gst-plugins-good 1.5.91

From: Sebastian Dröge <install-module master gnome org>
To: FTP Releases <ftp-release-list gnome org>
Subject: gst-plugins-good 1.5.91
Date: Mon, 21 Sep 2015 09:46:18 +0000 (UTC)
ChangeLog
=========

2015-09-18  Sebastian Dröge <slomo coaxion net>

        * configure.ac:
          releasing 1.5.91

2015-09-18 11:50:31 +0200  Sebastian Dröge <sebastian centricular com>

        * po/zh_CN.po:
          po: Update translations

2015-09-17 10:50:01 +0900  Eunhae Choi <eunhae1 choi samsung com>

        * gst/avi/gstavidemux.c:
          avidemux: Fix taglist leak
          gst_tag_list_insert() does not take ownership of the inserted taglist.
          https://bugzilla.gnome.org/show_bug.cgi?id=755138

2015-09-16 07:05:36 +1000  Jan Schmidt <jan centricular com>

        * gst/audioparsers/gstaacparse.c:
          aacparse: Skip LOAS AAC until a valid config is seen.
          It's normal when dropping into the middle of a stream to
          not always have the config available immediately, so skip LOAS
          until a valid config is seen without either setting invalid
          caps or erroring out.
          https://bugzilla.gnome.org/show_bug.cgi?id=751386

2015-09-13 15:41:38 +0200  Mark Nauwelaerts <mnauw users sourceforge net>

        * gst/rtpmanager/gstrtpjitterbuffer.c:
          rtpjitterbuffer: reset just a bit more upon flush_stop

2015-09-13 15:40:09 +0200  Mark Nauwelaerts <mnauw users sourceforge net>

        * gst/rtpmanager/gstrtpjitterbuffer.c:
          rtpjitterbuffer: remove dead struct member

2015-09-11 17:09:28 +0900  Vineeth TM <vineeth tm samsung com>

        * gst/udp/gstmultiudpsink.c:
          multiudpsink: fix GError memory leak when hostname resolution fails
          https://bugzilla.gnome.org/show_bug.cgi?id=754869

2015-09-10 15:26:54 -0300  Thiago Santos <thiagoss osg samsung com>

        * gst/matroska/ebml-write.c:
          matroskamux: drop HEADER flag from output buffers
          Drop HEADER flag from output buffers if they are not indeed
          headers.
          Fixes resending of headers in tcp connection handling
          https://bugzilla.gnome.org/show_bug.cgi?id=754768

2015-09-10 16:00:50 +0100  Tim-Philipp Müller <tim centricular com>

        * gst/matroska/ebml-write.c:
          matroskamux: fix matroskamux ! matroskademux
          Don't carry over DISCONT flags from the input buffers to the
          output buffer, or the demuxer might reset its state when it
          receives the first data buffer just after parsing the simple
          block header, and then expect sane data to follow.
          Fixes matroskamux ! demux erroring out.
          https://bugzilla.gnome.org/show_bug.cgi?id=754768
          https://bugzilla.gnome.org/show_bug.cgi?id=657805

2015-09-09 12:51:40 -0700  Martin Kelly <martin surround io>

        * gst/rtsp/README:
          rtsp: fix small README typo
          https://bugzilla.gnome.org/show_bug.cgi?id=754807

2015-09-04 19:45:37 +0100  Tim-Philipp Müller <tim centricular com>

        * gst/audioparsers/gstwavpackparse.c:
          wavpackparse: set both pts and dts so baseparse doesn't make up wrong dts after seeks
          https://bugzilla.gnome.org/show_bug.cgi?id=752106

2015-09-04 19:34:41 +0100  Tim-Philipp Müller <tim centricular com>

        * gst/audioparsers/gstflacparse.c:
          flacparse: set both pts and dts so baseparse doesn't make up wrong dts after a seek
          flac contains the sample offset in the frame header, so after a seek
          without index flacparse will know the exact position we landed on and
          timestamp buffers accordingly. It only set the pts though, which means
          the baseparse-set dts which was set to the seek position prevails, and
          since the seek was based on an estimate, there's likely a discrepancy
          between where we wanted to land and where we did land, so from here on
          that dts/pts difference will be maintained, with dts possibly multiple
          seconds ahead of pts, which is just wrong. The easiest way to fix this
          is to just set both pts and dts based on the sample offset, but perhaps
          parsed audio should just not have dts set at all.
          https://bugzilla.gnome.org/show_bug.cgi?id=752106

2015-09-06 16:33:02 +0100  Tim-Philipp Müller <tim centricular com>

        * docs/plugins/gst-plugins-good-plugins.args:
        * docs/plugins/gst-plugins-good-plugins.signals:
          docs: remove properties and signals that no longer exist
          https://bugzilla.gnome.org/show_bug.cgi?id=726443

2013-10-11 15:13:00 +0000  George Chriss <gschriss gmail com>

        * gst/flv/gstflvmux.c:
          flvmux: Make the element count in arrays not include end
          One-line removal of tags_written++
          This should fix rtmp output to crtmpserver, and hopefully
          noone is expecting that the element count includes the end
          element, as different bits of documentation say different
          things about whether it should or not.
          https://bugzilla.gnome.org/show_bug.cgi?id=661624

2015-07-30 00:59:15 +1000  Jan Schmidt <jan centricular com>

        * gst/flv/gstflvmux.c:
        * gst/flv/gstflvmux.h:
          flvmux: Store incoming bitrate tags and send in the metadata
          Apparently the Microsoft Azure RTMP server requires that the
          videodatarate and audiodatarate metadata be provided, so
          set those, even if it's to 0. Use the actual input bitrate
          tags if available.

2015-09-04 00:06:29 +1000  Jan Schmidt <jan centricular com>

        * gst/rtsp/gstrtspsrc.c:
          rtspsrc: Don't parse key data more than needed.
          When an auxilliary streams are present in the SDP media,
          there's no need to re-parse the SDP attributes multiple
          times.

2015-09-03 20:56:55 +1000  Jan Schmidt <jan centricular com>

        * gst/rtsp/gstrtspsrc.c:
          rtspsrc: Fix SRTP + RTX, auth access, a leak, and an invalid memory access.
          In parse_keymgmt(), don't mutate the input string that's been passed
          as const, especially since we might need the original value again if
          the same key info applies to multiple streams (RTX, for example).
          When a resource is 404, and we have auth info - retry with the auth
          info the same as if we had receive unauthorised, in case the resource
          isn't even visible until credentials are supplied.
          Fix a memory leak handling Mikey data.
          When generating a random keystring, don't overrun the 30 byte
          buffer by generating 32 bytes into it.

2015-09-04 15:18:05 +0300  Sebastian Dröge <sebastian centricular com>

        * gst/udp/gstudpsrc.c:
          udpsrc: Fix build with GLib < 2.44
          G_IO_ERROR_CONNECTION_CLOSED was added in 2.44.

2015-09-04 12:01:52 +0300  Sebastian Dröge <sebastian centricular com>

        * gst/udp/gstudpsrc.c:
          udpsrc: Ignore G_IO_ERROR_CONNECTION_CLOSED when receiving data
          This happens on Windows if we use the same socket for sending packets,
          and the remote sends ICMP port/host unreachable messages.
          https://bugzilla.gnome.org/show_bug.cgi?id=754534

2015-09-02 21:12:41 +0300  Sebastian Dröge <sebastian centricular com>

        * gst/rtp/gstrtptheoradepay.c:
        * gst/rtp/gstrtpvorbisdepay.c:
          rtpvorbis/theoradepay: Fix handling of fragmented packets
          This was broken in b1089fb520 by not considering the full packet length of a
          fragmented packet but only the length of the first one.
          https://bugzilla.gnome.org/show_bug.cgi?id=754417

2015-09-01 15:39:22 -0400  Olivier Crête <olivier crete collabora com>

        * gst/dtmf/gstdtmfsrc.c:
        * gst/dtmf/gstrtpdtmfsrc.c:
          dtmfsrc: Reply to latency query

2015-08-31 16:42:30 -0400  Olivier Crête <olivier crete collabora com>

        * tests/check/elements/rtpsession.c:
          tests: Fix rtpsession test failure
          The time of the first RTCP packet is semi-random, so
          sometimes it was produced before enough packets from
          the second SSRC were received. First drop queued RTCP
          packets, then advance the clock enough to ensure
          that at least one new RTCP packet is produced.
          https://bugzilla.gnome.org/show_bug.cgi?id=750731

2015-08-31 13:56:04 +0200  Stefan Sauer <ensonic users sf net>

        * tests/check/elements/level.c:
          level: improve the test for multi-channel mode
          Change the test to verify the read-index for multiple messages per buffer.
          See https://bugzilla.gnome.org/show_bug.cgi?id=754144

2015-08-31 12:46:52 +0200  Jan Alexander Steffens (heftig) <jan steffens gmail com>

        * gst/matroska/matroska-demux.c:
          matroskademux: Align raw video frames to 32 bytes
          Outputting unaligned video frames causes videoscale et al to
          crash when attempting SIMD-accelerated conversion.
          https://bugzilla.gnome.org/show_bug.cgi?id=736965

2015-08-26 23:16:46 +0200  Stefan Sauer <ensonic users sf net>

        * gst/level/gstlevel.c:
          level: fix level calculations for mutliple channels
          This was broken with 7b90bf32150897a141a29a12ecab555d8c5b7fab.

2015-08-27 10:28:55 +0530  Ravi Kiran K N <ravi kiran samsung com>

        * gst/smpte/gstsmpte.c:
          smpte: Fix memory leak
          In gst_smpte_collected(), check upfront if input formats are same
          or not. This avoids allocation of in1 and in2 buffers and
          subsequent memory leak when input formats do not match.
          https://bugzilla.gnome.org/show_bug.cgi?id=754153

2015-08-21 11:52:19 +0100  Tim-Philipp Müller <tim centricular com>

        * tests/check/elements/souphttpsrc.c:
          tests: souphttpsrc: don't try to connect to dead radio server

2015-08-21 16:29:16 +0900  Vineeth TM <vineeth tm samsung com>

        * gst/rtsp/gstrtspsrc.c:
          rtspsrc: Trivial fix to check correct condition
          When checking for describe method, because of missing parentheses, wrong
          condition is being checked, which will result in wrong behavior.
          https://bugzilla.gnome.org/show_bug.cgi?id=753912

2015-08-21 13:19:02 +0900  Vineeth TM <vineeth tm samsung com>

        * gst/matroska/matroska-read-common.c:
          matroska: read: fix tag list memory leak
          gst_toc_entry_merge_tags makes a new ref of the taglist, so it should
          be unref'ed as soon as the tags are merged to the tocentry
          https://bugzilla.gnome.org/show_bug.cgi?id=753904

2015-08-21 12:20:59 +0900  Vineeth TM <vineeth tm samsung com>

        * ext/wavpack/gstwavpackdec.c:
          wavpackdec: fix taglist memory leak
          When passing the taglist to gst_audio_decoder_merge_tags, the reference is increased
          by audiodecoder and the caller should free the taglist being passed.
          https://bugzilla.gnome.org/show_bug.cgi?id=753903

2015-08-20 14:45:33 +0200  Jean-Michel Hautbois <jean-michel hautbois veo-labs com>

        * sys/v4l2/gstv4l2transform.c:
          v4l2transform: fix pad closing
          Signed-off-by: Jean-Michel Hautbois <jean-michel hautbois veo-labs com>
          https://bugzilla.gnome.org/show_bug.cgi?id=753875



Download
========
https://download.gnome.org/sources/gst-plugins-good/1.5/gst-plugins-good-1.5.91.tar.xz (3.03M)
  sha256sum: bdd892935e9b6be5d87ec62de147e941c5ba5ec1510ca80e60faa9b961f5140a
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