gst-plugins-base 1.4.4


2014-11-06  Sebastian Dröge <slomo coaxion net>

          releasing 1.4.4

2014-11-06 09:39:08 +0000  Tim-Philipp Müller <tim centricular com>

        * tests/check/
          tests: dist header file needed for ABI checks on powerpc32
          Fixes 'make check' on debian powerpc32 buildbot:
          libs/libsabi.c:95:26: fatal error: struct_ppc32.h: No such file or directory

2014-10-01 15:04:09 -0700  Aleix Conchillo Flaqué <aleix oblong com>

        * gst-libs/gst/rtsp/gstrtspconnection.c:
          rtspconnection: call watch notify before freeing any watch resources
          This gives control to the notify function allowing it to finish other
          watch related functionality.

2014-10-20 15:31:29 +0200  Sebastian Dröge <sebastian centricular com>

        * gst-libs/gst/app/gstappsink.c:
          appsink: Fix gst_app_sink_pull() docs to transfer full for the return value
          Also we get a GstSample, not a GstBuffer here.

2014-10-13 22:24:31 -0300  Thiago Santos <thiago sousa santos collabora com>

        * gst-libs/gst/audio/gstaudiodecoder.c:
          audiodecoder: should post DECODE errors and not ENCODE
          Fix error code for audio decoder

2014-10-10 12:14:17 +0300  Heinrich Fink <hfink toolsonair com>

        * gst/playback/gstplaysink.c:
          playsink: Use correct property enum value for video-filter property installation

2014-10-07 12:10:42 +0400  Andrei Sarakeev <sarakusha gmail com>

        * gst/playback/gstdecodebin2.c:
          decodebin: Only emit the drain signal for the main decode chain, not any subchains

2014-10-04 23:09:19 +0300  Sebastian Dröge <sebastian centricular com>

        * gst-libs/gst/video/gstvideoencoder.c:
          videoencoder: Stop storing if we received EOS
          This was never reset when going from PAUSED->READY and resulted
          in encoders being not reusable after EOS. They just rejected any
          buffer because they received EOS in their previous life.
          The flag wasn't used anywhere except for rejecting buffers after
          EOS, and this is now handled by GstPad directly.

2014-10-02 00:14:03 +0200  Aurélien Zanelli <aurelien zanelli darkosphere fr>

        * ext/vorbis/gstvorbisdeclib.c:
          vorbisdec: don't reorder streams with channels count greater than eight
          vorbis_reorder_map is defined for eight channels max. If we have more
          than eight channels, it's the application which shall define the order.
          Since we set audio position to none, we just interleave all the channels
          without any particular reordering.

2014-10-01 11:16:30 +0200  Aurélien Zanelli <aurelien zanelli parrot com>

        * gst-libs/gst/video/gstvideoencoder.c:
          videoencoder: release frame in finish_frame when no output state is configured
          Otherwise, frame is leaked.

2014-09-23 14:14:36 -0300  Thiago Santos <thiagoss osg samsung com>

        * gst-libs/gst/audio/gstaudiosink.c:
          audiosink: compensate for segment restart with clock's time_offset
          When playing chained data the audio ringbuffer is released and
          then acquired again. This makes it reset the segbase/segdone
          variables, but the next sample will be scheduled to play in
          the next position (right after the sample from the previous media)
          and, as the segdone is at 0, the audiosink will wait the duration
          of this previous media before it can write and play the new data.
          What happens is this:
          pointer at 0, write to 698-1564, diff 698, segtotal 20, segsize 1764, base 0
          it will have to wait the length of 698 samples before being able to write.
          In a regular sample playback it looks like:
          pointer at 677, write to 696-1052, diff 19, segtotal 20, segsize 1764, base 0
          In this case it will write to the next available position and it
          doesn't need to wait or fill with silence.
          This solution is borrowed from pulsesink that resets the clock to
          start again from 0, which makes it reset the time_offset to the time
          of the last played sample. This is used to correct the place of
          writing in the ringbuffer to the new start (0 again)

======== (2.51M)
  sha256sum: 49cd9e8f23c416b1607b43837a09833fa03e0106929d81ead2ddfde6c0ade44b

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