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gst-plugins-good 1.8.1

ChangeLog
=========

2016-04-20  Sebastian Dröge <slomo coaxion net>

        * configure.ac:
          releasing 1.8.1

2016-04-20 15:31:19 +0300  Sebastian Dröge <sebastian centricular com>

        * po/da.po:
          po: Update translations

2016-04-15 00:46:56 -0700  Aleix Conchillo Flaqué <aleix oblong com>

        * gst/rtsp/gstrtspsrc.c:
        * gst/rtsp/gstrtspsrc.h:
          rtspsrc: add srtp rollover counters from mikey crypto sessions
          The server can send multiple crypto sessions, one for each SSRC with its
          own rollover counter. We parse this information and pass it to the SRTP
          decoder via the "request-key" signal.
          https://bugzilla.gnome.org/show_bug.cgi?id=730540

2016-04-15 10:44:02 -0400  Xavier Claessens <xavier claessens collabora com>

        * gst/multifile/gstsplitmuxsink.c:
          spitmuxsink: Avoid creating small file at EOS
          When EOS is reached, the current file get closed and the last
          GOP in the mq was written in a new file.
          https://bugzilla.gnome.org/show_bug.cgi?id=765072

2016-04-16 02:17:26 +1000  Jan Schmidt <jan centricular com>

        * ext/pulse/pulsesink.c:
          Revert "pulsesink: uncork if needed upon commit"
          This reverts commit 0dd46accf6d282ff07065852bd91c85c78af3394.
          With some audiosinks, starting the ringbuffer on the first commit
          causes audio glitches at startup by starting to output segments
          from the ringbuffer before it has been filled / fully prerolled. This
          doesn't usually happen with pulsesink because we map the pulseaudio
          ringbuffer directly, but we should keep things consistent with
          other sinks with regards to startup latency, plus it gives more
          headway to avoid glitching, should the initial 2nd segment take
          more than 10ms to generate.
          https://bugzilla.gnome.org/show_bug.cgi?id=657076

2016-04-15 19:59:15 +0300  Sebastian Dröge <sebastian centricular com>

        * gst/audiofx/gstscaletempo.c:
          scaletempo: S16 uses S32 temporary buffers, float/double their own type
          Make sure to allocate not only a S16 buffer for S16 but a twice as big one to
          hold S32.
          https://bugzilla.gnome.org/show_bug.cgi?id=765116

2016-04-13 18:45:07 +0100  Damian Ziobro <damian xmementoit com>

        * gst/multifile/gstsplitmuxsink.c:
        * gst/multifile/gstsplitmuxsink.h:
          splitmuxsink: Add max_files_number property
          https://bugzilla.gnome.org/show_bug.cgi?id=744612

2016-04-12 10:15:39 +0300  Sebastian Dröge <sebastian centricular com>

        * gst/rtpmanager/rtpjitterbuffer.c:
          rtpjitterbuffer: Fix rtp_jitter_buffer_get_ts_diff() fill level calculation
          The head of the queue is the oldest packet (as in lowest seqnum), the tail is
          the newest packet. To calculate the fill level, we should calculate tail-head
          while considering wraparounds. Not the other way around.
          Other code is already doing this in the correct order.
          https://bugzilla.gnome.org/show_bug.cgi?id=764889

2016-04-11 08:33:17 +0900  Seungha Yang <sh yang lge com>

        * gst/isomp4/qtdemux.c:
          qtdemux: Fix parsing segment duration of empty edit list box
          For empty edit list, segment-duration in edit list box should not be
          used for segment event.
          https://bugzilla.gnome.org/show_bug.cgi?id=764870

2016-04-12 09:41:00 +0000  Paolo Pettinato <ppettina cisco com>

        * 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

2016-04-12 15:01:28 +0300  Sebastian Dröge <sebastian centricular com>

        * gst/deinterlace/gstdeinterlace.c:
          deinterlace: Drain the field history if the caps are changing
          Otherwise we will use fields from the old caps with everything set up for the
          new caps, causing crashes and worse.
          Also don't do anything if the same caps are set twice.

2016-03-31 02:15:04 +1100  Jan Schmidt <jan centricular com>

        * gst/multifile/gstsplitmuxsink.c:
          splitmux: Handle a hang draining out at EOS
          Make sure that all data is drained out when the reference pad
          goes EOS. Fixes a problem where data that arrives on other
          pads after the reference pad finishes can stall forever and
          never pass EOS.
          https://bugzilla.gnome.org/show_bug.cgi?id=763711

2016-03-18 15:45:01 -0400  Xavier Claessens <xavier claessens collabora com>

        * gst/multifile/gstsplitmuxsink.c:
          splitmuxsink: Fix occasional deadlock when ending file with subtitle
          Deadlock occurs when splitting files if one stream received no buffer during
          the first GOP of the next file. That can happen in that scenario for example:
          1) The first GOP of video is collected, it has a duration of 10s.
          max_in_running_time is set to 10s.
          2) Other streams catchup and we receive the first subtitle buffer at ts=0 and
          has a duration of 1min.
          3) We receive the 2nd subtitle buffer with a ts=1min. in_running_time is set to
          1min. That buffer is blocked in handle_mq_input() because
          max_in_running_time is still 10s.
          4) Since all in_running_time are now > 10s, max_out_running_time is now set to
          10s. That first GOP gets recorded into the file. The muxer pop buffers out
          of the mq, when it tries to pop a 2nd subtitle buffer it blocks because the
          GstDataQueue is empty.
          5) A 2nd GOP of video is collected and has a duration of 10s as well.
          max_in_running_time is now 20s. Since subtitle's in_running_time is already
          1min, that GOP is already complete.
          6) But let's say we overran the max file size, we thus set state to
          SPLITMUX_STATE_ENDING_FILE now. As soon as a buffer with ts > 10s (end of
          previous GOP) arrives in handle_mq_output(), EOS event is sent downstream
          instead. But since the subtitle queue is empty, that's never going to
          happen. Pipeline is now deadlocked.
          To fix this situation we have to:
          - Send a dummy event through the queue to wakeup output thread.
          - Update out_running_time to at least max_out_running_time so it sends EOS.
          - Respect time order, so we set out_running_tim=max_in_running_time because
          that's bigger than previous buffer and smaller than next.
          https://bugzilla.gnome.org/show_bug.cgi?id=763711

2016-01-31 11:08:38 +1100  Sebastian Dröge <sebastian centricular com>

        * 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

2016-03-16 20:17:55 +0100  Havard Graff <havard graff gmail com>

        * gst/flv/gstflvdemux.c:
        * tests/check/elements/flvdemux.c:
          flvdemux: don't emit pad-added until caps are ready
          In other words, gst_pad_get_current_caps should never return NULL
          in a pad-added callback from the demuxer.
          Added tests for the two special cases with AAC and H.264 where this
          would happen every time.
          https://bugzilla.gnome.org/show_bug.cgi?id=763780

2016-02-29 23:40:03 -0300  Thiago Santos <thiagoss osg samsung com>

        * 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

2016-03-24 15:14:23 +0900  Jimmy Ohn <yongjin ohn lge com>

        * gst/isomp4/qtdemux.c:
          qtdemux: Fix qtdemux memory leak in src_convert function
          If we don't find the index of the sample correctly in src_convert function,
          we have to unref about the qtdemux before returning value.
          So, I have modify it about instead pass qtdemux as a parameter into
          src_convert function.
          https://bugzilla.gnome.org/show_bug.cgi?id=763973

2015-11-04 14:51:19 +0900  Jihae Yi <jihae yi samsung com>

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

2016-03-24 19:23:12 -0400  Nicolas Dufresne <nicolas dufresne collabora com>

        * ext/vpx/gstvpxdec.c:
          vpxdec: Use threads on multi-core systems
          This is a redo of commit b848c1b6ffd1e508228820a013f94fb445e4777f. The
          code was lost when the elements where ported to use a baseclass.
          https://bugzilla.gnome.org/show_bug.cgi?id=764169



Download
========
https://download.gnome.org/sources/gst-plugins-good/1.8/gst-plugins-good-1.8.1.tar.xz (3.06M)
  sha256sum: 2103e17921d67894e82eafdd64fb9b06518599952fd93e625bfbc83ffead0972
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