gst-plugins-good 1.2.1



ChangeLog
=========

2013-11-09  Sebastian Dröge <slomo coaxion net>

        * configure.ac:
          releasing 1.2.1

2013-11-09 12:01:55 +0100  Sebastian Dröge <sebastian centricular com>

        * po/de.po:
        * po/id.po:
        * po/sr.po:
          po: Update translations

2013-11-08 17:59:24 +0100  Philippe Normand <philn igalia com>

        * gst/wavenc/gstwavenc.c:
          wavenc: generate a non-empty data header
          Restore the behavior of the element to the state before commit
          db29522a430e44450415ca3676abd1b77ee923d9. A non-empty header is
          generated and when the EOS event is received the header is generated
          again, this time with the correct size.
          https://bugzilla.gnome.org/show_bug.cgi?id=711699

2013-10-07 14:27:21 -0300  Thiago Santos <ts santos partner samsung com>

        * ext/soup/gstsouphttpsrc.c:
        * ext/soup/gstsouphttpsrc.h:
          souphttpsrc: do not emit EOS when connection drops
          If the pipeline is stalled for too long, souphttpsrc will block and
          stop fetching data from the network. This can cause the connection to
          drop and souphttpsrc would handle it as an EOS. This patch makes it
          persist and try to fetch more data until the end of the content length
          or until receiving an error that it is beyong limits in case the content
          is unknown.
          https://bugzilla.gnome.org/show_bug.cgi?id=683536

2013-10-25 11:30:36 -0300  Thiago Santos <ts santos partner samsung com>

        * gst/isomp4/qtdemux.c:
          qtdemux: check if the end_time is defined before using it
          Avoids sending EOS too soon because of overflow. Can happen on
          fragmented mp4 playback.

2013-10-25 18:22:00 -0300  Thiago Santos <ts santos partner samsung com>

        * gst/isomp4/qtdemux.c:
          qtdemux: handle fragmented files with mdat before moofs
          Assume a file with atoms in the following order: moov, mdat, moof,
          mdat, moof ...
          The first moov usually doesn't contain any sample entries atoms (or
          they are all set to 0 length), because the real samples are signaled
          at the moofs. In push mode, qtdemux parses the moov and then finds the mdat,
          but then it has 0 entries and assumes it is EOS.
          This patch makes it continue parsing in case it is a fragmented file so that
          it might find the moofs and play the media.
          https://bugzilla.gnome.org/show_bug.cgi?id=710623

2013-10-25 11:42:37 -0300  Thiago Santos <ts santos partner samsung com>

        * gst/isomp4/qtdemux.c:
        * gst/isomp4/qtdemux.h:
          qtdemux: When using a buffered mdat, store all received data for later use
          In push mode, when qtdemux can't use a seek to skip the mdat buffer it has
          to buffer it for later use.
          The issue is that after parsing the next moov/moof, there might be some
          trailing bytes from the next atom in the file. This data was being discarded
          along with the already parsed moov/moof and playback would fail to continue
          after the contents of this moov/moof are played.
          This is particularly bad on fragmented files that have the mdat before the
          corresponding moof. So you'd get:
          mdat|moof|mdat|moof ...
          When a moof was received, it usually came with some extra bytes that would
          belong to the next mdat (because upstream doesn't care about atoms alignment).
          So those bytes were being discarded and playback would fail.
          This patch makes qtdemux store those extra bytes to reuse them later after the
          mdat is emptied.
          https://bugzilla.gnome.org/show_bug.cgi?id=710623

2013-11-07 09:49:55 +0100  Sebastian Dröge <sebastian centricular com>

        * gst/udp/gstmultiudpsink.c:
          multiudpsink: Also use the bind-port property if no bind-address was given

2013-11-07 00:51:12 +0100  Andoni Morales Alastruey <ylatuya gmail com>

        * sys/osxaudio/gstosxcoreaudiohal.c:
          osxaudiosink: fix segfault when we can't get the channels layout

2013-11-05 17:26:49 +0100  Sebastian Dröge <sebastian centricular com>

        * gst/rtp/gstrtpvp8pay.c:
          rtpvp8pay: Make Picture ID mode configurable and default to no picture ID
          Some implementations (linphone) only support no picture at all in the
          stream and will fail if one is provided.
          https://bugzilla.gnome.org/show_bug.cgi?id=711497

2013-11-02 22:50:47 +0100  Rico Tzschichholz <ricotz ubuntu com>

        * gst/rtsp/Makefile.am:
        * gst/rtsp/gstrtspsrc.h:
          rtsp: Add missing gio-2.0 deps and includes

2013-11-01 18:10:51 +0000  Olivier Crête <olivier crete collabora com>

        * configure.ac:
          Revert "configure: Require gst-plugins-base >= 1.2.1 for the TLS validation check flags in 
GstRTSPConnection"
          Version 1.2.1 doesn't exist yet, re-apply when it does
          This reverts commit c98380985db3483ea78a8e738d544d1201d8ed1e.

2013-11-01 18:31:36 +0100  Sebastian Dröge <sebastian centricular com>

        * gst/audiofx/audioiirfilter.c:
          audioiirfilter: Fix initialization coefficient handling
          Broke unit test.

2013-11-01 16:59:11 +0100  Sebastian Dröge <sebastian centricular com>

        * configure.ac:
          configure: Require gst-plugins-base >= 1.2.1 for the TLS validation check flags in GstRTSPConnection

2013-10-31 14:05:43 -0700  Aleix Conchillo Flaque <aleix oblong com>

        * gst/rtsp/gstrtspsrc.c:
        * gst/rtsp/gstrtspsrc.h:
          rtspsrc: allow setting tls certificate validation flags
          Added a new property "tls-validation-flags". If the url transport is
          TLS, the validation flags will be set to the rtsp connection.
          https://bugzilla.gnome.org/show_bug.cgi?id=711230

2013-10-31 22:43:49 +0100  Sebastian Dröge <sebastian centricular com>

        * gst/audiofx/audiofxbaseiirfilter.c:
        * gst/audiofx/audioiirfilter.c:
          audioiirfilter: Don't crash if no filter coefficients are provided
          ...and by default use a identity filter.
          https://bugzilla.gnome.org/show_bug.cgi?id=710215

2013-10-31 19:15:12 +0100  Sebastian Dröge <sebastian centricular com>

        * ext/wavpack/gstwavpackenc.c:
          wavpackenc: Fix writing of MD5 sums and other metadata blocks
          These don't have the FINAL_BLOCK flag set.

2013-10-14 16:23:25 +0200  Ognyan Tonchev <ognyan axis com>

        * gst/udp/gstmultiudpsink.c:
          multiudpsink: Fix memory leak
          Unmap all GstMemory of the current buffer when flushing.
          https://bugzilla.gnome.org/show_bug.cgi?id=710110

2013-10-12 20:37:41 +0100  Tim-Philipp Müller <tim centricular net>

        * gst/flv/gstflvmux.c:
          flvmux: require stream-format=raw for mpeg-2 too, but don't require framed field
          raw implies that it's framed already. Fixes .. ! faac ! flvmux

2013-10-10 13:52:35 +0200  Sebastian Dröge <slomo circular-chaos org>

        * ext/dv/gstdvdec.c:
        * ext/dv/gstdvdec.h:
          dvdec: Don't send segment event before caps
          https://bugzilla.gnome.org/show_bug.cgi?id=709728

2013-10-09 17:46:33 +0200  Sebastian Dröge <slomo circular-chaos org>

        * ext/dv/gstdvdemux.c:
          dvdemux: Send stream-start, caps and segment events in the right order
          https://bugzilla.gnome.org/show_bug.cgi?id=709728

2013-10-08 11:28:04 +0200  Sebastian Dröge <slomo circular-chaos org>

        * gst/wavenc/gstwavenc.c:
          wavenc: A-Law and Mu-Law don't have width/depth/signed caps fields
          https://bugzilla.gnome.org/show_bug.cgi?id=709614

2013-10-07 12:54:11 +0200  Sebastian Dröge <slomo circular-chaos org>

        * gst/deinterlace/tvtime/greedyh.c:
          deinterlace: Fix handling of planar video formats in greedyh method
          https://bugzilla.gnome.org/show_bug.cgi?id=709507

2013-10-04 13:34:09 +0200  Peter Korsgaard <peter korsgaard com>

        * sys/v4l2/gstv4l2bufferpool.c:
          v4l2bufferpool: O_CLOEXEC needs _GNU_SOURCE
          On some systems (E.G. uClibc and older Glibc versions), O_CLOEXEC is only
          defined when _GNU_SOURCE is specified, so do so.
          _GNU_SOURCE needs to be defined before any system headers are included,
          so move the fcntl.h section up.
          https://bugzilla.gnome.org/show_bug.cgi?id=709423

2013-10-04 14:42:59 -0700  Reynaldo H. Verdejo Pinochet <r verdejo partner samsung com>

        * gst/matroska/matroska-mux.c:
          matroska: Do not write SegmentUID for WebM mux
          WebM spec states SegmentUID is Unsupported. Files produced
          with gstreamer without this change will spit an error like
          this when passed to mkvalidator:
          ERR201: Invalid 'SegmentUID' for profile 'webm' in Info at 192

2013-10-03 22:38:43 +0200  Mathieu Duponchelle <mduponchelle1 gmail com>

        * gst/videomixer/videoconvert.c:
          videomixer: Update videoconvert copy
          https://bugzilla.gnome.org/show_bug.cgi?id=709390

2013-10-03 21:36:34 +0200  Mathieu Duponchelle <mduponchelle1 gmail com>

        * gst/videomixer/videomixer2.c:
          videomixer: Check if the pad needs reconfiguration in collected
          https://bugzilla.gnome.org/show_bug.cgi?id=709384

2013-10-03 11:59:25 +0200  Sebastian Dröge <slomo circular-chaos org>

        * gst/isomp4/qtdemux.c:
          qtdemux: Add support for the mp2v fourcc for MPEG-2 video
          https://bugzilla.gnome.org/show_bug.cgi?id=709270

2013-10-04 12:11:56 +0200  Wim Taymans <wim taymans collabora co uk>

        * gst/rtpmanager/gstrtpjitterbuffer.c:
          jitterbuffer: fix race in flush-start/flush-stop
          When flush-stop arrives before we process the result of the _push() in the
          loop function, we might pause even though we are not flushing anymore. Fix this
          race by waiting for the srcpad loop function to completely pause after doing the
          flush-start.

2013-10-03 14:39:35 +0100  Matthieu Bouron <matthieu bouron collabora com>

        * ext/jpeg/gstjpegdec.c:
          jpegdec: Relax sink caps
          Since jpegdec already parse the jpeg stream, the sink caps could be
          relaxed. This will allow jpegdec to be selected in more case and in
          particular when the jpeg typefinder does not find the width and height.
          https://bugzilla.gnome.org/show_bug.cgi?id=709352

2013-10-02 15:56:53 +0200  Ognyan Tonchev <ognyan axis com>

        * gst/matroska/matroska-demux.c:
          matroskademux: Fix memory leak
          https://bugzilla.gnome.org/show_bug.cgi?id=709266

2013-09-30 12:24:32 +0200  Ognyan Tonchev <ognyan axis com>

        * gst/rtp/gstrtpgstpay.c:
          rtpgstpay: Fix memory leak
          We were leaking the GList nodes of the pending buffers.
          https://bugzilla.gnome.org/show_bug.cgi?id=709079

2013-09-30 12:31:00 +0200  Wim Taymans <wim taymans collabora co uk>

        * gst/rtpmanager/gstrtpjitterbuffer.c:
        * gst/rtpmanager/rtpjitterbuffer.h:
          rtpjitterbuffer: fix race when updating the next_seqnum
          If we were not waiting for the missing seqnum when we insert the lost packet
          event in the jitterbuffer, we end up not updating the next_seqnum and wait
          forever for the lost packets to arrive. Instead, keep track of the amount of
          packets contained by the jitterbuffer item and update the next expected
          seqnum only after pushing the buffer/event. This makes sure we correctly handle
          GAPS in the sequence numbers.

2013-09-30 12:30:23 +0200  Wim Taymans <wim taymans collabora co uk>

        * gst/rtpmanager/gstrtpjitterbuffer.c:
          rtpjitterbuffer: small debug improvement

2013-09-30 11:53:08 +0200  Wim Taymans <wim taymans collabora co uk>

        * gst/rtpmanager/rtpjitterbuffer.c:
          rtpjitterbuffer: reset skew does not reset clock-rate
          Don't reset the clock-rate when we reset the skew correction algorithm.
          Reset the skew correction algorithm when we change the clock-rate.

2013-09-30 11:16:32 +0200  Wim Taymans <wim taymans collabora co uk>

        * gst/rtpmanager/gstrtpjitterbuffer.c:
          rtpjitterbuffer: pause timer when PAUSED
          Also pause the timer when we go to the PAUSED state. It is possible that we
          don't have a clock or base-time in PAUSED to perform the timeouts.

2013-09-30 11:15:25 +0200  Wim Taymans <wim taymans collabora co uk>

        * gst/rtpmanager/gstrtpjitterbuffer.c:
          rtpjitterbuffer: improve debug

2013-09-27 15:05:04 +0200  Wim Taymans <wim taymans collabora co uk>

        * gst/rtsp/gstrtspsrc.c:
          rtspsrc: also go into the loop function after connect
          When we have opened the stream, go into the loop function so that we can
          receive messages from the server.

2013-09-26 16:20:04 +0200  Wim Taymans <wim taymans collabora co uk>

        * gst/rtpmanager/rtpjitterbuffer.c:
          rtpjitterbuffer: don't calculate skew without rtptime
          Skip trying to calculate the skew when we don't have an rtptime.
          It causes problems when lost packet events are placed in the jitterbuffer.

2013-09-25 17:42:02 +0200  Wim Taymans <wim taymans collabora co uk>

        * gst/rtsp/gstrtspsrc.c:
          rtspsrc: disable checks when linking pads
          We know the pad links will work (and we don't check the return value
          anyway).

2013-09-25 17:36:15 +0200  Wim Taymans <wim taymans collabora co uk>

        * gst/rtpmanager/gstrtpbin.c:
          rtpbin: avoid some pad link checks
          Link pads without checks, we know it will work.

2013-09-24 04:02:09 +0200  Wim Taymans <wim taymans collabora co uk>

        * gst/rtpmanager/gstrtpjitterbuffer.c:
          rtpjitterbuffer: calculate some stats

2013-09-23 17:05:44 +0200  Wim Taymans <wim taymans collabora co uk>

        * gst/rtpmanager/gstrtpjitterbuffer.c:
          rtpjitterbuffer: move send_lost_event function
          Move the send_lost_event function to the do_lost_event handling, there is no
          need to have a separate function.

2013-10-03 18:33:01 +0100  Tim-Philipp Müller <tim centricular net>

        * sys/v4l2/gstv4l2object.c:
          v4l2src: print probed caps as caps again in debug log
          This got lost during refactoring.

2013-09-26 20:41:26 +0200  Hans Månsson <hansm axis com>

        * gst/isomp4/gstqtmuxmap.c:
          mp4mux: Do not require framerate in peer video caps
          Remove the framerate restriction on the caps.
          Reference: https://bugzilla.gnome.org/show_bug.cgi?id=708864

2013-09-16 11:20:51 -0300  Thiago Santos <thiago sousa santos collabora com>

        * gst/isomp4/qtdemux.c:
          qtdemux: add code to parse creation time earlier than 1970
          Use g_date_time seconds manipulation to allow to cover the quicktime
          spec for creation_time. It uses seconds since 1904.
          Both paths could be done using the generic approach of seconds since
          1904 with GDateTime handling, but the first path using seconds from
          1970 should be more commonly found and avoids a few objects creation and
          ref/unref, so keep it there for performance.
          Additionally, the code for handling seconds since 1970 changed from >
          to >= because having 0 seconds since 1970 is also a valid case for that
          path to handle.
          https://bugzilla.gnome.org/show_bug.cgi?id=707975

2013-09-21 00:55:26 +0200  Matej Knopp <matej knopp gmail com>

        * gst/matroska/matroska-demux.c:
          matroskademux: update stream->pos when sending buffers so that gap events are not sent unnecessarily
          https://bugzilla.gnome.org/show_bug.cgi?id=708505

2013-09-27 12:53:06 +0200  Matej Knopp <matej knopp gmail com>

        * gst/matroska/matroska-demux.c:
          matroskademux: move the check for subtitle buffer being null terminated before validating UTF-8
          https://bugzilla.gnome.org/show_bug.cgi?id=707933

2013-09-25 12:55:21 +0200  Sebastian Dröge <slomo circular-chaos org>

        * gst/isomp4/gstqtmux.c:
          qtmux: Don't error out if downstream is not seekable for non-fragmented variants
          Doing so would be a regression over 1.0 and breaks the unit test.
          However the result will be most likely unusable, so let's post
          a warning message on the bus.

2013-09-24 17:24:26 +0100  Tim-Philipp Müller <tim centricular net>

        * README:
        * common:
          Automatic update of common submodule
          From 6b03ba7 to 7412249



Download
========
https://download.gnome.org/sources/gst-plugins-good/1.2/gst-plugins-good-1.2.1.tar.xz (2.74M)
  sha256sum: 660fa02dbe01086fcf702d87acc0ba5dde2559d6a11ecf438874afe504c50517



[Date Prev][Date Next]   [Thread Prev][Thread Next]   [Thread Index] [Date Index] [Author Index]