gst-plugins-good 1.12.1


2017-06-20  Sebastian Dröge <slomo coaxion net>

          releasing 1.12.1

2017-06-20 11:08:32 +0300  Sebastian Dröge <sebastian centricular com>

        * po/bg.po:
        * po/da.po:
        * po/de.po:
        * po/el.po:
        * po/fr.po:
        * po/hr.po:
        * po/hu.po:
        * po/nb.po:
        * po/pl.po:
        * po/ru.po:
        * po/sr.po:
        * po/sv.po:
        * po/tr.po:
        * po/uk.po:
        * po/vi.po:
        * po/zh_CN.po:
          po: Update translations

2017-06-13 17:40:19 +0300  Vivia Nikolaidou <vivia ahiru eu>

        * gst/multifile/gstsplitmuxsink.c:
          splitmux: Drop allocation queries
          They can cause us to deadlock, while we're waiting for a new frame and
          upstream is waiting for the allocation query to be answered before
          sending a frame

2017-06-15 10:40:51 +0300  Sebastian Dröge <sebastian centricular com>

        * gst/rtsp/gstrtspsrc.c:
        * gst/rtsp/gstrtspsrc.h:
          rtspsrc: Use a mutex for protecting against concurrent send/receives
          We currently send data to the RTSP connection from multiple threads:
          whenever a command is to be handled and whenever RTCP is generated. This
          can cause data corruption or worse if both happen at the same time.
          As such, protect gst_rtsp_connection_send() and gst_rtsp_connection_receive()
          calls with a mutex. While this means that we hold a mutex during the IO
          operation, this is not actually a problem as the IO operation can be
          interrupted (gst_rtsp_connection_flush()) at any time and is blocking by
          itself anyway.

2017-06-15 11:50:44 +0300  Sebastian Dröge <sebastian centricular com>

        * gst/isomp4/atoms.c:
          qtmux: Un-merge the last two stsc entries after serializing
          The last entry will most likely get new samples added to it in "robust"
          muxing mode, changing the samples_per_chunk and thus making it wrong to
          keep the last two entries merged. It will run into an assertion later
          when adding a new sample to the chunk.
          Thanks to gdiener cardinalpeak com for the analysis of the bug and
          proposal for a solution.

2017-06-14 00:09:25 +0300  Sebastian Dröge <sebastian centricular com>

        * gst/wavparse/gstwavparse.c:
          wavparse: Actually clip to upstream size instead of size of the data chunk
          There might be other chunks after the data chunk, so clipping the chunk
          size with the data size can lead to a negative number and all following
          calculations go wrong and cause crashes or worse.
          This was introduced in 3ac119bbe2c360e28c087cf3852ea769d611b120.

2017-05-30 22:23:10 +0200  Juan Navarro <juan navarro gmx es>

        * gst/rtpmanager/rtpsession.c:
          rtpsession: print value of unknown RTCP Payload Type
          This adds printing the actual value of any unknown RTCP PT
          to the already existing WARNING log message.

2017-06-02 11:30:15 +0100  Tim-Philipp Müller <tim centricular com>

        * gst/rtp/gstrtph265depay.c:
          rtph265depay: fix caps leak

2017-05-24 11:33:05 +0530  vijay <vijay palaniswamy in bosch com>

        * gst/audioparsers/gstaacparse.c:
          aacparse : Fix, Caps were not set while reusing aacparse
          While reusing aacparse caps were not set.This fix enables aacparse to reuse in same pipeline.

2017-05-16 12:56:15 +0300  Vivia Nikolaidou <vivia ahiru eu>

        * gst/isomp4/gstqtmux.c:
          qtmux: Do not check timecode data for mp4 container
          Timecode trak is only supported for mov right now, not for mp4. That
          code would otherwise create an invalid trak if the muxed video contained
          timecode metadata.

2017-05-10 15:58:41 +0200  Sebastian Dröge <sebastian centricular com>

        * gst/isomp4/gstqtmux.c:
          qtmux: Lateness is in QT timescale, diff in GstClockTime
          Print the right one in debug output to get meaningful numbers.

2017-05-09 11:41:25 +0200  Sebastian Dröge <sebastian centricular com>

        * ext/vpx/gstvpxdec.c:
          vpxdec: Set fb->priv to NULL after freeing just in case

2017-05-08 15:22:00 +0000  Dustin Spicuzza <dustin virtualroadside com>

        * sys/directsound/gstdirectsoundsink.c:
        * sys/directsound/gstdirectsoundsink.h:
          directsoundsink: Use GstClock API instead of Sleep() for waiting
          It's more accurate and allows cancellation.

2017-05-08 15:05:45 +0000  Tim-Philipp Müller <tim centricular com>

        * ext/vpx/gstvp9dec.c:
          vpx: fix build against older libvpx versions
          Such as 1.3.0 as on raspbian.

2017-05-03 23:23:10 +0530  Nirbheek Chauhan <nirbheek centricular com>

        * sys/directsound/gstdirectsoundsink.c:
          directsoundsink: Fix corner case causing large CPU usage
          We were unnecessarily looping/goto-ing repeatedly when we had exactly
          the amount of data as the free space, and also when the free space was
          too small. This, as it turns out, is a very common scenario with
          Directsound on Windows.
          We have to do polling here because the event notification API that
          Directsound exposes cannot be used with live playback since all events
          must be registered in advance with the capture buffer, you cannot
          add/remove them once playback has begun. Directsoundsrc had the same
          See also:

======== (3.33M)
  sha256sum: 121e8e46a7f0e622f09ec9be012607b89d737dd72d48b0f2f0680821ae2cf54b

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