gst-plugins-good 1.9.90

[Sebastian Dröge <install-module master gnome org>]

ChangeLog
=========

2016-09-30  Sebastian Dröge <slomo coaxion net>

        * configure.ac:
          releasing 1.9.90

2016-09-30 11:43:54 +0300  Sebastian Dröge <sebastian centricular com>

        * po/el.po:
          po: Update translations

2016-09-30 13:22:32 +0530  Arun Raghavan <arun osg samsung com>

        * tests/check/pipelines/tagschecking.c:
          tests: Fix tagschecking failure due to missing PTS
          qtmux now needs the PTS (commit a993883b7), so let's make sure we
          produce one with our buffers.
          https://bugzilla.gnome.org/show_bug.cgi?id=772228

2016-09-28 23:03:58 +0300  Sebastian Dröge <sebastian centricular com>

        * gst/isomp4/gstqtmux.c:
          qtmux: Don't calculate PTS offset and DTS with GST_CLOCK_TIME_NONE
          Just error out if there is no valid PTS.
          https://bugzilla.gnome.org/show_bug.cgi?id=772143

2016-09-29 17:37:28 +0300  Sebastian Dröge <sebastian centricular com>

        * gst/isomp4/qtdemux_types.c:
          qtdemux: Add JPEG2000 ihdr atom to the list of known ones
          Otherwise qtdemux is always going to complain about it being unknown.

2016-09-29 10:19:56 +0300  Sebastian Dröge <sebastian centricular com>

        * gst/matroska/matroska-mux.c:
          matroskamux: Always write the default frame duration for VP8/9 too
          The WebM spec allows this now, and it allows us to guess a framerate.
          See https://bugzilla.gnome.org/show_bug.cgi?id=772141 and
          also https://bugzilla.gnome.org/show_bug.cgi?id=654379

2016-09-27 15:26:19 -0400  Olivier Crête <olivier crete collabora com>

        * gst/rtp/gstrtph264depay.c:
        * gst/rtp/gstrtph265depay.c:
          rtph26[45]depay: Don't handle NALs inside STAP units twice
          They've already been handled before pushing them into the adapter.

2016-09-27 12:39:12 +0100  Tim-Philipp Müller <tim centricular com>

        * tests/check/meson.build:
          meson: tests: fix vp8 availability checks
          Those variables are not defined if vp8 was not found.

2016-09-27 10:23:38 +0100  Tim-Philipp Müller <tim centricular com>

        * gst/multifile/gstmultifilesink.c:
          Revert "multifilesink: streamline the file-switch code a bit"
          This reverts commit f1ceaab02f3f557e23b77b14771a575788f92bb4.
          This broke atomic file writes in "buffer" mode. It did make
          sure that any streamheaders are prepended to each file in
          buffer mode as well, but that's not really needed in practice,
          whereas atomic file writes are, so let's restore the status
          quo ante for now since this was primarily a code cleanup anyway,
          and if anyone needs to streamheaders in buffer mode too they
          can make a patch to implement that differently. Re-implementing
          the atomic writes in the element also seems way too much work.
          https://bugzilla.gnome.org/show_bug.cgi?id=766990

2016-09-27 10:22:57 +0100  Tim-Philipp Müller <tim centricular com>

        * gst/multifile/gstmultifilesink.c:
          Revert "multifilesink: close file on write error with next-file mode is set to buffer"
          This reverts commit 84e441d2685cf223d348a95be0c5ba693bbf6624.
          This will no longer be needed once we revert f1ceaab02.

2016-09-26 13:22:29 -0300  Thibault Saunier <thibault saunier osg samsung com>

        * tests/check/meson.build:
          meson: Add gst-plugins-base plugins directories to be used by tests

2016-09-26 14:30:00 +0100  Tim-Philipp Müller <tim centricular com>

        * ext/vpx/meson.build:
        * meson.build:
        * tests/check/getpluginsdir:
        * tests/check/meson.build:
          meson: add unit tests
          Only works properly in an installed setup currently, most
          likely won't work with a subprojects setup yet.

2016-09-24 09:36:24 +0100  Tim-Philipp Müller <tim centricular com>

        * meson.build:
        * po/meson.build:
          meson: hook up translations

2016-09-08 17:30:41 +0530  Arun Raghavan <arun arunraghavan net>

        * ext/pulse/pulsesrc.c:
          pulsesrc: Don't negotiate to less than two segments
          GstAudioRingBuffer doesn't needs us to have at least 2 segments. We make
          sure that if our buffer parameters are such that the maxlength is not at
          least 2x fragsize, we still request the ringbuffer to keep that much
          space so it continues to work.
          https://bugzilla.gnome.org/show_bug.cgi?id=770446

2016-09-24 23:22:01 +0530  Arun Raghavan <arun arunraghavan net>

        * gst/rtp/gstrtpsbcpay.c:
        * gst/rtp/gstrtpsbcpay.h:
          rtpsbcpay: Fix timestamping
          We were just picking the timestamp of the last buffer pushed into our
          adapter before we had enough data to push out.
          This fixes things to figure out how large each frame is and what
          duration it covers, so we can set both the timestamp and duration
          correctly.
          Also adds some DISCONT handling.

2016-07-12 18:14:52 +0200  Georg Lippitsch <glippitsch toolsonair com>

        * gst/isomp4/gstqtmux.c:
          qtmux: Fix fourcc for ProRes Proxy
          This is apco, according to
          https://wiki.multimedia.cx/index.php?title=Apple_ProRes
          https://bugzilla.gnome.org/show_bug.cgi?id=769048

2016-09-18 20:55:31 +0100  Tim-Philipp Müller <tim centricular com>

        * ext/vpx/meson.build:
          meson: fix build with vpx 1.3.x
          vpx >= 1.4.0 is optional

2016-09-15 18:19:35 +0200  Sebastian Dröge <sebastian centricular com>

        * gst/rtsp/gstrtspsrc.c:
          rtspsrc: Use new bin suppressed flags API for managing the element flags

2016-09-15 09:52:31 +0100  Tim-Philipp Müller <tim centricular com>

        * ext/jack/gstjackaudioclient.c:
        * gst/rtp/dboolhuff.c:
        * gst/rtpmanager/rtpsession.c:
        * gst/videofilter/gstvideoflip.c:
          ext, gst: fix indentation

2016-09-15 09:52:17 +0100  Tim-Philipp Müller <tim centricular com>

        * tests/check/elements/flvmux.c:
        * tests/check/elements/rtph263.c:
        * tests/check/elements/rtpjitterbuffer.c:
        * tests/check/elements/rtpsession.c:
        * tests/check/elements/rtpvp9.c:
          tests: fix indentation

2016-08-11 11:04:22 -0600  Thomas Bluemel <tbluemel control4 com>

        * gst/rtpmanager/gstrtpjitterbuffer.c:
          rtpjitterbuffer: Fix calculating next_seqnum when dropping old buffers from a full queue.
          Fixes calculating the next sequence number when a ITEM_TYPE_LOST with more than one
          definitely lost packets is encountered.
          https://bugzilla.gnome.org/show_bug.cgi?id=769757

2016-08-11 23:07:44 +0200  Havard Graff <havard graff gmail com>

        * gst/rtpmanager/gstrtpjitterbuffer.c:
        * tests/check/elements/rtpjitterbuffer.c:
          rtpjitterbuffer: improved rtx-rtt averaging
          The basic idea is this:
          1. For *larger* rtx-rtt, weigh a new measurement as before
          2. For *smaller* rtx-rtt, be a bit more conservative and weigh a bit less
          3. For very large measurements, consider them "outliers"
          and count them a lot less
          The idea being that reducing the rtx-rtt is much more harmful then
          increasing it, since we don't want to be underestimating the rtt of the
          network, and when using this number to estimate the latency you need for
          you jitterbuffer, you would rather want it to be a bit larger then a bit
          smaller, potentially losing rtx-packets. The "outlier-detector" is there
          to prevent a single skewed measurement to affect the outcome too much.
          On wireless networks, these are surprisingly common.
          https://bugzilla.gnome.org/show_bug.cgi?id=769768

2016-08-05 12:51:59 +0200  Stian Selnes <stian pexip com>

        * gst/rtpmanager/gstrtpjitterbuffer.c:
        * tests/check/elements/rtpjitterbuffer.c:
          rtpjitterbuffer: Detect whether to assume equidistant spacing when loss
          Assuming equidistant packet spacing when that's not true leads to more
          loss than necessary in the case of reordering and jitter. Typically this
          is true for video where one frame often consists of multiple packets
          with the same rtp timestamp. In this case it's better to assume that the
          missing packets have the same timestamp as the last received packet, so
          that the scheduled lost timer does not time out too early causing the
          packets to be considered lost even though they may arrive in time.
          https://bugzilla.gnome.org/show_bug.cgi?id=769768

2016-07-27 10:39:50 +0200  Stian Selnes <stian pexip com>

        * gst/rtpmanager/gstrtpjitterbuffer.c:
        * tests/check/elements/rtpjitterbuffer.c:
          rtpjitterbuffer: Don't request rtx if 'now' is past retry period
          There is no need to schedule another EXPECTED timer if we're already
          past the retry period. Under normal operation this won't happen, but if
          there are more timers than the jitterbuffer is able to process in
          real-time, scheduling more timers will just make the situation worse.
          Instead, consider this packet as lost and move on. This scenario can
          occur with high loss rate, low rtt and high configured latency.
          https://bugzilla.gnome.org/show_bug.cgi?id=769768

2016-07-26 18:01:48 +0200  Stian Selnes <stian pexip com>

        * gst/rtpmanager/gstrtpjitterbuffer.c:
        * tests/check/elements/rtpjitterbuffer.c:
          rtpjitterbuffer: Fix lost duration when gap after lost timer
          This patch fixes an issue with the estimated gap duration when there is
          a gap immediately after a lost timer has been processed. Previously
          there was a discrepancy beteen the gap in seqnum and gap in dts which
          would cause wrong calculated duration. The issue would only be seen with
          retranmission enabled since when it's disabled lost timers are only
          created when a packet is received and the actual gap length and last dts
          is known.
          https://bugzilla.gnome.org/show_bug.cgi?id=769768

2016-07-19 01:11:58 +0200  Havard Graff <havard graff gmail com>

        * gst/rtpmanager/gstrtpjitterbuffer.c:
          rtpjitterbuffer: Expose rtx-deadline as a property
          The default -1 gives the old behavior.
          https://bugzilla.gnome.org/show_bug.cgi?id=769768

2016-08-11 12:02:19 +0200  Havard Graff <havard graff gmail com>

        * gst/rtpmanager/gstrtpjitterbuffer.c:
        * tests/check/elements/rtpjitterbuffer.c:
          rtpjitterbuffer: Improved expected-timer handling when gap > 0
          https://bugzilla.gnome.org/show_bug.cgi?id=769768

2016-08-11 11:51:50 +0200  Stian Selnes <stian pexip com>

        * gst/rtpmanager/gstrtpjitterbuffer.c:
        * tests/check/elements/rtpjitterbuffer.c:
          rtpjitterbuffer: Major improvements for RTX stats
          Stats should also be collected for unsuccessful packets.
          rtx-rtt is very important for determining the necessary configured
          latency on the jitterbuffer. It's especially important to be able to
          increase the latency when retransmitted packets arrive too late and are
          considered lost. This patch includes these late packets in the
          calculation of the various rtx stats, making them more correct and
          useful.
          Also in the case where the original packet arrives after a NACK is sent,
          the received RTX packet should update the stats since it provides useful
          information about RTT.
          The RTT is only updated if and only if all requested retranmissions are
          received. That way the RTT is guaranteed to make sense. If not we don't
          know which request the packet is a response to and the RTT may be bogus.
          A consequence of this patch is that RTT is not updated for a request
          when one of the RTX packets for that seqnum is lost, but that since
          measured RTT will be more accurate.
          The implementation store the RTX information from the timed out timers
          and use this when the retransmitted packet arrives. For performance
          these timers are stored separately from the "normal" timers in order to
          not impact performance (see attached performance test).
          https://bugzilla.gnome.org/show_bug.cgi?id=769768

2016-08-11 11:02:44 +0200  Havard Graff <havard graff gmail com>

        * gst/rtpmanager/gstrtpjitterbuffer.c:
        * tests/check/elements/rtpjitterbuffer.c:
          rtpjitterbuffer: Add and expose more stats and increase testing of it
          Add num-pushed and num-lost.
          Expose num-late, num-duplicates and avg-jitter.
          https://bugzilla.gnome.org/show_bug.cgi?id=769768

2016-07-07 10:20:02 +0200  Stian Selnes <stian pexip com>

        * gst/rtpmanager/gstrtprtxreceive.c:
          rtxreceive: Set buffer flag for retransmitted packets
          https://bugzilla.gnome.org/show_bug.cgi?id=769768

2016-07-09 23:47:41 +0200  Havard Graff <havard graff gmail com>

        * gst/rtpmanager/gstrtpjitterbuffer.c:
          rtpjitterbuffer: Option to disable rtx-delay-reorder
          When disabled we can save some iterations over timers.
          There is probably an argument for rtx-delay-reorder to exist, but
          for normal operations, handling jitter (reordering) is something a
          jitterbuffer should do, and this variable feels like functionality that
          is not "in-sync" with what the jitterbuffer is trying to achieve.
          Example: You have 50ms jitter on your network, and are receiving
          audio packets with 10ms durations. An audio packet should not be
          considered late until its rtx-timeout has expired (and hence a rtx-event
          is sent), but with rtx-delay-reorder, events will be sent pretty much
          all the time due to the jitter on the network.
          Point being: The jitterbuffer should adapt its size to the measured network
          jitter, and then rtx-delay-reorder needs to adapt as well, or simply
          get out of the way and let the other (better) rtx-mechanisms do their job.
          Also change find_timer to only use seqnum as an argument, since there
          will only ever be one timer per seqnum at any given time. In the
          one case where the type matters, the caller simply checks the type.
          https://bugzilla.gnome.org/show_bug.cgi?id=769768

2016-09-14 09:58:41 -0400  Olivier Crête <olivier crete collabora com>

        * gst/rtp/gstrtph263pay.c:
          rtph263pay: Fix double free from coverity
          CID #1372887

2016-09-14 09:58:37 -0400  Olivier Crête <olivier crete collabora com>

        * gst/rtp/gstrtph263pay.c:
          rtph263pay: Indent as per gst-indent

2016-09-14 11:30:41 +0200  Sebastian Dröge <sebastian centricular com>

        * configure.ac:
          configure: Depend on gstreamer 1.9.2.1

2016-09-14 10:17:02 +0900  Wonchul Lee <wonchul lee collabora com>

        * gst/autodetect/gstautodetect.c:
          autodetect: Use gst_bin_set_suppressed_flags() API
          https://bugzilla.gnome.org/show_bug.cgi?id=771395

2016-09-09 15:36:12 +0200  Thomas Scheuermann <Thomas Scheuermann barco com>

        * ext/jack/gstjackaudioclient.c:
          jack: Fix pipeline hang when jack changes sample rate or buffer size
          If jackd changes the buffer size or sample rate, jackaudiosink hangs
          and can't be stopped. This also happens if jack is configured as slave
          and a gstreamer pipeline is started on the slave machine while the jack
          master isn't running yet. If the the jack master is started it changes
          the buffer size / sample rate and jackaudiosink can't be stopped.
          This fix calls jack_shutdown_cb when jack_sample_rate_cb or
          jack_buffer_size_cb is called.
          https://bugzilla.gnome.org/show_bug.cgi?id=771272

2016-09-12 20:08:36 +0200  Sebastian Dröge <sebastian centricular com>

        * gst/deinterlace/gstdeinterlace.c:
          deinterlace: Fix field ordering for reverse playback
          And actually calculate the field duration instead of a frame duration so
          that we can properly timestamp output frames in fields=all mode.
          This is probably still broken for reverse playback in telecine mode.

2016-09-12 09:02:00 +0000  Thomas Klausner <tk giga or at>

        * gst/udp/gstudpsrc.c:
          udpsrc: Fix compilation on NetBSD
          https://bugzilla.gnome.org/show_bug.cgi?id=771278

2016-09-10 20:51:10 +1000  Jan Schmidt <jan centricular com>

        * autogen.sh:
        * common:
          Automatic update of common submodule
          From b18d820 to f980fd9

2016-09-09 14:02:25 +0200  Xabier Rodriguez Calvar <calvaris igalia com>

        * gst/isomp4/qtdemux.c:
          qtdemux: offset is irrelevant when no crypto info
          Cause later it will try to use the crypto info array to get an index and
          attach on of the positions as buffer's crypto info.
          https://bugzilla.gnome.org/show_bug.cgi?id=770951

2016-09-10 09:53:57 +1000  Jan Schmidt <jan centricular com>

        * autogen.sh:
        * common:
          Automatic update of common submodule
          From f49c55e to b18d820

2016-09-07 15:33:30 -0400  Nicolas Dufresne <nicolas dufresne collabora com>

        * sys/osxaudio/Makefile.am:
          osxaudio: Distribute device provider files
          Those where missing the the dev release tarballs for 1.9.2 which
          prevented building from tarball on OSX platform

2016-09-06 09:49:39 +0200  Xabier Rodriguez Calvar <calvaris igalia com>

        * gst/isomp4/qtdemux.c:
          qtdemux: Fix crash with no cenc aux offset
          https://bugzilla.gnome.org/show_bug.cgi?id=770951

2016-09-05 09:39:33 +0100  Vincent Penquerc'h <vincent penquerch collabora co uk>

        * gst/audioparsers/gstaacparse.c:
          aacparse: parse a bit more of the humongous LOAS data
          https://bugzilla.gnome.org/show_bug.cgi?id=769278

2016-09-05 09:39:08 +0100  Vincent Penquerc'h <vincent penquerch collabora co uk>

        * gst/audioparsers/gstaacparse.c:
          aacparse: make it clear when a potential LOAS frame is not one
          https://bugzilla.gnome.org/show_bug.cgi?id=769278

2016-09-05 09:38:26 +0100  Vincent Penquerc'h <vincent penquerch collabora co uk>

        * gst/audioparsers/gstaacparse.c:
          aacparse: add a few comments to anchor parsing to the spec
          https://bugzilla.gnome.org/show_bug.cgi?id=769278

2016-09-05 09:37:02 +0100  Vincent Penquerc'h <vincent penquerch collabora co uk>

        * gst/audioparsers/gstaacparse.c:
        * gst/audioparsers/gstaacparse.h:
          aacparse: improve channel/rate handling
          Keep track of the last parsed channels/rate fields so they can be
          used even if the element was not yet configured.
          https://bugzilla.gnome.org/show_bug.cgi?id=769278

2016-09-05 09:35:53 +0100  Vincent Penquerc'h <vincent penquerch collabora co uk>

        * gst/audioparsers/gstaacparse.c:
          aacparse: fix varlength number reading as per spec
          https://bugzilla.gnome.org/show_bug.cgi?id=769278

2016-09-05 09:35:02 +0100  Vincent Penquerc'h <vincent penquerch collabora co uk>

        * gst/audioparsers/gstaacparse.c:
          aacparse: strip uneeded static arrays slack
          https://bugzilla.gnome.org/show_bug.cgi?id=769278

2016-07-18 19:18:58 -0400  Olivier Crête <olivier crete collabora com>

        * gst/rtp/gstrtpmp4adepay.c:
        * gst/rtp/gstrtpmp4adepay.h:
          rtpmp4adepay: Only declare a stream to be framed once a marker bit has been seen
          This may cause a few packets to be processed by the parser, but it's
          better than never pushing out buffers from a slightly broken stream
          where no marker bits are set.

2016-09-06 14:25:42 +0300  Sebastian Dröge <sebastian centricular com>

        * ext/dv/gstdvdemux.c:
          dvdemux: Fix timestamping in reverse playback mode
          This is only supported right now if after a demuxer that supports reverse
          playback, e.g. with DV container inside AVI container.

2016-09-05 12:23:54 -0300  Thibault Saunier <thibault saunier osg samsung com>

        * meson.build:
          meson: Bump version to 1.9.2

2015-06-26 20:13:17 +0200  Mathieu Duponchelle <mathieu duponchelle opencreed com>

        * gst/isomp4/GstQTMux.prs:
        * gst/isomp4/Makefile.am:
        * gst/isomp4/gstqtmux.c:
          qtmux: Implement the preset interface.
          + And provide a "youtube" preset, which based on
          https://support.google.com/youtube/answer/1722171 sets
          faststart to True.
          https://bugzilla.gnome.org/show_bug.cgi?id=751559

2016-09-01 12:27:35 +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-1.9.90.tar.xz (3.23M)
  sha256sum: 75a1ac367785964a5767e31e447967d1aa51ca5af5f4822a977d4c1b3a7a2306