gst-plugins-good 1.1.3



ChangeLog
=========

2013-07-29  Sebastian Dröge <sebastian droege collabora co uk>

        * configure.ac:
          releasing 1.1.3

2013-07-29 12:12:41 +0200  Sebastian Dröge <slomo circular-chaos org>

        * gst/avi/gstavidemux.c:
        * gst/flv/gstflvdemux.c:
        * gst/isomp4/qtdemux.c:
        * gst/matroska/matroska-demux.c:
          gst: Don't swap start/stop for negative rates in the SEGMENT query

2013-07-29 11:18:40 +0200  Matej Knopp <matej knopp gmail com>

        * gst/isomp4/qtdemux.c:
          qtdemux: Check for data size when parsing h264 codec data from strf atom

2013-07-29 10:53:54 +0200  Sebastian Dröge <slomo circular-chaos org>

        * gst/matroska/matroska-demux.c:
          matroskademux: Implement SEGMENT query

2013-07-29 10:53:47 +0200  Sebastian Dröge <slomo circular-chaos org>

        * gst/flv/gstflvdemux.c:
          flvdemux: Implement SEGMENT query

2013-07-29 10:50:59 +0200  Sebastian Dröge <slomo circular-chaos org>

        * gst/avi/gstavidemux.c:
          avidemux: Implement SEGMENT query

2013-07-27 18:10:22 +0200  Matej Knopp <matej knopp gmail com>

        * gst/isomp4/qtdemux.c:
        * gst/isomp4/qtdemux_fourcc.h:
          qtdemux: Support H264 fourcc
          https://bugzilla.gnome.org/show_bug.cgi?id=704996

2013-07-28 18:09:33 +0200  Sebastian Dröge <slomo circular-chaos org>

        * ext/flac/gstflacenc.c:
          flacenc: Fix handling of image tags
          The caps should be used to get the mimetype and there is
          only an info structure for the GstSample if the image-type
          is not NONE.

2013-07-28 18:04:32 +0200  Sebastian Dröge <slomo circular-chaos org>

        * ext/flac/gstflacenc.c:
          flacenc: Don't crash if there is no image tag information
          https://bugzilla.gnome.org/show_bug.cgi?id=705018

2013-07-28 17:38:56 +0200  Sebastian Dröge <slomo circular-chaos org>

        * gst/avi/gstavidemux.c:
          avidemux: Fix duration reporting in push mode
          https://bugzilla.gnome.org/show_bug.cgi?id=700933

2013-07-28 17:32:27 +0200  Sebastian Dröge <slomo circular-chaos org>

        * gst/avi/gstavidemux.c:
          avidemux: Don't forget unmapping and unreffing buffer

2013-07-26 21:06:17 +0200  Matej Knopp <matej knopp gmail com>

        * gst/avi/gstavidemux.c:
          avidemux: unmap buffer
          https://bugzilla.gnome.org/show_bug.cgi?id=704951

2013-07-26 22:31:41 +0200  Wim Taymans <wim taymans collabora co uk>

        * gst/rtpmanager/rtpsession.c:
          session: don't make buffer writable prematurely
          There is no reason to make the SR buffer writable at this point. This is better
          delayed until needed.

2013-07-26 22:25:50 +0200  Wim Taymans <wim taymans collabora co uk>

        * gst/rtpmanager/rtpsession.c:
          session: ignore RTCP for inactive sources

2013-07-26 22:25:17 +0200  Wim Taymans <wim taymans collabora co uk>

        * gst/rtpmanager/rtpsession.c:
          session: small cleanup

2013-07-26 17:17:31 +0200  Wim Taymans <wim taymans collabora co uk>

        * gst/rtpmanager/rtpsession.c:
        * gst/rtpmanager/rtpsession.h:
        * gst/rtpmanager/rtpsource.h:
          session: handle partial RTCP report blocks
          When we have more SSRCs to report than what fit in an RTCP packet, use a
          generation counter to make sure all of them end up in a packet eventually.

2013-07-26 17:23:10 +0200  Wim Taymans <wim taymans collabora co uk>

        * gst/rtpmanager/rtpsession.c:
          session: create SSRC before doing session cleanup
          Make the internal source before we do session cleanup

2013-07-26 17:21:08 +0200  Wim Taymans <wim taymans collabora co uk>

        * gst/rtpmanager/rtpsession.c:
          session: reorganize the report block code

2013-07-26 16:02:01 +0200  Matej Knopp <matej knopp gmail com>

        * gst/matroska/matroska-demux.c:
          matroskademux: fix memory leak in check_subtitle_buffer
          https://bugzilla.gnome.org/show_bug.cgi?id=704921

2013-07-26 14:21:40 +0200  Wim Taymans <wim taymans collabora co uk>

        * gst/rtpmanager/rtpsession.c:
          session: refactor active and sender checks

2013-07-26 12:06:35 +0200  Wim Taymans <wim taymans collabora co uk>

        * gst/rtpmanager/rtpsession.c:
          session: remove internal sources on timeout
          When an internal source times out and becomes a receiver, remove it.

2013-07-26 11:47:56 +0200  Wim Taymans <wim taymans collabora co uk>

        * gst/rtpmanager/rtpsession.c:
          session: create an internal source for RTCP
          When we need to do RTCP and we don't have an internal source yet,
          make one.

2013-07-26 10:47:28 +0200  Wim Taymans <wim taymans collabora co uk>

        * gst/rtpmanager/rtpsession.c:
        * gst/rtpmanager/rtpsession.h:
        * gst/rtpmanager/rtpsource.c:
          session: remove old code to change SSRC
          Remove code used to change the SSRC after a collision. We now send
          a RECONFIGURE event upstream to make the upstream element change the SSRC.

2013-07-26 10:42:44 +0200  Wim Taymans <wim taymans collabora co uk>

        * gst/rtpmanager/rtpsource.c:
          source: don't update packet SSRC
          Remove the code to update the SSRC in packets, it can never be called now that
          we always use a source with matching packet SSRC.

2013-07-26 10:24:22 +0200  Wim Taymans <wim taymans collabora co uk>

        * gst/rtpmanager/rtpsession.c:
        * gst/rtpmanager/rtpsession.h:
          session: delay allocation of internal source
          Allocate the internal source when we receive a caps with the SSRC or when we see
          a buffer with the SSRC.

2013-07-26 10:00:58 +0200  Wim Taymans <wim taymans collabora co uk>

        * gst/rtpmanager/gstrtpsession.c:
        * gst/rtpmanager/rtpsession.c:
          session: generate reconfigure on collision
          When we detect a collision, change the SSRC that we suggest upstream
          and trigger RECONFIGURE. This should make upstream select a new SSRC.

2013-07-26 09:37:24 +0200  Wim Taymans <wim taymans collabora co uk>

        * gst/rtpmanager/rtpsession.c:
        * gst/rtpmanager/rtpsession.h:
          session: produce RTCP for all internal sources
          Loop over all the internal sources and produce RTCP. We also need
          to queue the RTCP packets and send them when we are finished.

2013-07-26 01:40:20 +0200  Wim Taymans <wim taymans collabora co uk>

        * gst/rtpmanager/rtpsession.c:
        * gst/rtpmanager/rtpsession.h:
          session: deprecate internal source and ssrc properties
          Deprecate the internal source and internal ssrc properties. There might
          be more than one internal source.

2013-07-26 01:29:08 +0200  Wim Taymans <wim taymans collabora co uk>

        * gst/rtpmanager/rtpsession.c:
          session: internal sources don't use probation

2013-07-26 01:24:07 +0200  Wim Taymans <wim taymans collabora co uk>

        * gst/rtpmanager/gstrtpsession.c:
        * gst/rtpmanager/rtpsession.c:
          session: give caps to session
          Let the session parse the caps and update its SSRC when needed.

2013-07-26 01:14:04 +0200  Wim Taymans <wim taymans collabora co uk>

        * gst/rtpmanager/gstrtpsession.c:
        * gst/rtpmanager/rtpsession.c:
        * gst/rtpmanager/rtpsession.h:
          session: make method to suggest available SSRC
          Make a method to suggest the best available SSRC. This is the SSRC of the last
          created internal source and is used to instruct upstream to produce this
          SSRC.

2013-07-26 01:01:49 +0200  Wim Taymans <wim taymans collabora co uk>

        * gst/rtpmanager/rtpsession.c:
        * gst/rtpmanager/rtpsession.h:
          session: keep SDES and set on new internal sources
          Keep track of the SDES ourselves and set it on all newly created
          internal sources.

2013-07-26 00:48:25 +0200  Wim Taymans <wim taymans collabora co uk>

        * gst/rtpmanager/rtpsession.c:
          session: make method to make internal sources
          Add a method to obtain an internal source and use it to create
          our internal source

2013-07-26 00:29:41 +0200  Wim Taymans <wim taymans collabora co uk>

        * gst/rtpmanager/rtpsession.c:
        * gst/rtpmanager/rtpstats.h:
          session: count internal sources and how many are senders

2013-07-26 00:14:29 +0200  Wim Taymans <wim taymans collabora co uk>

        * gst/rtpmanager/gstrtpsession.c:
        * gst/rtpmanager/rtpsession.c:
        * gst/rtpmanager/rtpsession.h:
          rtpsession: separate BYE marking and scheduling
          First mark sources with BYE and then schedule the BYE RTCP message.

2013-07-25 23:56:46 +0200  Wim Taymans <wim taymans collabora co uk>

        * gst/rtpmanager/rtpsession.c:
          session: get SSRC from RTCP packet itself
          Get the SSRC from the RTCP packet instead.

2013-07-25 23:51:34 +0200  Wim Taymans <wim taymans collabora co uk>

        * gst/rtpmanager/rtpsession.c:
          session: fix bandwidth calculation
          We iterate over all sources and the internal one is also in the
          hashtable so avoid adding it twice.

2013-07-25 23:38:08 +0200  Wim Taymans <wim taymans collabora co uk>

        * gst/rtpmanager/rtpsession.c:
          session: add some docs

2013-07-25 23:11:05 +0200  Wim Taymans <wim taymans collabora co uk>

        * gst/rtpmanager/rtpsession.c:
          session: Rearrange RTCP reporting a little
          Make a function to generate an RTCP packet for a source, pass the source as a
          parameter.
          Move timeout of collisions to session cleanup phase.

2013-07-25 22:39:04 +0200  Wim Taymans <wim taymans collabora co uk>

        * gst/rtpmanager/rtpsession.c:
          session: move check for is_early around
          Move the check for the early RTCP to where it is needed and used.

2013-07-25 17:35:02 +0200  Wim Taymans <wim taymans collabora co uk>

        * gst/rtpmanager/rtpsession.c:
          session: parse packet outside of the session lock

2013-07-25 17:34:06 +0200  Wim Taymans <wim taymans collabora co uk>

        * gst/rtpmanager/rtpsession.c:
          session: do nicer checks for internal sources

2013-07-25 17:15:37 +0200  Wim Taymans <wim taymans collabora co uk>

        * gst/rtpmanager/rtpsession.c:
        * gst/rtpmanager/rtpsession.h:
        * gst/rtpmanager/rtpsource.c:
        * gst/rtpmanager/rtpsource.h:
          session: let source keep track if it sent BYE

2013-07-25 17:06:22 +0200  Wim Taymans <wim taymans collabora co uk>

        * gst/rtpmanager/rtpsource.c:
          source: reset more

2013-07-25 16:49:41 +0200  Wim Taymans <wim taymans collabora co uk>

        * gst/rtpmanager/rtpsession.c:
        * gst/rtpmanager/rtpsession.h:
        * gst/rtpmanager/rtpsource.c:
        * gst/rtpmanager/rtpsource.h:
          source: also use the source for bye_reason
          Store the BYE reason in our internal source object. Rename the methods on the
          source object a little because now the BYE can be received in RTCP or
          set when the session wants to send BYE.

2013-07-25 16:24:04 +0200  Wim Taymans <wim taymans collabora co uk>

        * gst/rtpmanager/rtpsession.c:
        * gst/rtpmanager/rtpsession.h:
        * gst/rtpmanager/rtpsource.c:
        * gst/rtpmanager/rtpsource.h:
          session: configure sdes with structure only
          Remove code to configure the SDES with methods and types, only
          allow configuration with GstStructure

2013-07-25 15:56:39 +0200  Wim Taymans <wim taymans collabora co uk>

        * gst/rtpmanager/rtpsession.c:
          session: refactor add and find source
          Make functions to find and add a source to the hashtable.

2013-07-25 15:43:11 +0200  Wim Taymans <wim taymans collabora co uk>

        * gst/rtpmanager/gstrtpsession.c:
        * gst/rtpmanager/rtpsession.c:
        * gst/rtpmanager/rtpsession.h:
          session: remove source from sync_rtcp
          We don't need to know the sender source of the session in the
          callback, the SR packet is for all participants in the session.

2013-07-24 14:18:14 +0200  Wim Taymans <wim taymans collabora co uk>

        * gst/rtpmanager/gstrtpjitterbuffer.c:
          jitterbuffer: add some more debug

2013-07-15 17:11:45 +0100  Vincent Penquerc'h <vincent penquerch collabora co uk>

        * gst/audioparsers/Makefile.am:
        * gst/audioparsers/gstaacparse.c:
        * gst/audioparsers/gstaacparse.h:
          aacparse: allow conversion from ADTS to raw AAC
          Some muxers (eg, qtmux) only support raw AAC, so this allows linking
          an encoder that outputs ADTS only to those muxers.
          The conversion is simple (omit the first 7 or 9 bytes of the frame),
          but has to be done in pre_push instead of handle_frame as 1.0 does
          not seem to allow skipping bytes there as 0.10 used to.
          Other conversions are not supported (yet).

2013-07-15 17:15:44 +0100  Vincent Penquerc'h <vincent penquerch collabora co uk>

        * gst/audioparsers/gstaacparse.c:
          aacparse: fix object_type parsing off-by-one in ADTS frame
          According to http://wiki.multimedia.cx/index.php?title=ADTS,
          the value stored in ADTS headers is one less than the object
          type of the AAC stream.
          A look at ffmpeg shows it also adds 1 to the value read off
          the ADTS header.
          Note that this might break other things that happen to have
          an inverse off by one to match the existing code.

2013-07-25 11:13:01 -0300  Thiago Santos <thiago sousa santos collabora com>

        * gst/avi/gstavidemux.c:
          avidemux: fix seqnum handling for seeks
          Use the same seqnum as the seek for flushes/segments that are
          caused by the seek. Also do the same for segment events
          Fixes #676242

2013-07-25 01:39:58 -0300  Thiago Santos <thiago sousa santos collabora com>

        * gst/matroska/matroska-demux.c:
        * gst/matroska/matroska-demux.h:
          matroskademux: fix seqnum handling for seeks
          Use the same seqnum as the seek for flushes/segments that are
          caused by the seek. Also do the same for segment events
          Fixes #676242

2013-07-25 01:11:31 -0300  Thiago Santos <thiago sousa santos collabora com>

        * gst/isomp4/qtdemux.c:
          qtdemux: correctly handle seqnum for seeks and segments
          Use the same seqnum on messages and events for derived events.
          Fixed for flushes / stream-start / segment after a seek, and segment
          after a segment.
          Fixes #676242

2013-07-12 20:01:42 +0200  Arnaud Vrac <avrac freebox fr>

        * ext/soup/gstsouphttpsrc.c:
          souphttpsrc: always ignore HEAD errors
          https://bugzilla.gnome.org/show_bug.cgi?id=704241

2013-07-25 14:26:07 +0200  Sebastian Dröge <slomo circular-chaos org>

        * ext/jpeg/gstjpegenc.c:
          jpegenc: Clean up reset/start/stop handling

2013-07-25 14:13:10 +0200  Sebastian Dröge <slomo circular-chaos org>

        * ext/jpeg/gstjpegdec.c:
        * ext/jpeg/gstjpegdec.h:
          jpegdec: Use base class error handling function instead of replicating it here

2013-07-25 14:12:56 +0200  Sebastian Dröge <slomo circular-chaos org>

        * ext/jpeg/gstjpegdec.c:
          jpegdec: Clean up handling of reset/start/stop

2013-07-25 10:41:22 +0100  Tim-Philipp Müller <tim muller collabora co uk>

        * tests/files/id3-407349-1.tag:
        * tests/files/id3-407349-2.tag:
        * tests/files/id3-447000-wcop.tag:
          tests: fix test ID3 tags up not to rely on dodgy typefinding code
          Change 0xff 0xfb 'mp3' marker to 'fLaC' marker, so we can fix
          the typefinder.
          https://bugzilla.gnome.org/show_bug.cgi?id=681368

2013-07-25 08:22:45 +0200  Alessandro Decina <alessandro d gmail com>

        * sys/osxaudio/gstosxaudiosink.c:
          osxaudiosink: intersect the probed caps with the filter passed to get_caps()

2013-07-24 14:17:45 +0200  Wim Taymans <wim taymans collabora co uk>

        * gst/rtpmanager/gstrtpbin.c:
          bin: fix compilation

2013-07-24 12:42:31 +0200  Wim Taymans <wim taymans collabora co uk>

        * gst/rtp/gstrtpvrawdepay.c:
          vrawdepay: fix UYVP format

2013-07-24 12:41:58 +0200  Wim Taymans <wim taymans collabora co uk>

        * gst/rtp/gstrtpvrawpay.c:
          vrawpay: fix UYVP format

2013-07-24 12:41:44 +0200  Wim Taymans <wim taymans collabora co uk>

        * gst/rtp/gstrtpvrawpay.c:
          vrawpay: fix caps

2013-07-24 10:49:03 +0200  Wim Taymans <wim taymans collabora co uk>

        * gst/rtpmanager/gstrtpjitterbuffer.c:
          rtpjitterbuffer: fix locking
          Take the lock earlier so that we do things that follow with the right
          locking.

2013-07-23 17:40:02 +0200  Wim Taymans <wim taymans collabora co uk>

        * gst/rtpmanager/rtpsession.c:
          rtpsession: don't use invalid times in RTCP timeouts
          An invalid timeout can be calculated when we disabled RTCP by setting the
          bandwidth to 0. Make sure all code can handle this case.
          Fixes https://bugzilla.gnome.org/show_bug.cgi?id=674626

2013-07-23 17:38:20 +0200  Wim Taymans <wim taymans collabora co uk>

        * gst/rtpmanager/rtpsession.c:
          rtpsession: lock session when changing bandwidth
          Take the session lock when changing the bandwidth properties so that we don't
          end up with inconsistent behaviour.

2013-07-23 17:37:05 +0200  Wim Taymans <wim taymans collabora co uk>

        * gst/rtpmanager/rtpsession.c:
          session: reset some RTCP variables
          The early_send time was set to 0 and always triggering an early RTCP packet.

2013-07-23 15:03:31 +0200  Edward Hervey <edward collabora com>

        * gst/isomp4/qtdemux.c:
          qtdemux: Add all the mpeg XDCAM variants
          This should cover all known XDCAM variants (which are all mpeg2 video)
          Fixes #672227

2013-07-03 18:41:42 +0200  Carlos Rafael Giani <dv pseudoterminal org>

        * gst/rtpmanager/gstrtpbin.c:
        * gst/rtpmanager/gstrtpbin.h:
          rtpbin: added custom downstream sync event
          rtpbin can now send a custom in-band downstream event which informs
          downstream that the bin has received an RTCP SR packet. This is useful
          for applications which want to drop the initial unsynchronized received
          RTP packets.
          Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703560
          Signed-off-by: Carlos Rafael Giani <dv pseudoterminal org>

2013-07-22 18:00:16 +0100  Tim-Philipp Müller <tim muller collabora co uk>

        * gst/deinterlace/gstdeinterlace.c:
          deinterlace: fix on-the-fly changing of "mode" and "fields" properties
          We call setcaps() to reconfigure ourselves, but we need to pass
          the current *sink* caps, not the source caps then. Also fix a
          caps leak.
          https://bugzilla.gnome.org/show_bug.cgi?id=641599

2013-07-22 15:23:39 +0200  Sebastian Dröge <slomo circular-chaos org>

        * gst/wavparse/gstwavparse.c:
          wavparse: Add support for group-id in the stream-start event

2013-07-22 15:23:20 +0200  Sebastian Dröge <slomo circular-chaos org>

        * gst/rtsp/gstrtspsrc.c:
          rtspsrc: Add support for group-id in the stream-start event

2013-07-22 15:23:11 +0200  Sebastian Dröge <slomo circular-chaos org>

        * gst/rtpmanager/gstrtpsession.c:
          rtpsession: Add support for group-id in the stream-start event

2013-07-22 15:22:55 +0200  Sebastian Dröge <slomo circular-chaos org>

        * gst/matroska/matroska-demux.c:
        * gst/matroska/matroska-demux.h:
          matroskademux: Add support for group-id in the stream-start event

2013-07-22 15:22:47 +0200  Sebastian Dröge <slomo circular-chaos org>

        * gst/isomp4/qtdemux.c:
        * gst/isomp4/qtdemux.h:
          qtdemux: Add support for group-id in the stream-start event

2013-07-22 15:22:36 +0200  Sebastian Dröge <slomo circular-chaos org>

        * gst/flv/gstflvdemux.c:
        * gst/flv/gstflvdemux.h:
          flvdemux: Add support for group-id in the stream-start event

2013-07-22 15:22:16 +0200  Sebastian Dröge <slomo circular-chaos org>

        * gst/avi/gstavidemux.c:
        * gst/avi/gstavidemux.h:
          avidemux: Add support for group-id in the stream-start event

2013-07-22 15:21:49 +0200  Sebastian Dröge <slomo circular-chaos org>

        * ext/dv/gstdvdemux.c:
        * ext/dv/gstdvdemux.h:
          dvdemux: Add support for group-id in the stream-start event

2013-07-19 22:59:15 +0200  Mathieu Duponchelle <mathieu duponchelle epitech eu>

        * gst/videomixer/videomixer2.c:
          videomixer: use gst_util_uint64_scale*_round.
          There could be a case where:
          1) you do a new set_caps after buffers have been processed.
          2) ts_offset gets set to a different value, eg 0.033333333
          3) your pads get EOS, but the check dor that doesn't work
          because you use ts_offset + a truncated value < segment.stop
          4) so in the next collected, you end up comparing for example:
          0.9999999999 > 1., which is false and means you don't send EOS.
          Also adds scale_round in two other places where it potentially could
          have caused problems.

2013-07-15 17:55:19 -0400  Olivier Crête <olivier crete collabora com>

        * gst/isomp4/qtdemux.c:
        * gst/isomp4/qtdemux_fourcc.h:
          qtdemux: Add WRLE support

2013-07-19 19:35:26 +0100  Tim-Philipp Müller <tim muller collabora co uk>

        * gst/isomp4/qtdemux.c:
        * gst/isomp4/qtdemux_fourcc.h:
          qtdemux: make files from Vivotek camera play
          Skip tracks of 'vivo' subtype with empty stsd instead of
          erroring out saying that the file is broken.
          https://bugzilla.gnome.org/show_bug.cgi?id=699791

2013-07-19 17:14:06 +0100  Tim-Philipp Müller <tim muller collabora co uk>

        * gst/isomp4/gstqtmux.c:
          qtmux: when streaming don't try to seek when stopping
          It might cause errors in sinks that are not seekable and
          have reported this (like e.g. fdsink)
          https://bugzilla.gnome.org/show_bug.cgi?id=696228

2013-07-19 17:26:54 +0200  Wim Taymans <wim taymans collabora co uk>

        * gst/isomp4/qtdemux.c:
          qtdemux: simplify some helpers
          Some helper functions are not needed anymore or can be simplified.

2013-07-19 17:12:37 +0200  Wim Taymans <wim taymans collabora co uk>

        * gst/isomp4/qtdemux.c:
          qtdemux: for non-raw video, move palette in caps
          We only need to append the palette to raw video buffers, non-raw video has the
          palette in the caps still.
          Fixes https://bugzilla.gnome.org/show_bug.cgi?id=704292

2013-07-19 01:49:20 +0200  Arnaud Vrac <avrac freebox fr>

        * gst/isomp4/qtdemux.c:
          qtdemux: nitpicking in esds parsing

2013-07-19 01:49:07 +0200  Arnaud Vrac <avrac freebox fr>

        * gst/isomp4/qtdemux.c:
          qtdemux: set proper caps for mpeg-1 audio
          Remove AAC specific fields from mpeg-1 audio caps, remove assumption
          that the mpeg1 audio layer is 3, and set `parsed' field.
          https://bugzilla.gnome.org/show_bug.cgi?id=704548

2013-06-17 21:27:37 +0200  Arnaud Vrac <avrac freebox fr>

        * ext/vpx/gstvp8dec.h:
        * ext/vpx/gstvp8enc.h:
        * ext/vpx/gstvp9dec.h:
        * ext/vpx/gstvp9enc.h:
          vpx: fix compilation when encoder or decoder headers are not installed
          https://bugzilla.gnome.org/show_bug.cgi?id=704547

2013-07-16 20:41:15 -0400  Nicolas Dufresne <nicolas dufresne collabora com>

        * tests/check/elements/videocrop.c:
          videocrop: Fix unit for GRAY16 formats

2013-07-16 22:17:17 +0200  Arnaud Vrac <avrac freebox fr>

        * gst/isomp4/qtdemux.c:
          qtdemux: remove chapter stream
          Remove all streams that are actually table of contents, since we will
          never need the data after parsing them.

2013-07-16 21:59:37 +0200  Arnaud Vrac <avrac freebox fr>

        * gst/isomp4/qtdemux.c:
          qtdemux: send gap event for sparse streams in push mode
          This allows to pre-roll at least if the next subtitle buffer
          is far away.

2013-07-16 21:56:07 +0200  Arnaud Vrac <avrac freebox fr>

        * gst/isomp4/qtdemux.c:
          qtdemux: do not use indexes from sparse stream when seeking in push mode
          This makes seeking more accurate in push mode, since the previous
          keyframe on a sparse stream might be far away.

2013-07-16 21:04:07 +0200  Arnaud Vrac <avrac freebox fr>

        * gst/isomp4/qtdemux.c:
          qtdemux: advertise subtitle streams as sparse

2013-07-17 17:11:44 +0200  Arnaud Vrac <avrac freebox fr>

        * gst/matroska/matroska-demux.c:
          mastrokademux: do not push discont buffers if they aren't discont
          Unset the discont flag instead of posssibly pushing a buffer with
          a flag that's still set.
          https://bugzilla.gnome.org/show_bug.cgi?id=682110

2013-07-17 15:10:00 +0200  Wim Taymans <wim taymans collabora co uk>

        * gst/isomp4/qtdemux.c:
          qtdemux: extract the palette from stsd
          Sometimes a palette is inside the stsd, extract it instead of always using
          the default one

2013-07-17 14:30:16 +0200  Sebastian Dröge <slomo circular-chaos org>

        * gst/goom2k1/gstgoom.c:
          goom2k1: Fix event handling and negotiate as soon as possible

2013-07-17 14:27:57 +0200  Sebastian Dröge <slomo circular-chaos org>

        * gst/goom/gstgoom.c:
          goom: Fix event handling and negotiate as soon as possible

2013-07-11 19:45:17 +0200  Andoni Morales Alastruey <ylatuya gmail com>

        * sys/osxvideo/osxvideosink.m:
          osxvideosink: warn about the future deprecation of the "embed" property

2013-07-17 09:56:01 +0200  Wim Taymans <wim taymans collabora co uk>

        * gst/isomp4/qtdemux.c:
          qtdemux: add support for WRAW
          Fixes https://bugzilla.gnome.org/show_bug.cgi?id=704292

2013-07-17 09:54:58 +0200  Wim Taymans <wim taymans collabora co uk>

        * gst/isomp4/qtdemux.c:
          qtdemux: palette is appended to buffers, not in caps
          Fix the palette handling, in 1.0 we append the palette to the buffer instead of
          placing it on the caps.
          See also https://bugzilla.gnome.org/show_bug.cgi?id=704292

2013-07-16 15:37:49 -0400  Olivier Crête <olivier crete collabora com>

        * gst/rtp/gstrtpgstpay.c:
        * gst/rtp/gstrtpmp2tpay.c:
        * gst/rtp/gstrtpmp4gpay.c:
        * gst/rtp/gstrtpmp4vpay.c:
        * gst/rtp/gstrtpmpapay.c:
        * gst/rtp/gstrtpmpvpay.c:
          rtp: Use gst_adapter_take_buffer_fast() where possible in RTP payloaders

2013-07-15 16:24:07 +0200  Arnaud Vrac <avrac freebox fr>

        * gst/isomp4/qtdemux.c:
          qtdemux: reset segment on flush stop
          cca2f555d14 introduces a regression, where the demux segment is not
          reset on flush stop, so the next upstream segment event will calculate
          an invalid base time on the new segment to be sent downstream.
          https://bugzilla.gnome.org/show_bug.cgi?id=704255

2013-07-06 17:20:49 +0200  Matej Knopp <matej knopp gmail com>

        * gst/isomp4/qtdemux.c:
        * gst/isomp4/qtdemux.h:
          qtdemux: offset samples according to edit list
          https://bugzilla.gnome.org/show_bug.cgi?id=700264

2013-07-14 12:50:13 +1200  Douglas Bagnall <douglas halo gen nz>

        * tests/examples/spectrum/spectrum-example.c:
          level: Fix the spectrum example for 1.0
          The "message" property has been replaced by "post-messages".
          Pre-patch output:
          (test_spectrum:23101): GLib-GObject-WARNING **: g_object_set_valist:
          object class `GstSpectrum' has no property named `message'
          New spectrum message, endtime 0:00:00.100000000
          (test_spectrum:23101): GStreamer-CRITICAL **:
          gst_value_list_get_value: assertion `GST_VALUE_HOLDS_LIST (value)' failed
          [...]
          Post-patch:
          New spectrum message, endtime 0:00:00.100000000
          band 0 (freq 400): magnitude -65.988777 dB phase 1.533397
          band 1 (freq 1200): magnitude -65.545563 dB phase -0.780900
          band 2 (freq 2000): magnitude -64.791946 dB phase -0.799611
          band 3 (freq 2800): magnitude -64.556175 dB phase -0.063615
          [...]
          https://bugzilla.gnome.org/show_bug.cgi?id=704179

2013-07-13 20:56:26 +0200  Matej Knopp <matej knopp gmail com>

        * gst/audioparsers/gstaacparse.c:
          aacparse: be less verbose when parsing LOAS streams
          https://bugzilla.gnome.org/show_bug.cgi?id=704162

2013-07-12 12:31:39 +0200  Wim Taymans <wim taymans collabora co uk>

        * ext/pulse/pulsesink.h:
          sink: alaw/mulaw caps don't have a layout property

2013-07-12 12:27:53 +0200  Wim Taymans <wim taymans collabora co uk>

        * ext/pulse/pulseutil.c:
          pulse: relax mulaw and alaw format checks
          The audio library considers them as encoded formats and does not fill in the
          sample width. The audio ringbuffers identifies the format as alaw/mulaw and that
          is always 8 bits.

2013-07-11 16:13:05 +0200  Matej Knopp <matej knopp gmail com>

        * gst/isomp4/qtdemux.c:
        * gst/isomp4/qtdemux.h:
        * gst/isomp4/qtdemux_fourcc.h:
        * gst/isomp4/qtdemux_types.c:
          qtdemux: unselect instead of ignoring disabled track, detect chapter track
          https://bugzilla.gnome.org/show_bug.cgi?id=704007

2013-07-11 20:41:23 -0300  Thiago Santos <thiago sousa santos collabora com>

        * ext/soup/gstsouphttpsrc.c:
          souphttpsrc: ignore errors from HEAD request
          HEAD requests are used to check the server headers to see if it
          seekable. Ignore errors from those requests as they shouldn't be
          critical.
          https://bugzilla.gnome.org/show_bug.cgi?id=704053

2013-07-12 03:24:08 +0800  Kyosuke Nekomura <supercatexpert gmail com>

        * gst/audiofx/audioecho.c:
          audioecho: Fix handling of delay property in PLAYING/PAUSED state
          https://bugzilla.gnome.org/show_bug.cgi?id=703901

2013-07-09 17:56:57 -0400  Olivier Crête <olivier crete collabora com>

        * gst/rtpmanager/gstrtpmux.c:
          rtpmux: Enable proxy caps on the src pads

2013-07-11 16:57:15 +0200  Sebastian Dröge <slomo circular-chaos org>

        * configure.ac:
          Back to development



Download
========
http://download.gnome.org/sources/gst-plugins-good/1.1/gst-plugins-good-1.1.3.tar.xz (2.70M)
  sha256sum: 1f516cb39e24ba5544a503bef246f9cf5b002dfbcbc2c475c6b8d5c23bc140a6



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