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gst-plugins-base 1.10.3

ChangeLog
=========

2017-01-30  Sebastian Dröge <slomo coaxion net>

        * configure.ac:
          releasing 1.10.3

2017-01-30 13:30:51 +0200  Sebastian Dröge <sebastian centricular com>

        * po/fr.po:
        * po/nb.po:
        * po/sr.po:
          po: Update translations

2017-01-30 12:35:04 +0200  Sebastian Dröge <sebastian centricular com>

        * gst-libs/gst/audio/audio-resampler-x86-sse41.c:
          audio-resampler: Fix integer overflow in clamping code
          https://bugzilla.gnome.org/show_bug.cgi?id=777921

2017-01-20 12:41:16 +0200  Sebastian Dröge <sebastian centricular com>

        * gst-libs/gst/riff/riff-media.c:
          riff-media: Don't divide block align by zero channels
          https://bugzilla.gnome.org/show_bug.cgi?id=777525

2017-01-20 08:02:38 +0200  Sebastian Dröge <sebastian centricular com>

        * gst/subparse/samiparse.c:
          samiparse: Check that the string has a non-zero length before overwriting the last byte with '\0'
          https://bugzilla.gnome.org/show_bug.cgi?id=777502

2017-01-15 18:42:34 +0100  Sebastian Dröge <sebastian centricular com>

        * gst-libs/gst/riff/riff-media.c:
          riff-media: Don't recurse in for nested WAVEFORMATEX
          There was already a check for that, but it failed because
          subformat_guid[0] is a guint32 and that is then casted implicitely to a
          guint16 when recursing... just that we checked the uncasted value.
          This caused an infinite recursion and thus stack overflow.
          https://bugzilla.gnome.org/show_bug.cgi?id=777265

2017-01-15 18:31:56 +0100  Sebastian Dröge <sebastian centricular com>

        * gst-libs/gst/riff/riff-media.c:
          riff-media: Check for valid channels/rate before using the values
          Otherwise we might divide by zero or otherwise create invalid caps.
          https://bugzilla.gnome.org/show_bug.cgi?id=777262

2017-01-11 18:24:38 +0200  Sebastian Dröge <sebastian centricular com>

        * gst-libs/gst/video/video-converter.c:
          video-converter: Fix crashes in fast-paths when converting interlaced formats with different 
vertical subsampling
          E.g. the following pipelines fail because chroma values after the last
          line are read (note: 486 % 4 == 2):
          gst-launch-1.0 videotestsrc ! 
"video/x-raw,interlace-mode=interleaved,width=720,height=486,format=UYVY" ! videoconvert ! 
"video/x-raw,format=I420" ! fakesink
          gst-launch-1.0 videotestsrc ! 
"video/x-raw,interlace-mode=interleaved,width=720,height=486,format=I420" ! videoconvert ! 
"video/x-raw,format=UYVY" ! fakesink
          gst-launch-1.0 videotestsrc ! 
"video/x-raw,interlace-mode=interleaved,width=720,height=486,format=I420" ! videoconvert ! 
"video/x-raw,format=AYUV" ! fakesink

2017-01-10 08:57:51 -0300  Thibault Saunier <thibault saunier osg samsung com>

        * gst-libs/gst/pbutils/encoding-profile.c:
          pbutils: Fix annotation in gst_encoding_profile_set_preset

2017-01-09 21:25:26 +1100  Jan Schmidt <jan centricular com>

        * gst-libs/gst/video/video.c:
          gst_video_guess_framerate: Don't throw away all precision
          When operating on framerates near 10000fps, at least keep 1
          digit of precision for calculations

2017-01-04 11:21:51 -0300  Thibault Saunier <thibault saunier osg samsung com>

        * gst/encoding/gstencodebin.c:
          encodebin: Fix stream_group_free when creating it went bad
          Avoiding trying to use NULL pointers

2016-12-30 17:55:18 +0100  Mark Nauwelaerts <mnauw users sourceforge net>

        * gst/playback/gstplaysink.c:
          playsink: do not link to audio or video filter using padname
          ... as a sinkpad need not be called "sink", and it is not the case
          for e.g. timeoverlay (and friends).
          Fixes https://bugzilla.gnome.org/show_bug.cgi?id=776623

2017-01-02 12:54:32 +0000  Tim-Philipp Müller <tim centricular com>

        * gst/encoding/gstencodebin.c:
          encodebin: fix queue property types when setting

2015-11-25 11:30:42 +0000  Stuart Weaver <stuart weaver datapath co uk>

        * gst-libs/gst/rtsp/gstrtspurl.c:
          rtsp-url: unescape special chars in user/pass part of URL
          This way special characters such as '@' can be used in
          usernames or passwords, e.g.
          rtsp://view:%40dm%4An@<IP-ADDR>/media/camera1
          will now parse username and password into:
          User: view
          Pass: @dm:n
          https://bugzilla.gnome.org/show_bug.cgi?id=758389

2016-09-02 15:23:18 +0200  Carlos Rafael Giani <dv pseudoterminal org>

        * gst/audiotestsrc/gstaudiotestsrc.c:
          audiotestsrc: Fix incorrect start of tick waveform
          Make sure ticks start with an accumulator value of 0 by incrementing it
          after filling in samples instead of before and by resetting the accumulator
          every time a tick begins. This prevents it from being discontinuous at the
          beginning of the tick.
          https://bugzilla.gnome.org/show_bug.cgi?id=774050

2016-12-22 18:47:19 +0100  Nicolas Dechesne <nicolas dechesne linaro org>

        * tools/gst-play.c:
          tools: gst-play: set GST_GL_XINITHREADS
          This ensure that XInitThreads is called and so gl contexts are properly
          initialized.
          https://bugzilla.gnome.org/show_bug.cgi?id=776403

2016-12-21 00:11:06 +1100  Jan Schmidt <jan centricular com>

        * gst/playback/gstparsebin.c:
          parsebin: Ignore failure to send sticky events
          When plugging and then exposing a parser, don't fail
          if it fails to send sticky events. The most likely
          reason is that things were flushed due to the app
          immediately doing a seek, but we can't detect flushing
          separately to other error conditions without a
          gst_pad_send_event_full() core function that returns
          a GstFlowReturn.

2016-12-15 16:29:02 +0200  Sebastian Dröge <sebastian centricular com>

        * gst/playback/gstdecodebin2.c:
          decodebin: For adaptive streaming, ensure to put the buffering multiqueue after a parser or demuxer
          There are cases when there is no demuxer involved that could do the
          buffering, e.g. HLS with raw MP3 or AAC. In this case we want to place
          the buffering multiqueue after the parser.
          Before this change, we've considered the first element after the
          adaptive streaming demuxer as a parser. This is not always true, e.g.
          id3demux. Instead we now wait until we actually have a parser (or
          decoder).
          Fixes playback on such HLS streams.

2016-12-09 17:36:47 +0200  Sebastian Dröge <sebastian centricular com>

        * gst-libs/gst/tag/gstxmptag.c:
          xmptag: Don't leak the namespace string if there are multiple
          https://bugzilla.gnome.org/show_bug.cgi?id=775887

2016-12-09 17:57:52 +1100  Jan Schmidt <jan centricular com>

        * gst-libs/gst/tag/id3v2.c:
          id3v2: Add missing overrun check for frame sizes
          When frames claim to have a footer, ensure they
          are large enough to contain one to avoid an invalid
          read overrun.
          Spotted by Joshua Yabut

2016-12-06 16:29:23 +0200  Sebastian Dröge <sebastian centricular com>

        * gst-libs/gst/tag/gsttagdemux.c:
          tagdemux: Fix crash when shutting down element during getrange()
          Ensure that nothing is in any of the streaming thread functions
          anymore when going from PAUSED to READY. While the parent's state change
          function has deactivated all pads, there is nothing preventing
          downstream from activating our srcpad again and calling the getrange()
          function. Although we're in READY!
          https://bugzilla.gnome.org/show_bug.cgi?id=775687

2016-11-04 16:41:05 +0000  Vincent Penquerc'h <vincent penquerch collabora co uk>

        * ext/opus/gstopusdec.c:
          opusdec: fix 120 ms buffers being wrongly emitted
          Using the max 120 ms buffer size to ensure we have enough space
          for decoded data meant that Opus could actually return 120 ms'
          worth of data.
          https://bugzilla.gnome.org/show_bug.cgi?id=771723

2016-09-26 10:50:52 +0100  Vincent Penquerc'h <vincent penquerch collabora co uk>

        * ext/opus/gstopusdec.c:
          opusdec: fix "buffer too small" error
          Always supply a buffer with max size to the decoder, as we
          can't really decide how many samples will be in the lost packet
          based on the timestamps we get.
          https://bugzilla.gnome.org/show_bug.cgi?id=771723

2016-10-06 11:44:11 +0100  Vincent Penquerc'h <vincent penquerch collabora co uk>

        * ext/opus/gstopusdec.c:
          opusdec: interpret zero duration as unknown
          This fixes missing audio when we get buffers with zero
          duration, denoting unknown duration. When several such
          buffers are received in a row, they're all at the same
          timestamp, with zero duration.
          https://bugzilla.gnome.org/show_bug.cgi?id=771723

2016-11-29 16:26:22 +0100  Jan Alexander Steffens (heftig) <jan steffens gmail com>

        * tests/check/elements/multifdsink.c:
          multifdsink: Add a test involving a slow client
          https://bugzilla.gnome.org/show_bug.cgi?id=774908

2016-11-23 14:35:04 +0100  Jan Alexander Steffens (heftig) <jan steffens gmail com>

        * gst/tcp/gstmultihandlesink.c:
          multihandlesink: Update bufpos in a separate pass
          If a client gets dropped and the iteration gets restarted, bufpos is
          incremented again for all clients that preceded the dropped one, causing
          havoc.
          Adjust the bufpos for all clients first before trying to drop any.
          https://bugzilla.gnome.org/show_bug.cgi?id=774908

2016-11-29 15:30:43 +0100  Jan Alexander Steffens (heftig) <jan steffens gmail com>

        * gst/tcp/gstmultihandlesink.c:
          multihandlesink: Fix buffers-queued being off by one
          max_buffer_usage is the index of the oldest buffer in the queue,
          starting at zero, not the number of buffers queued.
          find_limits returns the index of the oldest buffer that satisfies the
          limits in its min_idx parameter, not the number of buffers needed. Fix
          this use too in order to keep passing the tests that read
          buffers-queued.
          https://bugzilla.gnome.org/show_bug.cgi?id=775351

2016-12-01 15:12:59 +0200  Sebastian Dröge <sebastian centricular com>

        * ext/ogg/gstoggdemux.c:
          oggdemux: Don't end up ignoring caps just because there are no headers for this stream
          https://bugzilla.gnome.org/show_bug.cgi?id=775459

2016-12-01 19:57:47 +0200  Sebastian Dröge <sebastian centricular com>

        * gst/subparse/gstssaparse.c:
          ssaparse: Free initialization section before storing the next one
          If getting multiple caps events.
          https://bugzilla.gnome.org/show_bug.cgi?id=775480



Download
========
https://download.gnome.org/sources/gst-plugins-base/1.10/gst-plugins-base-1.10.3.tar.xz (2.92M)
  sha256sum: e6299617d705a0cbfb535107c1d3a8fc0f0967f14193a8c5c7583f46a88b1710
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