ChangeLog
=========

2018-03-28 14:19:50 +0100  Tim-Philipp Müller <t...@centricular.com>

        * ChangeLog:
        * NEWS:
        * RELEASE:
        * configure.ac:
        * gst-plugins-base.doap:
        * meson.build:
          Release 1.12.5

2018-03-28 14:19:50 +0100  Tim-Philipp Müller <t...@centricular.com>

        * docs/plugins/inspect/plugin-adder.xml:
        * docs/plugins/inspect/plugin-alsa.xml:
        * docs/plugins/inspect/plugin-app.xml:
        * docs/plugins/inspect/plugin-audioconvert.xml:
        * docs/plugins/inspect/plugin-audiorate.xml:
        * docs/plugins/inspect/plugin-audioresample.xml:
        * docs/plugins/inspect/plugin-audiotestsrc.xml:
        * docs/plugins/inspect/plugin-cdparanoia.xml:
        * docs/plugins/inspect/plugin-encoding.xml:
        * docs/plugins/inspect/plugin-gio.xml:
        * docs/plugins/inspect/plugin-libvisual.xml:
        * docs/plugins/inspect/plugin-ogg.xml:
        * docs/plugins/inspect/plugin-opus.xml:
        * docs/plugins/inspect/plugin-pango.xml:
        * docs/plugins/inspect/plugin-pbtypes.xml:
        * docs/plugins/inspect/plugin-playback.xml:
        * docs/plugins/inspect/plugin-rawparse.xml:
        * docs/plugins/inspect/plugin-subparse.xml:
        * docs/plugins/inspect/plugin-tcp.xml:
        * docs/plugins/inspect/plugin-theora.xml:
        * docs/plugins/inspect/plugin-typefindfunctions.xml:
        * docs/plugins/inspect/plugin-videoconvert.xml:
        * docs/plugins/inspect/plugin-videorate.xml:
        * docs/plugins/inspect/plugin-videoscale.xml:
        * docs/plugins/inspect/plugin-videotestsrc.xml:
        * docs/plugins/inspect/plugin-volume.xml:
        * docs/plugins/inspect/plugin-vorbis.xml:
        * docs/plugins/inspect/plugin-ximagesink.xml:
        * docs/plugins/inspect/plugin-xvimagesink.xml:
          Update docs

2018-03-17 06:33:38 +0100  Edward Hervey <edw...@centricular.com>

        * ext/ogg/gstoggstream.c:
          oggstream: protect against out-of-bounds read
          We need at least 17 bytes of data for a valid flac header
          oss-fuzz #6974

2018-02-28 23:12:39 -0500  Nicolas Dufresne <nicolas.dufre...@collabora.com>

        * gst-libs/gst/video/gstvideodecoder.c:
          videodecoder: Reset QoS time after pushing segment
          This fixes playbin gapless playback. An ancient QoS time was used and
          would lead to all frames being dropped.
          https://bugzilla.gnome.org/show_bug.cgi?id=668995

2018-02-12 16:26:01 +0100  Edward Hervey <edw...@centricular.com>

        * gst-libs/gst/tag/id3v2.c:
          id3v2: re-fix handling of v2.4 extended headers
          The various id3v2 specs handle the extended header sizes differently
          (because hey, it wouldn't be fun otherwise).
          http://id3.org/id3v2.3.0 states:
          "Where the 'Extended header size', currently 6 or 10 bytes, excludes
          itself."
          http://id3.org/id3v2.4.0-structure states:
          Extended header size   4 * %0xxxxxxx
          Number of flag bytes       $01
          Extended Flags             $xx
          Where the 'Extended header size' is the size of the whole extended
          header, stored as a 32 bit synchsafe integer. An extended header can
          thus never have a size of fewer than six bytes.
          So in id3v2.4.0 it's the *whole* extended header size (a-la ISOBMFF
          atom), whereas in id3v2.3.0 it's the extended header size *excluding*
          those 4 initial bytes.
          And for other versions, god knows..
          Fixes regression introduced in commit da607005.
          https://bugzilla.gnome.org/show_bug.cgi?id=792983

2018-02-15 01:14:52 +0100  Mathieu Duponchelle <math...@centricular.com>

        * gst-libs/gst/audio/gstaudiopack.orc:
          gstaudiopack.orc: pack_u32be_swap: actually swap
          Fixes:
          gst-launch-1.0 audiotestsrc ! audio/x-raw, format=U32BE ! \
          audioconvert ! autoaudiosink

2018-02-14 14:37:52 -0500  Nicolas Dufresne <nicolas.dufre...@collabora.com>

        * docs/libs/gst-plugins-base-libs-docs.sgml:
          doc: Add per version newly added API indexes

2018-02-14 14:11:47 -0500  Nicolas Dufresne <nicolas.dufre...@collabora.com>

        * gst-libs/gst/allocators/gstfdmemory.h:
        * gst-libs/gst/video/video-color.h:
          doc: Remove extra . after Since marker

2018-02-14 14:16:14 -0500  Nicolas Dufresne <nicolas.dufre...@collabora.com>

        * gst-libs/gst/allocators/gstdmabuf.h:
          doc: Fix since marker in dmabuf to match a stable release

2018-02-14 14:39:32 -0500  Nicolas Dufresne <nicolas.dufre...@collabora.com>

        * gst/playback/gsturidecodebin.c:
          doc: Remove obsolete Since 0.10.X marks

2018-02-13 08:36:30 +0100  Edward Hervey <edw...@centricular.com>

        * ext/vorbis/gstvorbisdec.c:
        * ext/vorbis/gstvorbisdec.h:
          vorbisdec: Improve "new headers while initialized" handling
          If new headers arrive after we are initialized, we need to make
          sure that they are indeed valid.
          A vorbis bitstream always begins with three header packets and must
          be in order.
          Also some streams have unframed (invalid?) headers that might
          confuse and disrupt the decoding process.
          Therefore if ever we see new headers, we accumulate them and once
          we get a non-header packet we check them to make sure that:
          * We have at least 3 headers
          * They are the expected ones (identification, comments and setup)
          * They are in order
          * Any other "header" is ignored
          If those conditions are met, we reset and reconfigure the decoder
          https://bugzilla.gnome.org/show_bug.cgi?id=784530

2018-01-25 18:39:11 +0000  Tim-Philipp Müller <t...@centricular.com>

        * gst/subparse/gstsubparse.c:
          subparse: fix pushing out of last chunk if last line has no newline
          With playbin the last subtitle chunk would not get displayed
          if the last chunk was missing a newline at the end. This is
          because streamsynchronizer will hold back the EOS event until
          the audio and video streams are finished too, so subparse
          would never forcefully push out the last chunk until the very
          end when it is too late.
          We get a STREAM_GROUP_DONE event from streamsynchronizer however,
          so handle that like EOS and force out any remaining text then.
          https://bugzilla.gnome.org/show_bug.cgi?id=771853

2018-01-17 14:35:11 +0100  Edward Hervey <edw...@centricular.com>

        * ext/theora/gsttheoradec.c:
          theoradec: Check for valid width/height
          If width or height are zero ... there's no video :)

2017-10-31 15:04:47 +0530  Ashish Kumar <kr.ash...@samsung.com>

        * gst/playback/gstplaybackutils.c:
          playback-utils: Fix caps leak on failure
          https://bugzilla.gnome.org/show_bug.cgi?id=789358

2018-01-03 15:31:04 +0100  Edward Hervey <edw...@centricular.com>

        * gst/typefind/gsttypefindfunctions.c:
          typefind: Fix mp3 typefinding with multiple different headers
          (yes, this has never worked since it was introduced, don't worry)
          If we want to actually detect layer/channels/samplerate changes,
          it would be better to:
          * not reset the various prev_* variables at every iteration.
          * and actually store the values when they change
          CID #206079
          CID #206080
          CID #206081

2017-12-08 10:33:10 +0100  Edward Hervey <edw...@centricular.com>

        * ext/ogg/gstoggstream.c:
          oggdemux: Check encoder name is valid
          Encoder names should be valid utf-8, if not just ignore them

2017-12-08 08:00:07 +0100  Edward Hervey <edw...@centricular.com>

        * gst/typefind/gsttypefindfunctions.c:
          typefind: Avoid overflow calculation (image/quicktime)
          The qt typefinder uses guint64 values for offset and size calculation
          but the typefinder system only supports gint64 values.
          Make sure we don't end up using potentially overflowing values.

2017-12-08 08:00:07 +0100  Edward Hervey <edw...@centricular.com>

        * gst/typefind/gsttypefindfunctions.c:
          typefind: Avoid overflow calculation
          The qt typefinder uses guint64 values for offset and size calculation
          but the typefinder system only supports gint64 values.
          Make sure we don't end up using potentially overflowing values.

2017-12-15 10:50:44 +0900  Dongil Park <dongil.p...@lge.com>

        * gst/playback/gstplaybin3.c:
          playbin3: Fix accessing invalid index in GstStream when received 
select-stream event
          If select-stream event was send to playbin3 as missing any GstStream 
of ES type
          (V or A or TEX) of collection then, playbin will access to invalid 
address of
          GstStream due to invalid index limit. This caused SIGSEGV.
          https://bugzilla.gnome.org/show_bug.cgi?id=791638



Download
========
https://download.gnome.org/sources/gst-plugins-base/1.12/gst-plugins-base-1.12.5.tar.xz
 (2.95M)
  sha256sum: 8fd9f25b65f3286f43530868b501a4e7cdc3f1568be78c75da716cd2559b712e

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