gst-plugins-good 1.9.2



ChangeLog
=========

2016-09-01  Sebastian Dröge <slomo coaxion net>

        * configure.ac:
          releasing 1.9.2

2016-09-01 11:23:33 +0300  Sebastian Dröge <sebastian centricular com>

        * po/af.po:
        * po/az.po:
        * po/bg.po:
        * po/ca.po:
        * po/cs.po:
        * po/da.po:
        * po/de.po:
        * po/el.po:
        * po/en_GB.po:
        * po/eo.po:
        * po/es.po:
        * po/eu.po:
        * po/fi.po:
        * po/fr.po:
        * po/gl.po:
        * po/hr.po:
        * po/hu.po:
        * po/id.po:
        * po/it.po:
        * po/ja.po:
        * po/lt.po:
        * po/lv.po:
        * po/mt.po:
        * po/nb.po:
        * po/nl.po:
        * po/or.po:
        * po/pl.po:
        * po/pt_BR.po:
        * po/ro.po:
        * po/ru.po:
        * po/sk.po:
        * po/sl.po:
        * po/sq.po:
        * po/sr.po:
        * po/sv.po:
        * po/tr.po:
        * po/uk.po:
        * po/vi.po:
        * po/zh_CN.po:
        * po/zh_HK.po:
        * po/zh_TW.po:
          po: Update translations

2016-09-01 10:59:51 +0300  Sebastian Dröge <sebastian centricular com>

        * tests/examples/equalizer/demo.c:
        * tests/examples/spectrum/demo-audiotest.c:
        * tests/examples/spectrum/demo-osssrc.c:
          tests/examples: #define GDK_DISABLE_DEPRECATION_WARNINGS
          We use gdk_cairo_create() which is deprecated since 3.22.

2016-08-31 05:50:44 +1000  Jan Schmidt <jan centricular com>

        * sys/osxvideo/Makefile.am:
        * sys/osxvideo/cocoawindow.h:
        * sys/osxvideo/osxvideosink.h:
          osxvideo: Remove QuickTime references.
          QuickTime.h is no longer available on OS X 10.12 (Sierra),
          and both the header and the framework seem unnecessary
          for compilation - at least as of 10.11 (El Capitan).
          https://bugzilla.gnome.org/show_bug.cgi?id=770526

2016-08-19 11:11:03 -0700  Thibault Saunier <thibault saunier osg samsung com>

        * ext/dv/gstdvdemux.c:
        * ext/gdk_pixbuf/gstgdkpixbufdec.c:
        * gst/avi/gstavidemux.c:
        * gst/debugutils/rndbuffersize.c:
        * gst/flv/gstflvdemux.c:
        * gst/imagefreeze/gstimagefreeze.c:
        * gst/isomp4/qtdemux.c:
        * gst/matroska/matroska-demux.c:
        * gst/matroska/matroska-parse.c:
        * gst/multifile/gstsplitmuxsrc.c:
        * gst/rtsp/gstrtspsrc.c:
        * gst/wavparse/gstwavparse.c:
          Use the new API to post flow ERROR messages on the bus
          https://bugzilla.gnome.org/show_bug.cgi?id=770158

2016-08-26 21:32:07 +0200  Josep Torra <n770galaxy gmail com>

        * tests/check/elements/.gitignore:
          gitignore: ignore qtdemux, rtph261 and rtpvp9 tests

2016-08-26 21:22:16 +0200  Josep Torra <n770galaxy gmail com>

        * tests/check/Makefile.am:
          tests: use GST_NET_LIBS instead of hardcoded -lgstnet
          Fixes build in OSX when running 'make check' in gst-uninstalled.

2016-08-26 21:14:47 +0200  Josep Torra <n770galaxy gmail com>

        * tests/check/elements/rtp-payloading.c:
          tests: remove a wrong 'const' specifier
          Fixes "error: duplicate 'const' declaration specifier"

2016-08-26 21:11:59 +0200  Josep Torra <n770galaxy gmail com>

        * configure.ac:
        * tests/check/Makefile.am:
          build: silence error about pthread for 'make check' in osx
          Fixes "clang: error: argument unused during compilation: '-pthread'"

2016-08-26 20:31:10 +0300  Sebastian Dröge <sebastian centricular com>

        * tests/check/Makefile.am:
          vp9enc: Fix build of unit test by letting it link to libgstvideo

2016-08-26 12:06:35 -0400  Olivier Crête <olivier crete collabora com>

        * gst/rtpmanager/gstrtpmux.c:
        * gst/rtpmanager/gstrtpmux.h:
          Revert "rtpmux: fix PROP_TIMESTAMP_OFFSET range problems"
          This broke API, so we need a better solution!
          This reverts commit c7579d31a6e9d788e94b83258309063d0aae481e.

2016-06-08 15:06:28 +0200  Stian Selnes <stian pexip com>

        * gst/rtp/gstrtpvp9depay.c:
        * tests/check/Makefile.am:
        * tests/check/elements/rtpvp9.c:
          rtpvp9depay: Support flexible mode

2016-06-06 17:03:36 +0200  Stian Selnes <stian pexip com>

        * ext/vpx/gstvp9enc.c:
        * tests/check/Makefile.am:
        * tests/check/elements/vp9enc.c:
          vp9enc: Fix leak of vpx_image_t

2016-05-06 13:33:22 +0200  Stian Selnes <stian pexip com>

        * gst/rtp/gstrtph263pdepay.c:
        * tests/check/elements/rtph263.c:
          rtph263pdepay: Don't try to push empty frame
          If the result of depayloading is an empty frame, just drop it. This is
          likely the result of a buggy payloader.

2016-05-06 16:06:53 +0200  Havard Graff <havard graff gmail com>

        * gst/rtpmanager/gstrtpmux.c:
        * gst/rtpmanager/gstrtpmux.h:
          rtpmux: fix PROP_TIMESTAMP_OFFSET range problems
          It could not set the offset for the full guint32 range.

2016-05-06 09:44:42 +0200  Havard Graff <havard graff gmail com>

        * gst/rtpmanager/gstrtpbin.c:
        * gst/rtpmanager/gstrtpbin.h:
          rtpbin: introduce max-streams property
          To be able to cap the number of allowed streams for one session.
          This is useful for preventing DoS attacks, where a sender can change
          SSRC for every buffer, effectively bringing rtpbin to a halt.
          https://bugzilla.gnome.org/show_bug.cgi?id=770292

2016-03-31 00:10:49 +0200  Havard Graff <havard graff gmail com>

        * gst/rtpmanager/rtpsource.c:
          rtpsource: reordered packets are very normal, and should not be a warning

2016-02-05 14:19:25 +0100  Havard Graff <havard graff gmail com>

        * gst/rtpmanager/rtpsession.c:
          rtpsession: degrade g_warning to GST_ERROR
          So we don't blow up while investigating

2016-02-04 14:16:40 +0100  Stian Selnes <stian pexip com>

        * gst/rtp/gstrtph263pdepay.c:
        * tests/check/elements/rtph263.c:
          rtph263pdepay: Fix picture header for non-writable payload
          Under certain conditions gst_rtp_buffer_get_payload() returns a copy of
          the payload. In this case the payload modifications will not affect the
          rtp buffer. So instead of modifying the payload buffer directly we
          should modify the buffer that actually gets pushed on the adapter.

2015-11-19 11:50:47 +0100  Stian Selnes <stian pexip com>

        * gst/rtp/gstrtph261depay.c:
        * tests/check/Makefile.am:
        * tests/check/elements/rtph261.c:
          rtph261depay: Fix check of valid payload length
          Packets with no H.261 payload should be dropped to avoid invalid
          write/reads.

2015-11-09 10:06:21 +0100  Stian Selnes <stian pexip com>

        * gst/rtp/gstrtph263pay.c:
        * tests/check/elements/rtph263.c:
          rtph263pay: Fix double free, invalid reads and leak

2014-06-30 15:43:58 +0200  Stian Selnes <stian pexip com>

        * gst/rtpmanager/rtpsession.c:
          rtpsession: sanity check RTT before ignoring PLI/FIR

2014-06-30 15:07:45 +0200  Stian Selnes <stian pexip com>

        * gst/rtpmanager/rtpsession.c:
          rtpsession: handle sdes messages with non-utf8 more gracefully

2014-06-17 08:52:50 +0200  Stian Selnes <stian selnes gmail com>

        * gst/rtp/gstrtph263pay.c:
          rtph263pay: change log level on bitstream parsing messages

2016-07-07 11:13:18 +0200  Mikhail Fludkov <misha pexip com>

        * tests/check/elements/rtprtx.c:
          tests/rtprtx: refactor the tests to use gstharness
          The functionality of all the tests was kept exactly the same. Some tests
          were renamed:
          test_push_forward_seq -> test_rtxsend_rtxreceive
          test_drop_one_sender -> test_rtxsend_rtxreceive_with_packet_loss
          test_drop_multiple_sender -> test_multi_rtxsend_rtxreceive_with_packet_loss
          test_rtxreceive_data_reconstruction was testing that retransmitted
          buffer produced by rtxsend was correctly transformed to the original
          buffer by rtxreceive. Now we are checking for this in all the tests
          where both rtxsend & rtxreceive are involved. That's why the test was
          removed.

2016-08-25 15:52:36 +0200  Jonas Holmberg <jonashg axis com>

        * gst/rtp/gstrtph265pay.c:
          rtph265pay: Set RTP marker bit
          Set the RTP marker bit on the last RTP packet of an H.265 access unit.
          https://bugzilla.gnome.org/show_bug.cgi?id=770394

2016-07-26 19:39:58 +0200  Xabier Rodriguez Calvar <calvaris igalia com>

        * gst/videofilter/gstvideoflip.c:
        * gst/videofilter/gstvideoflip.h:
          videoflip: added GstVideoDirection interface
          It implements now this interface with its video-direction
          property. Values are changed to GstVideoOrientationMethod but they have
          the same value than the originals.
          https://bugzilla.gnome.org/show_bug.cgi?id=768687

2015-11-06 10:39:16 +0100  Havard Graff <havard graff gmail com>

        * gst/rtpmanager/gstrtpsession.c:
          gstrtpsession: refactor duplicate code into a function
          Less code, easier to read, more consistent.
          https://bugzilla.gnome.org/show_bug.cgi?id=770293

2016-08-23 17:06:44 +0100  Vincent Penquerc'h <vincent penquerch collabora co uk>

        * gst/rtpmanager/gstrtpbin.c:
          rtpbin: fix typo in max-misorder-time property name

2016-08-22 00:05:52 +0100  Tim-Philipp Müller <tim centricular com>

        * gst/multifile/gstsplitmuxsink.c:
          splitmuxsink: fix printf format compiler warning in debug message
          On 32-bit x86: gstsplitmuxsink.c:966:31: warning: format ‘%u’ expects
          argument of type ‘unsigned int’, but argument 9 has type
          ‘guint64 {aka long long unsigned int}’

2016-08-12 21:12:30 +0530  Nirbheek Chauhan <nirbheek centricular com>

        * .gitignore:
        * config.h.meson:
        * ext/cairo/meson.build:
        * ext/dv/meson.build:
        * ext/flac/meson.build:
        * ext/gdk_pixbuf/meson.build:
        * ext/jack/meson.build:
        * ext/jpeg/meson.build:
        * ext/libpng/meson.build:
        * ext/meson.build:
        * ext/pulse/meson.build:
        * ext/shout2/meson.build:
        * ext/soup/meson.build:
        * ext/speex/meson.build:
        * ext/taglib/meson.build:
        * ext/vpx/meson.build:
        * ext/wavpack/meson.build:
        * gst/alpha/meson.build:
        * gst/apetag/meson.build:
        * gst/audiofx/meson.build:
        * gst/audioparsers/meson.build:
        * gst/auparse/meson.build:
        * gst/autodetect/meson.build:
        * gst/avi/meson.build:
        * gst/cutter/meson.build:
        * gst/debugutils/meson.build:
        * gst/deinterlace/meson.build:
        * gst/dtmf/meson.build:
        * gst/effectv/meson.build:
        * gst/equalizer/meson.build:
        * gst/flv/meson.build:
        * gst/flx/meson.build:
        * gst/goom/meson.build:
        * gst/goom2k1/meson.build:
        * gst/icydemux/meson.build:
        * gst/id3demux/meson.build:
        * gst/imagefreeze/meson.build:
        * gst/interleave/meson.build:
        * gst/isomp4/meson.build:
        * gst/law/meson.build:
        * gst/level/meson.build:
        * gst/matroska/meson.build:
        * gst/meson.build:
        * gst/monoscope/meson.build:
        * gst/multifile/meson.build:
        * gst/multipart/meson.build:
        * gst/replaygain/meson.build:
        * gst/rtp/meson.build:
        * gst/rtpmanager/meson.build:
        * gst/rtsp/meson.build:
        * gst/shapewipe/meson.build:
        * gst/smpte/meson.build:
        * gst/spectrum/meson.build:
        * gst/udp/meson.build:
        * gst/videobox/meson.build:
        * gst/videocrop/meson.build:
        * gst/videofilter/meson.build:
        * gst/videomixer/meson.build:
        * gst/wavenc/meson.build:
        * gst/wavparse/meson.build:
        * gst/y4m/meson.build:
        * meson.build:
        * meson_options.txt:
        * sys/directsound/meson.build:
        * sys/meson.build:
        * sys/v4l2/meson.build:
        * sys/ximage/meson.build:
        * tests/check/meson.build:
        * tests/meson.build:
          Add support for Meson as alternative/parallel build system
          https://github.com/mesonbuild/meson
          With contributions from:
          Tim-Philipp Müller <tim centricular com>
          Jussi Pakkanen <jpakkane gmail com> (original port)
          Highlights of the features provided are:
          * Faster builds on Linux (~40-50% faster)
          * The ability to build with MSVC on Windows
          * Generate Visual Studio project files
          * Generate XCode project files
          * Much faster builds on Windows (on-par with Linux)
          * Seriously fast configure and building on embedded
          ... and many more. For more details see:
          http://blog.nirbheek.in/2016/05/gstreamer-and-meson-new-hope.html
          http://blog.nirbheek.in/2016/07/building-and-developing-gstreamer-using.html
          Building with Meson should work on both Linux and Windows, but may
          need a few more tweaks on other operating systems.

2016-08-20 16:59:30 +0800  Jie Jiang <jiangjie nudt edu cn>

        * gst/multifile/gstsplitmuxsink.c:
        * gst/multifile/gstsplitmuxsink.h:
          Fixed splitmuxsink 32-bit overflow bug
          Extend the byte tracking counters to 64-bit on
          all platforms, instead of using gsize, which overflows
          after 4GB.
          https://bugzilla.gnome.org/show_bug.cgi?id=770019

2016-08-19 17:18:16 +0300  Vivia Nikolaidou <vivia toolsonair com>

        * gst/isomp4/atoms.c:
          isomp4: Fix coverity warning
          If atom_copy_data fails to write anything, return 0
          CID #1371458

2016-04-09 07:51:03 +0530  Nirbheek Chauhan <nirbheek centricular com>

        * sys/v4l2/gstv4l2deviceprovider.c:
        * sys/v4l2/v4l2-utils.c:
          v4l2: consistently check #ifdef HAVE_GUDEV instead of #if
          Both work with autotools but they definitely don't mean the same thing, cause
          problems with other build systems, and are bad form. Existence should always be
          checked with #ifdef or #if defined.

2016-04-19 10:53:05 +0530  Nirbheek Chauhan <nirbheek centricular com>

        * sys/directsound/gstdirectsoundsink.c:
        * sys/directsound/gstdirectsoundsink.h:
          directsound: port away from old DirectX API
          D3DX has been deprecated for the last 4 years and latest versions of
          Windows no longer ship headers for it. This is fine as long as you're
          building with Cerbero's Wine-based DirectX headers, but sucks if you
          want to build against the actual Windows SDK.
          We were just using it to get error strings anyway, so just use the
          generic error string API.

2016-08-18 12:02:01 +0100  Tim-Philipp Müller <tim centricular com>

        * gst/audioparsers/gstflacparse.c:
          Revert "flacparse: Add maximum bitrate tag"
          This reverts commit c703ab69f526092bb26cce41ca691a896c8383d8.
          https://bugzilla.gnome.org/show_bug.cgi?id=769392

2016-08-18 09:57:51 +0300  Sebastian Dröge <sebastian centricular com>

        * tests/check/elements/rtpjitterbuffer.c:
          rtpjitterbuffer: Fix unit test by disabling adaptive misorder/dropout calculations
          Need to set max-misorder-time and max-dropout-time to 0 so the
          jitterbuffer does not base them on packet rate calculations.
          If it does, out gap is big enough to be considered a new stream and
          we wait for a few consecutive packets just to be sure
          https://bugzilla.gnome.org/show_bug.cgi?id=751311

2016-08-09 12:55:59 +0300  Vivia Nikolaidou <vivia ahiru eu>

        * gst/multifile/gstsplitmuxsink.c:
        * gst/multifile/gstsplitmuxsink.h:
          splitmuxsink: Add option to split at exactly max-size-time
          Will try to request a keyframe from the encoder to be sent at the target
          running time.
          https://bugzilla.gnome.org/show_bug.cgi?id=769664

2016-08-09 20:16:16 +0300  Vivia Nikolaidou <vivia ahiru eu>

        * gst/multifile/gstsplitmuxsink.c:
          splitmuxsink: Allow time and bytes to reach their respective thresholds
          https://bugzilla.gnome.org/show_bug.cgi?id=769664

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

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

2016-06-24 16:32:37 +0300  Vivia Nikolaidou <vivia toolsonair com>

        * gst/isomp4/atoms.c:
        * gst/isomp4/atoms.h:
        * gst/isomp4/fourcc.h:
        * gst/isomp4/gstqtmux.c:
        * gst/isomp4/gstqtmux.h:
          qtmux: Added support for writing timecode track
          https://bugzilla.gnome.org/show_bug.cgi?id=767950

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

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

2016-08-10 11:45:13 -0600  Thomas Bluemel <tbluemel control4 com>

        * gst/rtpmanager/gstrtpjitterbuffer.c:
        * gst/rtpmanager/rtpstats.c:
          rtpjitterbuffer: Actually calculate the packet rate for max-dropout and max-misorder calculations.
          https://bugzilla.gnome.org/show_bug.cgi?id=751311

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

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

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

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

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

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

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

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

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

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

2016-08-04 00:36:28 -0300  Thiago Santos <thiagossantos gmail com>

        * gst/rtpmanager/gstrtpjitterbuffer.c:
          rtpjitterbuffer: avoid unref of null buffer
          The current 'l' pointer will be NULL when the loop
          is interrupted with a 'break' statement. Need to have
          it advance to the next list item before interrupting.

2016-08-02 14:01:14 +0200  Carlos Rafael Giani <dv pseudoterminal org>

        * gst/wavparse/Makefile.am:
        * gst/wavparse/gstwavparse.c:
          wavparse: Add tags for container format and bitrate for uncompressed PCM
          The PCM bitrate is added to help downstream elements (like uridecodebin)
          figure out a proper network buffer size
          https://bugzilla.gnome.org/show_bug.cgi?id=769390

2016-08-01 18:52:26 +0200  Carlos Rafael Giani <dv pseudoterminal org>

        * gst/audioparsers/gstflacparse.c:
          flacparse: Add maximum bitrate tag
          https://bugzilla.gnome.org/show_bug.cgi?id=769392

2016-07-28 17:58:16 +0300  Sebastian Dröge <sebastian centricular com>

        * gst/isomp4/qtdemux.c:
          qtdemux: When receiving a DISCONT buffer that does not point to a sample, remember the offset
          And don't just reset everything. This makes sure that we can continue to
          handle data in the following scenario:
          moov: discont
          moof: discont
          mdat: continuous
          Previously this would fail because the offset would be the accumulated offset
          from moov and moof at the mdat position, while the buffer offset might be
          something completely different.

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

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

2016-03-03 11:35:06 +0000  Vincent Penquerc'h <vincent penquerch collabora co uk>

        * ext/soup/gstsouphttpsrc.c:
          souphttpsrc: include http-status-code in error message details
          https://bugzilla.gnome.org/show_bug.cgi?id=763038

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

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

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

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

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

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

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

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

2016-07-12 17:30:56 +0200  Guillaume Desmottes <guillaume desmottes collabora co uk>

        * tests/check/elements/qtdemux.c:
          tests: qtdemux: fix element and pad leak
          https://bugzilla.gnome.org/show_bug.cgi?id=768739

2016-07-12 16:45:36 +0200  Guillaume Desmottes <guillaume desmottes collabora co uk>

        * tests/check/elements/audiofirfilter.c:
        * tests/check/elements/audioiirfilter.c:
        * tests/check/elements/rtp-payloading.c:
        * tests/check/elements/videobox.c:
        * tests/check/pipelines/effectv.c:
          tests: fix bus leaks
          gst_bus_add_signal_watch() takes a ref on the bus which should be
          released using gst_bus_remove_signal_watch().
          https://bugzilla.gnome.org/show_bug.cgi?id=768739

2016-07-14 03:07:11 +0800  Ting-Wei Lan <lantw src gnome org>

        * configure.ac:
          configure: Call AG_GST_PKG_CONFIG_PATH to set GST_PKG_CONFIG_PATH
          GST_PKG_CONFIG_PATH is used in docs/plugins directory, so
          AG_GST_PKG_CONFIG_PATH must be called to set it.
          https://bugzilla.gnome.org/show_bug.cgi?id=768787

2016-07-12 07:39:58 +0200  Edward Hervey <edward centricular com>

        * ext/soup/gstsouphttpsrc.c:
          souphttpsrc: Don't drop final bytes of a range request
          At the end of a range request, we don't want to return GST_FLOW_EOS otherwise
          the last bytes we just read will be dropped by basesrc.
          Instead just return GST_FLOW_OK (which was set just before) and let basesrc
          handle the fact we are at the end of the segment.

2016-07-11 18:30:18 -0400  Nicolas Dufresne <nicolas dufresne collabora com>

        * sys/v4l2/gstv4l2deviceprovider.c:
          v4l2provider: Fix device type detection
          The type detection would lead to assertion as it would try
          to create a device without having found any type for it. It
          also didn't detect MPLANE devices properly.

2016-07-11 18:29:01 -0400  Nicolas Dufresne <nicolas dufresne collabora com>

        * sys/v4l2/gstv4l2object.c:
          v4l2object: Don't assert when used by the monitor
          The monitor sets the object->element object as a GstObject. This
          works for debug traces, but will assert for ELEMENT_ERROR. This
          was the only case where that could happen. Add a check for that.

2016-07-11 17:38:00 -0400  Nicolas Dufresne <nicolas dufresne collabora com>

        * sys/v4l2/gstv4l2object.c:
          v4l2object: Indent very long line

2016-07-12 00:42:02 +0300  Sebastian Dröge <sebastian centricular com>

        * ext/soup/gstsouphttpsrc.c:
          souphttpsrc: At the end of a range request, read another time to finalize the request
          If we're at the end of a range request, read again to let libsoup
          finalize the request. This allows to reuse the connection again later,
          otherwise we would have to cancel the message and close the connection.

2016-07-11 21:13:47 +0200  Stefan Sauer <ensonic users sf net>

        * common:
          Automatic update of common submodule
          From f363b32 to f49c55e

2016-07-11 19:57:18 +0300  Sebastian Dröge <sebastian centricular com>

        * ext/soup/gstsouphttpsrc.c:
          souphttpsrc: Fix keep-alive handling
          We have to get rid of the message on EOS when the complete stream is read to
          remember that we successfully finished handling this specific message.
          Otherwise we will cancel it later and close the connection instead of reusing
          it at a later time.
          It might also make sense to reuse connections if a non-200 response is
          received. As long as there was no connection error, the HTTP connection should
          be re-usable.

2016-07-11 12:05:06 -0400  Nicolas Dufresne <nicolas dufresne collabora com>

        * configure.ac:
          Also enable V4L2 probe on aarch64 (aka ARM 64bit)

2016-07-11 11:59:19 -0400  Olivier Crête <olivier crete collabora com>

        * tests/examples/rtp/client-PCMA.c:
          rtp example: Fix leak
          Also stop fetching the internal source as this
          functionality has been broken.

2016-07-08 14:58:37 -0400  Nicolas Dufresne <nicolas dufresne collabora com>

        * configure.ac:
          Enable v4l2 probe on Linux/ARM
          Most of those have V4L2 drivers these days enabling it make sure that it
          this code is enabled in major distribution, hence that HW accelerated
          decoder/encoder can be used on platforms that support it. The probes are
          slightly increasing the first init of gstreamer library, though the
          result is cached in the registry for later use.

2016-07-11 09:46:49 +0200  Jonas Holmberg <jonashg axis com>

        * gst/rtp/gstrtph265pay.c:
        * tests/check/elements/rtp-payloading.c:
          rtph265pay: Accept array_completeness=1
          When parsing NAL unit type in codec_data, check the 6bits of
          NAL_unit_type only and do not require the array_completeness bit to be
          0, since the default and mandatory value of array_completeness is 1 for
          hvc1.
          https://bugzilla.gnome.org/show_bug.cgi?id=768653

2016-07-10 21:35:06 -0400  Nicolas Dufresne <nicolas dufresne collabora com>

        * sys/v4l2/v4l2_calls.c:
          v4l2: Also copy device_caps in gst_v4l2_dup
          This fixes regression where M2M error out saying they have no output
          format (the V4L2 CAPTURE side).
          https://bugzilla.gnome.org/show_bug.cgi?id=768195

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

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

2016-07-08 17:28:19 +0000  Luis de Bethencourt <luisbg osg samsung com>

        * sys/v4l2/v4l2_calls.c:
          v4l2: fix multiplanar capture
          After switching to using V4L2_CAP_DEVICE_CAPS we lost support for
          multiplanar device types. After some research, it looks like
          vcap.capabilities treated the multiplanar flag of output and capture
          devices equally, but not the new device_caps.
          https://bugzilla.gnome.org/show_bug.cgi?id=768195

2016-07-08 14:56:30 +0200  Mats Lindestam <matslm axis com>

        * gst/multipart/multipartmux.c:
        * gst/multipart/multipartmux.h:
          multipartmux: Use PTS and DTS instead of timestamp
          And pass-through both of them.
          Based on a patch by Göran Jönsson <goranjn axis com>
          https://bugzilla.gnome.org/show_bug.cgi?id=767900

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

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

2016-06-23 15:30:19 +0200  Edward Hervey <edward centricular com>

        * gst/isomp4/qtdemux.c:
          qtdemux: Let upstream events go through upstream
          There's no real reason to avoid sending QOS/NAVIGATION events upstrea.
          Some elements might want to have that information.

2016-06-23 15:22:56 +0200  Edward Hervey <edward centricular com>

        * gst/avi/gstavidemux.c:
          avidemux: Let upstream events go through upstream
          There's no real reason to avoid sending QOS/NAVIGATION events upstrea.
          Some elements might want to have that information.

2016-06-23 15:17:36 +0200  Edward Hervey <edward centricular com>

        * ext/dv/gstdvdemux.c:
          dvdemux: Let upstream events go through upstream
          There's no real reason to avoid sending QOS/NAVIGATION events upstrea.
          Some elements might want to have that information.
          Also remove downstream-only CAPS event handling and minimize code

2016-07-07 23:53:54 +0100  Luis de Bethencourt <luisbg osg samsung com>

        * sys/v4l2/gstv4l2.c:
          v4l2: fix v4l2 probe build error
          A typo in gst_v4l2_probe_and_register() caused a build error when building
          with --enable-v4l2-probe. Fixing it.
          gstv4l2.c: In function 'gst_v4l2_probe_and_register':
          gstv4l2.c:150:25: error: 'struct v4l2_capability' has no member named 'capabilitites'
          device_caps = vcap.capabilitites;

2016-07-01 22:53:33 -0700  Reynaldo H. Verdejo Pinochet <reynaldo osg samsung com>

        * sys/v4l2/gstv4l2src.c:
          v4l2src: use gst_caps_intersect_full in negotiate()
          Instead of reimplementing the GST_CAPS_INTERSECT_FIRST
          interection mode.
          https://bugzilla.gnome.org/show_bug.cgi?id=768195

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

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

2016-07-07 18:24:59 +0300  Sebastian Dröge <sebastian centricular com>

        * gst/matroska/matroska-mux.c:
          matroskamux: Remove suspicious checks for pads being active and linked
          We should add all pads, no matter if they are linked or active or not at this
          point. Skipping some that are not will cause different behaviour than with
          other muxers.

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

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

2016-07-07 18:14:43 +0300  Sebastian Dröge <sebastian centricular com>

        * gst/matroska/matroska-mux.c:
          matroskamux: Consistently use gst_matroska_mux_set_codec_id() for setting the codec id

2016-07-04 09:50:11 +0200  Jonas Holmberg <jonashg axis com>

        * gst/rtp/gstrtph265depay.c:
        * gst/rtp/gstrtph265pay.c:
        * gst/rtp/gstrtph265pay.h:
        * tests/check/elements/rtp-payloading.c:
          rtph265pay/depay: Sync against RFC 7798
          Handle sprop-vps, sprop-sps and sprop-pps in caps instead of
          sprop-parameter-sets.
          rtph265pay works with byte-stream and hvc1 formats but not hev1 yet. It
          handles profile-id, tier-flag and level-id in caps query.
          https://bugzilla.gnome.org/show_bug.cgi?id=753760

2016-07-06 09:25:00 +0200  Jan Alexander Steffens (heftig) <jan steffens gmail com>

        * gst/flv/gstflvdemux.c:
        * gst/flv/gstflvdemux.h:
          flvdemux: Push nominal bitrate tags
          Add per-stream tag lists, which are used to send nominal
          bitrate tags. When remuxing FLV => FLV, this now passes
          through the upstream bitrate.
          https://bugzilla.gnome.org/show_bug.cgi?id=768440

2016-07-06 09:24:49 +0200  Jan Alexander Steffens (heftig) <jan steffens gmail com>

        * gst/flv/gstflvdemux.c:
        * gst/flv/gstflvdemux.h:
          flvdemux: Refactor metadata tag handling
          The FLV header cannot be trusted to indicate video or
          audio presence, as the comments already mention. Don't
          delay pushing tags waiting for streams that might never
          appear.
          Tags are now pushed immediately after they change:
          - After parsing an onMetaData script object
          - After negotiating caps on a pad
          https://bugzilla.gnome.org/show_bug.cgi?id=768440

2016-07-06 12:44:10 +0100  Luis de Bethencourt <luisbg osg samsung com>

        * gst/isomp4/qtdemux.c:
          qtdemux: fix AAC codec_data values
          As seen in the parent switch for object_type_id, the 4 possible values are
          0x40, 0x66, 0x67 and 0x68. Fixing the nested switch to match these values.
          Looks like it was a typo making them decimal instead of hexadecimal.
          CID 1363328

2016-07-06 13:51:03 +0300  Sebastian Dröge <sebastian centricular com>

        * configure.ac:
          Back to development



Download
========
https://download.gnome.org/sources/gst-plugins-good/1.9/gst-plugins-good-1.9.2.tar.xz (3.23M)
  sha256sum: 3a27b7f770432c1275b555efa8df1762762446d68565bfeff71500a31925d8c6



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