a copy of the original attributes list instead of just
  a ref, since pango_attr_list_filter can remove elements from it.

2013-02-07 23:06:16 +1100  Jan Schmidt 

  

gst-plugins-good 1.0.5

[Tim-Philipp Müller]

News


Changes since 1.0.4:

  * rtspsrc: fix regression that make rtspsrc hang when stopping
  * audio/video parsers: fix negotiation with encoders in some transcoding 
cases
  * cairo: port cairooverlay to 1.0
  * jpegenc: pass flow returns upstream
  * qtdemux: fix pixel-aspect-ratio of some files with ProRes video
  * cairo: port cairooverlay to 1.0

Bugs fixed since 1.0.4:

  * 690184 : [parsers] Return too strict caps on the sinkpads
  * 690476 : regression: unable to stop rtsp stream
  * 691098 : v4l2src: breaks because of changed ioctl error return code 
from driver on failed control query in 3.8-rc1

Changes since 1.0.3:

  * deinterleave: properly set srcpad channel position
  * osxvideosink: Fix resizing the Cocoa window on receiving new caps
  * rtspsrc fixes
  * shout2send: also accept audio/webm in addition to video/webm
  * videobox: fix border filling for planar YUV formats
  * webmmux: fix linking to shout2send
  * v4l2: fix build on FreeBSD

Bugs fixed since 1.0.3:

  * 684312 : rtspsrc: mutex blocks going to NULL state
  * 684991 : videobox: sometimes incorrectly crops producing visual 
artifacts, and green instead of black border
  * 687469 : mulawdec: integer overrun with large chunks of data
  * 689732 : osxvideosink can't resize Cocoa window on receiving caps
  * 689336 : webmmux: can't link webmux to shout2send

ChangeLog
=
http://download.gnome.org/sources/gst-plugins-good/1.0/gst-plugins-good-1.0.5.changes
  (14.5K)

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  sha256sum: 53bd0b848e7896f1b22548ca4e8be56cf425e5100e121c472dff8272d5140bc5

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gstreamer 1.0.5

[Tim-Philipp Müller]

News


Changes since 1.0.4:

  * add GST_BIN_FLAG_NO_RESYNC flag that disables a resync when
an element is added, removed or linked in the bin; this is
interesting for complex bins that dynamically add elements
to themselves and want to manage the state of those elements
without interference from state resync threads (which may
cause deadlocks)
  * add OTHER clock type

Bugs fixed since 1.0.4:

  * 690915 : Replace AM_CONFIG_HEADER with AC_CONFIG_HEADERS in configure.ac

API additions since 1.0.4:

  * GST_BIN_FLAG_NO_RESYNC
  * GST_CLOCK_TYPE_OTHER

Changes since 1.0.3:

  * basesrc: fix potential leaks when re-activating in a different mode
  * bindings: make all pad probe types work with bindings
  * bindings: fix gst_event_parse_stream_start() annotation, fixing crash
  * documentation improvements

Bugs fixed since 1.0.3:

  * 687845 : filesrc: leaks file descriptor if activation in pull mode fails
  * 688804 : Getting error trying to add a pad_probe
  * 688934 : Gst.Event.parse_stream_start seems to not function correctly

ChangeLog
=
http://download.gnome.org/sources/gstreamer/1.0/gstreamer-1.0.5.changes  (6.40K)

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gst-plugins-base 1.0.5

[Tim-Philipp Müller]

News


Changes since 1.0.4:

  * alsasrc: don't output buffers without timestamps or with bogus 
timestamps
  * discoverer, decodebin: fix state change re-sync race that might lead to 
deadlocks
  * video: fix crashes with and frame sizes of A420 video format

Bugs fixed since 1.0.4:

  * 691244 : unable to play flv file with VP6 alpha video

Changes since 1.0.3:

  * playbin: fix occasional not-negotiated errors when switching 
visualisations
  * ssaparse: ignore invalid UTF-8 in SSA/ASS subtitles init sections in 
matroska files
  * streamsynchronizer: better timestamp and gap handling at EOS, fixing 
potential OOM in baseaudiosink
  * bindings: fix annotation for gst_app_src_push_buffer(), fixing crash
  * bindings: add several missing annotations for GstRtspMessage API

Bugs fixed since 1.0.3:

  * 679976 : playbin: switching visualisations results in not-negotiated 
flow erros
  * 689814 : Memory allocated by GstVideoDecoder base class is not freed
  * 689873 : gstrtspmessage: Several functions are missing introspection 
annotations...

ChangeLog
=
http://download.gnome.org/sources/gst-plugins-base/1.0/gst-plugins-base-1.0.5.changes
  (9.37K)

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gstreamer 1.0.3

[Tim-Philipp Müller]

News


Changes since 1.0.2:

  * bufferpool: fix deadlock
  * baseparse: forward stream-start event in push mode, fixing issues with 
streamsynchronizer
  * basesink: reset START_TIME when needed, fixing position reporting after 
seeking beyond end

Bugs fixed since 1.0.2:

  * 666053 : baseparse: prevent excessively high memory usage with large 
streams
  * 687896 : bufferpool: deadlock in GstBufferPool
  * 688477 : inputselector: fix clock leak in wait_running_time
  * 688511 : Windows compilers without pthreads still have testrtpool.c 
being built in tests
  * 688655 : baseparse: stream-start events not sent in push mode
  * 685870 : Regression with position when seeking beyond the media duration

ChangeLog
=
http://download.gnome.org/sources/gstreamer/1.0/gstreamer-1.0.3.changes  (2.60K)

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http://download.gnome.org/sources/gstreamer/1.0/gstreamer-1.0.3.tar.xz (2.97M)
  sha256sum: 69ac49e2ae9cd2ea5f789de641aeb338bb8731961acfced08ae1c76a797fe6b1

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gst-plugins-good 1.0.3

[Tim-Philipp Müller]

News


Changes since 1.0.2:

  * rtspsrc: numerous improvements
  * build fix for gst-plugins-base installed in non-default prefix
  * multifilesink: post messages in max-size mode as well
  * vp8dec: improve robustness on decoding errors, e.g. for videocalls over 
RTP

Bugs fixed since 1.0.2:

  * 639420 : RTSP setup, add client_ports
  * 686837 : imagefreeze: improve caps negotiation
  * 686985 : [pulsesrc] Assertion 'm' failed at 
pulse/thread-mainloop.c:166, function pa_threaded_mainloop_lock(). Aborting.
  * 687013 : auparse,level,videocrop,y4m,caca: Fix missing 
GST_PLUGINS_BASE_LIBS dependency in Makefile.am
  * 687154 : examples: error in level plugin example code
  * 687330 : videobox, videomixer: height obtained using _WIDTH macros
  * 687464 : speexdec: Don't unmap or finish_frame an invalid GstBuffer
  * 688382 : rtspsrc doesn't work with the interleaved protocols

ChangeLog
=
http://download.gnome.org/sources/gst-plugins-good/1.0/gst-plugins-good-1.0.3.changes
  (10.7K)

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gst-plugins-base 1.0.3

[Tim-Philipp Müller]

News


Changes since 1.0.2:

  * typefind: detect isml ftyp as iso-fragmented video/quicktime
  * typefinding improvements fixing playback of some wavpack files
  * textoverlay rendering fixes
  * gobject-introspection annotation fixes

Bugs fixed since 1.0.2:

  * 686276 : rtsp: http tunneling does not work
  * 687030 : ogg: crash checking header of empty ogg packet
  * 687055 : exiftag: fix use after free and memory leak
  * 687057 : vorbistag: fix memory leak
  * 687421 : GstRTSPConnection: a number of methods not annotated correctly
  * 687459 : textoverlay:forward allocation queries instead of discarding
  * 687473 : rtspconnection: remove extra 'return'
  * 687620 : GstRTSPMessage: fix GI annotations
  * 687666 : textoverlay: aborts when shading is enabled but not supported 
for the current video format
  * 687674 : typefinding: some WavPack files are mis-detected as AAC and 
fail to play
  * 687991 : videodecoder: add getter for QoS proportion
  * 687994 : rtsp: missing g-i annotation for gst_rtsp_message_set_body
  * 688151 : pbutils missing description for Opus codec
  * 686841 : pango: fix shadow text color

API additions since 1.0.2:

  * gst_video_decoder_get_qos_proportion()

ChangeLog
=
http://download.gnome.org/sources/gst-plugins-base/1.0/gst-plugins-base-1.0.3.changes
  (11.9K)

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gstreamer 1.0.2

[Tim-Philipp Müller]

News


Changes since 1.0.1:

  * Parallel installability with 0.10.x series
  * collectpads: ensure all timestamps are in same time domain
  * gobject-introspection: fix GstSample annotations
  * add GstElement::post_message() vfunc
  * info: allow gst_debug_add_log_function() to be called before gst_init()
  * event: allow GST_CLOCK_TIME_NONE as duration for GAP events
  * documentation: new chapters and updates for the application development 
manual and the plugin writer's guide

Bugs fixed since 1.0.1:

  * 686456 : g_type_init is no longer required
  * 686658 : allocator: fix memory leak in _fallback_mem_copy
  * 686659 : preset: remove variable not read
  * 686723 : basesrc: generic/states test in -base now hangs with appsrc
  * 686787 : testrtpool must be linked to pthreads

ChangeLog
=
http://download.gnome.org/sources/gstreamer/1.0/gstreamer-1.0.2.changes  (11.1K)

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  sha256sum: 10c59112bb01a274eb33950300da13d25fd23a7494781bf1b2328372fdb6ee06

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