+=== release 1.5.1 ===
+
+2015-06-07 Sebastian Dröge <slomo@coaxion.net>
+
+ * configure.ac:
+ releasing 1.5.1
+
+2015-06-07 10:32:38 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/rtpsession.c:
+ * gst/rtpmanager/rtpsession.h:
+ * tests/check/elements/rtpsession.c:
+ rtpsession: Override the SSRC from the packets' SSRC if none was given via caps or property
+
+2015-06-07 09:35:38 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * po/af.po:
+ * po/az.po:
+ * po/bg.po:
+ * po/ca.po:
+ * po/cs.po:
+ * po/da.po:
+ * po/de.po:
+ * po/el.po:
+ * po/en_GB.po:
+ * po/eo.po:
+ * po/es.po:
+ * po/eu.po:
+ * po/fi.po:
+ * po/fr.po:
+ * po/gl.po:
+ * po/hr.po:
+ * po/hu.po:
+ * po/id.po:
+ * po/it.po:
+ * po/ja.po:
+ * po/lt.po:
+ * po/lv.po:
+ * po/mt.po:
+ * po/nb.po:
+ * po/nl.po:
+ * po/or.po:
+ * po/pl.po:
+ * po/pt_BR.po:
+ * po/ro.po:
+ * po/ru.po:
+ * po/sk.po:
+ * po/sl.po:
+ * po/sq.po:
+ * po/sr.po:
+ * po/sv.po:
+ * po/tr.po:
+ * po/uk.po:
+ * po/vi.po:
+ * po/zh_CN.po:
+ * po/zh_HK.po:
+ * po/zh_TW.po:
+ po: Update translations
+
+2015-06-05 15:32:10 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
+
+ * sys/v4l2/gstv4l2object.c:
+ v4l2: Don't warn when optional CID are not implement
+ gst_v4l2_get_attributre() shall only be used when the CID is expected
+ to be supported. Otherwise, we get unwanted warning posted to the bus.
+
+2015-06-05 16:43:08 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/gstrtpsession.c:
+ * gst/rtpmanager/rtpsession.c:
+ * gst/rtpmanager/rtpsession.h:
+ rtpsession: Only suggest our internal ssrc if it's not a random one and was selected as internal ssrc
+ https://bugzilla.gnome.org/show_bug.cgi?id=749581
+
+2015-06-04 14:18:01 +0900 Vineeth TM <vineeth.tm@samsung.com>
+
+ * gst/interleave/interleave.c:
+ interleave: error when channel-positions-from-input=False
+ self->channels is being incremented only when
+ channel-positions-from-input is set as TRUE. So in case of FALSE
+ self->func is not set and hence creating assertion error.
+ Hence removing the condition to increment self->channels.
+ https://bugzilla.gnome.org/show_bug.cgi?id=744211
+
+2015-06-05 10:33:11 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ rtpjitterbuffer: Add support for receiving reduced size RTCP
+ It worked before but gave warnings, now we just ignore RTCP
+ packets that don't start with a SR. As all we're interested
+ in here are SRs.
+
+2015-06-03 12:22:42 +0200 Jose Antonio Santos Cadenas <santoscadenas@gmail.com>
+
+ * gst/rtpmanager/gstrtpssrcdemux.c:
+ rtpssrcdemux: Add support for reduce size rtcp
+ According to RFC 5506, reduce size packages can be sent, this
+ packages may not be compound, so we need to add support for
+ getting ssrc from other types of packages.
+ https://bugzilla.gnome.org/show_bug.cgi?id=750327
+
+2015-06-03 13:14:44 +0200 Jose Antonio Santos Cadenas <santoscadenas@gmail.com>
+
+ * gst/rtpmanager/rtpsession.c:
+ rtpsession: Add support for receiving reduced size rtcp
+ See RFC 5506
+ https://bugzilla.gnome.org/show_bug.cgi?id=750332
+
+2015-06-04 16:09:41 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/audioparsers/gstaacparse.c:
+ aacparse: Add support for channel configurations 11, 12 and 14 and 7 actually has 8 channels
+ ISO/IEC 14496-3:2009/PDAM 4 added 11, 12 and 14.
+
+2015-06-03 08:57:57 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
+
+ * gst/rtp/gstasteriskh263.c:
+ asteriskh263: Un-rank clashing depayloader
+ This depayloader clash with the standard one for H263p. It produces an
+ H263p stream with a modified header. It uses encoding-name that is the
+ same as H263p (H263-1998) though the resulting ES is not decodable or
+ parsable in GStreamer, making it unsuable in dynamic pipeline. This
+ patch unrank this specialized depayloader since it can only be used in
+ custom pipeline.
+ https://bugzilla.gnome.org/show_bug.cgi?id=739935
+
+2015-06-02 18:09:48 +0100 Luis de Bethencourt <luis.bg@samsung.com>
+
+ * gst/goom2k1/gstgoom.c:
+ * gst/goom2k1/gstgoom.h:
+ goom2k1: remove variables not needed anymore
+ https://bugzilla.gnome.org/show_bug.cgi?id=742875
+
+2015-06-02 17:52:46 +0100 Luis de Bethencourt <luis.bg@samsung.com>
+
+ * gst/goom2k1/Makefile.am:
+ * gst/goom2k1/gstaudiovisualizer.c:
+ * gst/goom2k1/gstaudiovisualizer.h:
+ * gst/goom2k1/gstgoom.c:
+ * gst/goom2k1/gstgoom.h:
+ goom2k1: rebase to use the audiovisualizer class
+ Rebase to have goom2k1 using the common GstAudioVisualizer class
+ https://bugzilla.gnome.org/show_bug.cgi?id=742875
+
+2015-06-02 17:29:36 +0100 Luis de Bethencourt <luis.bg@samsung.com>
+
+ * gst/goom/Makefile.am:
+ * gst/goom/gstaudiovisualizer.c:
+ * gst/goom/gstaudiovisualizer.h:
+ * gst/goom/gstgoom.c:
+ * gst/goom/gstgoom.h:
+ goom: rebase to use the audiovisualizer class
+
+2015-06-02 16:27:24 +0200 Edward Hervey <edward@centricular.com>
+
+ * tests/check/elements/aacparse.c:
+ * tests/check/elements/ac3parse.c:
+ * tests/check/elements/apev2mux.c:
+ * tests/check/elements/aspectratiocrop.c:
+ * tests/check/elements/audioamplify.c:
+ * tests/check/elements/audiochebband.c:
+ * tests/check/elements/audiocheblimit.c:
+ * tests/check/elements/audiodynamic.c:
+ * tests/check/elements/audioinvert.c:
+ * tests/check/elements/audiowsincband.c:
+ * tests/check/elements/audiowsinclimit.c:
+ * tests/check/elements/avimux.c:
+ * tests/check/elements/equalizer.c:
+ * tests/check/elements/flacparse.c:
+ * tests/check/elements/id3v2mux.c:
+ * tests/check/elements/jpegdec.c:
+ * tests/check/elements/jpegenc.c:
+ * tests/check/elements/matroskamux.c:
+ * tests/check/elements/mpegaudioparse.c:
+ * tests/check/elements/rganalysis.c:
+ * tests/check/elements/rglimiter.c:
+ * tests/check/elements/rgvolume.c:
+ * tests/check/elements/rtpbin.c:
+ * tests/check/elements/rtpsession.c:
+ * tests/check/elements/spectrum.c:
+ * tests/check/elements/videobox.c:
+ * tests/check/elements/videocrop.c:
+ * tests/check/elements/videofilter.c:
+ * tests/check/elements/wavpackdec.c:
+ * tests/check/elements/wavpackenc.c:
+ * tests/check/elements/wavpackparse.c:
+ * tests/check/elements/y4menc.c:
+ * tests/check/pipelines/simple-launch-lines.c:
+ * tests/check/pipelines/tagschecking.c:
+ * tests/check/pipelines/wavpack.c:
+ check: Use GST_CHECK_MAIN () macro everywhere
+ Makes source code smaller, and ensures we go through common initialization
+ path (like the one that sets up XML unit test output ...)
+
+2015-05-26 14:47:31 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/rtpsession.c:
+ * gst/rtpmanager/rtpsession.h:
+ rtpsession: Only schedule a timer when we actually have to send RTCP
+ Otherwise we will have 10s-100s of thread wakeups in feedback profiles, create
+ RTCP packets, etc. just to suppress them in 99% of the cases (i.e. if no
+ feedback is actually pending and no regular RTCP has to be sent).
+ This improves CPU usage and battery life quite a lot.
+ https://bugzilla.gnome.org/show_bug.cgi?id=746543
+
+2015-05-22 13:44:03 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/rtpsession.c:
+ rtpsession: Remove useless goto
+ https://bugzilla.gnome.org/show_bug.cgi?id=746543
+
+2015-05-21 12:54:47 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * tests/examples/rtp/Makefile.am:
+ * tests/examples/rtp/client-H264-rtx.sh:
+ * tests/examples/rtp/client-rtpaux.c:
+ * tests/examples/rtp/server-VTS-H264-rtx.sh:
+ * tests/examples/rtp/server-rtpaux.c:
+ examples: Set RTP profile to AVPF for rtpaux examples
+ https://bugzilla.gnome.org/show_bug.cgi?id=746543
+
+2015-05-04 16:41:50 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp/gstrtspsrc.c:
+ rtspsrc: Set RTP profile on the rtpsession objects
+ https://bugzilla.gnome.org/show_bug.cgi?id=746543
+
+2015-05-21 14:13:56 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/gstrtpbin.c:
+ * gst/rtpmanager/gstrtpbin.h:
+ rtpbin: Add rtp-profile property for setting the default profile of newly created sessions
+ https://bugzilla.gnome.org/show_bug.cgi?id=746543
+
+2015-05-04 11:51:41 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/rtpsession.c:
+ rtpsession: Only put RRs and full SDES into regular RTCP packets
+ If we may suppress the packet due to the rules of RFC4585 (i.e. when
+ below the t-rr-int), we can send a smaller RTCP packet without RRs
+ and full SDES. In theory we could even send a minimal RTCP packet
+ according to RFC5506, but we don't support that yet.
+ https://bugzilla.gnome.org/show_bug.cgi?id=746543
+
+2015-05-04 13:51:50 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/rtpsession.c:
+ * gst/rtpmanager/rtpsession.h:
+ rtpsession: Keep track of tp/tn and t_rr_last separately
+ Otherwise we can't properly schedule RTCP in feedback profiles as we need to
+ distinguish the time when we last checked for sending RTCP (tp) but might have
+ suppressed it, and the time when we last actually sent a non-early RTCP
+ packet.
+ This together with the other changes should now properly implement RTCP
+ scheduling according to RFC4585, and especially allow us to send feedback
+ packets a lot if needed but only send regular RTCP packets every once in a
+ while.
+ https://bugzilla.gnome.org/show_bug.cgi?id=746543
+
+2015-05-04 11:42:08 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/gstrtpsession.c:
+ * gst/rtpmanager/rtpsession.c:
+ * gst/rtpmanager/rtpsession.h:
+ * gst/rtpmanager/rtpsource.h:
+ * gst/rtpmanager/rtpstats.c:
+ * gst/rtpmanager/rtpstats.h:
+ rtpsession: Add property for selecting RTP profile (AVP/AVPF/etc)
+ And modify our RTCP scheduling algorithm accordingly. We now can send more
+ RTCP packets if needed for feedback, but will throttle full RTCP packets by
+ rtcp-min-interval (t-rr-int from RFC4585).
+ In non-feedback mode, rtcp-min-interval is Tmin from RFC3550, which is
+ statically set to 1s or 0s by RFC4585. Tmin defines how often we should
+ send RTCP packets at most.
+ https://bugzilla.gnome.org/show_bug.cgi?id=746543
+
+2015-05-30 17:41:05 -0400 Olivier Crête <olivier.crete@collabora.com>
+
+ * gst/law/mulaw-decode.c:
+ mulawdec: Let baseclass estimate bitrate
+ This makes playback directly from a file work with the right caps.
+
+2015-05-27 16:31:23 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/udp/gstdynudpsink.c:
+ * gst/udp/gstdynudpsink.h:
+ dynudpsink: keep GCancellable fd around instead of re-creating it constantly
+ And create it only when starting the element.
+
+2015-05-27 15:55:56 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/udp/gstmultiudpsink.c:
+ * gst/udp/gstmultiudpsink.h:
+ udpsink, multiudpsink: keep GCancellable fd around instead of re-creating it constantly
+ Otherwise we constantly create/close event file descriptors,
+ every time we call g_socket_condition_timed_wait() or
+ g_socket_send_message(s)(), i.e. a lot. Which is not
+ particularly good for performance.
+ Can't create GCancellable in ::start() here because it's used
+ in client_new() which may be called via the add-client action
+ signal which may be called before the element is up and running.
+
+2015-05-19 18:13:16 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/udp/gstudpsrc.c:
+ * gst/udp/gstudpsrc.h:
+ udpsrc: keep GCancellable fd around instead of re-creating it constantly
+ Otherwise we constantly create/close event file descriptors,
+ every single time we call g_socket_condition_timed_wait() or
+ g_socket_receive_message(), i.e. twice per packet received!
+ This was not particularly good for performance.
+ Also only create GCancellable on start-up.
+
+2015-05-26 15:33:37 +0100 Luis de Bethencourt <luis.bg@samsung.com>
+
+ * gst/matroska/matroska-read-common.c:
+ matroska: overwritten value assignment
+ curpos is set and immediately after, set again. Remove the redundant
+ assignment.
+ https://bugzilla.gnome.org/show_bug.cgi?id=749909
+
+2015-05-23 13:47:17 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/rtp/gstrtpvrawdepay.c:
+ rtpvrawdepay: don't shadow existing outbuf variable
+ And fix unref of the wrong one which will contain NULL
+ in an error code path.
+
+2015-05-23 13:23:22 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/rtp/gstrtpvrawdepay.c:
+ * gst/rtp/gstrtpvrawdepay.h:
+ rtpvrawdepay: map/unmap output frame only once, not for every input packet
+ Map output buffer after creating it and keep it mapped
+ until we're done with it instead of mapping/unmapping
+ it for every single input buffer.
+
+2015-05-25 08:47:47 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: remove fixme from 2006
+ It has been verified by use over time.
+
+2015-05-23 14:36:41 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: fix reverse playback of fragmented media
+ qtdemux creates a samples array and gets the timestamps for buffers by
+ accumulating their durations. When doing reverse playback of fragments,
+ accumulating samples will lead to wrong timestamps as the timestamps
+ should go decreasing from fragment to fragment and the accumulation
+ will produce wrong results.
+ In this case, when receiving a discont for fragmented reverse playback,
+ the previous samples information should be flushed before new data
+ is processed.
+
+2015-05-23 01:03:18 +0900 Jimmy Ohn <yongjin.ohn@lge.com>
+
+ * gst/multifile/gstsplitfilesrc.c:
+ splitfilesrc: Implement binary search in find_part_for_offset
+ Implement binary search using gst_util_array_binary_search
+ https://bugzilla.gnome.org/show_bug.cgi?id=749690
+
+2015-05-21 13:26:53 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/rtpsession.c:
+ rtpsession: Don't crash if we receive FIR/PLI from a source we don't know
+
+2015-05-21 09:35:58 +0200 Santiago Carot-Nemesio <sancane@gmail.com>
+
+ * gst/rtpmanager/rtpsession.c:
+ rtpsession: Fix collection of statistics
+ Stats should be collected on the media rtp source not in the
+ sender one.
+ https://bugzilla.gnome.org/show_bug.cgi?id=749669
+
+2015-04-20 10:07:30 +0200 Edward Hervey <edward@centricular.com>
+
+ * gst/multifile/gstmultifilesink.c:
+ * gst/multifile/gstmultifilesink.h:
+ multifilesink: Add a new max-duration file switching mode
+ This new mode ensures that files will never exceed a certain duration
+ based on incoming buffer PTS (and duration if present)
+ Note:
+ * You need timestamped buffers (duh). If some of the incoming buffers don't
+ have PTS, then it will just accept them in the current file
+
+2015-04-17 16:18:32 +0200 Edward Hervey <edward@centricular.com>
+
+ * gst/multifile/gstmultifilesink.c:
+ multifilesink: streamline the file-switch code a bit
+ Use the same functions regardless of the mode we are using
+
+2015-04-02 13:35:18 +0100 Edward Hervey <edward@centricular.com>
+
+ * gst/multifile/gstmultifilesink.c:
+ * gst/multifile/gstmultifilesink.h:
+ multifilesink: add "aggregate-gops" property to process GOPs as a whole
+ This property can be used in combination with next-file=max-size
+ (and perhaps a future next-file=max-duration) to make sure that
+ each file part starts cleanly with a key frame and the appropriate headers.
+ In order for this property to work correctly, upstream elements should make
+ sure than any headers that need to be written in a standalone file are:
+ 1) in the streamheader caps field
+ 2) and/or in the stream as one or more buffers marked with GST_BUFFER_FLAG_HEADER
+ that are just before the keyframe buffer
+ This is useful for MPEG-TS/MPEG-PS file segmenting in
+ combination with mpegtsmux or mpegpsmux.
+ Original patch by: Tim-Philipp Müller <tim@centricular.com>
+
+2015-05-20 16:37:22 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp/gstrtspsrc.h:
+ rtspsrc: Use single-include header for the RTSP library
+
+2014-10-24 23:47:21 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/udp/gstdynudpsink.c:
+ * gst/udp/gstmultiudpsink.c:
+ * gst/udp/gstudpsrc.c:
+ udp: don't use soon-to-be-deprecated g_cancellable_reset()
+ From the API documentation: "Note that it is generally not
+ a good idea to reuse an existing cancellable for more
+ operations after it has been cancelled once, as this
+ function might tempt you to do. The recommended practice
+ is to drop the reference to a cancellable after cancelling
+ it, and let it die with the outstanding async operations.
+ You should create a fresh cancellable for further async
+ operations."
+ https://bugzilla.gnome.org/show_bug.cgi?id=739132
+
+2015-05-18 20:13:01 +0200 Stefan Sauer <ensonic@users.sf.net>
+
+ * gst/audiofx/audiochebband.c:
+ * gst/audiofx/audiocheblimit.c:
+ * gst/cutter/gstcutter.c:
+ * gst/equalizer/gstiirequalizernbands.c:
+ * gst/multifile/gstmultifilesink.c:
+ Revert "doc: Workaround gtkdoc issue"
+ This reverts commit 1797c8f8b12d7f4c7a9444c94f34f4d08ec85945.
+ This is fixed by the gtk-doc 1.23 release.
+ <para> cannot contain <refsect2>:
+ http://www.docbook.org/tdg/en/html/para.html
+ http://www.docbook.org/tdg/en/html/refsect2.html
+
+2015-05-18 16:40:21 +0200 Nicola Murino <nicola.murino@gmail.com>
+
+ * gst/rtp/gstrtpg726pay.c:
+ rtpg726pay: fix caps leak
+ https://bugzilla.gnome.org/show_bug.cgi?id=749544
+
+2015-05-18 16:34:13 +0200 Nicola Murino <nicola.murino@gmail.com>
+
+ * gst/rtp/gstrtpg726depay.c:
+ rtpg726depay: don't leak input buffer
+ https://bugzilla.gnome.org/show_bug.cgi?id=749543
+
+2015-05-18 17:38:31 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/rtpsource.c:
+ rtpsource: Queue bad packets instead of dropping them
+ So we can send them out once we found the next, consecutive sequence number in
+ case one is following.
+
+2015-05-18 17:38:14 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/rtpsource.c:
+ rtpsource: Use g_queue_foreach() to unref all buffers in queues
+
+2015-05-18 17:19:31 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/rtpsource.c:
+ rtpsource: Refactor seqnum comparison code a bit
+
+2015-05-18 17:08:53 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/rtpsource.c:
+ rtpsource: Allow sequence number wraparound during probation
+
+2015-05-18 17:07:23 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/rtpsource.c:
+ rtpsource: Make sequence number comparison code more readable
+ ... by using gst_rtp_buffer_compare_seqnum() and signed integers
+ instead of implictly using effects of integer over/underflows.
+
+2015-04-22 18:54:06 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ rtpjitterbuffer: When detecting a huge seqnum gap, wait for 5 consecutive packets before resetting everything
+ It might just be a late retransmission or spurious packet from elsewhere, but
+ resetting everything would mean that we will cause a noticeable hickup. Let's
+ get some confidence first that the sequence numbers changed for whatever
+ reason.
+ https://bugzilla.gnome.org/show_bug.cgi?id=747922
+
+2015-05-16 23:37:06 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
+
+ * gst/audiofx/audiochebband.c:
+ * gst/audiofx/audiocheblimit.c:
+ * gst/cutter/gstcutter.c:
+ * gst/equalizer/gstiirequalizernbands.c:
+ * gst/multifile/gstmultifilesink.c:
+ doc: Workaround gtkdoc issue
+ With gtkdoc 1.22, the XML generator fails when a itemizedlist is
+ followed by a refsect2. Workaround the issue by wrapping the
+ refsect2 into para.
+
+2015-01-23 13:57:40 +0100 Stefan Sauer <ensonic@users.sf.net>
+
+ * gst/isomp4/qtdemux_types.c:
+ qtdemux: avoid wrong warnings on unknown node types
+ Add 'name' and 'mean' fourccs, as we handle them. Right now each use would
+ trigger a warning.
+
+2015-05-08 19:13:00 +0200 Nicola Murino <nicola.murino@gmail.com>
+
+ * gst/rtp/gstrtpg726depay.c:
+ * gst/rtp/gstrtpg726depay.h:
+ rtpg726depay: add block_align to output caps
+ It is needed to correctly negotiate caps with matroskamux
+ and most other muxers.
+ https://bugzilla.gnome.org/show_bug.cgi?id=749129
+
+2015-05-12 13:41:58 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/audiofx/audiofxbasefirfilter.c:
+ audiofxbasefirfilter: Fix time-domain convolution with >1 channels
+ input_samples is the number of frames, but we used it as the number of
+ samples.
+ https://bugzilla.gnome.org/show_bug.cgi?id=747204
+
+2015-05-12 12:13:16 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * ext/vpx/gstvp8enc.c:
+ * ext/vpx/gstvp9enc.c:
+ vp[89]enc: Properly convert between GStreamer and encoder timebase
+ ... by switching numerator and denominator when scaling.
+ https://bugzilla.gnome.org/show_bug.cgi?id=749122
+
+2015-05-11 13:33:26 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * ext/vpx/gstvp8enc.c:
+ * ext/vpx/gstvp9enc.c:
+ vp[89]enc: Don't set timebase from the framerate
+ The framerate very often is just an indication of the ideal framerate, not the
+ actual framerate of the stream. By just using the framerate, we confuse the
+ rate control algorithm algorithm as multiple frames will map to the same PTS
+ or have durations of 0.
+ https://bugzilla.gnome.org/show_bug.cgi?id=749122
+
+2015-05-10 14:21:04 +0200 Mark Nauwelaerts <mnauw@users.sourceforge.net>
+
+ * tests/check/elements/wavpackparse.c:
+ tests: wavpackparse: fix unit test
+ See also https://bugzilla.gnome.org/show_bug.cgi?id=738237
+
+2015-05-10 11:05:00 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * ext/shout2/gstshout2.c:
+ * ext/vpx/gstvp8dec.c:
+ * ext/vpx/gstvp8enc.c:
+ * ext/vpx/gstvp9dec.c:
+ * ext/vpx/gstvp9enc.c:
+ * gst/rtp/gstrtpL16depay.c:
+ * gst/rtp/gstrtpL16pay.c:
+ * gst/rtp/gstrtpL24depay.c:
+ * gst/rtp/gstrtpL24pay.c:
+ * gst/rtp/gstrtpac3pay.c:
+ * gst/rtp/gstrtpamrpay.c:
+ * gst/rtpmanager/gstrtpmux.c:
+ * tests/check/pipelines/wavenc.c:
+ * tests/examples/rtp/client-PCMA.c:
+ * tests/examples/rtp/server-alsasrc-PCMA.c:
+ docs: update example pipelines in element docs
+ Mostly gst-launch -> gst-launch-1.0
+ Use autovideosink/autoaudiosink more often.
+ Sprinkle some converters here and there.
+
+2015-05-09 19:48:55 +0200 Piotr Drąg <piotrdrag@gmail.com>
+
+ * po/POTFILES.in:
+ po: update POTFILES.in
+ https://bugzilla.gnome.org/show_bug.cgi?id=749163
+
+2015-05-10 10:52:18 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/multifile/gstsplitmuxsrc.c:
+ splitmuxsrc: minor error message clean-up
+ Don't put filename in error message shown to user.
+
+2015-05-07 16:25:36 +0200 Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
+
+ * gst/audioparsers/gstflacparse.c:
+ flacparse: fix buffer leak when stored to seektable
+ Fix a leak with the
+ validate.file.playback.change_state_intensive.samples_multimedia_cx_flac_Yesterday_flac
+ scenario.
+ https://bugzilla.gnome.org/show_bug.cgi?id=749072
+
+2015-05-07 17:10:37 +0900 Paul Hyunil <paul.hyunil@lge.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: fix example pipeline in docs
+ The gst-launch script for example launch line to test qtdemux is
+ missing a queue before the decodebins, otherwise the gst-launch-1.0
+ command won't work.
+ https://bugzilla.gnome.org/show_bug.cgi?id=749054
+
+2015-05-07 14:51:45 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/rtpsession.c:
+ Revert "rtpsession: Also report internal sources in on-new-ssrc and on-ssrc-active"
+ This reverts commit d22ec496328e6ba8edbf2d071d5608b2af2831e8.
+ Application code might expect that it only gets external sources on those
+ signals, and get confused by this. If anything we would need to add new
+ signals.
+
+2015-03-25 15:27:34 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/rtpsession.c:
+ rtpsession: Also report internal sources in on-new-ssrc and on-ssrc-active
+ Without this it seems impossible for an application to easily get notified
+ about the internal ssrcs that are created, e.g. sender sources, and also
+ to know when they are active and produce RTCP packets.
+ https://bugzilla.gnome.org/show_bug.cgi?id=746747
+
+2015-05-04 19:26:14 +0200 Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
+
+ * ext/jpeg/gstjpegdec.c:
+ jpegdec: fix frame leaks in handle_frame() implementation
+ handle_frame() is supposed to consume @frame, so if we don't call
+ gst_video_decoder_drop_frame() or gst_video_decoder_finish_frame() we have to
+ release it manually.
+ https://bugzilla.gnome.org/show_bug.cgi?id=748909
+
+2015-05-04 16:50:38 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp/gstrtspsrc.c:
+ rtspsrc: Fix up last commit
+
+2015-05-04 16:46:02 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp/gstrtspsrc.c:
+ rtspsrc: Only do RTX when using a feedback profile
+
+2015-05-04 13:50:31 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/rtpsession.c:
+ rtpsession: The stats min_interval is in seconds, not nanoseconds
+ We have to scale it to compare it against our clock times.
+
+2015-05-04 11:38:27 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/rtpsession.c:
+ rtpsession: Only return TRUE if early feedback was requested already and it's early enough
+
+2015-04-30 15:42:34 +0100 Luis de Bethencourt <luis.bg@samsung.com>
+
+ * gst/matroska/matroska-parse.c:
+ matroska: remove unused property enum items
+
+2015-04-30 12:13:59 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: fix buffer leak on eos in push mode
+ Based on patch by Guillaume Desmottes.
+ scenario: validate.http.playback.seek_with_stop.raw_h264_1_mp4
+ https://bugzilla.gnome.org/show_bug.cgi?id=748617
+
+2015-04-29 19:41:29 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Check for sizes of the rdrf (redirect) atom before accessing the data and use g_strndup() instead of g_strdup()
+ Thanks to Ralph Giles for reporting this.
+
+2015-04-29 15:52:27 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp/gstrtspsrc.c:
+ rtspsrc: Only enable retransmissions if there is retransmission info in the SDP
+ Otherwise we're going to send early RTCP and NACKs in non-feedback sessions
+ too, which will confuse servers.
+ https://bugzilla.gnome.org/show_bug.cgi?id=748627
+
+2015-02-11 18:09:24 +0530 Ravi Kiran K N <ravi.kiran@samsung.com>
+
+ * ext/dv/gstdvdemux.c:
+ dvdemux: extract recording time
+ Extracts the recorded time of the dv file from
+ the metadata and puts it into the global tags.
+ https://bugzilla.gnome.org/show_bug.cgi?id=743657
+
+2015-04-28 15:59:25 +0200 Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
+
+ * gst/matroska/matroska-demux.c:
+ matroskademux: fix seek event leak
+ gst_matroska_demux_handle_seek_event() doesn't consume the
+ event so we have to unref it.
+ https://bugzilla.gnome.org/show_bug.cgi?id=748584
+
+2015-04-28 15:42:49 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/matroska/matroska-demux.c:
+ matroska-demux: Send pending tags when adding a new pad
+ We might've parsed those tags before already and tried to push them to
+ non-existing pads before. Now let's do it for real.
+
+2015-04-23 18:57:37 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/rtpstats.c:
+ rtpstats: Average RTCP packet size is in bytes, bandwidths in bits
+ We need to convert the size to bits for our calculations.
+ https://bugzilla.gnome.org/show_bug.cgi?id=747863
+
+2015-04-23 18:53:39 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/rtpstats.c:
+ rtpstats: Use the same lower limit for RTCP bandwidth to stop sending RTCP everywhere
+ https://bugzilla.gnome.org/show_bug.cgi?id=747863
+
+2015-04-14 18:41:07 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/gstrtpsession.c:
+ * gst/rtpmanager/rtpsession.c:
+ rtpsession: Use bandwidth calculation by default instead of some arbitrary hardcoded value
+ https://bugzilla.gnome.org/show_bug.cgi?id=747863
+
+2015-04-23 18:49:37 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/rtpsession.c:
+ rtpsession: Bandwidth is supposed to be in bits/s, not bytes/s
+ https://bugzilla.gnome.org/show_bug.cgi?id=747863
+
+2015-04-27 16:36:27 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * tests/check/elements/rtpjitterbuffer.c:
+ rtpjitterbuffer: Fix RTX unit test
+ The calculations were a bit off everywhere, even before the changes done
+ recently to the delay for RTX of expected future packets. It only worked by
+ accident, but now the calculations are all correct again. Hopefully.
+
+2015-04-27 11:22:11 +0100 Luis de Bethencourt <luis.bg@samsung.com>
+
+ * gst/avi/gstavimux.c:
+ * gst/debugutils/breakmydata.c:
+ * gst/debugutils/cpureport.c:
+ * gst/debugutils/gstnavseek.c:
+ * gst/debugutils/progressreport.c:
+ * gst/debugutils/rndbuffersize.c:
+ * gst/dtmf/gstrtpdtmfdepay.c:
+ * gst/flv/gstindex.c:
+ * gst/goom/gstgoom.c:
+ * gst/goom2k1/gstgoom.c:
+ * gst/id3demux/gstid3demux.c:
+ * gst/isomp4/gstrtpxqtdepay.c:
+ * gst/law/mulaw-decode.c:
+ * gst/law/mulaw-encode.c:
+ * gst/matroska/matroska-demux.c:
+ * gst/matroska/matroska-mux.c:
+ * gst/matroska/matroska-parse.c:
+ * gst/multifile/gstmultifilesrc.c:
+ * gst/multipart/multipartmux.c:
+ * gst/rtp/gstrtpamrdepay.c:
+ * gst/rtp/gstrtpceltdepay.c:
+ * gst/rtp/gstrtpdvdepay.c:
+ * gst/rtp/gstrtpg723depay.c:
+ * gst/rtp/gstrtpg729depay.c:
+ * gst/rtp/gstrtpmp4vpay.c:
+ * gst/rtp/gstrtppcmadepay.c:
+ * gst/rtp/gstrtppcmudepay.c:
+ * gst/rtp/gstrtpqcelpdepay.c:
+ * gst/rtp/gstrtpspeexdepay.c:
+ * gst/rtpmanager/gstrtpmux.c:
+ * gst/videocrop/gstaspectratiocrop.c:
+ * gst/videocrop/gstvideocrop.c:
+ * gst/videofilter/gstvideotemplate.c:
+ * gst/y4m/gsty4mencode.c:
+ Rename property enums from ARG_ to PROP_
+ Property enum items should be named PROP_ for consistency and readability.
+
+2015-04-25 02:49:58 +0300 Ilya Konstantinov <ilya.konstantinov@gmail.com>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ rtpjitterbuffer: Fix "stats" property docs
+ https://bugzilla.gnome.org/show_bug.cgi?id=748436
+
+2015-04-26 17:54:52 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * Android.mk:
+ * gst/alpha/Makefile.am:
+ * gst/apetag/Makefile.am:
+ * gst/audiofx/Makefile.am:
+ * gst/auparse/Makefile.am:
+ * gst/autodetect/Makefile.am:
+ * gst/avi/Makefile.am:
+ * gst/cutter/Makefile.am:
+ * gst/debugutils/Makefile.am:
+ * gst/deinterlace/Makefile.am:
+ * gst/dtmf/Makefile.am:
+ * gst/effectv/Makefile.am:
+ * gst/equalizer/Makefile.am:
+ * gst/flv/Makefile.am:
+ * gst/flx/Makefile.am:
+ * gst/goom/Makefile.am:
+ * gst/goom2k1/Makefile.am:
+ * gst/icydemux/Makefile.am:
+ * gst/id3demux/Makefile.am:
+ * gst/imagefreeze/Makefile.am:
+ * gst/interleave/Makefile.am:
+ * gst/isomp4/Makefile.am:
+ * gst/law/Makefile.am:
+ * gst/level/Makefile.am:
+ * gst/matroska/Makefile.am:
+ * gst/monoscope/Makefile.am:
+ * gst/multifile/Makefile.am:
+ * gst/multipart/Makefile.am:
+ * gst/replaygain/Makefile.am:
+ * gst/rtp/Makefile.am:
+ * gst/rtpmanager/Makefile.am:
+ * gst/rtsp/Makefile.am:
+ * gst/shapewipe/Makefile.am:
+ * gst/smpte/Makefile.am:
+ * gst/spectrum/Makefile.am:
+ * gst/udp/Makefile.am:
+ * gst/videobox/Makefile.am:
+ * gst/videocrop/Makefile.am:
+ * gst/videofilter/Makefile.am:
+ * gst/videomixer/Makefile.am:
+ * gst/wavenc/Makefile.am:
+ * gst/wavparse/Makefile.am:
+ * gst/y4m/Makefile.am:
+ Remove obsolete Android build cruft
+ This is not needed any longer.
+
+2015-04-24 13:55:08 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/videocrop/gstvideocrop.c:
+ videocrop: print the property values when set
+ Instead of printing the currently used values. The log is meant
+ to show what the properties changed to, not what is being currently
+ used.
+
+2015-04-24 17:01:10 +0100 Luis de Bethencourt <luis.bg@samsung.com>
+
+ * gst/alpha/gstalpha.c:
+ * gst/audiofx/audiokaraoke.c:
+ * gst/deinterlace/gstdeinterlace.c:
+ * gst/multifile/gstmultifilesink.c:
+ * gst/rtp/gstrtpg726depay.c:
+ * gst/rtp/gstrtpg726pay.c:
+ * gst/rtp/gstrtpgstpay.c:
+ * gst/rtp/gstrtph264pay.c:
+ * gst/rtp/gstrtpjpegpay.c:
+ * gst/rtpmanager/gstrtpbin.c:
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ * gst/rtpmanager/gstrtprtxqueue.c:
+ * gst/rtpmanager/gstrtprtxreceive.c:
+ * gst/rtpmanager/gstrtprtxsend.c:
+ * gst/rtpmanager/gstrtpsession.c:
+ * gst/rtpmanager/rtpsession.c:
+ * gst/rtpmanager/rtpsource.c:
+ * gst/rtsp/gstrtspsrc.c:
+ * gst/smpte/gstsmpte.c:
+ * gst/smpte/gstsmptealpha.c:
+ * gst/udp/gstmultiudpsink.c:
+ * gst/udp/gstudpsrc.c:
+ remove unused enum items PROP_LAST
+ This were probably added to the enums due to cargo cult programming and are
+ unused. Removing them.
+
+2015-04-24 00:30:35 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/level/gstlevel.c:
+ level: fix infinite loop for very low interval values
+ https://bugzilla.gnome.org/show_bug.cgi?id=745515
+
+2015-04-23 16:08:54 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/check/Makefile.am:
+ tests: define GST_CHECK_TEST_ENVIRONMENT_BEACON
+ Make sure the test environment is set up.
+ https://bugzilla.gnome.org//show_bug.cgi?id=747624
+
+2015-04-23 16:08:32 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * configure.ac:
+ configure: bump automake requirement to 1.14 and autoconf to 2.69
+ This is only required for builds from git, people can still
+ build tarballs if they only have older autotools.
+ https://bugzilla.gnome.org//show_bug.cgi?id=747624
+
+2015-04-23 16:06:57 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * .gitignore:
+ Update .gitignore
+
+2015-04-23 09:55:59 +0200 Jesper Larsen <knorr.jesper@gmail.com>
+
+ * gst/rtsp/gstrtspsrc.c:
+ rtspsrc: Fix RTCP caps leak
+ https://bugzilla.gnome.org//show_bug.cgi?id=748353
+
+2015-04-22 20:24:20 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ rtpjitterbuffer: When request retransmissions for future packets, consider the packet spacing in the extra delay
+ We now take the maximum of 2*jitter and 0.5*packet_spacing for the extra
+ delay. If jitter is very low, this should prevent unnecessary retransmission
+ requests to some degree.
+ https://bugzilla.gnome.org/show_bug.cgi?id=748041
+
+2015-04-22 19:41:07 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ rtpjitterbuffer: Take a running average of the packet spacings instead of just the latest
+ https://bugzilla.gnome.org/show_bug.cgi?id=748041
+
+2015-04-13 11:20:40 +0200 Miguel París Díaz <mparisdiaz@gmail.com>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ rtpjitterbuffer: Add "rtx-next-seqnum" property
+ If this is set to FALSE, rtpjitterbuffer will not request retransmissions for
+ future packets based on when they are estimated to arrive.
+ See also https://bugzilla.gnome.org/show_bug.cgi?id=748041
+ https://bugzilla.gnome.org/show_bug.cgi?id=739868
+
+2015-04-22 19:29:34 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/gstrtprtxreceive.c:
+ rtxreceive: Put debug output for retransmission requests at the right place
+ Before it was only ever printed once for every time a ssrc was associated with
+ a specific stream.
+
+2015-04-22 18:05:24 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * sys/v4l2/gstv4l2object.c:
+ v4l2: don't add the same interlace mode twice
+ Some drivers modify the interlace mode to progressive, no matter what
+ input you give them, make sure that we don't add the same interlace mode
+ twice.
+
+2015-04-21 16:34:21 +0100 Luis de Bethencourt <luis.bg@samsung.com>
+
+ * gst/equalizer/gstiirequalizer.c:
+ equalizer: fix dynamic changes on bands
+ When we are in passthrough, the transform function doesn't run and if the
+ passthrough check is in this function it will never be deactivated. Fix this by
+ checking directly whenever a gain is changed.
+ Also set the passthrough to TRUE at init because the gains default to 0, so we
+ can passthrough until any gain property is changed.
+ https://bugzilla.gnome.org/show_bug.cgi?id=748068
+
+2015-04-22 10:30:52 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * INSTALL:
+ Remove INSTALL file
+ autotools automatically generate this, and when using different versions
+ for autogen.sh there will always be changes to a file tracked by git.
+
+2015-04-22 10:30:14 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * LICENSE_readme:
+ Remove LICENSE_readme
+ It's completely outdated and just confusing, better if people are
+ forced to look at the actual code in question than trusting this file.
+
+2015-04-21 15:21:33 +0100 Luis de Bethencourt <luis.bg@samsung.com>
+
+ * sys/v4l2/v4l2_calls.c:
+ v4l2: cast unused return to void
+ Quell unchecked return value defect by casting the return value to void and
+ making it explicit it is going to be ignored.
+ CID #206031
+
+2015-04-17 13:08:02 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * ext/vpx/gstvp8dec.c:
+ vp8dec: optimize vpx image to gstbuffer copy when strides match
+ Solving this FIXME. Copy the full plane when strides are the same
+
+2015-04-16 15:11:05 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * ext/vpx/gstvp9dec.c:
+ vp9dec: optimize vpx image to gstbuffer copy when strides match
+ Solving this FIXME. Copy the full plane when strides are the same
+
+2015-04-17 13:32:54 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * gst/audioparsers/gstac3parse.c:
+ ac3parse: fix memory leak
+
+2015-04-17 06:51:46 +0000 Alex O'Konski <alexanderokonski@gmail.com>
+
+ * gst/icydemux/gsticydemux.c:
+ icydemux: Fix segfault if metadata-interval is 0
+ Prevents an extra unref of GstBuffer when passing a non-icy stream through
+ icydemux with metadata-interval set to 0.
+ Reproducible with:
+ gst-launch-1.0 filesrc location=~/testsong.mp3 ! \
+ 'application/x-icy,metadata-interval=(int)0' ! icydemux ! decodebin ! wavenc ! \
+ filesink location=~/testsong.wav
+ https://bugzilla.gnome.org/show_bug.cgi?id=748024
+
+2015-04-17 11:54:23 +0530 Ravi Kiran K N <ravi.kiran@samsung.com>
+
+ * gst/audiofx/audioamplify.c:
+ * gst/audiofx/audiodynamic.c:
+ audiofx: fix typo in example pipelines
+ Fix typo in example pipelines
+ https://bugzilla.gnome.org/show_bug.cgi?id=748022
+
+2015-04-15 18:22:37 +0300 Ilya Konstantinov <ilya.konstantinov@gmail.com>
+
+ * sys/osxaudio/gstosxcoreaudiohal.c:
+ osxaudio: fix spelling in debug message
+ https://bugzilla.gnome.org//show_bug.cgi?id=747936
+
+2015-04-16 16:33:44 +0100 Luis de Bethencourt <luis.bg@samsung.com>
+
+ * tests/examples/equalizer/demo.c:
+ tests: selectable amount of bands in equalizer demo
+ Adding an option in the equalizer demo to make the number of bands selectable.
+
+2015-04-16 15:31:25 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/gstrtprtxsend.c:
+ * gst/rtpmanager/rtpsource.c:
+ rtpsource/rtprtxsend: Also pass correct seqnum-offset and payload to the RTX rtpsource
+ https://bugzilla.gnome.org/show_bug.cgi?id=747394
+
+2015-04-06 12:56:50 +0530 Arun Raghavan <arun@centricular.com>
+
+ * gst/rtpmanager/gstrtprtxsend.c:
+ * gst/rtpmanager/rtpsession.c:
+ rtpsession: Track RTX ssrc caps
+ This is needed so that we can generate SR for RTX stream correctly (the
+ clock rate is required).
+ https://bugzilla.gnome.org/show_bug.cgi?id=747394
+
+2015-04-14 13:56:38 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/gstrtprtxsend.c:
+ rtprtxsend: Copy over timestamps from the orignal buffers to the RTX buffers
+ https://bugzilla.gnome.org/show_bug.cgi?id=747394
+
+2015-04-16 16:01:50 +0100 Luis de Bethencourt <luis.bg@samsung.com>
+
+ * tests/examples/equalizer/demo.c:
+ tests: switch equalizer demo to play from uri
+ Switch the equalizer-nbands demo to use uridecodebin, so users can listen to
+ something more pleasant than white noise. If anybody misses the white noise
+ a uri handler to audiotestsrc can be used.
+
+2015-04-16 11:17:38 +0100 Luis de Bethencourt <luis.bg@samsung.com>
+
+ * tests/examples/equalizer/demo.c:
+ tests: improve readability of equalizer demo
+ Rename variable name to make it more readable, add comments for the three
+ scales created per block, and set the window title.
+
+2015-04-15 17:32:37 +0100 Luis de Bethencourt <luis.bg@samsung.com>
+
+ * tests/examples/equalizer/demo.c:
+ tests: add missing license header for equalizer demo
+
+2015-04-16 13:09:19 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: fix tag list leaks on error paths
+
+2015-04-16 12:23:38 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: fix tag list leak on unknown stream type
+
+2015-04-09 13:19:49 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * tests/check/gst-plugins-good.supp:
+ suppressions: ignore an apparent bug in strtod
+ A buffer overread.
+ https://bugzilla.gnome.org/show_bug.cgi?id=747554
+
+2015-04-15 11:07:27 +0200 George Kiagiadakis <george.kiagiadakis@collabora.com>
+
+ * gst/multifile/gstsplitmuxsink.c:
+ splitmuxsink: do not access property variable without the object lock, use the local stack copy instead
+
+2015-04-14 18:45:44 +0200 George Kiagiadakis <george.kiagiadakis@collabora.com>
+
+ * gst/multifile/gstsplitmuxsink.c:
+ splitmuxsink: add probe on the multiqueue's sink pad instead of the ghost pad
+ because _release_pad tries to release it from ctx->sinkpad, which is
+ multiqueue's sink pad, and currently fails because the probe is not
+ installed there
+
+2015-04-14 19:08:24 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/gstrtprtxreceive.c:
+ * gst/rtpmanager/gstrtprtxsend.c:
+ rtprtx*: Fix typos
+
+2015-04-14 17:24:46 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/rtpsession.c:
+ rtpsession: Not sending early RTCP now because of dithering means we send it with the next compound packet
+
+2015-04-14 16:27:18 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/rtpsession.c:
+ rtpsession: Improve debug output a bit if we can't allow early feedback
+
+2015-04-07 18:00:53 -0400 Olivier Crête <olivier.crete@collabora.com>
+
+ * gst/rtp/gstrtpvp8depay.c:
+ rtpvp8depay: When dropping intra packet, request keyframe
+ https://bugzilla.gnome.org/show_bug.cgi?id=747208
+
+2015-04-13 20:25:00 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/rtpjitterbuffer.c:
+ rtpjitterbuffer: Change resyncing GST_WARNING to GST_INFO
+ This also happens in the very beginning when we receive the first packet, a
+ warning would be very confusing here. In all places where we should warn about
+ this, we would've printed a warning already before.
+
+2015-04-02 13:26:41 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/multifile/gstmultifilesink.c:
+ multifilesink: minor docs improvement
+
+2014-11-06 12:08:03 +0100 Miguel París Díaz <mparisdiaz@gmail.com>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ rtpjitterbuffer: Add "rtx-max-retries" property
+ This property allows to limit the maximum number of retransmission
+ for a specific packet.
+ https://bugzilla.gnome.org/show_bug.cgi?id=739868
+
+2014-11-04 15:00:52 +0100 Miguel París Díaz <mparisdiaz@gmail.com>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ rtpjitterbuffer: Fix expected_dts calc in calculate_expected
+ Right above we consider lost_packet packets, each of them having duration,
+ as lost and triggered their timers immediately. Below we use expected_dts
+ to schedule retransmission or schedule lost timers for the packets that
+ come after expected_dts.
+ As we just triggered lost_packets packets as lost, there's no point in
+ scheduling new timers for them and we can just skip over all lost packets.
+ https://bugzilla.gnome.org/show_bug.cgi?id=739868
+
+2015-03-20 18:21:57 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ rtpjitterbuffer: Make the next output buffer discont after resetting the jitterbuffer
+ Resetting the jitterbuffer drops all packets and other things, and will cause
+ a discontinuity in the packets received by the depayloaders. They should now
+ also flush anything they had pending as the new data will start at a different
+ position.
+ https://bugzilla.gnome.org/show_bug.cgi?id=739868
+
+2015-04-10 09:17:26 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Update segment.start after key-unit seek
+ When doing key uint seek, qtdemux calls gst_qtdemux_adjust_seek
+ to get proper offset. And then this offset is set to
+ segment.position and segment.time in gst_qtdemux_perform_seek but
+ segment.start is not updated.
+ After that, application sends segment query,
+ qtdemux sets start and stop to query using gst_segment_to_stream_time. Due
+ to the wrong value in segment.start, the stop position is smaller than
+ it should.
+ https://bugzilla.gnome.org/show_bug.cgi?id=746822
+
+2015-04-07 16:12:40 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/isomp4/gstqtmux.c:
+ qtmux: remove useless variable do_pts
+ We always write the CTTS in qtmux. Ideally we only want to do that
+ for streams that need DTS, it should be present on the track information
+ rather than be decided based on each buffer
+
+2015-04-07 00:53:35 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/isomp4/gstqtmux.c:
+ qtmux: remove subtraction that makes PTS/DTS start from 0
+ As qt uses durations, it doesn't matter, only the difference
+ between consecutive buffers is important. Also, collectpads
+ already replaces PTS/DTS with the running times for them.
+
+2015-04-06 22:36:43 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * tests/check/elements/qtmux.c:
+ tests: qtmux: add tests to verify it handles non-0 segments
+ Both input streams in this test have a segment.start = 10s, so
+ output should start from 0 anyway.
+ Another test has both starting at non-0 segments, but the running
+ time of both streams should still start from 0
+
+2015-04-06 20:03:19 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * tests/check/elements/qtmux.c:
+ tests: qtmux: simple muxing test
+ Adds a new simple test that verifies that data is properly muxed
+ and preserved. PTS, DTS, duration and caps are verified.
+
+2015-04-10 10:59:26 +0530 Ravi Kiran K N <ravi.kiran@samsung.com>
+
+ * gst/smpte/gstsmpte.h:
+ smpte: remove unused fields
+ Remove the fields - format and fps from smpte
+ as they are unused.
+ https://bugzilla.gnome.org/show_bug.cgi?id=747597
+
+2015-04-10 10:29:47 +0530 Ravi Kiran K N <ravi.kiran@samsung.com>
+
+ * tests/check/Makefile.am:
+ * tests/check/elements/.gitignore:
+ * tests/check/elements/alpha.c:
+ tests: add test suite for alpha
+ Added test suite for alpha element with test cases
+ 1. alpha
+ 2. chroma keying
+ https://bugzilla.gnome.org/show_bug.cgi?id=747595
+
+2015-04-09 12:58:46 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * tests/check/gst-plugins-good.supp:
+ suppressions: add a well known zlib inflate bug
+
+2015-04-09 12:58:26 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * gst/multifile/gstsplitmuxsink.c:
+ splitmuxsink: fix mutex leak
+
+2015-04-09 12:58:04 +1000 Jan Schmidt <jan@centricular.com>
+
+ * tests/check/elements/rtprtx.c:
+ tests: Fix rtprtx test by handling buffer lists
+ Commit #1018aa made rtprtxsend handle buffer lists, breaking
+ the test which probes for buffers, but not buffer lists.
+ Use a utility function to run the probe callback on each buffer
+ in the list in turn and remove any buffers that are dropped.
+
+2015-04-01 11:15:38 +1100 Jan Schmidt <jan@centricular.com>
+
+ * gst/isomp4/gstqtmux.c:
+ * gst/isomp4/gstqtmux.h:
+ isomp4: Refactor various state variables into a mux_mode var
+ Instead of checking various state variables around the muxer,
+ track the current muxing mode in a single 'mux_mode' enum.
+ Add some implementation notes about the different mux modes
+
+2015-04-08 16:40:02 +0200 Edward Hervey <edward@centricular.com>
+
+ * common:
+ * tests/check/Makefile.am:
+ tests: Use AM_TESTS_ENVIRONMENT
+ Needed by the new automake test runner
+
+2015-04-08 11:17:31 +0200 Edward Hervey <bilboed@bilboed.com>
+
+ * gst/rtp/gstrtph263depay.c:
+ rtph263depay: Fix framesize parsing
+ The string passed to the parsing function only contains a framesize, and
+ not <pt> + <framesize>
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726416
+
+2015-03-20 12:18:37 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * gst/wavparse/gstwavparse.c:
+ wavparse: clip chunk size above the valid maximum (0x7fffffff)
+ https://bugzilla.gnome.org/show_bug.cgi?id=722567
+
+2015-03-20 09:07:35 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * gst/wavparse/gstwavparse.c:
+ wavparse: clip chunk length to available data (when known)
+ This prevents silly chunk lengths from possibly overflowing
+ (at least when we know the actual data length).
+ https://bugzilla.gnome.org/show_bug.cgi?id=722567
+
+2015-04-06 20:17:52 -0700 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Don't accumulate segment bases manually
+ gst_segment_do_seek() does that for us already, and doing it twice
+ will break non-flushing seeks in interesting ways. Leftover from 1.0
+ porting.
+ Also copy over segment offset and applied_rate, just in case.
+
+2015-04-06 19:08:10 -0700 Sebastian Dröge <sebastian@centricular.com>
+
+ * tests/icles/test-segment-seeks.c:
+ icles: Fix waiting for segment-done if it happens too fast
+ Sometimes we can get segment-done before we got async-done. If we waited
+ for async-done only, the segment-done would be dropped and we would wait
+ forever for it a few lines below.
+
+2015-04-06 18:55:08 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: stbl_index is valid from 0 onwards
+ It indicates the last sample parsed, not the next one to parse.
+ As it starts in -1, any value from 0 onwards means that it has
+ some valid data.
+
+2015-04-05 20:06:09 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * docs/plugins/gst-plugins-good-plugins-sections.txt:
+ * gst/rtpmanager/gstrtpbin.c:
+ * gst/rtpmanager/gstrtpbin.h:
+ docs: make GstRTCPSync enum show up in rtpbin docs
+ https://bugzilla.gnome.org/show_bug.cgi?id=747358
+
+2015-04-05 11:45:45 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * docs/plugins/gst-plugins-good-plugins-sections.txt:
+ docs: add RTPJitterBufferMode enum to rtpbin docs
+ https://bugzilla.gnome.org/show_bug.cgi?id=747358
+
+2015-04-04 11:55:00 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/multifile/gstmultifilesink.c:
+ multifilesink: close files before posting message
+ Makes sure the files were properly flushed and closed before
+ the message reaches the application
+
+2015-03-30 13:54:23 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * tests/check/elements/multifile.c:
+ tests: multifile: increment tests to check for multifile messages
+ Also verify that the multifilesink file messages are being correctly
+ posted to the bus
+
+2015-03-30 12:51:35 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * tests/check/elements/multifile.c:
+ tests: multifile: handle FIXME for proper checking when test finished
+ Use a GstBus and wait for EOS to finish the tests instead of
+ relying on sleeping
+
+2015-03-30 11:14:09 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/multifile/gstmultifilesink.c:
+ multifilesink: post file message on EOS
+ When multifilesink is operating in any mode other than one file
+ per buffer, the last file created won't have a file message posted
+ as multifilesink doesn't handle the EOS event.
+ This patch fixes it by using the last position to post a file
+ message when EOS is received. This should ensure at least the
+ time related data and the filename are posted to the application
+ or other elements
+ https://bugzilla.gnome.org/show_bug.cgi?id=747000
+
+2015-04-03 18:57:50 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * autogen.sh:
+ * common:
+ Automatic update of common submodule
+ From bc76a8b to c8fb372
+
+2015-04-03 02:08:50 +1100 Jan Schmidt <jan@centricular.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Guard against 64-bit overflow
+ For large-file atoms, guard against overflow in the size field,
+ which could make us jump backward in the file and cause
+ infinite loops.
+
+2015-04-01 23:46:13 +1100 Jan Schmidt <jan@centricular.com>
+
+ * gst/isomp4/gstqtmux.c:
+ * gst/isomp4/gstqtmux.h:
+ * tests/check/elements/qtmux.c:
+ isomp4: Make non-seekable downstream an error in normal mode
+ When not in fast-start or fragmented mode, we need to be able
+ to rewrite the size of the mdat atom, or else the output just
+ won't be playable - the mdat placeholder with size == 0 will
+ cover the rest of the file, including any moov atom we write out.
+ https://bugzilla.gnome.org/show_bug.cgi?id=708808
+
+2014-03-15 15:23:01 +0100 Sebastian Rasmussen <sebras@hotmail.com>
+
+ * gst/rtp/gstrtph263depay.c:
+ * gst/rtp/gstrtph263pay.c:
+ * tests/check/elements/rtp-payloading.c:
+ rtph263pay/-depay: add framesize SDP attribute
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726416
+
+2014-03-15 13:33:56 +0100 Sebastian Rasmussen <sebras@hotmail.com>
+
+ * gst/rtp/gstrtpjpegdepay.c:
+ * gst/rtp/gstrtpjpegpay.c:
+ rtpjpegpay/-depay: Remove incorrectly introduced framesize SDP attribute
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=726415
+
+2015-03-27 21:09:44 +0100 Peter Seiderer <ps.report@gmx.net>
+
+ * sys/v4l2/gstv4l2src.c:
+ * sys/v4l2/gstv4l2src.h:
+ v4l2src: device sequence/offset correction in case of renegotiation
+ The v4l2 device restarts the sequence counter in case of streamoff/streamon,
+ the GST offset values are supposed to increment strictly monotonic, so
+ adjust the sequence counter/offset values in case of caps
+ renegotiation.
+ https://bugzilla.gnome.org/show_bug.cgi?id=745441
+
+2014-11-14 14:18:51 +0100 Peter Seiderer <ps.report@gmx.net>
+
+ * sys/v4l2/gstv4l2src.c:
+ v4l2src: add frame loss detection
+ In case of v4l2 driver filled offset/sequence values add frame
+ loss detection (and write a warning message).
+ Move offset meta data setting and frame loss checking after the
+ timestamp adjustment code to get proper timestamps for the
+ warning message.
+ https://bugzilla.gnome.org/show_bug.cgi?id=745441
+
+2014-11-14 13:48:51 +0100 Peter Seiderer <ps.report@gmx.net>
+
+ * sys/v4l2/gstv4l2bufferpool.c:
+ * sys/v4l2/gstv4l2src.c:
+ v4l2: use v4l2 capture device sequence counter
+ Use the v4l2 capture device sequence counter for
+ setting the GstBuffer offset/offset_end values.
+ https://bugzilla.gnome.org/show_bug.cgi?id=745441
+
+2015-03-30 13:12:35 +0200 Tobias Modschiedler <tobias.modschiedler@cetitec.com>
+
+ * sys/v4l2/gstv4l2bufferpool.c:
+ * sys/v4l2/gstv4l2object.c:
+ v4l2: Ask the driver about its requirements for min_buffers before initiating buffer pool.
+ If propose_allocation() had not been called yet, it was possible that the driver was not asked at all.
+ In buffer pool: Consider minimum number of buffers requested by driver when setting config.
+ https://bugzilla.gnome.org/show_bug.cgi?id=746834
+
+2015-04-01 19:30:27 -0400 Olivier Crête <olivier.crete@collabora.com>
+
+ * gst/rtp/gstrtpvp8depay.c:
+ * gst/rtp/gstrtpvp8depay.h:
+ rtpvp8depay: Parse width/height/profile from keyframes
+ This makes it possible to mux the result into a container
+ such as matroska.
+ https://bugzilla.gnome.org/show_bug.cgi?id=747208
+
+2015-04-01 19:01:49 -0400 Olivier Crête <olivier.crete@collabora.com>
+
+ * ext/vpx/gstvp8enc.c:
+ vp8enc: Expose VP8 width/height limitations in the caps template
+ The VP8 format specification (RFC 6386 section 18.1) specifies
+ that the maximum size is 16383x16383.
+
+2015-03-31 00:20:13 +1100 Jan Schmidt <jan@centricular.com>
+
+ * gst/flv/gstflvdemux.c:
+ flv: When passing seek event upstream, hold a ref.
+ In case upstream can't handle the seek, make sure we
+ keep a ref on the event to attempt to handle it ourselves.
+
+2015-03-26 13:34:53 +0100 Guillaume Desmottes <guillaume.desmottes@collabora.co.uk>
+
+ * gst/matroska/matroska-read-common.c:
+ matroska: fix GValue leaks when parsing tags
+ gst_tag_list_add_value() doesn't consume the GValue we pass to it so there is
+ no point copying it.
+ https://bugzilla.gnome.org/show_bug.cgi?id=746810
+
+2015-03-23 20:58:25 +0100 Mark Nauwelaerts <mnauw@users.sourceforge.net>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: resurrect some flow return handling
+ https://bugzilla.gnome.org/show_bug.cgi?id=744572
+
+2015-03-23 20:57:56 +0100 Mark Nauwelaerts <mnauw@users.sourceforge.net>
+
+ * gst/flv/gstflvdemux.c:
+ flvdemux: resurrect some flow return handling
+ https://bugzilla.gnome.org/show_bug.cgi?id=744572
+
+2015-03-23 20:56:41 +0100 Mark Nauwelaerts <mnauw@users.sourceforge.net>
+
+ * gst/matroska/matroska-demux.c:
+ matroskademux: resurrect some flow return handling
+ https://bugzilla.gnome.org/show_bug.cgi?id=744572
+
+2015-03-27 18:58:31 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/matroska/matroska-demux.c:
+ * gst/matroska/matroska-ids.c:
+ * gst/matroska/matroska-ids.h:
+ * gst/matroska/matroska-read-common.c:
+ matroska: store stream tags and push as updated
+ New tags can be found on different parts of the file, so this patch
+ keeps the stream taglists around for the life cycle of the pad
+ and adds those new tags as found. Then a new tag is found, the
+ pad's is marked with a tags changed flag, making the element push
+ a new tag event on the next check. Before this, we were sending
+ only the newly found tags, as the element was losing its taglist
+ when pushing the event.
+
+2015-03-15 14:40:36 +0100 Ramiro Polla <ramiro.polla@collabora.co.uk>
+
+ * gst/matroska/matroska-demux.c:
+ matroskademux: send global tags incrementally
+ Instead of sending only new tags once they are found, merge the taglist
+ and send them incrementally.
+
+2015-03-14 17:07:05 +0100 Ramiro Polla <ramiro.polla@collabora.co.uk>
+
+ * gst/matroska/matroska-parse.c:
+ * gst/matroska/matroska-read-common.c:
+ * gst/matroska/matroska-read-common.h:
+ matroskaparse: send global tags
+ Global tags are already being read in matroskaparse, but they are not
+ currently being sent.
+ This patch makes global tags get sent incrementally whenever new ones
+ are found.
+ https://bugzilla.gnome.org/show_bug.cgi?id=746242
+
+2015-02-03 10:18:58 +0530 Vineeth T M <vineeth.tm@samsung.com>
+
+ * gst/effectv/gstquark.c:
+ quarktv: fix "planes" property range, a value of 0 is not allowed
+ When planes property is set to 0, the pipeline executes in
+ an infinite loop and never exits. Since planes must never
+ be 0, set the minimum value in the property description
+ to 1.
+ https://bugzilla.gnome.org/show_bug.cgi?id=743906
+
+2015-03-26 13:42:02 -0700 David Schleef <ds@schleef.org>
+
+ * gst/wavparse/gstwavparse.c:
+ wavparse: Fix up comments regarding DTS
+
+2015-03-25 15:11:34 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * gst/rtsp/gstrtspsrc.c:
+ * gst/rtsp/gstrtspsrc.h:
+ rtspsrc: Fix segment in TCP mode
+ It is expected that buffers are time-stamped with running time. Set
+ a segment accordingly. In this case we pick 0,-1 as this is what udpsrc
+ would do. Depayloaders will update the segment to reflect the playback
+ position.
+ https://bugzilla.gnome.org/show_bug.cgi?id=635701
+
+2015-03-26 12:21:25 -0700 David Schleef <ds@schleef.org>
+
+ * gst/wavparse/gstwavparse.c:
+ wavparse: be more strict about typefinding DTS
+ Code now matches comments.
+
+2015-03-25 15:10:53 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * gst/rtsp/gstrtspsrc.c:
+ rtspsrc: Remove useless function
+ This function didn't do anything special, let's not use a function for
+ that.
+
+2015-03-20 13:03:09 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ rtpjitter: Account for rtx_retry in overflow check
+ As rtx_retry is part of the substraction, we need to take it into
+ account, otherwise we may endup with a big value.
+
+2015-03-24 23:15:15 +0000 Julien Isorce <j.isorce@samsung.com>
+
+ * sys/osxvideo/cocoawindow.m:
+ osxvideosink: check for deprecated constants prior to OSX 10.10
+ cocoawindow.m:339:5: error: 'NSOpenGLPFAWindow'
+ is deprecated: first deprecated in OS X 10.9
+ cocoawindow.m:576:7: error: 'NSOpenGLPFAFullScreen'
+ is deprecated: first deprecated in OS X 10.6
+ cocoawindow.m:605:24: error: 'setFullScreen'
+ is deprecated: first deprecated in OS X 10.7
+
+2015-03-24 16:51:12 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
+
+ * gst/rtsp/gstrtspsrc.c:
+ rtspsrc: Fix seeking query
+ The segment start/stop in the query is meant to represent the seekable
+ portion of the stream. It does not match the segment start/stop. Instead
+ export 0 to duration.
+
+2015-03-24 16:18:53 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/flv/gstflvdemux.c:
+ flvdemux: Only set caps once if they don't change
+ Previously we were setting new caps with the same content for every H264 or
+ AAC codec_data we found in the stream, spamming everything and causing
+ renegotiations.
+
+2015-03-24 12:46:19 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/flv/gstflvdemux.c:
+ flvdemux: Don't create AAC/H264 caps without codec_data
+ Instead delay creating the caps until we read the codec_data from the stream,
+ or fail if we get normal data before the codec_data.
+ AAC raw caps and H264 avc caps always need codec_data, setting caps on the pad
+ without them is going to make negotiation fail most of the time. Even if we
+ later set new caps with the codec_data, that's usually going to be too late.
+ https://bugzilla.gnome.org/show_bug.cgi?id=746682
+
+2015-03-24 15:39:22 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/flv/gstflvdemux.c:
+ flvdemux: Fix indention
+
+2015-03-22 13:23:44 +0200 Ilya Konstantinov <ilya.konstantinov@gmail.com>
+
+ * sys/osxaudio/gstosxcoreaudio.h:
+ osxaudio: Fix string format warning on 32-bit
+ UInt32 (Darwin, not C99's uint32_t) is 'unsigned long' on 32-bit
+ platforms.
+
+2015-03-21 17:50:40 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/gstrtpsession.c:
+ rtpsession: Fix another instance of sticky event misordering warnings
+ Make sure that the sync_src pad has caps before the segment event.
+ Otherwise we might get a segment event before caps from the receive
+ RTCP pad, and then later when receiving RTCP packets will set caps.
+ This will results in a sticky event misordering warning
+ This fixes warnings in the rtpaux unit test but also in the
+ rtpaux and rtx examples in tests/examples/rtp
+ https://bugzilla.gnome.org/show_bug.cgi?id=746445
+
+2015-03-21 17:18:47 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/gstrtpsession.c:
+ rtpsession: Also start the RTCP send thread when receiving RTP or RTCP
+ Before we only started it when either:
+ - there is no send RTP stream
+ or
+ - we received an RTP packet for sending
+ This could mean that if the send RTP pads are connected but never receive any
+ RTP data, and the same session is also used for receiving RTP/RTCP, we would
+ never start the RTCP thread and would never send RTCP for the receiving part
+ of the session.
+ This can be reproduced with a pipeline like:
+ gst-launch-1.0 rtpbin name=rtpbin \
+ udpsrc port=5000 ! "application/x-rtp, media=video, clock-rate=90000, encoding-name=H264" ! rtpbin.recv_rtp_sink_0 \
+ udpsrc port=5001 ! rtpbin.recv_rtcp_sink_0 \
+ rtpbin.send_rtcp_src_0 ! fakesink name=rtcp_fakesink silent=false async=false sync=false \
+ rtpbin.recv_rtp_src_0_2553225531_96 ! decodebin ! xvimagesink \
+ fakesrc ! valve drop=true ! rtpbin.send_rtp_sink_0 \
+ rtpbin.send_rtp_src_0 ! fakesink name=rtp_fakesink silent=false async=false sync=false -v
+ Before this change the rtcp_fakesink would never send RTCP for the receiving
+ part of the session (i.e. no receiver reports!), after the change it does.
+ And before and after this change it would send RTCP for the receiving part of
+ the session if the sender part was omitted (the last two lines).
+
+2015-03-19 11:54:12 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/gstrtprtxsend.c:
+ rtprtxsend: Add support for buffer lists
+
+2015-03-19 11:39:38 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/gstrtprtxqueue.c:
+ rtprtxqueue: Implement support for buffer lists
+
+2015-03-18 17:32:36 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * gst/rtsp/gstrtspsrc.c:
+ rtspsrc: Improve trace readability
+ Change the command number into strings.
+
+2015-01-20 10:18:56 +0100 Jan Alexander Steffens (heftig) <jsteffens@make.tv>
+
+ * gst/flv/gstflvdemux.c:
+ * gst/flv/gstflvdemux.h:
+ flvdemux: Don't repeatedly warn after no_more_pads (v2)
+ This can get rather spammy for such a high log level.
+ Only warn once per stream.
+ https://bugzilla.gnome.org/show_bug.cgi?id=746274
+
+2015-03-16 11:23:52 +0100 Jan Alexander Steffens (heftig) <jsteffens@make.tv>
+
+ * gst/flv/gstflvdemux.c:
+ flvdemux: Introduce constant for no-more-pads threshold
+ https://bugzilla.gnome.org/show_bug.cgi?id=746274
+
+2015-01-20 10:18:29 +0100 Jan Alexander Steffens (heftig) <jsteffens@make.tv>
+
+ * gst/flv/gstflvdemux.c:
+ flvdemux: Fix warning to contain 'video'
+ https://bugzilla.gnome.org/show_bug.cgi?id=746274
+
+2015-03-11 21:25:40 +0100 Nicola Murino <nicola.murino@gmail.com>
+
+ * gst/matroska/matroska-demux.c:
+ * gst/matroska/matroska-ids.h:
+ matroskademux: for dts only stream set pts=dts for intra only formats
+ https://bugzilla.gnome.org/show_bug.cgi?id=745192
+
+2015-03-14 16:39:09 +0100 Ramiro Polla <ramiro.polla@collabora.co.uk>
+
+ * gst/matroska/matroska-demux.c:
+ * gst/matroska/matroska-read-common.c:
+ matroskademux: fix sending of tags
+ * Fix critical when new tags are found after segment event has already
+ been sent.
+ * Send global tags before stream tags.
+ * Split sending of tags out of gst_matroska_demux_send_event() into its
+ own function.
+ https://bugzilla.gnome.org/show_bug.cgi?id=745973
+
+2015-03-13 18:26:06 +0000 Ramiro Polla <ramiro.polla@collabora.co.uk>
+
+ * gst/rtsp/gstrtspsrc.c:
+ rtspsrc: properly escape percent sign in documentation
+
+2015-03-13 18:26:44 +0000 Ramiro Polla <ramiro.polla@collabora.co.uk>
+
+ * gst/rtpmanager/gstrtpdtmfmux.c:
+ rtpdtmfmux: properly escape percent sign in documentation
+
+2015-03-13 18:48:03 +0000 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * sys/v4l2/gstv4l2src.c:
+ * sys/v4l2/gstv4l2src.h:
+ v4l2src: delay renegotiation until it is likely buffers were reclaimed
+ Allow renegotiation to happen when buffers have returned after an allocation
+ query. As the allocation query is serialized, all buffers from the pool
+ should have returned and we can stop it to create a new one for the
+ new format
+ https://bugzilla.gnome.org/show_bug.cgi?id=682770
+
+2015-03-13 18:47:55 +0000 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * sys/v4l2/gstv4l2object.c:
+ * sys/v4l2/gstv4l2object.h:
+ v4l2object: add gst_v4l2_object_try_format
+ Similar to set_format but it uses TRY_FMT instead of S_FMT
+ https://bugzilla.gnome.org/show_bug.cgi?id=682770
+
+2015-03-13 18:38:42 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/udp/gstmultiudpsink.c:
+ multiudpsink: fix crash with GST_DEBUG enabled
+ g_inet_socket_address_get_address() does not give
+ us a ref to the address, so don't unref it.
+
+2015-03-12 13:49:56 +0000 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/level/gstlevel.c:
+ level: Don't read over the end of the input memory
+ Previously we advanced the in_data pointer by bps for every channel, and then
+ later again for block_size*bps. This caused us to be one sample further than
+ expected if an input buffer covered two analysis frames. And in the end lead
+ to completely bogus values reported by level.
+ https://bugzilla.gnome.org/show_bug.cgi?id=746065
+
+2015-03-12 01:37:08 +1100 Jan Schmidt <jan@centricular.com>
+
+ * sys/oss/gstossdmabuffer.c:
+ Remove a couple of superfluous trailing semi-colons
+
+2015-03-10 09:31:20 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/alpha/gstalpha.c:
+ * gst/avi/gstavidemux.c:
+ * gst/debugutils/gstpushfilesrc.c:
+ * gst/isomp4/gstisoff.c:
+ * gst/rtpmanager/rtpsession.c:
+ * gst/udp/gstmultiudpsink.c:
+ * sys/osxaudio/gstosxaudioringbuffer.c:
+ * sys/osxaudio/gstosxcoreaudiocommon.c:
+ Fix double semicolons
+
+2015-03-10 15:46:40 +1100 Jan Schmidt <jan@centricular.com>
+
+ * gst/multifile/gstsplitmuxsrc.c:
+ splitmux: Shut down element before downward state change
+ Make sure the state change won't hang trying to shut down pads
+ by making sure the streaming has stopped before chaining up.
+
+2015-03-09 22:58:05 +0200 Ilya Konstantinov <ilya.konstantinov@gmail.com>
+
+ * sys/osxaudio/gstosxcoreaudio.h:
+ osxaudio: stream format is an SPDIF-only field
+
+2015-03-09 22:53:41 +0200 Ilya Konstantinov <ilya.konstantinov@gmail.com>
+
+ * sys/osxaudio/gstosxaudiosrc.h:
+ osxaudio: fix spaces
+
+2015-03-09 22:52:46 +0200 Ilya Konstantinov <ilya.konstantinov@gmail.com>
+
+ * sys/osxaudio/gstosxaudiosrc.h:
+ osxaudio: add type check macro
+
+2015-03-09 22:51:51 +0200 Ilya Konstantinov <ilya.konstantinov@gmail.com>
+
+ * sys/osxaudio/gstosxcoreaudiocommon.c:
+ * sys/osxaudio/gstosxcoreaudiocommon.h:
+ * sys/osxaudio/gstosxcoreaudiohal.c:
+ osxaudio: rename gst_core_audio_set_channels_layout()
+ to gst_core_audio_get_channel_layout().
+
+2015-03-09 22:30:28 +0200 Ilya Konstantinov <ilya.konstantinov@gmail.com>
+
+ * sys/osxaudio/gstosxaudioringbuffer.c:
+ osxaudio: remove unused finalize
+
+2015-03-09 16:25:43 +0000 Luis de Bethencourt <luis.bg@samsung.com>
+
+ * ext/vpx/gstvp9enc.c:
+ vp9enc: remove duplicate declaration of function
+
+2015-03-09 16:22:29 +0000 Luis de Bethencourt <luis.bg@samsung.com>
+
+ * gst/rtp/gstrtph264depay.c:
+ rtph264depay: remove unused value
+ CID #1226474
+
+2015-03-09 16:14:34 +0000 Luis de Bethencourt <luis.bg@samsung.com>
+
+ * gst/rtp/gstrtph263pay.c:
+ rtph263pay: fix leak
+ CID 1212156
+
+2015-03-09 15:58:33 +0000 Luis de Bethencourt <luis.bg@samsung.com>
+
+ * gst/rtp/gstrtph263pay.c:
+ rtph263pay: remove uneeded variable
+ We just need to save the ebit information in case there is an error decoding.
+
+2015-03-09 16:46:02 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * ext/vpx/gstvp8enc.c:
+ * ext/vpx/gstvp9enc.c:
+ vp[89]enc: Reset the encoder when flushing
+ https://bugzilla.gnome.org/show_bug.cgi?id=745704
+
+2015-03-09 12:51:17 +0000 Luis de Bethencourt <luis.bg@samsung.com>
+
+ * gst/matroska/matroska-parse.c:
+ matroska: error mode if can't push buffer
+ If gst_pad_push() fails, inform and return flow error.
+
+2015-03-09 12:13:34 +0000 Luis de Bethencourt <luis.bg@samsung.com>
+
+ * gst/matroska/matroska-parse.c:
+ matroska: unused value
+ Value set in ret will be overwritten just before exiting the function.
+ CID #1226469
+
+2015-03-09 11:10:35 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ rtpjitterbuffer: Drop packets with sequence numbers before the seqnum-base
+ These are outside the expected range of sequence numbers and should be
+ clipped, especially for RTSP they might belong to packets from before a seek
+ or a previous stream in general.
+
+2014-02-27 10:52:16 +0100 Linus Svensson <linussn@axis.com>
+
+ * gst/rtsp/gstrtspsrc.c:
+ rtspsrc: Don't include payload type in the caps for framesize
+ When the sdp media attribute framesize are converted to caps
+ the <payload> should not be included.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=725335
+
+2015-03-09 10:05:14 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ rtpjitterbuffer: Don't forget to unlock the mutex when receiving GAPs in TCP streams
+
+2015-03-09 11:24:58 +0530 Arun Raghavan <arun@centricular.com>
+
+ * ext/pulse/pulsesink.c:
+ pulsesink: Make sure to filter caps in all cases during CAPS query
+ We were skipping the filter step while returning template caps, for
+ example.
+
+2015-03-08 21:15:53 +0000 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * sys/v4l2/gstv4l2bufferpool.c:
+ v4l2bufferpool: Don't update buffer for OUTPUT
+ For output device, we should not update the buffer with flags and
+ timestamp when we dequeue. The information in the v4l2_buffer is not
+ meaningful and it breaks the case where the buffer is rendered at
+ multiple places.
+ https://bugzilla.gnome.org/show_bug.cgi?id=745438
+
+2015-03-08 18:04:34 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * ext/soup/gstsouphttpclientsink.c:
+ souphttpclientsink: Implement cookies property
+
+2015-03-08 18:02:51 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * ext/soup/gstsouphttpclientsink.c:
+ souphttpclientsink: Implement automatic-redirect property
+
+2015-03-08 17:54:07 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * ext/soup/gstsouphttpclientsink.c:
+ souphttpclientsink: Implement proxy support
+ The properties were there before, but not used anywhere.
+
+2015-02-21 20:05:24 +0100 Mark Nauwelaerts <mnauw@users.sourceforge.net>
+
+ * gst/avi/gstavidemux.c:
+ avidemux: resurrect some flow return handling
+
+2015-03-04 10:27:17 +0100 Nicolas Huet <nicolas.huet@parrot.com>
+
+ * gst/audioparsers/gstaacparse.c:
+ aacparse: fix LOAS parsing issue
+ Fix missing index in syncword searching
+ https://bugzilla.gnome.org/show_bug.cgi?id=745585
+
+2015-03-05 17:54:43 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * sys/directsound/gstdirectsoundsink.c:
+ directsoundsink: fix modulo math with ringbuffer parameters
+ To get a multiple of bpf use a subtraction and not an addition
+ https://bugzilla.gnome.org/show_bug.cgi?id=745684
+
+2015-03-07 00:55:47 +1100 Jan Schmidt <jan@centricular.com>
+
+ * gst/multifile/gstsplitmuxsink.c:
+ splitmuxsink: Protect property variables with the object lock.
+ Use the object lock instead of the splitmux lock to protect
+ internal property variables, so they're not locked when
+ switching to a new file.
+ https://bugzilla.gnome.org/show_bug.cgi?id=744420
+
+2015-03-06 11:39:39 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * tests/check/elements/rtpjitterbuffer.c:
+ check: add jitterbuffer unit test
+ See https://bugzilla.gnome.org/show_bug.cgi?id=745539
+
+2015-03-05 09:18:52 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp/gstrtspsrc.c:
+ rtspsrc: Fix handling of interleaved (TCP) streams
+ We need to set up the transport in any case, not just if we have a container
+ stream or a non-interleaved stream. Only if we have an interleaved stream and
+ are retrying, we should not set up the stream again.
+ https://bugzilla.gnome.org/show_bug.cgi?id=745599
+
+2015-03-05 10:00:33 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * ext/vpx/gstvp8dec.c:
+ * ext/vpx/gstvp9dec.c:
+ vp[89]dec: Drop frames that have no output buffer because of errors
+ finish_frame() assumes that there is an output buffer.
+
+2015-03-05 09:56:23 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp/gstrtspsrc.c:
+ rtspsrc: Don't unref caps we don't own
+
+2015-03-05 09:46:17 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp/gstrtspsrc.c:
+ rtspsrc: Push RTCP caps on the RTCP pads
+ Otherwise we will get not-negotiated later from rtpbin, and will never be able
+ to send RTCP packets back to the server. Note that error flow returns from the
+ RTCP pads are ignored, that's why it didn't fail more visible before.
+
+2015-03-05 09:35:32 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp/gstrtspsrc.c:
+ rtspsrc: Make sure to send SEGMENT events on all pads
+
+2015-03-03 16:23:15 +0100 Santiago Carot-Nemesio <sancane@gmail.com>
+
+ * gst/rtpmanager/rtpsession.c:
+ * gst/rtpmanager/rtpsource.c:
+ * gst/rtpmanager/rtpstats.h:
+ rtp: Add Full Intra Request (FIR) packets to statistics
+ https://bugzilla.gnome.org/show_bug.cgi?id=745587
+
+2015-03-03 16:01:53 +0100 Santiago Carot-Nemesio <sancane@gmail.com>
+
+ * gst/rtpmanager/rtpsession.c:
+ * gst/rtpmanager/rtpsource.c:
+ * gst/rtpmanager/rtpstats.h:
+ rtp: Add Packet Loss Indication (PLI) to statistics
+ This is helpful to provide statistics in the format defined in
+ http://w3c.github.io/webrtc-stats/#dictionary-rtcrtpstreamstats-members.
+ https://bugzilla.gnome.org/show_bug.cgi?id=745587
+
+2015-03-03 19:19:50 +0100 Nicola Murino <nicola.murino@gmail.com>
+
+ * gst/matroska/matroska-mux.c:
+ * gst/matroska/matroska-mux.h:
+ matroskamux: Remove duration accumulation logic
+ Duration accumulation can cause rounding errors and generate wrong
+ duration with different buffers that share the same timestamp.
+ https://bugzilla.gnome.org/show_bug.cgi?id=745192
+
+2015-03-03 18:40:16 +0100 Nicola Murino <nicola.murino@gmail.com>
+
+ * gst/matroska/matroska-demux.c:
+ * gst/matroska/matroska-ids.c:
+ * gst/matroska/matroska-ids.h:
+ * gst/matroska/matroska-mux.c:
+ matroska: Add an helper method to get buffer timestamps
+ ... and replace GST_BUFFER_TIMESTAMP that always return PTS with this method
+ that return PTS or DTS based on stream type.
+ https://bugzilla.gnome.org/show_bug.cgi?id=745192
+
+2015-03-04 11:28:12 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/rtpsession.c:
+ rtpsession: Add explanation why we have space for 32 hash tables
+ And also create only one, there's no need yet to create all 32 until
+ we implement RFC2762.
+
+2015-03-04 11:26:57 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/rtpsession.c:
+ * gst/rtpmanager/rtpsession.h:
+ Revert "rtpsession: Do not use an array of maps if they are not being used"
+ This reverts commit 1591adf4cd843d13d8622a30c619425691a84128.
+ https://bugzilla.gnome.org/show_bug.cgi?id=745586#c1:
+ It's the beginning of an implementation of RFC 2762, which is needed for
+ large multicast groups. The implementation is not yet complete but why
+ not leave what is there and implement RFC 2762 instead?
+
+2015-03-04 10:35:12 +0100 Santiago Carot-Nemesio <sancane@gmail.com>
+
+ * gst/rtpmanager/rtpsession.c:
+ * gst/rtpmanager/rtpsession.h:
+ rtpsession: Do not use an array of maps if they are not being used
+ rtpsession declares an array of maps to store srrcs but only the
+ the key 0 is being used. This patch replaces the array of maps
+ for just one map and remove useless parameters in rtpsession
+ https://bugzilla.gnome.org/show_bug.cgi?id=745586
+
+2015-02-27 18:12:09 +0900 Jimmy Ohn <yongjin.ohn@lge.com>
+
+ * gst/avi/gstavidemux.c:
+ avidemux: remove not needed code
+ In gst_avi_demux_handle_src_query, there is not needed code.
+ We already check about stream is vbr or not at the upper line.
+ o, we don't need to check this condition becase stream is not
+ vbr 100% in this case.
+ https://bugzilla.gnome.org/show_bug.cgi?id=745276
+
+2015-03-03 23:25:35 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/icles/gdkpixbufoverlay-test.c:
+ tests: gdkpixbufoverlay-test: replace deprecated function
+ Just avoid using the deprecated function entirely,
+ it's easy enough. Defining the macro is not enough.
+
+2015-03-03 19:04:48 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/icles/gdkpixbufoverlay-test.c:
+ tests: gdkpixbufoverlay-test: fix compilation against newer gdk-pixbuf
+ gdk_pixbuf_new_from_inline() has been deprecated in favour
+ of GResource.
+
+2015-03-03 18:39:15 +0530 Arun Raghavan <arun@centricular.com>
+
+ * sys/osxaudio/gstosxaudiosrc.c:
+ osxaudiosrc: Allow caps renegotiation
+ The ringbuffer does allow renegotiation, so we do not have to report
+ fixed caps once it is acquired (based on a similar patch for the sink
+ side by Ilya Konstantinov <ilya.konstantinov@gmail.com>).
+
+2015-02-21 14:41:08 +0200 Ilya Konstantinov <ilya.konstantinov@gmail.com>
+
+ * sys/osxaudio/gstosxaudiosink.c:
+ osxaudiosink: Allow renegotiating caps
+ Once osxaudiosink's device is open, it fixates on the initial caps and
+ refuses to accept new caps. This is erroneous since the Audio Unit is
+ can accept a new ASBD, and GstAudioRingBuffer supports reconfiguration
+ as well.
+ https://bugzilla.gnome.org/show_bug.cgi?id=743925
+
+2015-03-02 12:04:00 +0100 Gwenole Beauchesne <gwenole.beauchesne@intel.com>
+
+ * sys/v4l2/gstv4l2bufferpool.c:
+ v4l2allocator: fix fd leak in DMABUF import mode.
+ Ensure gst_v4l2_buffer_pool_release_buffer() releases the associated
+ GstV4l2MemoryGroup. In particular, this allows for closing the DMABUF
+ handles prior to instantiating new ones.
+ https://bugzilla.gnome.org/show_bug.cgi?id=745443
+
+2015-03-02 15:06:09 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * ext/vpx/gstvp8enc.c:
+ vp8enc: Use 0 as duration for the EOS "frame"
+
+2015-03-02 15:02:20 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * ext/vpx/gstvp8enc.c:
+ * ext/vpx/gstvp8enc.h:
+ * ext/vpx/gstvp9enc.c:
+ * ext/vpx/gstvp9enc.h:
+ vp{8,9}enc: Tell the encoder about actual timestamps and durations of frames
+ ... instead of just counting frames. The values are supposed to be in timebase
+ units, not frame units. This fixes various quality problems with VP8/VP9
+ encoding and in general makes the encoder behave better.
+ Thanks to Nirbheek Chauhan for noticing this bug.
+
+2015-03-01 13:56:17 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
+
+ * ext/vpx/gstvp8dec.c:
+ * ext/vpx/gstvp9dec.c:
+ vpxdec: Fix calculation of width in bytes
+ Right now we only support I420, but vpx seems to support more formats.
+ This will prevent hard to find bug in the future.
+
+2015-03-01 13:52:50 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
+
+ * ext/vpx/gstvp8dec.c:
+ * ext/vpx/gstvp9dec.c:
+ vpxdec: Don't memcpy in frame map failed
+ This avoid a crash if mapping the frame failed.
+
+2015-03-01 13:48:45 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
+
+ * sys/v4l2/gstv4l2bufferpool.c:
+ v4l2bufferpool: Add missing break
+ This is cosmetic change.
+
+2015-03-01 13:46:18 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
+
+ * sys/v4l2/gstv4l2bufferpool.c:
+ * sys/v4l2/gstv4l2bufferpool.h:
+ v4l2: Workaround driver not setting field correctly
+ As it's very common, handle driver not setting field in buffers
+ by using the field value from the format. This workaround a long time
+ bug in UVC driver. For even buggier driver, we simply assume
+ progressive as before. We also only warn once, to avoid spamming.
+
+2015-02-28 18:10:06 +0100 Matej Knopp <matej.knopp@gmail.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: fix key unit seek
+ Unlike many other seek flags, the KEY_UNIT seek
+ flag is not copied over into the GstSegment,
+ since it's only relevant for the seek itself,
+ so we need to pass it explicitly to the seek
+ handler here.
+ https://bugzilla.gnome.org/show_bug.cgi?id=745339
+
+2015-02-27 09:38:01 +0100 Edward Hervey <bilboed@bilboed.com>
+
+ * docs/plugins/gst-plugins-good-plugins.args:
+ * docs/plugins/gst-plugins-good-plugins.hierarchy:
+ * docs/plugins/gst-plugins-good-plugins.interfaces:
+ * docs/plugins/inspect/plugin-isomp4.xml:
+ * docs/plugins/inspect/plugin-multifile.xml:
+ * docs/plugins/inspect/plugin-rtp.xml:
+ * docs/plugins/inspect/plugin-rtpmanager.xml:
+ * docs/plugins/inspect/plugin-shout2send.xml:
+ * docs/plugins/inspect/plugin-video4linux2.xml:
+ * docs/plugins/inspect/plugin-videofilter.xml:
+ * docs/plugins/inspect/plugin-wavenc.xml:
+ docs/plugins: Updates
+
+2015-02-26 23:41:47 +0100 Nicola Murino <nicola.murino@gmail.com>
+
+ * gst/matroska/matroska-demux.c:
+ * gst/matroska/matroska-mux.c:
+ matroskamux/demux: initialize dts_only
+ https://bugzilla.gnome.org/show_bug.cgi?id=745192
+
+2015-02-26 23:28:11 +0100 Nicola Murino <nicola.murino@gmail.com>
+
+ * gst/matroska/matroska-mux.c:
+ matroskamux: store DTS for V_MS/VFW/FOURCC streams
+ https://bugzilla.gnome.org/show_bug.cgi?id=745192
+
+2015-02-26 19:48:33 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/multifile/gstsplitmuxsink.c:
+ * gst/multifile/gstsplitmuxsrc.c:
+ multifile: attempt to fix docs build issue on build bot
+
+2015-02-27 00:41:46 +0530 Arun Raghavan <git@arunraghavan.net>
+
+ * gst/interleave/interleave.c:
+ interleave: Drop custom latency query handling
+ This is implemented by the default query handler now.
+
+2015-02-27 00:40:05 +0530 Arun Raghavan <git@arunraghavan.net>
+
+ * gst/videomixer/videomixer2.c:
+ videomixer: Drop custom latency querying logic
+ This is now implemented in the default latency query handler.
+
+2015-02-26 16:10:41 +0100 Sebastian Rasmussen <sebrn@axis.com>
+
+ * gst/rtp/gstrtpvorbispay.c:
+ rtpvorbispay: fix payloader description and author e-mail
+ https://bugzilla.gnome.org/show_bug.cgi?id=745226
+
+2014-09-05 16:34:26 +0200 Aurélien Zanelli <aurelien.zanelli@parrot.com>
+
+ * sys/v4l2/gstv4l2object.c:
+ * sys/v4l2/gstv4l2sink.c:
+ v4l2: query crop configuration after each call of S_CROP
+ S_CROP ioctl is write-only and the device can adjust crop rectangle so
+ we query back the crop configuration after each S_CROP to know what has
+ been done.
+ https://bugzilla.gnome.org/show_bug.cgi?id=736133
+
+2015-02-26 02:12:18 +0100 Matej Knopp <matej.knopp@gmail.com>
+
+ * gst/matroska/matroska-demux.c:
+ * gst/matroska/matroska-ids.h:
+ matroskademux: V_MS/VFW/FOURCC streams have DTS instead of PTS
+ When such stream is present demuxer should set DTS on buffers instead
+ of PTS. This is consistent with how VLC and libav/ffmpeg handle VFW
+ streams.
+ Sample file
+ https://s3.amazonaws.com/MatejK/Samples/Matroska-VFW-DTS-Only.mkv
+ https://bugzilla.gnome.org/show_bug.cgi?id=745192
+
+2015-02-25 16:45:11 -0800 Aleix Conchillo Flaqué <aleix@oblong.com>
+
+ * sys/v4l2/gstv4l2bufferpool.c:
+ v4l2bufferpool: Check corruption flag on the right buffer
+ We where checking the buffer we are copying to instead of the buffer we
+ are copying from.
+ https://bugzilla.gnome.org/show_bug.cgi?id=740040
+
+2015-01-19 15:29:24 +0100 Aurélien Zanelli <aurelien.zanelli@parrot.com>
+
+ * sys/v4l2/gstv4l2object.c:
+ v4l2object: set colorspace in caps for capture devices
+ This information is set by the driver for a capture device, and so could
+ be forwarded to pipeline by setting the colorimetry in caps.
+ https://bugzilla.gnome.org/show_bug.cgi?id=743186
+
+2014-10-06 17:30:06 +0200 Aurélien Zanelli <aurelien.zanelli@parrot.com>
+
+ * sys/v4l2/gstv4l2allocator.c:
+ * sys/v4l2/gstv4l2bufferpool.c:
+ * sys/v4l2/gstv4l2object.c:
+ * sys/v4l2/gstv4l2object.h:
+ v4l2bufferpool: fix import_userptr() in single-planar API when n_planes > 1
+ In the V4L2 single-planar API, when format is semi-planar/planar,
+ drivers expect the planes to be contiguous in memory.
+ So this commit change the way we handle semi-planar/planar format
+ (n_planes > 1) when we use the single-planar API (group->n_mem == 1).
+ To check that planes are contiguous and have expected size, ie: no
+ padding. We test the fact that plane 'i' start address + plane 'i'
+ expected size equals to plane 'i + 1' start address. If not, we return
+ in error.
+ Math are done in bufferpool rather than in allocator because the
+ former is aware of video info.
+ https://bugzilla.gnome.org/show_bug.cgi?id=738013
+
+2015-01-23 10:15:46 +0100 Aurélien Zanelli <aurelien.zanelli@parrot.com>
+
+ * sys/v4l2/gstv4l2allocator.c:
+ * sys/v4l2/gstv4l2allocator.h:
+ * sys/v4l2/gstv4l2bufferpool.c:
+ v4l2allocator: let bufferpool calculate image size when importing userptr
+ Offset are relative to the buffer and there is no guarantee substracting
+ them will give us the plane size. So we let bufferpool make the math as
+ it is more aware of video info than allocator and pass a size array to
+ allocator import function.
+ Pointed out by Nicolas Dufresne <nicolas.dufresne@collabora.com>
+ https://bugzilla.gnome.org/show_bug.cgi?id=738013
+
+2014-12-11 16:13:15 +0100 Philippe De Muyter <phdm@macqel.be>
+
+ * sys/v4l2/gstv4l2object.c:
+ v4l2object: recognize and distinguish all bayer arrangements
+ Up to now, v4l2src recognized only "bggr" amongst the bayer arrangements.
+ Recognize now also the "rggb", "gbrg" and "grbg" arrangements.
+ https://bugzilla.gnome.org/show_bug.cgi?id=742363
+
+2015-01-15 16:11:53 +0100 Aurélien Zanelli <aurelien.zanelli@parrot.com>
+
+ * sys/v4l2/gstv4l2bufferpool.c:
+ v4l2bufferpool: set v4l2_buffer.field when queuing buffer in an output device
+ According to the current specification, application must set this field
+ for an output device.
+ https://bugzilla.gnome.org/show_bug.cgi?id=743013
+
+2015-02-24 05:57:24 +0200 Ilya Konstantinov <ilya.konstantinov@gmail.com>
+
+ * sys/osxaudio/gstosxaudiosrc.c:
+ * sys/osxaudio/gstosxcoreaudio.c:
+ * sys/osxaudio/gstosxcoreaudio.h:
+ * sys/osxaudio/gstosxcoreaudiocommon.c:
+ * sys/osxaudio/gstosxcoreaudiocommon.h:
+ osxaudiosrc: iOS resampling causes stuttering
+ Fixes stuttering audio when iOS AU is resampling. To make AU resample,
+ one has to request a rate that differs from AVAudioSession's
+ sampleRate. The resampling itself is not the culprit, but rather our
+ API misuse.
+ AudioUnitRender modifies the mDataByteSize members with the
+ actual read bytes count. Therefore, they must be reinitialized
+ before each AudioUnitRender. (The buffers themselves can be
+ preallocated.)
+ The "stutter" was caused by one AudioUnitRender making the buffer
+ too small for other AudioUnitRender invocations, making them fail
+ with -50 (paramErr). By way of luck, when AU didn't resample, all
+ AudioUnitRender invocations read the same number of bytes.
+ (This patch addresses some non-interleaved audio concerns, but
+ at this moment the elements do not support non-interleaved audio
+ and non-interleaved is untested.)
+ https://bugzilla.gnome.org/show_bug.cgi?id=744922
+
+2015-02-22 01:49:52 +0100 Krzysztof Kotlenga <pocek@users.sf.net>
+
+ * gst/rtsp/gstrtspsrc.c:
+ rtspsrc: improve error message when unauthorized
+ Make use of NOT_AUTHORIZED error code instead of falling back to generic
+ READ error.
+ https://bugzilla.gnome.org/show_bug.cgi?id=601733
+
+2015-02-23 20:06:25 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * sys/ximage/ximageutil.c:
+ ximagesrc: remove pointless g_return_val_if_fail()
+ ximage won't ever be NULL here because the dispose
+ function is called via ximage->dispose().
+
+2015-02-23 19:40:25 +0100 Thibault Saunier <tsaunier@gnome.org>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: All segment resulting from a seek should have the same seqnum
+ https://bugzilla.gnome.org/show_bug.cgi?id=744983
+
+2015-02-19 23:12:31 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
+
+ * sys/v4l2/gstv4l2bufferpool.c:
+ * sys/v4l2/gstv4l2bufferpool.h:
+ * sys/v4l2/gstv4l2object.c:
+ v4l2: Enable copy when no known allocation params
+ When there is no allocation parameters in the query, enable copy
+ threshold. When this threshold is reached, the buffer pool will start
+ copying when the pool reaches a critical level. If the driver supports
+ CREATE_BUFS, this will be used instead.
+
+2015-02-19 23:08:34 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
+
+ * sys/v4l2/gstv4l2bufferpool.c:
+ v4l2bufferpool: Update allocator flags
+ When we hit emulated formats, we disable CREATE_BUFS since libv4l2
+ cope very badly with it. Also clear the allocator flags so we will
+ never try to allocate more buffers. This fixes failure when the copy
+ threshold is reached as we where calling CREATE_BUFS, which lead to
+ libv4l2 instability.
+
+2015-02-19 23:07:23 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
+
+ * sys/v4l2/gstv4l2bufferpool.c:
+ v4l2bufferpool: Use specific debug category
+ The pool has grown enough that it is now handy to seperate v4l2object
+ trace from v4l2bufferpool trace.
+
+2015-02-19 14:29:02 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * gst/rtp/gstrtpvp8pay.c:
+ rtpvp8pay: default encoding name to VP8
+ https://bugzilla.gnome.org/show_bug.cgi?id=737810
+
+2015-02-19 14:06:51 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * gst/rtp/gstrtpvp8pay.c:
+ rtpvp8pay: make caps writable before truncating them
+ https://bugzilla.gnome.org/show_bug.cgi?id=737810
+
+2015-02-05 10:29:26 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * gst/rtp/gstrtpvp8pay.c:
+ rtpvp8pay: negotiate encoding name
+ Chrome uses a different one than gstreamer.
+ https://bugzilla.gnome.org/show_bug.cgi?id=737810
+
+2015-02-19 12:35:07 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/gstrtpsession.c:
+ rtpsession: Send initial events on sync_rtcp pad when using RTP/RTCP muxing
+ Otherwise we will just send buffers on the pad without any events beforehand
+ and will get g_warnings() about that.
+
+2015-02-19 11:20:51 +0000 Luis de Bethencourt <luis.bg@samsung.com>
+
+ * ext/jack/gstjackaudiosrc.c:
+ jack: case missing break statement
+ commit b1098c2ea5eabea7af08ce51d22b867eaed2bbe2 added a new case in
+ gst_jack_audio_src_get_property() but forgot to add the break statement to it.
+
+2015-02-18 19:18:00 +0000 Luis de Bethencourt <luis.bg@samsung.com>
+
+ * sys/v4l2/v4l2_calls.c:
+ Revert "v4l2: fraction is reversed"
+ This reverts commit b91fe36644b15ae070d72b9e8a9c7087e82aef12.
+
+2015-02-18 17:49:29 +0000 Luis de Bethencourt <luis.bg@samsung.com>
+
+ * sys/v4l2/v4l2_calls.c:
+ v4l2: fraction is reversed
+ In the fraction 1 / 2. 1 is the numerator and 2 is the denominator.
+ The arguments of fraction gst_value_set_fractions() are value,
+ numerator and denominator.
+ Also, gst_value_set_fraction() fails if denominator is 0 for obvious
+ reasons.
+
+2015-02-17 20:26:55 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
+
+ * sys/v4l2/gstv4l2bufferpool.c:
+ v4l2pool: Deactivate other pool
+ When importing buffers from a downstream pool, we need to deactivate
+ that pool to ensure it will be usable again later. Relying on the
+ refcount to reach zero does not work, since elements like xvimagesink
+ keeps a reference on their proposed pool.
+
+2015-02-18 10:10:53 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/isomp4/gstqtmux.c:
+ * gst/isomp4/qtdemux.c:
+ qtmux: remove not needed condition
+ gst_buffer_replace can handle NULL inputs by itself
+
+2015-02-18 09:40:14 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: prefer the tfdt timestamp over the buffer's that is less accurate
+ The tfdt should be more accurate as the buffer timestamp is provided
+ by the fragmented format manifest and it might just be an approximation.
+
+2015-02-17 16:57:55 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ rtpjitterbuffer: When resetting the jitterbuffer because of packet discont, don't flush sticky events
+ We will otherwise flush away STREAM_START, CAPS or SEGMENT events and will
+ confuse downstream with buffers that come before such events.
+
+2015-02-17 12:20:57 +0100 hark <hark@puscii.nl>
+
+ * ext/jack/gstjackaudiosink.c:
+ * ext/jack/gstjackaudiosink.h:
+ * ext/jack/gstjackaudiosrc.c:
+ * ext/jack/gstjackaudiosrc.h:
+ jack: Add property port-pattern to specify which JACK ports to connect to
+ https://bugzilla.gnome.org/show_bug.cgi?id=690719
+
+2015-02-17 12:31:06 +0100 Edward Hervey <bilboed@bilboed.com>
+
+ * gst/isomp4/gstisoff.c:
+ * gst/isomp4/gstisoff.h:
+ * gst/isomp4/qtdemux.c:
+ isomp4: Redefine gst_isoff_ symbols to gst_isoff_qt_
+ We need different symbol names, because these symbols are also present
+ in the fragmented plugin ... which will cause conflicts when doing
+ static linking
+
+2015-02-16 14:31:05 +0000 Luis de Bethencourt <luis.bg@samsung.com>
+
+ * gst/goom2k1/lines.c:
+ goom2k1: use fractional part of float division
+
+2015-02-16 13:59:14 +0000 Luis de Bethencourt <luis.bg@samsung.com>
+
+ * gst/multifile/gstsplitmuxsink.c:
+ splitmuxsin: remove dead code
+ Every instance of goto beach has buf_info equal NULL. Don't check
+ for a condition that never happens.
+ CID #1268399
+
+2015-02-15 21:45:24 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
+
+ * tests/check/elements/splitmux.c:
+ splitmux-test: Parse error message
+ The test had a function to print the error, but was not parsing it.
+ This was causing warning about dbg_info being used uninitialized. If
+ the test was testing any errors, this would have crashed.
+
+2015-02-15 21:34:28 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
+
+ * gst/spectrum/gstspectrum.c:
+ spectrum: Fix min and max for bands property
+ The number of FFTs is calculated with the following formula:
+ guint nfft = 2 * bands - 2;
+ nfft is passed to gst_fft_f32_new() as the len argument and is of type
+ unsigned integer. This method required that len is at leas 1, then
+ maximum G_MAXINT, as other values would be negative. If we extrapolate
+ from the formula above it means we need "bands" to be between 2 and
+ ((guint)G_MAXINT + 2) / 2).
+ https://bugzilla.gnome.org/show_bug.cgi?id=744213
+
+2015-02-15 15:51:55 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
+
+ * sys/v4l2/gstv4l2allocator.c:
+ v4l2allocator: Fix freeing of shared memory
+ When memory (that has been shared using gst_memory_share()) are freed,
+ the memory (or the DMABUF FD) should not bee freed. These memories have
+ a parent. This also removes the extra _v4l2mem_free function and avoid
+ calling close twice on the DMABUF FD.
+ https://bugzilla.gnome.org/show_bug.cgi?id=744573
+
+2015-02-14 11:11:30 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: do not use sparse streams in push-based seeking
+ Using the sparse streams can make the push-based seeking return
+ too far in the stream. It also can lead to issues as the
+ sparse streams will be ignored when restarting playback and,
+ if the sparse stream is the one that has the earliest sample,
+ it will confuse qtdemux's offsets as one stream will have
+ an earlier offset than the demuxer's one which might lead to
+ early EOS.
+ https://bugzilla.gnome.org/show_bug.cgi?id=742661
+
+2015-02-13 19:43:16 +0900 Jimmy Ohn <yongjin.ohn@lge.com>
+
+ * ext/pulse/pulsesink.c:
+ pulsesink: Enhance code readability in pulsesink_query
+ In pulsesink_query function, we use a switch for the query
+ type. In the CAPS case, there is no 'break', instead we
+ return right away. Use a break and return at the end of
+ the function instead for better code readability.
+ https://bugzilla.gnome.org/show_bug.cgi?id=744461
+
+2015-02-13 20:40:48 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/multifile/gstsplitmuxsink.c:
+ splitmuxsink: flag as sink from the start
+
+2015-02-11 15:30:44 +0100 Philippe Normand <philn@igalia.com>
+
+ * gst/isomp4/Makefile.am:
+ * gst/isomp4/fourcc.h:
+ * gst/isomp4/gstisoff.c:
+ * gst/isomp4/gstisoff.h:
+ * gst/isomp4/qtdemux.c:
+ * gst/isomp4/qtdemux.h:
+ qtdemux: Initial 'sidx' atom parsing support
+ Parse the 'sidx' atom and update the total duration according to the
+ parser result. The isoff parser code is imported from
+ gst-plugins-bad's dashdemux and a gst_isoff_sidx_parser_add_data()
+ function was factored out of the gst_isoff_sidx_parser_add_buffer()
+ function.
+ https://bugzilla.gnome.org/show_bug.cgi?id=743578
+
+2015-02-11 05:06:45 +1100 Jan Schmidt <jan@centricular.com>
+
+ * gst/flv/Makefile.am:
+ * gst/flv/gstflvdemux.c:
+ flvdemux: Use gst_video_guess_framerate()
+ Use gst_video_guess_framerate() from libgstvideo to guess
+ sensible common framerates where possible from the
+ floating point fps in the stream.
+
+2015-02-11 13:53:02 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * ext/raw1394/gstdv1394src.c:
+ * ext/vpx/gstvp8enc.c:
+ * ext/vpx/gstvp9enc.c:
+ * gst/interleave/interleave.c:
+ * gst/rtsp/gstrtpdec.c:
+ * gst/videomixer/videomixer2.c:
+ Improve and fix LATENCY query handling
+ This now follows the design docs everywhere, especially the maximum latency
+ handling.
+ https://bugzilla.gnome.org/show_bug.cgi?id=744106
+
+2015-02-11 10:29:55 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/rtpsession.c:
+ rtpsession: Handle first RTCP packet and early feedback correctly
+ According to RFC 4585 section 3.5.3 step 1 we are not allowed to send
+ an early RTCP packet for the very first one. It must be a regular one.
+ Also make sure to not use last_rtcp_send_time in any calculations until
+ we actually sent an RTCP packet already. In specific this means that we
+ must not use it for forward reconsideration of the current RTCP send time.
+ Instead we don't do any forward reconsideration for the first RTCP packet.
+
+2015-02-10 18:53:53 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtp/gstrtph263depay.c:
+ rtph263depay: fix compilation with gcc 5.0
+
+2015-02-10 16:00:07 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/multifile/gstsplitmuxsink.c:
+ splitmuxsink: fix example pipeline properly
+ x264enc might not have a max-key-int property, but it
+ has a key-int-max property...
+
+2015-02-10 14:57:55 +0000 Luis de Bethencourt <luis.bg@samsung.com>
+
+ * gst/multifile/gstsplitmuxsrc.c:
+ splitmux: fix typo
+
+2015-02-10 14:56:23 +0000 Luis de Bethencourt <luis.bg@samsung.com>
+
+ * gst/multifile/gstsplitmuxsink.c:
+ splitmux: update example pipeline
+ Element x264enc doesn't have a max-key-int property
+
+2015-02-10 13:29:32 +0000 Luis de Bethencourt <luis.bg@samsung.com>
+
+ * gst/multifile/gstsplitmuxsink.c:
+ splitmux: fix memory leak
+ If execution goes to the beach in line 981, buf_info goes out of scope without
+ the memory being free'd. Handle this case.
+ CID #1268403
+
+2015-02-08 12:03:10 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/rtsp/gstrtspsrc.c:
+ rtspsrc: fix awkward if clause
+
+2015-02-07 01:41:49 +1100 Jan Schmidt <jan@centricular.com>
+
+ * gst/multifile/gstsplitmuxpartreader.c:
+ * gst/multifile/gstsplitmuxsink.c:
+ * tests/check/elements/splitmux.c:
+ splitmux: Add unit test for file splitting
+ Add a unit test for file splitting, and fix the leaks in the
+ splitmuxsink it found
+
+2015-02-06 14:43:22 +0000 Luis de Bethencourt <luis.bg@samsung.com>
+
+ * gst/wavparse/gstwavparse.c:
+ wavparse: fix which stop variable is used in assignment
+ Assignment is done to variable segment.stop when the intention was to assign to
+ local variable stop. Instead of overwriting it, the value is now clamped and
+ segment.stop is set to it soon after.
+ CID #1265773
+
+2015-02-07 00:19:36 +1100 Jan Schmidt <jan@centricular.com>
+
+ * gst/multifile/gstsplitmuxpartreader.c:
+ * gst/multifile/gstsplitmuxsrc.c:
+ * tests/check/elements/splitmux.c:
+ splitmux: Fix memory leaks until the test valgrinds clean
+
+2015-02-06 06:42:17 +1100 Jan Schmidt <jan@centricular.com>
+
+ * gst/multifile/gstsplitmuxpartreader.c:
+ splitmux: Handle early EOS during part preparation
+ Handle the case where a short file reaches EOS while we're still
+ waiting for no-more-pads, and make sure we continue to the internal
+ READY state for real playback to work properly later.
+
+2015-02-06 05:03:19 +1100 Jan Schmidt <jan@centricular.com>
+
+ * tests/files/splitvideo00.ogg:
+ * tests/files/splitvideo01.ogg:
+ * tests/files/splitvideo02.ogg:
+ tests: Change splitmux test video files
+ Avoid test failure by changing the stored video resolution
+ from 80x60 to 80x64, which needs bug 741030 to be fixed.
+
+2014-08-01 00:07:53 +1000 Jan Schmidt <jan@centricular.com>
+
+ * docs/plugins/Makefile.am:
+ * docs/plugins/gst-plugins-good-plugins-docs.sgml:
+ * docs/plugins/gst-plugins-good-plugins-sections.txt:
+ * docs/plugins/gst-plugins-good-plugins.hierarchy:
+ * docs/plugins/gst-plugins-good-plugins.interfaces:
+ * gst/multifile/Makefile.am:
+ * gst/multifile/gstmultifile.c:
+ * gst/multifile/gstsplitfilesrc.c:
+ * gst/multifile/gstsplitmuxpartreader.c:
+ * gst/multifile/gstsplitmuxpartreader.h:
+ * gst/multifile/gstsplitmuxsink.c:
+ * gst/multifile/gstsplitmuxsink.h:
+ * gst/multifile/gstsplitmuxsrc.c:
+ * gst/multifile/gstsplitmuxsrc.h:
+ * gst/multifile/gstsplitutils.c:
+ * gst/multifile/gstsplitutils.h:
+ * gst/multifile/test-splitmuxpartreader.c:
+ * tests/check/Makefile.am:
+ * tests/check/elements/.gitignore:
+ * tests/check/elements/splitmux.c:
+ * tests/files/splitvideo00.ogg:
+ * tests/files/splitvideo01.ogg:
+ * tests/files/splitvideo02.ogg:
+ splitmux: Implement new elements for splitting files at mux level.
+ Implement 2 new elements - splitmuxsink and splitmuxsrc.
+ splitmuxsink is a bin which wraps a muxer and takes 1 video stream,
+ plus audio/subtitle streams, and starts a new file
+ whenever necessary to avoid overrunning a threshold of either bytes
+ or time. New files are started at a keyframe, and corresponding audio
+ and subtitle streams are split at packet boundaries to match
+ video GOP timestamps.
+ splitmuxsrc is a corresponding source element which handles
+ the splitmux:// URL and plays back all component files,
+ reconstructing the original elementary streams as it goes.
+
+2015-02-04 16:32:14 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * tests/check/elements/souphttpsrc.c:
+ * tests/files/test-cert.pem:
+ * tests/files/test-key.pem:
+ tests: souphttpsrc: update ssl key/cert pair
+ Our ones were expired. The new ones were copied from libsoup's
+ tests files.
+ Also sets the property to use our own cert to validate the
+ server, otherwise the default system certs would be used
+ and it would fail.
+
+2015-02-04 02:25:44 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/rtp/gstrtph264depay.c:
+ rtph264depay: prevent trying to get 0 bytes from adapter
+ This causes an assertion and would lead to getting a NULL instead
+ of a buffer. Without proper checking this would easily lead to
+ a segfault
+ https://bugzilla.gnome.org/show_bug.cgi?id=737199
+
+2015-02-04 21:50:51 +1100 Jan Schmidt <jan@centricular.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Simple implementation of GST_SEGMENT_FLAG_TRICKMODE_KEY_UNITS
+ When the trickmode key-units flag is set on the segment, simply skip
+ any sample on a video stream that isn't a keyframe
+
+2015-02-03 17:35:52 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtsp/gstrtspsrc.c:
+ rtspsrc: fix container handling
+ We detect a container correctly now so we need to revert the weird
+ check there was before.
+ Use gst_rtspsrc_stream_push_event() to push the caps event on the
+ right pad.
+ See https://bugzilla.gnome.org/show_bug.cgi?id=739391
+
+2015-02-02 19:46:27 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/matroska/matroska-ids.h:
+ * gst/matroska/matroska-mux.c:
+ * gst/matroska/matroska-mux.h:
+ matroskamux: store and write stream tags
+ Separate global from stream tags storage and write them to the
+ appropriate tags entry in the output
+
+2015-02-02 13:35:59 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: parse stream tags
+ Keep global and stream tags separately and parse the udta node
+ that can be found under the trak atom. The udta will contain
+ stream specific tags and will be pushed as such
+ https://bugzilla.gnome.org/show_bug.cgi?id=692473
+
+2015-01-31 14:32:34 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/isomp4/gstqtmux.c:
+ * gst/isomp4/gstqtmux.h:
+ qtmux: store stream and container tags separately
+ Tags received via events, when marked as stream tags, will
+ be stored on that stream's trak atom instead of being stored
+ in the main tags atom. This allows the resulting file to have
+ global and stream tags stored.
+ https://bugzilla.gnome.org/show_bug.cgi?id=692473
+
+2015-01-31 13:14:44 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/isomp4/atoms.c:
+ * gst/isomp4/atoms.h:
+ * gst/isomp4/gstqtmux.c:
+ qtmux: refactor tags functions to accomodata UDTA at trak level
+ Refactor the functions that were bound to the 'moov' atom to
+ directly pass the desired 'udta' that should receive the tags.
+ This allows the tags to be written to 'udta' at the 'moov' or
+ the 'trak' level, creating tags that are for the container or
+ for a stream only.
+ https://bugzilla.gnome.org/show_bug.cgi?id=692473
+
+2015-01-31 10:47:40 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/isomp4/gstqtmux.c:
+ qtmux: map application name to _swr tag
+ It refers to the application name and version used to create the
+ file
+ https://bugzilla.gnome.org/show_bug.cgi?id=692473
+
+2015-01-31 02:30:40 +1100 Jan Schmidt <jan@centricular.com>
+
+ * gst/matroska/matroska-demux.c:
+ * gst/matroska/matroska-parse.c:
+ * gst/matroska/matroska-read-common.c:
+ * gst/matroska/matroska-read-common.h:
+ matroska: Fix seeking past the end of the file in reverse mode.
+ Snap to the end of the file when seeking past the end in reverse mode,
+ and also fix GST_SEEK_TYPE_END and GST_SEEK_TYPE_NONE handling
+ for the stop position by always seeking on a segment in stream time
+
+2015-01-30 18:22:31 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/rtpsession.c:
+ rtpsession: Fix signal name
+ This wasn't meant to be pushed at all yet, but now that it's there
+ already it won't hurt to make it correct at least.
+
+2015-01-30 16:56:35 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/rtpstats.h:
+ rtpstats: Fix typo in documentation
+
+2015-01-30 16:50:36 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/rtpsession.c:
+ * gst/rtpmanager/rtpsession.h:
+ rtpsession: Add new on-receiving-rtcp signal
+ This will be emitted whenever an RTCP packet is received. Different to
+ on-feedback-rtcp, this signal gets every complete RTCP packet and not
+ just the individual feedback packets.
+
+2015-01-28 14:02:15 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/isomp4/qtdemux.c:
+ * gst/isomp4/qtdemux.h:
+ qtdemux: simplify segment.base math
+ Remove a fix for heavily edited files added for fixing
+ https://bugzilla.gnome.org/show_bug.cgi?id=345830 to work
+ with seeks and proper gaps playback. The fix was replaced
+ for a more general solution that bases on using previous
+ segment's duration, just like it works for media segments
+ playback.
+ https://bugzilla.gnome.org/show_bug.cgi?id=743518
+
+2015-01-27 14:00:35 +0000 Luis de Bethencourt <luis.bg@samsung.com>
+
+ * gst/videomixer/videomixerorc-dist.c:
+ videomixer: update orc files
+
+2015-01-26 17:08:12 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Fix data dropping for fragmented streams
+ For fragmented streams with extra data at the end of the mdat
+ qtdemux was not dropping those bytes and would try to use
+ that extra data as the beginning of a new atom, causing the
+ stream to fail.
+ https://bugzilla.gnome.org/show_bug.cgi?id=743407
+
+2015-01-25 17:30:33 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/rtpsession.c:
+ rtpsession: Deprecate rtcp-immediate-feedback-threshold property
+ It had no effect since quite some time and also is not needed in general,
+ especially not to switch between immediate feedback mode and early feedback
+ mode. The latest understanding of the RFC is that from the endpoint point of
+ view, both modes are exactly the same. RTCP is only allowed to use the
+ bandwidth as given by the RFC constraints, as such it is only ever possible
+ to schedule a RTCP packet early but it's against the RFC to schedule more RTCP
+ packets.
+ The difference between immediate feedback mode and early feedback mode is that
+ the former guarantees that an RTCP packet can be sent for every event
+ "immediately", which means that the bandwidth calculations from the RFC have
+ resulted in an RTCP scheduling interval that is small enough. Early feedback
+ mode on the other hand means that we can schedule some packets early to make
+ that happen, but it's not guaranteed at all that it's possible to schedule
+ an RTCP packet per event (i.e. they need to be accumulated or dropped).
+
+2015-01-22 10:29:39 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/rtpsession.c:
+ rtpsession: Delay the next regular RTCP packet after early RTCP
+ This is required to not exceed the short term average RTCP bitrate when
+ using early feedback as compared to without early feedback.
+
+2015-01-22 10:28:52 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/rtpsession.c:
+ rtpsession: Add new send-rtcp-full signal
+ This indicates with a boolean return value if scheduling a new RTCP packet
+ within the requested delay was possible. Otherwise it behaves exactly like
+ send-rtcp. The only reason for adding a new signal is ABI compatibility.
+
+2015-01-20 00:32:00 +0000 Jimmy Ohn <yongjin.ohn@lge.com>
+
+ * ext/pulse/pulsesink.c:
+ pulsesink: Free format_info in query_getcaps
+ If we can not create probe stream in query_getcaps function, it will appear
+ memory leakage from format info.
+ The following patch prevent memory leakage in pulsesink.
+ https://bugzilla.gnome.org/show_bug.cgi?id=743178
+
+2015-01-23 17:35:51 +0000 Luis de Bethencourt <luis.bg@samsung.com>
+
+ * gst/matroska/matroska-read-common.c:
+ matroskademux: remove unnecessary check
+ No matter if gst_matroska_read_common_parse_index_cuetrack () returns that the
+ flow is OK or not, the check there will be a break from the switch. Removing the
+ check since the outcome is the same.
+ CID #1265762
+
+2015-01-23 15:16:25 +0100 Edward Hervey <bilboed@bilboed.com>
+
+ * gst/matroska/matroska-mux.c:
+ matroskamux: Avoid using freed variable
+ the name variable might have been attributed to pad_name, make sure we
+ free it only *after* pad_name has been used.
+ Coverity CID : 1265774
+
+2015-01-23 15:13:55 +0100 Edward Hervey <bilboed@bilboed.com>
+
+ * gst/avi/gstavimux.c:
+ avimux: Avoid using freed variable
+ the name variable might have been attributed to pad_name, make sure we
+ free it only *after* pad_name has been used.
+ Coverity CID : 1265775
+
+2014-11-14 12:59:31 +0100 Peter Seiderer <ps.report@gmx.net>
+
+ * sys/v4l2/gstv4l2object.c:
+ v4l2object: reuse caps framerate if not overwritten by v4l2 device
+ Enables duration setting in v4l2src.
+ https://bugzilla.gnome.org/show_bug.cgi?id=740403
+
+2015-01-22 10:29:24 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/rtpsession.c:
+ rtpsession: Fix indention
+
+2015-01-21 17:36:26 +0100 Edward Hervey <bilboed@bilboed.com>
+
+ * gst/isomp4/qtdemux_dump.c:
+ qtdemux_dump: Bypass even more code if debugging is disabled
+ And avoid using variables that won't exist when debugging is disabled
+
+2015-01-21 15:30:33 +0100 Edward Hervey <bilboed@bilboed.com>
+
+ * gst/isomp4/qtdemux_dump.c:
+ qtdemux: Only traverse/dump nodes if guaranteed to be used
+ __gst_debug_min is the "global" lowest debug level set. There's no
+ guarantee the qtdemux debug category is actually set at that level.
+
+2014-12-20 17:09:14 +0100 Edward Hervey <bilboed@bilboed.com>
+
+ * gst/matroska/ebml-read.c:
+ matroska: Avoid debugging below category threshold
+ This part alone was what made the matroska thread take a full core
+ on an android phone ...
+
+2015-01-21 09:55:30 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * ext/dv/gstsmptetimecode.c:
+ * ext/mikmod/mikmod_types.c:
+ * gst/audiofx/audiodynamic.c:
+ * gst/audiofx/audiopanorama.c:
+ * gst/effectv/gstradioac.c:
+ * gst/isomp4/atoms.c:
+ * gst/isomp4/gstqtmuxmap.c:
+ * gst/isomp4/qtdemux.c:
+ * gst/rtsp/gstrtspsrc.c:
+ * gst/videofilter/gstvideotemplate.c:
+ * gst/wavparse/gstwavparse.c:
+ Constify some static arrays everywhere
+
+2015-01-19 17:49:54 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: fix deadlock seeking in files without seek entries
+ A mutex unlock was missing.
+ https://bugzilla.gnome.org/show_bug.cgi?id=739975
+
+2015-01-19 12:34:25 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * gst/videomixer/blend.c:
+ videomixer: fix illegal memory access in blend function with negative ypos
+ https://bugzilla.gnome.org/show_bug.cgi?id=741115
+
+2015-01-13 16:49:34 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * sys/v4l2/gstv4l2videodec.c:
+ v4l2videodec: Proxy getcaps
+ Replace the sink_query with new getcaps() virtual and use the proxy
+ helper with the probed caps. This allow upstream element taking decision
+ base on what is supported downstream.
+
+2015-01-13 19:05:20 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/isomp4/fourcc.h:
+ * gst/isomp4/gstqtmux.c:
+ * gst/isomp4/gstqtmuxmap.c:
+ qtmux: Add support for v210
+
+2015-01-13 18:58:01 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: v210 is v210, not UYVY and yuv2 is YUY2, not I420
+ Also add a few other raw video formats we support: v308, v216
+ and add comments for a few others we don't support yet.
+ https://developer.apple.com/library/mac/technotes/tn2162/
+
+2015-01-12 15:56:29 +0100 Stefan Sauer <ensonic@users.sf.net>
+
+ * common:
+ Automatic update of common submodule
+ From f2c6b95 to bc76a8b
+
+2015-01-10 15:51:16 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * sys/osxvideo/cocoawindow.h:
+ * sys/osxvideo/cocoawindow.m:
+ * sys/osxvideo/osxvideosink.h:
+ * sys/osxvideo/osxvideosink.m:
+ osxvideosink: Disable hack for NSApp iteration with a special #define
+ The hack causes deadlocks and other interesting problems and it really
+ can only be fixed properly inside GLib. We will include a patch for
+ GLib in our builds for now that handles this, and hopefully at some
+ point GLib will also merge a proper solution.
+ A proper solution would first require to refactor the polling in
+ GMainContext to only provide a single fd, e.g. via epoll/kqueue
+ or a thread like the one added by our patch. Then this single
+ fd could be retrieved from the GMainContext and directly integrated
+ into a NSRunLoop.
+ https://bugzilla.gnome.org/show_bug.cgi?id=741450
+ https://bugzilla.gnome.org/show_bug.cgi?id=704374
+
+2015-01-08 21:07:05 +0100 Mark Nauwelaerts <mnauw@users.sourceforge.net>
+
+ * ext/pulse/pulsesink.c:
+ pulsesink: uncork if needed upon commit
+ ... to provide for a running clock.
+
+2015-01-09 16:59:53 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * sys/v4l2/gstv4l2videodec.c:
+ v4l2videodec: Prevent renegotiation
+ Renegotiation isn't supported, simply prevent it the way we do in
+ v4l2src.
+
+2015-01-06 13:54:25 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * sys/v4l2/gstv4l2videodec.c:
+ v4l2videodec: Don't unlock the stream lock twice
+
+2015-01-09 11:40:40 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: fix stream time conversion
+ Use the right macro to convert to the correct scale or the
+ segment information will be wrong
+ https://bugzilla.gnome.org/show_bug.cgi?id=742572
+
+2015-01-07 18:48:58 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
+
+ * sys/v4l2/gstv4l2allocator.c:
+ v4l2allocator: Add protection against driver bug
+ v4l2loopback driver has a this nasty bug that if the queue is larger
+ then 2 buffers, it returns random index on dqbuf. So far we assumed
+ that the index was always right, which would lead to memory being
+ unref twice, and eventually crash.
+
+2015-01-07 17:58:05 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
+
+ * sys/v4l2/gstv4l2allocator.c:
+ * sys/v4l2/gstv4l2allocator.h:
+ * sys/v4l2/gstv4l2bufferpool.c:
+ v4l2: Don't use allocator size to iterate
+ As the buffer array is fixed size and small, it's safer to simply
+ use this static size to cleanup the buffers. This is also more
+ consistent with the rest. The associated method is no longer
+ required and can be dropped.
+
+2015-01-07 17:55:14 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
+
+ * sys/v4l2/gstv4l2bufferpool.c:
+ v4l2bufferpool: Don't clean buffer array in dispose
+ This should already have been done, plus this code is incorrect
+ and may lead to crash.
+ https://bugzilla.gnome.org/show_bug.cgi?id=742074
+
+2015-01-07 17:48:31 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
+
+ * sys/v4l2/gstv4l2bufferpool.c:
+ v4l2bufferpool: Don't ref queued output buffer
+ This partly revert to the old 1.2 behavior. Instead of keeping a
+ reference to the output buffer queued, we simply release them but
+ don't forward it to GstBufferPool. This way, the buffer pool don't
+ need to be flushed to be stopped.
+ https://bugzilla.gnome.org/show_bug.cgi?id=742074
+
+2015-01-08 11:37:23 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
+
+ * sys/v4l2/gstv4l2bufferpool.c:
+ v4l2bufferpool: Never fail on streamoff
+ Failing streamoff prevents allocator from being disposed hence
+ lead to device FD leak. There is no known cases where streamoff
+ may fails for which we'd still be streaming. streamoff is known
+ to fail when a device is being unplugged (in which case errno
+ 19/ENODEV is set).
+ https://bugzilla.gnome.org/show_bug.cgi?id=732734
+
+2015-01-07 21:52:17 -0500 Brad Smith <brad@comstyle.com>
+
+ * configure.ac:
+ v4l2: Add support for detecting the presence of V4L2 support on OpenBSD
+ https://bugzilla.gnome.org/review?bug=742503
+
+2015-01-04 15:57:10 +0100 Matej Knopp <matej.knopp@gmail.com>
+
+ * gst/audioparsers/gstac3parse.c:
+ ac3parse: request at least 8 bytes to properly parse header
+ https://bugzilla.gnome.org/show_bug.cgi?id=742325
+
+2015-01-07 16:20:03 -0800 Michael Smith <michael.smith@rdio.com>
+
+ * gst/wavparse/gstwavparse.c:
+ wavparse: skip an additional uninteresting chunk type before the fmt chunk.
+
+2015-01-07 18:16:12 +0000 Luis de Bethencourt <luis.bg@samsung.com>
+
+ * gst/audiofx/audiodynamic.c:
+ audiodynamic: assert func_index is inside bounds
+ Bringing back the check removed in the previous commit but have that check be a
+ g_assert. Changing the function to static void since return can never be False,
+ because audio format will never be unkown.
+
+2015-01-07 17:31:39 +0000 Luis de Bethencourt <luis.bg@samsung.com>
+
+ * gst/audiofx/audiodynamic.c:
+ audiodynamic: remove always-true conditional
+ func_index is set by the sum of three ternary operators which add, 0:4, 0:2,
+ and 1:0. Minimum value would be 0+0+0=0, and maximum would be 4+2+1=7.
+ The conditional checking if func_index is >= 0 and < 8 will always be true.
+ Removing it.
+ CID 1226442
+
+2015-01-07 18:05:18 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ rtpjitterbuffer: If we get a gap with a buffer without DTS, error out
+ We (currently?) can't really handle gaps between RTP packets if they're not
+ properly timestamped. The current code would go into calculations with
+ GST_CLOCK_TIME_NONE and then cause assertions everywhere. It's probably
+ better to error out cleanly instead.
+
+2014-11-21 11:39:19 -0800 Aleix Conchillo Flaqué <aleix@oblong.com>
+
+ * gst/rtsp/gstrtspsrc.c:
+ rtspsrc: set PLAYING state after configuring caps
+ We set to PLAYING after we have configured the caps, otherwise we
+ might end up calling request_key (with SRTP) while caps are still
+ being configured, ending in a crash.
+ https://bugzilla.gnome.org/show_bug.cgi?id=740505
+
+2014-12-30 18:03:22 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/icles/gdkpixbufoverlay-test.c:
+ tests: gdkpixbufoverlay-test: remove outdated FIXME
+
+2014-12-30 17:19:42 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/check/elements/rtpcollision.c:
+ tests: rtpcollision: use alawenc/dec in these tests instead of Speex
+ They should always be built, while the speex elements are not.
+ Need to check for a smaller number of buffers then (7->4) because
+ speexenc will add 3 header buffers while alawenc will just output
+ as many buffers as it receives as input.
+ https://bugzilla.gnome.org/show_bug.cgi?id=742098
+
+2014-12-30 16:36:02 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/check/pipelines/simple-launch-lines.c:
+ tests: simple-launch-lines: only run jpeg/png tests if elements are available
+
+2014-12-30 16:26:58 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * ext/soup/gstsouphttpsrc.c:
+ souphttpsrc: Don't return a buffer when returning not GST_FLOW_OK
+ basesrc assumes that we don't return a buffer if
+ something else than OK is returned. It will just
+ leak any buffer we might accidentially provide
+ here.
+ This can potentially happen during flushing.
+ Maybe fixes https://bugzilla.gnome.org/show_bug.cgi?id=741993
+
+2014-12-30 14:52:42 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/check/elements/rtpaux.c:
+ tests: rtpaux: use alawenc/dec in these tests instead of Speex
+ They should always be built, while the speex elements are not.
+ https://bugzilla.gnome.org/show_bug.cgi?id=742098
+
+2014-12-29 15:35:19 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/matroska/matroska-demux.c:
+ matroskademux: Improve detection of being stuck at the same offset
+ Only error out if we read from the same position again and got the
+ same length. Just the same position is not necessarily enough.
+
+2014-12-29 15:00:02 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/matroska/matroska-demux.c:
+ matroskademux: Don't get stuck at the same offset when searching for clusters
+ This could happen if there is an invalid cluster with size 0, and in that
+ case just error out instead of looping forever.
+
+2014-12-25 21:32:40 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/isomp4/gstqtmux.c:
+ qtmux: fix ALAC muxing
+ Actually copy the codec data instead of copying nothing
+ and then bombing out because there's no data.
+ Fixes: gst-launch-1.0 audiotestsrc ! avenc_alac ! qtmux ! fakesink
+ https://bugzilla.gnome.org/show_bug.cgi?id=741783
+
+2014-12-25 15:48:04 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/rtpmanager/gstrtpptdemux.c:
+ rtpptdemux: just drop invalid rtp packets instead of erroring out
+ Apparently linphone sends an invalid RTP packet as very
+ first packet. We want to ignore that instead of erroring
+ out (same for any other invalid packets really).
+ https://bugzilla.gnome.org/show_bug.cgi?id=741398
+
+2014-12-25 15:44:15 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/rtpmanager/gstrtpptdemux.c:
+ rtpptdemux: fix 0.10-ism in docs
+
+2014-12-25 14:58:12 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/icles/gdkpixbufoverlay-test.c:
+ tests: gdkpixbufoverlay-test: use absolute positioning to fix demo
+ https://bugzilla.gnome.org/show_bug.cgi?id=739566
+
+2014-12-25 14:53:09 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * ext/gdk_pixbuf/gstgdkpixbufoverlay.c:
+ * ext/gdk_pixbuf/gstgdkpixbufoverlay.h:
+ gdkpixbufoverlay: add "positioning-mode" property to allow absolute positions
+ Set positioning-mode=pixels-absolute to allow positioning with
+ absolute coordinates, meaning negative x/y offsets will be
+ interpreted as being to the left/above the video frame instead
+ of being interpreted as relative to the right/bottom edge of
+ the video frame (which is a silly default, but that's how it is).
+ This means we can nicely slide images into and out of the frame,
+ see gdkpixbufoverlay-test.
+ https://bugzilla.gnome.org/show_bug.cgi?id=739566
+
+2014-12-22 15:33:51 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * sys/osxaudio/gstosxaudiosink.c:
+ * sys/osxaudio/gstosxaudiosrc.c:
+ osxaudio: Directly return the ringbuffer's caps if it is acquired
+
+2014-12-22 12:56:19 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * sys/osxaudio/gstosxaudiosink.c:
+ * sys/osxaudio/gstosxaudiosrc.c:
+ osxaudio: Put all audio formats into the template caps
+ We report the proper caps later from the get_caps() vfunc implementation after
+ probing the selected device.
+
+2014-12-22 12:56:05 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * sys/osxaudio/gstosxaudioringbuffer.c:
+ osxaudio: Also set the big endian flag for floating point samples
+
+2014-12-22 11:45:59 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * MAINTAINERS:
+ MAINTAINERS: Update my mail address
+
+2014-12-22 10:23:01 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * sys/osxaudio/gstosxaudiosink.c:
+ * sys/osxaudio/gstosxaudiosrc.c:
+ osxaudio: Fix deadlock and property change notification in device selection code
+ After creating the ringbuffer we have to set the device on the ringbuffer as
+ it defaults to kAudioDeviceUnknown. At this point it can't have changed to
+ anything else yet and we don't have to notify about changes to the sink/src
+ "device" property. It's also not a good idea because GstAudioBaseSrc has the
+ object lock taken while the ringbuffer is created, which might cause a
+ deadlock if something calls back into the element from "notify::device".
+ Once the base class is done with the NULL_TO_READY state change, it has opened
+ the device via the ringbuffer and this might have chosen a different device.
+ Especially if we initially used kAudioDeviceUnknown. Also notify about this
+ property change as initially intended by this code.
+
+2014-12-19 12:30:03 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * sys/v4l2/gstv4l2bufferpool.c:
+ v4l2pool: Update configuration size
+ We already update our copy of VideoInfo.size to proper size, now also
+ the configuration so the size matches on release.
+ https://bugzilla.gnome.org/show_bug.cgi?id=741420
+
+2014-12-19 10:57:29 +0100 Edward Hervey <bilboed@bilboed.com>
+
+ * gst/matroska/matroska-demux.c:
+ * gst/matroska/matroska-demux.h:
+ matroska-demux: Cache upstream length
+ Instead of constantly querying upstream, just cache the last duration,
+ and in the unlikelyness we might have gone over query again before
+ deciding we are EOS.
+ Cut 15% cpu off matroskademux streaming thread (srsly...)
+
+2014-12-17 17:36:18 +0000 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * gst/matroska/matroska-demux.c:
+ * gst/matroska/matroska-ids.c:
+ * gst/matroska/matroska-ids.h:
+ * gst/matroska/matroska-mux.c:
+ matroska: mux/demux the OpusHead header
+ This is meant to be so (https://wiki.xiph.org/MatroskaOpus - while
+ it is marked as a draft, this part was confirmed to be correct on
+ IRC), and allows one to determine whether a demuxed stream is
+ multistream or not, and thus set the multistream caps field
+ accordingly. In turn, this means downstream does not have to guess.
+ https://bugzilla.gnome.org/show_bug.cgi?id=740744
+
+2014-12-18 11:50:33 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp/gstrtspsrc.c:
+ rtspsrc: Don't dereference NULL if a suitable stream for the AUX element can't be found
+ CID 1258717
+
+2014-12-18 10:53:39 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * common:
+ Automatic update of common submodule
+ From ef1ffdc to f2c6b95
+
+2014-12-12 23:06:07 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/udp/gstmultiudpsink.c:
+ * gst/udp/gstmultiudpsink.h:
+ udpsink: allocate scratch space for render functions on the heap
+ and not the stack. Our allocations could get a bit too large
+ to be sure it's not going to cause trouble using the stack.
+
+2014-06-24 01:16:37 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/udp/gstmultiudpsink.c:
+ multiudpsink: re-use send_buffers() code path for render() function
+ It's like rendering a buffer list, just with one buffer.
+ Has the added advantage that if there are multiple clients
+ we can send the buffer to all the clients in one go.
+
+2014-06-24 01:15:25 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/udp/gstmultiudpsink.c:
+ * gst/udp/gstmultiudpsink.h:
+ multiudpsink: keep client list consistent during removals
+ We unlock and re-lock the client lock while emitting the
+ removed signal, which causes inconsistencies in the client
+ list vs. the client counts. Instead, remove the client from
+ the list already before emitting the signal and put it into
+ a temporary list of clients to be removed. That way things
+ look consistent to the streaming thread, but signal callbacks
+ can still do things like get stats from removed clients.
+
+2014-06-24 00:56:27 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/udp/gstmultiudpsink.c:
+ multiudpsink: fix client count after removal
+
+2014-06-23 18:43:21 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/udp/gstmultiudpsink.c:
+ multiudpsink: keep client list sorted by socket family
+ We make use of in the send_buffers() function if we
+ need to use different sockets to send to IPv4 and
+ IPv6 destinations.
+
+2014-06-20 11:36:19 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/udp/gstmultiudpsink.c:
+ * gst/udp/gstmultiudpsink.h:
+ multiudpsink: add sendmmsg-ready render_list function prototype
+ Add prototype for a render_list() function that can use a
+ sendmmsg-style g_socket_send_messages() function once it lands
+ in GLib. We can use this infrastructure to send multiple buffers
+ made up by multiple memories to multiple clients in one go, which
+ drastically reduces the number of syscalls made when sending
+ high-bitrate video streams.
+ https://bugzilla.gnome.org/show_bug.cgi?id=732152
+
+2014-06-19 19:16:01 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/udp/gstmultiudpsink.c:
+ * gst/udp/gstmultiudpsink.h:
+ multiudpsink: make udp client structure refcounted
+ Use the refcount for memory management and keep track
+ of the number of duplicate clients in a separate
+ variable. This will be useful later, and means we
+ don't have to hold the OBJECT_LOCK all the time.
+ https://bugzilla.gnome.org/show_bug.cgi?id=732866
+
+2014-06-19 18:31:05 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/udp/gstmultiudpsink.c:
+ * gst/udp/gstmultiudpsink.h:
+ multiudpsink: keep count of number of unique and non-unique IPv4 and IPv6 clients
+ This will come in handy later.
+
+2014-12-16 15:00:22 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * sys/v4l2/gstv4l2bufferpool.c:
+ v4l2bufferpool: Disable create_buf with libv4l2
+ Libv4l2 does not work with CREATE_BUFS. Instead of failing on random
+ error caused by libv4l2, disable CREATE_BUFS when an emulated format is
+ detected.
+
+2014-12-09 17:39:12 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * sys/v4l2/gstv4l2allocator.c:
+ v4l2allocator: Add protection against broken libv4l2
+ It looks like libv4l2 support for CREATE_BUF is incomplete. That
+ combine with existing bugs may lead to crash in GStreamer. These
+ check will make it robust by:
+ - Checking create buf index isn't an already in used index
+ - Checking that the index out of QUERYBUF matches the requested
+ index
+
+2014-12-16 16:37:24 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp/gstrtspsrc.c:
+ rtspsrc: Add something to the debug logs if an RTX AUX element can't be added
+ ... because the application already has a signal handler set up here.
+
+2014-11-21 14:13:34 +1100 Matthew Waters <matthew@centricular.com>
+
+ * gst/rtsp/gstrtspsrc.c:
+ * gst/rtsp/gstrtspsrc.h:
+ rtspsrc: add retransmission support according to RFC4588
+ Based on the client-rtpaux example
+
+2014-12-16 13:25:01 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * sys/osxvideo/osxvideosink.m:
+ osxvideosink: clear rectangle structures before use
+
+2014-12-09 15:09:56 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * sys/v4l2/gstv4l2object.c:
+ v4l2object: Always set format
+ Right now we try to be clever by detecting if device format have
+ changed or not, and skip setting format in this case. This is valid
+ behaviour with V4L2, but it's also very error prone. The rational
+ for not setting these all the time is for speed, though I can't
+ measure any noticeable gain on any HW I own. Also, until recently,
+ we where doing get/set on the format for each format we where
+ probing, making it near to impossible that the format would match.
+ This also fixes bug where we where skipping frame-rate setting if
+ format didn't change.
+ https://bugzilla.gnome.org/show_bug.cgi?id=740636
+
+2014-12-15 18:30:01 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * gst/videocrop/gstvideocrop.c:
+ videocrop: Remove todo about caps filter
+ The filter is already interected.
+
+2014-12-15 18:19:05 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * gst/videocrop/gstvideocrop.c:
+ * gst/videocrop/gstvideocrop.h:
+ videocrop: Make sure new crop is applied
+ Since "basetransform: Fix caps equality check" commit a7f357,
+ set_info() will not be called anymore if crop didn't change
+ the caps. This is fixed by setting "need_update" boolean when
+ cropping properties has been changed, and then applying these
+ if they where not applied before rendering the next frame. This
+ patch also fixed the locking, dropping un-needed custom lock,
+ and no holding needless lock while doing the operation as we
+ already hold the streaming lock.
+ https://bugzilla.gnome.org/show_bug.cgi?id=740787
+
+2014-12-12 18:10:35 +0530 Arun Raghavan <git@arunraghavan.net>
+
+ * sys/osxaudio/gstosxaudiosink.c:
+ osxaudiosink: Prefer filter caps order while getting caps
+ https://bugzilla.gnome.org/show_bug.cgi?id=740987
+
+2014-12-09 13:38:26 +0530 Arun Raghavan <git@arunraghavan.net>
+
+ * sys/osxaudio/gstosxaudiosink.c:
+ osxaudiosink: Add some error handling around channel layout parsing
+ For now we just spit a warning and ignore the channel layout if we can't
+ support it.
+ https://bugzilla.gnome.org/show_bug.cgi?id=740987
+
+2014-12-08 22:38:22 +0530 Arun Raghavan <git@arunraghavan.net>
+
+ * sys/osxaudio/gstosxaudiosink.c:
+ * sys/osxaudio/gstosxaudiosrc.c:
+ osxaudio: Take lock around sink/source before accessing the ringbuffer
+ https://bugzilla.gnome.org/show_bug.cgi?id=740987
+
+2014-12-01 21:06:27 +0530 Arun Raghavan <git@arunraghavan.net>
+
+ * sys/osxaudio/gstosxaudiosink.c:
+ * sys/osxaudio/gstosxaudiosink.h:
+ * sys/osxaudio/gstosxaudiosrc.c:
+ * sys/osxaudio/gstosxcoreaudio.c:
+ * sys/osxaudio/gstosxcoreaudio.h:
+ * sys/osxaudio/gstosxcoreaudioremoteio.c:
+ osxaudiosrc: Probe channel layout too
+ https://bugzilla.gnome.org/show_bug.cgi?id=740987
+
+2014-12-01 20:32:04 +0530 Arun Raghavan <git@arunraghavan.net>
+
+ * sys/osxaudio/gstosxaudiosink.c:
+ osxaudiosink: Only fix up channels/layout for PCM caps while probing
+ It's unlikely that setting a channel layout will do much for AC3/DTS
+ streams. If we find at some point that it does make sense, we can
+ perform the structure copying unconditionally (i.e., the current code is
+ wrong, since AC3/DTS will get two structures now - one with the channel
+ layout, one without).
+ https://bugzilla.gnome.org/show_bug.cgi?id=740987
+
+2014-12-01 19:41:35 +0530 Arun Raghavan <git@arunraghavan.net>
+
+ * sys/osxaudio/gstosxaudiosrc.c:
+ * sys/osxaudio/gstosxaudiosrc.h:
+ * sys/osxaudio/gstosxcoreaudio.c:
+ * sys/osxaudio/gstosxcoreaudio.h:
+ osxaudiosrc: Implement caps probing
+ https://bugzilla.gnome.org/show_bug.cgi?id=740987
+
+2014-12-01 19:29:57 +0530 Arun Raghavan <git@arunraghavan.net>
+
+ * sys/osxaudio/gstosxcoreaudiohal.c:
+ osxaudio: Bind audio device to audio unit early
+ We want to bind the device during open so that subsequent format queries
+ on the audio unit are as specific as possible from that point onwards.
+ https://bugzilla.gnome.org/show_bug.cgi?id=740987
+
+2014-11-29 23:16:30 +0530 Arun Raghavan <git@arunraghavan.net>
+
+ * sys/osxaudio/gstosxaudiosink.c:
+ osxaudiosink: Fix up caps querying a bit
+ This should make caps queries correct in PAUSED and higher as well.
+ https://bugzilla.gnome.org/show_bug.cgi?id=740987
+
+2014-11-28 22:32:36 +0530 Arun Raghavan <git@arunraghavan.net>
+
+ * sys/osxaudio/gstosxaudiosrc.c:
+ * sys/osxaudio/gstosxcoreaudio.c:
+ osxaudio: Move osxaudiosrc-specific code out of the generic path
+ Avoids one layering violation (GstCoreAudio referring to
+ GstOsxAudioSrc).
+ https://bugzilla.gnome.org/show_bug.cgi?id=740987
+
+2014-11-28 22:23:17 +0530 Arun Raghavan <git@arunraghavan.net>
+
+ * sys/osxaudio/gstosxaudioringbuffer.c:
+ * sys/osxaudio/gstosxaudioringbuffer.h:
+ * sys/osxaudio/gstosxaudiosink.c:
+ * sys/osxaudio/gstosxaudiosrc.c:
+ * sys/osxaudio/gstosxcoreaudio.c:
+ * sys/osxaudio/gstosxcoreaudio.h:
+ * sys/osxaudio/gstosxcoreaudiohal.c:
+ * sys/osxaudio/gstosxcoreaudioremoteio.c:
+ osxaudio: Clean up a GstCoreAudio -> GstOsxAudioSrc/Sink reference
+ Now that device selection has no sink/source-specific bits, we can have
+ generic device selection for this path. We do need to now track state
+ changes so we can look up the final device_id once the device is open,
+ though.
+ https://bugzilla.gnome.org/show_bug.cgi?id=740987
+
+2014-11-28 19:40:52 +0530 Arun Raghavan <git@arunraghavan.net>
+
+ * sys/osxaudio/gstosxaudiosink.c:
+ osxaudiosink: Move device caps probing to get_caps()
+ This should be preferred to running the probe at device open time.
+ https://bugzilla.gnome.org/show_bug.cgi?id=740987
+
+2014-11-28 18:37:02 +0530 Arun Raghavan <git@arunraghavan.net>
+
+ * sys/osxaudio/gstosxcoreaudiohal.c:
+ osxaudio: Make some debug code compile conditionally
+ https://bugzilla.gnome.org/show_bug.cgi?id=740987
+
+2014-11-28 15:06:35 +0530 Arun Raghavan <git@arunraghavan.net>
+
+ * sys/osxaudio/gstosxaudioringbuffer.c:
+ * sys/osxaudio/gstosxaudioringbuffer.h:
+ * sys/osxaudio/gstosxaudiosink.c:
+ * sys/osxaudio/gstosxaudiosrc.c:
+ osxaudio: Move device selection to ringbuffer->open_device()
+ This is conceptually the right thing to do, and allows us to correctly
+ catch errors in device selection as well, which we could not do while
+ creating the ringbuffer.
+ https://bugzilla.gnome.org/show_bug.cgi?id=740987
+
+2014-11-28 14:34:34 +0530 Arun Raghavan <git@arunraghavan.net>
+
+ * sys/osxaudio/gstosxaudiosink.c:
+ * sys/osxaudio/gstosxaudiosrc.c:
+ * sys/osxaudio/gstosxcoreaudio.c:
+ * sys/osxaudio/gstosxcoreaudio.h:
+ * sys/osxaudio/gstosxcoreaudiohal.c:
+ * sys/osxaudio/gstosxcoreaudioremoteio.c:
+ osxaudio: Consolidate input and output code paths a bit
+ https://bugzilla.gnome.org/show_bug.cgi?id=740987
+
+2014-11-21 11:54:18 +0100 Thibault Saunier <tsaunier@gnome.org>
+
+ * gst/deinterlace/gstdeinterlace.c:
+ Deinterlace: in query_caps return only supported formats if filter is interlaced
+ In some cases the currently set GstVideoInfo is not interlaced, but
+ upstream caps are interlaced and the info is passed in the filter,
+ we should take that info into account and make sure that we do not
+ consider that case as a "pass through" case.
+ https://bugzilla.gnome.org/show_bug.cgi?id=741407
+
+2014-12-12 11:06:17 +0100 Edward Hervey <bilboed@bilboed.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Fix debug statement
+ It was using the non-increasing offset variable, which made that statement
+ not so useful :)
+
+2014-12-12 11:03:15 +0100 Edward Hervey <bilboed@bilboed.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Add macros for the various timescale conversions
+ This helps make the code more readable and avoid future bad usage of
+ scaling function argument order.
+
+2014-12-11 10:16:06 +0100 Patrick Radizi <patrickr@axis.com>
+
+ * gst/rtp/gstrtph264pay.c:
+ rtph264pay: fix potential crash when shutting down
+ A race condition in the state change function may cause buffers
+ to be unreffed while they are still used by the streaming thread
+ in gst_rtp_h264_pay_send_sps_pps() resulting in a crash. Chain
+ up to the parent class first in the state change function to
+ make sure streaming has stopped and only then free those buffers.
+ https://bugzilla.gnome.org/show_bug.cgi?id=741381
+
+2014-12-12 00:42:06 +1100 Jan Schmidt <jan@centricular.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Copy flags of the overall segment to output segments
+ Preserve the segment flags of the overall demux segment on the output
+ segments for each pad.
+
+2014-12-09 02:43:00 +0100 Matej Knopp <matej.knopp@gmail.com>
+
+ * gst/isomp4/gstqtmux.c:
+ qtmux: use 64bit chunk_offset
+ https://bugzilla.gnome.org/show_bug.cgi?id=741279
+
+2014-12-10 17:39:17 +0100 Edward Hervey <bilboed@bilboed.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Fix rounding errors in duration update
+ Make sure we store updated segment stop/duration with the same
+ granularity as the duration timescale.
+ And add more debug
+
+2014-12-10 16:55:44 +0100 Edward Hervey <bilboed@bilboed.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Update duration when we get more information
+ When dealing with fragmented files, we will get more accurate duration
+ information via the mfra and moof atoms.
+ In order for playback to not stop at the initial duration (from the
+ moov atom), we need to check and update the various duration variables
+ when we find more information.
+ Fixes playback of fragmented files in pull mode
+
+2014-12-10 15:08:40 +0100 Edward Hervey <bilboed@bilboed.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Remove variable assignments never read
+ As detected by clang/scan-build
+
+2014-12-10 14:56:06 +0100 Edward Hervey <bilboed@bilboed.com>
+
+ * gst/isomp4/qtdemux.c:
+ * gst/isomp4/qtdemux.h:
+ qtdemux: Use GstClockTime for nanosecond-based time variables/fields
+ Avoids confusion with timescaled-based variables and bytes (offset)
+ variables.
+ And use GST_CLOCK_TIME_NONE where applicable
+
+2014-12-03 14:47:05 +0100 Edward Hervey <bilboed@bilboed.com>
+
+ * gst/debugutils/gstpushfilesrc.c:
+ * gst/debugutils/gstpushfilesrc.h:
+ pushfilesrc: Add TIME SEGMENT capability
+ Adds a new set of properties to make pushfilesrc output a TIME SEGMENT
+ (instead of the filesrc BYTE SEGMENT).
+ When time-segment is set to True the following will happen:
+ * Seeks are refused (data starts from the beginning of the file)
+ * The BYTE segment will be replaced by a TIME segment with the values
+ specified in the various properties
+ * The first outgoing buffer will have a timestamp set on it (by default
+ it has a value of GST_CLOCK_TIME_NONE)
+
+2014-12-10 11:35:29 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/audioparsers/gstaacparse.c:
+ aacparse: Also only unref caps if they're not NULL
+
+2014-12-10 11:34:42 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/audioparsers/gstaacparse.c:
+ aacparse: gst_pad_get_allowed_caps() will return NULL if there is no peer
+
+2014-12-09 16:38:38 +0100 Thibault Saunier <tsaunier@gnome.org>
+
+ * ext/vpx/gstvp8enc.c:
+ * ext/vpx/gstvp9enc.c:
+ vpXenc: CLOCK_TIME_NONE is not a valid min_latency value
+ We should just use 0 if we do not have the information
+
+2014-12-03 17:26:56 +0100 Thibault Saunier <tsaunier@gnome.org>
+
+ * gst/rtpmanager/gstrtpsession.c:
+ rtpsession: Use an empty iterator in iterate_internal_link when no links
+ And not a NULL Iterator, so it is consistent with the way it usually
+ works and avoid user to need a different code paths to handle that.
+
+2014-12-09 14:01:50 +0100 Aurélien Zanelli <aurelien.zanelli@parrot.com>
+
+ * sys/v4l2/gstv4l2bufferpool.c:
+ v4l2bufferpool: set buffer interlace flags when field is V4L2_FIELD_INTERLACED
+ If v4l2_buffer.field is V4L2_FIELD_INTERLACED, we set corresponding
+ GstVideoBuffer flags depending on the video standard.
+ According to V4L2 specification, M/NTSC transmits the bottom field
+ first, all other standards the top field first.
+ https://bugzilla.gnome.org/show_bug.cgi?id=737603
+
+2014-12-08 21:26:18 +0100 Patrick Radizi <patrickr@axis.com>
+
+ * gst/rtp/gstrtph264pay.c:
+ rtph264pay: Fixes buffer leak when using SPS/PPS
+ Fixes a buffer leak that would occurr if the pipeline was shutdown
+ while a SPS/PPS header was being created.
+ https://bugzilla.gnome.org/show_bug.cgi?id=741271
+
+2014-12-09 04:43:29 +0100 Mathieu Duponchelle <mathieu.duponchelle@opencreed.com>
+
+ * gst/effectv/gstaging.c:
+ agingtv: fix memcpy when no color aging requested.
+ video_size is the size in pixels, actual size of the memcpy
+ has to be stride * height.
+
+2014-12-07 17:33:51 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
+
+ * sys/v4l2/gstv4l2bufferpool.c:
+ v4l2: Workaround libv4l2 RW emulation bug
+ When libv4l2 emulates RW mode on top of MMAP devices, the queues are
+ only initialized on first read. The problem is that poll() will fail
+ if called before the queues are initialized and streaming. Workaround
+ this by doing a zero size read when pool is started in that IO mode.
+ https://bugzilla.gnome.org/show_bug.cgi?id=740633
+
+2014-12-07 17:27:37 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
+
+ * sys/v4l2/gstv4l2bufferpool.c:
+ v4l2: Fix RW io mode
+ In RW, allocator can be null, max_buffers can be zero, and we need not
+ to wait while the queue is empty since there is no queue.
+ https://bugzilla.gnome.org/show_bug.cgi?id=740633
+
+2014-12-03 16:40:49 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * sys/v4l2/gstv4l2bufferpool.c:
+ v4l2bufferpool: Cleanup uneeded check and cases
+ There is nothing in between the break and the "done:" anymore, plus
+ USERPTR and DMABUF_IMPORT case is exactly the same.
+
+2014-12-03 17:07:49 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * sys/v4l2/gstv4l2bufferpool.c:
+ v4l2pool: Fix CREATE_BUFS support for capture
+ This patch fixes CREATE_BUFS support for capture devices. Initially we
+ would only try and allocate more buffers when the copy threshold
+ is reached. When the threshold was not set (needed) it would never
+ happen. Another problem is that on capture side, acquire returns
+ filled buffer, hence need to pool. We need to set a special flag to
+ force allocation to happen.
+ https://bugzilla.gnome.org/show_bug.cgi?id=741134
+
+2014-12-03 16:27:59 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * sys/v4l2/gstv4l2allocator.c:
+ v4l2allocator: Fix CREATE_BUF probing
+ Current for every memory type we where probing MMAP CREATE_BUFS ioct.
+ https://bugzilla.gnome.org/show_bug.cgi?id=741134
+
+2014-11-18 16:52:40 +0100 Nicola Murino <nicola.murino@gmail.com>
+
+ * gst/matroska/matroska-demux.c:
+ matroskademux: set framerate 0/1 when duration is not known
+ https://bugzilla.gnome.org/show_bug.cgi?id=740130
+
+2014-12-04 17:25:55 +1100 Jan Schmidt <jan@centricular.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: More fixes for reverse playback
+ When seeking or finding the previous keyframe, do
+ comparisons against targets and segments using composition time
+ to correctly decide which sample times match.
+
+2014-12-03 11:12:55 +0100 Thibault Saunier <tsaunier@gnome.org>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ rtpjitterbuffer: Use an empty iterator in iterate_internal_link when no links
+ We used to setup an iterator with 1 GValue set with a NULL object
+ pointer which is not the normal way to do that. Instead we should make
+ sure that the first call to gst_iterator_next returns GST_ITERATOR_DONE.
+
+2014-12-03 13:20:57 +1100 Jan Schmidt <jan@centricular.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Handle seeks past EOS as a seek to the end
+ Fix reverse playback of every frame by making seeks past/to EOS
+ find the last segment and start there.
+
+2014-12-02 15:33:25 -0500 Olivier Crête <olivier.crete@collabora.com>
+
+ * gst/rtp/gstrtpmpadepay.c:
+ rtpmpadepay: Relax caps to allow any clock-rate
+ Some Wowza setups seem to send an invalid non-90000 clock-rate.
+
+2014-12-01 21:04:02 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: don't use GST_CLOCK_TIME_NONE in non GstClockTime variables
+ Use -1 instead as those are gint64/guint64 variables and not GstClockTime
+
+2014-11-07 17:06:49 +0100 Aurélien Zanelli <aurelien.zanelli@parrot.com>
+
+ * sys/v4l2/gstv4l2allocator.h:
+ v4l2allocator: fix gst_v4l2_allocator_stop prototype
+ gst_v4l2_allocator_stop returns a GstV4l2Return, not a gboolean.
+ https://bugzilla.gnome.org/show_bug.cgi?id=739792
+
+2014-11-07 16:41:52 +0100 Aurélien Zanelli <aurelien.zanelli@parrot.com>
+
+ * sys/v4l2/gstv4l2bufferpool.c:
+ v4l2bufferpool: unref pool when v4l2_allocator_new() fails
+ https://bugzilla.gnome.org/show_bug.cgi?id=739791
+
+2014-11-30 17:52:47 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
+
+ * sys/v4l2/v4l2_calls.h:
+ v4l2: Remove last include to linux/videodev2.h
+ We now use and update our internal copy so we no longer have to ifdef
+ the entire code for features and defines that where added over the
+ years.
+ https://bugzilla.gnome.org/show_bug.cgi?id=740905
+
+2014-08-24 13:38:08 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/isomp4/qtdemux.c:
+ * gst/isomp4/qtdemux.h:
+ qtdemux: implement seeking in fragmented mp4 files in pull mode based on the mfra table
+
+2014-11-29 15:25:51 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: use track fragment decoding time (tfdt) in parse_trun() for interpolation
+ As fallback if we don't have any existing samples
+ as reference point yet.
+ Based on patch by David Corvoysier <david.corvoysier@orange.com>
+
+2014-11-29 14:37:25 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/isomp4/qtdemux.c:
+ * gst/isomp4/qtdemux.h:
+ qtdemux: parse mfra random access box for fragmented mp4 files
+ If it's present, and we operate in pull mode.
+
+2014-08-15 14:58:26 +0200 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: stop parsing headers for fragmented mp4s at the first moof
+ Currently during header parsing, we scan through the entire file
+ and skip every moof+mdat chunk for fragmented mp4s, which makes
+ start-up incredibly slow. Instead, just stop at the first moof
+ chunk when have a moov, and start exposing the streams, so we
+ can go and start handling the moofs for real.
+
+2014-11-29 13:59:35 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/icles/.gitignore:
+ * tests/icles/Makefile.am:
+ * tests/icles/gdkpixbufoverlay-test.c:
+ tests: add interactive gdkpixbufoverlay test
+ Just need to fix the coordinate system now so
+ that negative offsets are actually negative
+ and not flipped to position things from the
+ opposite border.
+
+2014-11-29 13:53:03 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * ext/gdk_pixbuf/gstgdkpixbufoverlay.c:
+ * ext/gdk_pixbuf/gstgdkpixbufoverlay.h:
+ gdkpixbufoverlay: add "pixbuf" property
+ So we can set a GdkPixbuf directly instead of
+ reading it from an image file on the file system.
+
+2014-11-29 13:23:50 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * ext/gdk_pixbuf/Makefile.am:
+ * ext/gdk_pixbuf/pixbufscale.c:
+ * ext/gdk_pixbuf/pixbufscale.h:
+ gdkpixbuf: remove pixbufscale code that was never ported
+ Don't think we'll need this again.
+
+2014-11-29 18:35:42 -0500 Olivier Crête <olivier.crete@collabora.com>
+
+ * gst/rtpmanager/gstrtprtxreceive.c:
+ rtprtxreceive: Use offset when copying header
+ The header is not always at the start of the packet, so we need to compute
+ the offset first.
+
+2014-11-28 13:12:46 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * ext/taglib/gstapev2mux.cc:
+ apev2mux: write APE tags at end for wavpack files
+ http://www.wavpack.com/file_format.txt:
+ "Both the APEv2 tags and/or ID3v1 tags must come at the end of the
+ WavPack file, with the ID3v1 coming last if both are present."
+ WavPack files that contain APEv2 tags at the beginning of the files
+ are unplayable on players that use FFmpeg (like VLC) and most other
+ software (except Banshee). Players that use libwavpack directly can
+ play the files because it skips the tags, but does not recognize the
+ tag data at that location.
+ https://bugzilla.gnome.org/show_bug.cgi?id=711437
+
+2014-11-28 10:41:55 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/icles/.gitignore:
+ * tests/icles/Makefile.am:
+ * tests/icles/test-segment-seeks.c:
+ tests: add interactive test for gapless playback using SEGMENT seeks
+ Not working too well yet, there are glitches even with WAV or FLAC.
+ https://bugzilla.gnome.org/show_bug.cgi?id=692368
+
+2014-11-26 10:33:09 +0300 Andrei Sarakeev <sarakusha@gmail.com>
+
+ * gst/videocrop/gstaspectratiocrop.c:
+ * gst/videocrop/gstaspectratiocrop.h:
+ aspectratiocrop: Handle resolution changes properly
+ When an caps-event is received, we must immediately change the crop
+ to videocrop correctly changed caps-event dimension, otherwise the
+ videocrop will first use the previous value of the crop that when
+ resizing video to a smaller resolution may cause an error.
+ https://bugzilla.gnome.org/show_bug.cgi?id=740671
+
+2014-11-27 17:10:53 +0100 Edward Hervey <bilboed@bilboed.com>
+
+ * common:
+ Automatic update of common submodule
+ From 7bb2bce to ef1ffdc
+
+2014-11-27 11:20:36 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/icles/test-accurate-seek.c:
+ test: use gst_util_uint64_scale_round() for timestamp to sample calculation
+
+2014-11-27 11:16:35 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/icles/.gitignore:
+ * tests/icles/Makefile.am:
+ * tests/icles/test-accurate-seek.c:
+ tests: add interactive test for accurate seeking
+ For some audio formats.
+ https://bugzilla.gnome.org/show_bug.cgi?id=655276
+
+2014-11-26 16:04:26 +0100 Edward Hervey <bilboed@bilboed.com>
+
+ * gst/isomp4/qtdemux.c:
+ isomp4: Check presence of mfhd in moof
+ The 'mfhd' atom is mandatory in 'moof'. We can later on check whether
+ the fragment number properly increases
+
+2014-11-26 15:59:36 +0100 Edward Hervey <bilboed@bilboed.com>
+
+ * gst/isomp4/qtdemux_dump.c:
+ isomp4: Fix mfro and tfra atom dumping
+ mfro was skipping the version/flags
+ tfra had wrong byte_reader return value checks
+
+2014-11-26 15:58:26 +0100 Edward Hervey <bilboed@bilboed.com>
+
+ * gst/isomp4/qtdemux_dump.c:
+ * gst/isomp4/qtdemux_dump.h:
+ * gst/isomp4/qtdemux_types.c:
+ isomp4: Add mfhd atom dumping
+
+2014-11-27 00:15:02 +1100 Jan Schmidt <jan@centricular.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Handle empty segments when seeking in reverse play.
+ Empty segments in an edit list have a media_start time of -1,
+ as they don't actually play any media. Allow for that when
+ aligning to the reference stream in reverse play.
+
+2014-11-24 10:36:54 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
+
+ * sys/v4l2/gstv4l2allocator.c:
+ Revert "v4l2allocator: Remove unused variable"
+ This reverts commit ad4480d53408a4d97ab531174ef37f258f3253c0.
+
+2014-11-24 10:36:30 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
+
+ * sys/v4l2/gstv4l2allocator.c:
+ * sys/v4l2/gstv4l2allocator.h:
+ * sys/v4l2/gstv4l2bufferpool.c:
+ * sys/v4l2/gstv4l2object.c:
+ * sys/v4l2/gstv4l2object.h:
+ Revert "v4l2: move vb_queue probing from allocator to v4l2object"
+ This reverts commit ec6b8b84af719d828ddd91c724e715c0b4a556bc.
+
+2014-11-24 10:33:29 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
+
+ * sys/v4l2/gstv4l2bufferpool.c:
+ * sys/v4l2/gstv4l2object.c:
+ Revert "v4l2object: allow to automatic selection of dmabuf"
+ This reverts commit e6c2ad5571e5dedb212287efe238eb450032cd4f.
+
+2014-11-23 16:34:15 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * REQUIREMENTS:
+ REQUIREMENTS: update a little
+ People actually look at that it seems.
+
+2014-11-23 16:22:12 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/icydemux/Makefile.am:
+ icydemux: does not need to link against zlib
+
+2014-11-22 21:28:35 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * configure.ac:
+ * ext/speex/gstspeexdec.h:
+ * ext/speex/gstspeexenc.h:
+ speex: remove support for ancient speex versions
+
+2014-11-21 11:21:18 +0100 Branislav Katreniak <bkatreniak@nuvotechnologies.com>
+
+ * ext/soup/gstsouphttpsrc.c:
+ souphttpsrc: log connection events at info level
+ https://bugzilla.gnome.org/show_bug.cgi?id=739305
+
+2014-10-20 13:00:37 +0200 Miguel París Díaz <mparisdiaz@gmail.com>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ rtpjitterbuffer: ensure rtx_retry_period >= 0
+ https://bugzilla.gnome.org/show_bug.cgi?id=739344
+
+2014-11-21 11:44:24 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
+
+ * sys/v4l2/gstv4l2allocator.c:
+ v4l2allocator: Remove unused variable
+ this was introduced by commit ec6b8b
+ https://bugzilla.gnome.org/show_bug.cgi?id=699382
+
+2014-11-16 12:34:17 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
+
+ * sys/v4l2/gstv4l2bufferpool.c:
+ * sys/v4l2/gstv4l2bufferpool.h:
+ * sys/v4l2/gstv4l2src.c:
+ * sys/v4l2/gstv4l2transform.c:
+ * sys/v4l2/gstv4l2videodec.c:
+ v4l2: Handle corrupted buffer with empty payload
+ This allow skipping buffer flagged with ERROR that has no payload.
+ This is typical behaviour when a recovererable error occured during
+ capture in the driver, but that no valid data was ever written into that
+ buffer. This patch also translate V4L2_BUF_FLAG_ERROR into
+ GST_BUFFER_FLAG_CORRUPTED. Hence decoding error produce
+ by decoder due to missing frames will now be correctly marked. Finally,
+ this fixes a buffer leak when EOS is reached.
+ https://bugzilla.gnome.org/show_bug.cgi?id=740040
+
+2014-11-21 16:36:15 +0100 Benjamin Gaignard <benjamin.gaignard@linaro.org>
+
+ * sys/v4l2/gstv4l2bufferpool.c:
+ * sys/v4l2/gstv4l2object.c:
+ v4l2object: allow to automatic selection of dmabuf
+ If the v4l2 queue support dmabuf select this buffer pool mode
+ and update the query with allocator.
+ This patch only concern exporting dmabuf and not importing dmabuf
+ fd from downstream element.
+ https://bugzilla.gnome.org/show_bug.cgi?id=699382
+
+2014-11-21 16:13:05 +0100 Benjamin Gaignard <benjamin.gaignard@linaro.org>
+
+ * sys/v4l2/gstv4l2allocator.c:
+ * sys/v4l2/gstv4l2allocator.h:
+ * sys/v4l2/gstv4l2bufferpool.c:
+ * sys/v4l2/gstv4l2object.c:
+ * sys/v4l2/gstv4l2object.h:
+ v4l2: move vb_queue probing from allocator to v4l2object
+ The goal is to make those information available in v4l2_object
+ to be able later to select the best allocation method for the pool
+ https://bugzilla.gnome.org/show_bug.cgi?id=699382
+
+2014-11-20 22:42:59 +0530 Arun Raghavan <git@arunraghavan.net>
+
+ * gst/rtpmanager/gstrtpbin.h:
+ rtpbin: Fix up new_jitterbuffer signal prototype
+
+2014-11-20 20:19:25 +0530 Arun Raghavan <git@arunraghavan.net>
+
+ * gst/rtpmanager/gstrtpbin.c:
+ rtpbin: Document how to control per-SSRC retransmission
+
+2014-11-20 20:18:45 +0530 Arun Raghavan <git@arunraghavan.net>
+
+ * docs/design/design-rtpretransmission.txt:
+ doc: Trivial spelling and consistency update
+
+2014-11-20 13:14:14 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtp/gstrtpgstdepay.c:
+ * gst/rtp/gstrtpgstpay.c:
+ rtpgstpay: put 0-byte at the end of events
+ Put a 0-byte at the end of the event string. Does not break ABI because
+ old depayloaders will skip the 0 byte (which is included in the length).
+ Expect a 0-byte at the end of the event string or a ; for old
+ payloaders.
+ See https://bugzilla.gnome.org/show_bug.cgi?id=737591
+
+2014-11-20 12:40:28 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtp/gstrtpgstdepay.c:
+ rtpgstdepay: avoid buffer overread.
+ Check that a caps event string is 0 terminated and the event string is
+ terminated with a ; to avoid buffer overreads.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=737591
+
+2014-11-20 10:45:07 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/isomp4/gstqtmuxmap.c:
+ qtmux: don't limit max video resolution to 4096x4096
+ MAX isn't entirely correct as upper limit either,
+ it should really be MAXUINT32, but it's unlikely
+ to be a problem in the near future.
+ https://bugzilla.gnome.org/show_bug.cgi?id=740407
+
+2014-11-19 15:06:00 -0800 Aleix Conchillo Flaqué <aleix@oblong.com>
+
+ * gst/rtsp/gstrtspsrc.c:
+ rtspsrc: fix leak for mikey base64 decoded key-mgmt
+ https://bugzilla.gnome.org/show_bug.cgi?id=740392
+
+2014-11-20 09:01:38 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/videofilter/gstvideobalance.c:
+ videobalance: fix unhandled format in passthrough
+ In passthrough we can handle all formats.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=740387
+
+2014-11-19 16:12:38 +0100 Jan Alexander Steffens (heftig) <jsteffens@make.tv>
+
+ * gst/flv/gstflvdemux.c:
+ flvdemux: Restrict resyncing to TS regressions
+ The behavior of resyncing video and audio indepen-
+ dently can cause A/V desyncs. Lets restrict resyncs
+ to jumps backward for now.
+ https://bugzilla.gnome.org/show_bug.cgi?id=736397
+
+2014-11-17 23:16:03 +1100 Matthew Waters <matthew@centricular.com>
+
+ * gst/videomixer/videomixer2.c:
+ * gst/videomixer/videomixer2.h:
+ videomixer: fix up QoS handling for live sources
+ Only attempt adaptive drop when we are not live
+ https://bugzilla.gnome.org/show_bug.cgi?id=739996
+
+2014-11-10 22:34:39 +0100 Henning Heinold <henning@itconsulting-heinold.de>
+
+ * tests/examples/rtp/client-PCMA.py:
+ * tests/examples/rtp/server-alsasrc-PCMA.py:
+ examples: port python rtp PCMA client/server tests to 1.0
+ https://bugzilla.gnome.org/show_bug.cgi?id=739930
+
+2014-06-04 12:11:10 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * ext/flac/gstflacdec.c:
+ flacdec: set the channel positions using the appropriate API
+ This avoids _set_format setting the unpositioned flag when passed
+ NULL as channel positions, as it would not be cleared when setting
+ actual channel positions later.
+
+2014-11-01 22:39:41 +0100 Aurélien Zanelli <aurelien.zanelli@darkosphere.fr>
+
+ * ext/vpx/gstvp8enc.c:
+ * ext/vpx/gstvp9enc.c:
+ vpx: mark arnr-type properties as deprecated and set them to no-op
+ ARNR type control in libvpx has been deprecated so this commit mark the
+ vp8enc and vp9enc associated properties as deprecated and change their
+ behavior to just display a warning message.
+ https://bugzilla.gnome.org/show_bug.cgi?id=739476
+
+2014-11-10 13:16:01 +0530 Arun Raghavan <git@arunraghavan.net>
+
+ * gst/rtpmanager/gstrtpbin.c:
+ rtpmanager: Trivial typo fix
+
+2014-11-09 11:04:33 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/matroska/matroska-mux.c:
+ matroska-mux: Use G_DEFINE_TYPE() to register the pad instead of manually registering it
+
+2014-11-06 15:37:28 +0100 Göran Jönsson <goranjn@axis.com>
+
+ * gst/matroska/matroska-mux.c:
+ matroskamux: make GstMatroskamuxPad get_type() function thread-safe
+ https://bugzilla.gnome.org/show_bug.cgi?id=739722
+
+2014-11-07 16:11:24 +0100 Aurélien Zanelli <aurelien.zanelli@parrot.com>
+
+ * sys/v4l2/gstv4l2allocator.c:
+ v4l2allocator: fix error message if allocator is already active
+ https://bugzilla.gnome.org/show_bug.cgi?id=739789
+
+2014-11-06 21:21:40 -0500 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
+
+ * sys/v4l2/gstv4l2bufferpool.c:
+ v4l2bufferpool: Improve buffer validation
+ Improve buffer validation by making sure each memory are the right
+ one and that each memory is writable. This fixes tearing issues in
+ case downstream uses gst_buffer_make_writable() or other type
+ of GstBuffer copy where memory are only reffed.
+ https://bugzilla.gnome.org/show_bug.cgi?id=739754
+
+2014-11-06 21:38:43 +0100 Josep Torra <n770galaxy@gmail.com>
+
+ * gst/rtsp/Makefile.am:
+ rtsp: fix build in gst-uninstalled setup
+
+2014-10-29 18:44:43 +0100 Thibault Saunier <tsaunier@gnome.org>
+
+ * gst/imagefreeze/gstimagefreeze.c:
+ * gst/imagefreeze/gstimagefreeze.h:
+ imagefreeze: Handle seqnums
+ https://bugzilla.gnome.org/show_bug.cgi?id=739366
+
+2014-11-04 08:18:41 +0530 Vineeth T M <vineeth.tm@samsung.com>
+
+ * ext/libpng/gstpngdec.c:
+ * ext/libpng/gstpngdec.h:
+ pngdec: change parse logic
+ Right now in parse logic the signature is checked every time the parse function
+ is called, and the whole data is the scanned each and every time, even though the
+ data is scanned in the previous instance. Changing the logic such that, we skip
+ the bytes which are already scanned in the previous instances of parse. This
+ helps in avoiding multiple scan of already scanned data/signature.
+ https://bugzilla.gnome.org/show_bug.cgi?id=737708
+
+2014-11-03 15:26:06 +0100 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/videomixer/videomixer2.c:
+ videomixer2: reverse order of params for converter
+
+2014-11-03 11:44:28 +0100 Aurélien Zanelli <aurelien.zanelli@parrot.com>
+
+ * sys/v4l2/gstv4l2bufferpool.c:
+ v4l2bufferpool: fix typo in flags
+ https://bugzilla.gnome.org/show_bug.cgi?id=739549
+
+2014-11-02 23:33:23 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * sys/v4l2/gstv4l2object.c:
+ v4l2src: fix a couple of minor leaks
+
+2014-11-02 19:42:03 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/goom2k1/gstgoom.c:
+ * gst/goom2k1/gstgoom.h:
+ goom2k1: post QoS messages when dropping frames due to QoS
+
+2014-11-02 19:29:52 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/goom/gstgoom.c:
+ * gst/goom/gstgoom.h:
+ goom: post QoS messages when dropping frames due to QoS
+
+2014-11-02 19:02:35 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/matroska/matroska-mux.c:
+ matroskamux: tweak writing app tag string a little
+
+2014-11-02 16:51:23 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * ext/jpeg/gstjpegdec.c:
+ * gst/isomp4/gstqtmux.c:
+ * gst/level/gstlevel.c:
+ * gst/udp/gstmultiudpsink.c:
+ * gst/udp/gstudpsrc.c:
+ Sprinkle some G_PARAM_DEPRECATED and #ifndef GST_REMOVE_DEPRECATED
+
+2014-11-02 16:58:07 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/check/elements/level.c:
+ tests: don't use deprecated property in level unit test
+
+2014-11-02 13:06:33 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ rtpjitterbuffer: implement get/set for new rtx-min-retry-timeout property
+ Properties are so much more useful if you can actually set
+ and get their values.
+
+2014-10-30 17:41:19 +0000 Simon Farnsworth <simon.farnsworth@onelan.co.uk>
+
+ * sys/v4l2/gstv4l2bufferpool.c:
+ * sys/v4l2/gstv4l2object.c:
+ * sys/v4l2/gstv4l2src.c:
+ v4l2: Clean up interlace support
+ Rather than try and guess interlace support as part of checking supported
+ sizes, look for interlace support specifically in its own function.
+ As a cleanup, use V4L2_FIELD_ANY when probing sizes, which should result in
+ the driver doing the right thing.
+ With my capture setup, this gets me the following sample caps:
+ For 1080i resolution:
+ video/x-raw, format=(string)YUY2, width=(int)1920, height=(int)1080, pixel-aspect-ratio=(fraction)1/1, interlace-mode=(string)interleaved, framerate=(fraction){ 25/1, 30/1 }
+ For 720p resolution:
+ video/x-raw, format=(string)YUY2, width=(int)1280, height=(int)720, pixel-aspect-ratio=(fraction)1/1, interlace-mode=(string)progressive, framerate=(fraction){ 50/1, 60/1 }
+ For 576i/p resolution (both possible at the point of query):
+ video/x-raw, format=(string)YUY2, width=(int)720, height=(int)576, pixel-aspect-ratio=(fraction)1/1, interlace-mode=(string){ progressive, interleaved }, framerate=(fraction){ 25/1, 50/1 }
+ This, in turn, makes 576i work correctly; with the old code,
+ the caps would be interlace-mode=progressive for interlaced video.
+ https://bugzilla.gnome.org/show_bug.cgi?id=726194
+
+2014-11-01 12:18:02 +0100 Aurélien Zanelli <aurelien.zanelli@darkosphere.fr>
+
+ * ext/vpx/gstvp8utils.h:
+ vpx: remove compatibility defines
+ We are guaranteed to have VPX_IMG_FMT_I420, VPX_PLANE_Y,
+ VPX_PLANE_U and VPX_PLANE_V as we require libvpx > 1.1.0.
+ https://bugzilla.gnome.org/show_bug.cgi?id=739476
+
+2014-11-01 15:33:23 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * configure.ac:
+ * ext/wavpack/gstwavpackcommon.c:
+ * ext/wavpack/gstwavpackdec.c:
+ * ext/wavpack/gstwavpackenc.c:
+ wavpack: remove support for ancient API version
+
+2014-11-01 10:14:31 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
+
+ * gst/rtp/gstrtpvp8depay.c:
+ * gst/rtp/gstrtpvp8pay.c:
+ rtpvp8: Use VP8 encoding name
+ Both Firefox and Chrome uses VP8 as the encoding in their SDP.
+ Adding this now defacto standard name removes the need for special
+ case in SDP parsing code.
+ https://bugzilla.gnome.org/show_bug.cgi?id=737810
+
+2014-11-01 11:59:26 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/rtp/gstrtpmp2tpay.c:
+ rtpmp2tpay: fix up template caps so we can output the default pt 33
+ Add fixed payload type for mp2t to template caps as well, so
+ our output caps match the advertised default pt. Fixes a
+ regression from 1.2.
+ There's still something wrong with caps negotiation though,
+ rtpmp2tpay payload=96 ! fakesink will not output caps with
+ payload=96.
+
+2014-10-30 15:37:36 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
+
+ * gst/rtsp/gstrtspsrc.c:
+ rtspsrc: mikey related memory leaks
+ https://bugzilla.gnome.org/show_bug.cgi?id=739430
+
+2014-06-10 10:04:07 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * ext/speex/gstspeexenc.c:
+ * ext/speex/gstspeexenc.h:
+ speexenc: update output segment stop time to match clipped samples
+ This will let oggmux generate a granpos on the last page that properly
+ represents the clipped samples at the end of the stream.
+
+2014-06-10 10:59:13 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * ext/flac/gstflacenc.c:
+ * ext/flac/gstflacenc.h:
+ flacenc: update output segment stop time to match clipped samples
+ This will let oggmux generate a granpos on the last page that properly
+ represents the clipped samples at the end of the stream.
+
+2014-10-07 15:29:33 +0200 Aurélien Zanelli <aurelien.zanelli@parrot.com>
+
+ * sys/v4l2/gstv4l2bufferpool.c:
+ v4l2bufferpool: cleanly handle streamon failure for output device
+ On streamon failure, the queued buffer is not released from the
+ bufferpool class point of view because it is queued to the driver and
+ the flush logic is not performed since we are not in streaming state.
+ It causes the v4l2 bufferpool to always return that stop method failed
+ and to leak v4l2 objects and buffers.
+ This commit solve this by performing the flush logic in error case, ie
+ flushing the allocator and restoring queued buffer state to non-queued.
+ https://bugzilla.gnome.org/show_bug.cgi?id=738102
+
+2014-10-08 10:31:21 +0200 Aurélien Zanelli <aurelien.zanelli@parrot.com>
+
+ * sys/v4l2/gstv4l2bufferpool.c:
+ v4l2bufferpool: implement dispose method
+ Unref objects in dispose method rather than in finalize in order to
+ prevent circular reference.
+ https://bugzilla.gnome.org/show_bug.cgi?id=738102
+
+2014-10-08 10:35:14 +0200 Aurélien Zanelli <aurelien.zanelli@parrot.com>
+
+ * sys/v4l2/gstv4l2bufferpool.c:
+ v4l2bufferpool: check that allocator is non null when stopping pool
+ Otherwise, we could dereference NULL allocator when the stop method is
+ called by the GstBufferPool's finalize method.
+ https://bugzilla.gnome.org/show_bug.cgi?id=738102
+
+2014-10-09 12:15:05 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * sys/v4l2/gstv4l2sink.c:
+ v4l2sink: Implement unlock/unlock_stop
+ This will prevent deadlocks, but will also properly flush the pool and allocator
+ when going to READY state. It should also fix issues reported on mailing list
+ when seeking is performed.
+ https://bugzilla.gnome.org/show_bug.cgi?id=738152
+
+2014-10-28 21:32:06 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * ext/pulse/pulsedeviceprovider.h:
+ * sys/v4l2/gstv4l2deviceprovider.h:
+ * sys/v4l2/gstv4l2tuner.h:
+ pulse, v4l2: add missing G_END_DECLS in some places
+
+2014-10-27 17:57:20 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * common:
+ Automatic update of common submodule
+ From 84d06cd to 7bb2bce
+
+2014-10-27 11:08:20 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * tests/check/elements/aacparse.c:
+ aacparse: Fix unit test now that we always have profile/level in the caps
+
+2014-10-26 14:55:49 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * Makefile.am:
+ Parallelise 'make check-valgrind'
+ Some of the RTP unit tests are very flaky and will
+ fail more often with the CPU maxed out fully. Those
+ tests need to be fixed in any case though, they also
+ fail on slower machines and also occasionally with
+ normal 'make check'.
+
+2014-10-26 11:47:25 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/audioparsers/gstaacparse.c:
+ aacparse: Always set profile/level on the caps
+ We have the information already, so why not use it?
+
+2014-10-25 12:36:02 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ rtpjitterbuffer: fix crash on some 32-bit systems
+ Make sure to pass right number of bits to gst_structure_new()
+ which is a vararg function.
+ Fixes elements/rtpaux unit test on ppc32.
+
+2014-10-25 00:56:02 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/check/elements/rgvolume.c:
+ tests: fix rgvolume test on big-endian systems
+
+2014-10-25 00:53:39 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/check/Makefile.am:
+ * tests/check/elements/mulawdec.c:
+ * tests/check/elements/mulawenc.c:
+ tests: fix mulawdec/mulawenc test for big endian systems
+
+2014-10-24 23:48:30 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/interleave/interleave.c:
+ interleave: intersect result with filter caps in caps query
+ Fixes crash in audiotestsrc because of an unsupported format
+ getting negotiated on big-endian systems with
+ audiotestsrc ! interleave ! audioconvert ! wavenc
+
+2014-10-23 15:46:13 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * ext/pulse/pulsedeviceprovider.c:
+ * ext/pulse/pulsedeviceprovider.h:
+ pulse: remove some unused typedefs
+
+2014-10-22 15:28:44 +0200 Ananda <ananda@latelier23.com>
+
+ * ext/speex/gstspeexdec.c:
+ * ext/speex/gstspeexenc.c:
+ speex: Fix segfault when resetting the codecs multiple times
+ https://bugzilla.gnome.org/show_bug.cgi?id=738793
+
+2014-10-22 22:50:54 +0530 Arun Raghavan <arun@accosted.net>
+
+ * ext/pulse/pulsesink.c:
+ pulsesink: Temporarily disable stream status posting
+ We need a mechanism in PulseAudio to allow running code outside the
+ mainloop lock. Then we'd be able to post to the bus (taking the
+ GST_OBJECT_LOCK), without worrying about locking order with the mainloop
+ lock, which is the current cause of deadlocks while trying to post the
+ stream status messages.
+ https://bugzilla.gnome.org/show_bug.cgi?id=736071
+
+2014-10-22 15:04:24 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ rtpjitterbuffer: limit the retry frequency
+ When the RTT and jitter are very low (such as on a local network), the
+ calculated retransmission timeout is very small. Set some sensible lower
+ boundary to the timeout by adding a new property. We use the packet
+ spacing as a lower boundary by default.
+
+2014-10-22 13:40:58 +0200 Miguel París Díaz <mparisdiaz@gmail.com>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ gstrtpjitterbuffer: add "rtx-min-delay" property
+ This property is useful to set a min time to wait before sending a
+ retransmission event.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=735378
+
+2014-10-22 13:29:48 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ jitterbuffer: Refactor code
+ Refactor some code dealing with calculating various timeouts.
+ See https://bugzilla.gnome.org/show_bug.cgi?id=735378
+
+2014-10-10 19:50:06 +0200 Miguel París Díaz <mparisdiaz@gmail.com>
+
+ * gst/rtpmanager/rtpsession.c:
+ * gst/rtpmanager/rtpsession.h:
+ rtpsession: fix Early Feedback Transmission
+ In early retransmission we are allowed to schedule 1 regular RTCP packet
+ at an earlier time. When we do that, we need to set allow_early to FALSE
+ and ignore/drop (or merge) all future requests for early transmission.
+ We now first check if we can schedule an early RTCP and if we can,
+ actually prepare the data for the next RTCP interval.
+ After we send the next regular RTCP after the early RTCP, we set
+ allow_early to TRUE again to allow more early requests.
+ Remove the condition for the immediate feedback for now.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=738319
+
+2014-10-21 13:01:32 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * common:
+ Automatic update of common submodule
+ From a8c8939 to 84d06cd
+
+2014-10-21 13:10:24 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ rtpjitterbuffer: make debug line less confusing
+
+2014-10-21 12:58:13 +0200 Stefan Sauer <ensonic@users.sf.net>
+
+ * README:
+ * common:
+ Automatic update of common submodule
+ From 36388a1 to a8c8939
+
+2014-07-02 17:50:35 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/rtpmanager/rtpjitterbuffer.c:
+ * gst/rtpmanager/rtpjitterbuffer.h:
+ jitterbuffer: rework resync handling
+ Add a need-resync state, this is when we need to try to lock on to a
+ time/RTPtime pair.
+ Always check the RTP timestamps and if they go backwards, mark ourselves
+ as need-resync.
+ Only resync when need-resync is TRUE and we have a valid time. Otherwise
+ we keep the old values. This avoids locking on to an invalid time and
+ causing us to timestamp everything with -1.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=730417
+
+2014-10-03 17:28:06 -0700 Aleix Conchillo Flaqué <aleix@oblong.com>
+
+ * gst/rtsp/gstrtspsrc.c:
+ rtspsrc: set full stream caps on internal src TCP pads
+ Set the complete stream caps on the TCP internal src pads. Otherwise,
+ ptdemux will not properly detect the caps change.
+ https://bugzilla.gnome.org/show_bug.cgi?id=737868
+
+2014-10-17 22:23:27 +0200 Sjoerd Simons <sjoerd@luon.net>
+
+ * gst/rtpmanager/gstrtpmux.c:
+ * tests/check/elements/rtpmux.c:
+ rtpmux: Don't set PROXY_CAPS flag on the src pad
+ rtpmux behaves like a funnel in that it forwards whatever upstream is
+ sending buffers. So setting proxy caps doesn't make sense as the
+ upstream don't have to have compatible caps, thus resulting in an empty
+ caps set as a result of a caps query. Instead set fixed caps just
+ as funnel does.
+ https://bugzilla.gnome.org/show_bug.cgi?id=738722
+
+2014-10-20 11:57:38 +0530 Vineeth T M <vineeth.tm@samsung.com>
+
+ * gst/videobox/gstvideobox.c:
+ videobox: critical error when element properties set as max/min
+ left, right, top, bottom can be set from range of -2147483648 to 2147483647
+ when i launch the videobox element with that values, it gives a critical error
+ (gst-check-1.0:29869): GStreamer-CRITICAL **: gst_value_set_int_range_step: assertion 'start < end' failed
+ This happens because min cannot be equal to max.
+ https://bugzilla.gnome.org/show_bug.cgi?id=738838
+
+2014-10-15 17:45:24 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/rtp/Makefile.am:
+ * gst/rtp/gstrtp.c:
+ * gst/rtp/gstrtph265depay.c:
+ * gst/rtp/gstrtph265depay.h:
+ * gst/rtp/gstrtph265pay.c:
+ * gst/rtp/gstrtph265pay.h:
+ Revert "rtp: add h265 RTP payloader + depayloader"
+ This reverts commit d06ba9051f904a7eb482c07a97a1827169158663.
+ This breaks the build, as it depends on parser API in -bad.
+
+2014-10-15 17:34:50 +0200 Jurgen Slowack <jurgen.slowack@barco.com>
+
+ * gst/rtp/Makefile.am:
+ * gst/rtp/gstrtp.c:
+ * gst/rtp/gstrtph265depay.c:
+ * gst/rtp/gstrtph265depay.h:
+ * gst/rtp/gstrtph265pay.c:
+ * gst/rtp/gstrtph265pay.h:
+ rtp: add h265 RTP payloader + depayloader
+
+2014-10-05 21:24:27 +0200 Peter G. Baum <peter@dr-baum.net>
+
+ * gst/wavenc/gstwavenc.c:
+ * gst/wavenc/gstwavenc.h:
+ wavenc: Support RF64 format
+ https://bugzilla.gnome.org/show_bug.cgi?id=725145
+
+2014-10-11 11:18:42 +1100 David Sansome <me@davidsansome.com>
+
+ * gst/equalizer/gstiirequalizer.c:
+ equalizer: Don't call iirequalizer's transform_ip in passthrough mode
+ It tries to map the read-only buffer with GST_MAP_READWRITE and crashes.
+ https://bugzilla.gnome.org/show_bug.cgi?id=737886
+
+2014-10-10 18:30:07 -0400 Olivier Crête <olivier.crete@ocrete.ca>
+
+ * gst/rtpmanager/rtpsource.c:
+ * gst/rtpmanager/rtpsource.h:
+ rtpsource: Rename seqnum-base to seqnum-offset in caps
+ This was modified back in 1.0 in GstRtpBasePayload
+
+2014-10-10 18:11:19 -0400 Olivier Crête <olivier.crete@ocrete.ca>
+
+ * gst/dtmf/gstrtpdtmfsrc.c:
+ * tests/check/elements/dtmf.c:
+ rtpdtmfsrc: clock-base and seqnum-base -> timestamp-offset and seqnum-offset
+ These were renamed in GstRTPBasePayload in 1.0
+
+2014-10-10 17:30:24 -0400 Olivier Crête <olivier.crete@ocrete.ca>
+
+ * gst/rtpmanager/gstrtpmux.c:
+ * gst/rtpmanager/gstrtpmux.h:
+ * tests/check/elements/rtpmux.c:
+ rtpmux: clock-base and seqnum-base -> timestamp-offset and seqnum-offset
+ These were renamed in GstRTPBasePayload in 1.0
+
+2014-10-06 14:23:22 +0100 Luis de Bethencourt <luis.bg@samsung.com>
+
+ * gst/goom2k1/filters.c:
+ goom2k1: removing block of code that does nothing
+ The loop in zoomFilterSetResolution is meant to change the values in the
+ zf->firedec[] array. Each iteration writes the value of decc onto the arrya,
+ but no conditions that change the value of decc are ever met and the array is
+ filled with zero for each element. Which is the initial state of the
+ array before the loop begins.
+ The loop does nothing.
+ https://bugzilla.gnome.org/show_bug.cgi?id=728353
+
+2014-10-04 17:17:13 +0200 Stefan Sauer <ensonic@users.sf.net>
+
+ * gst/rtpmanager/rtpjitterbuffer.c:
+ rtpjitterbuffer: don't log all clock_rate changes as warnings.
+ We never initialize clock_rate explicitly, therefore it is 0 by default. The
+ parameter is a uint32 and the only caller ensure that it is >0, therefore it
+ won't become -1 ever.
+
+2014-10-02 14:26:08 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
+
+ * ext/soup/gstsouphttpclientsink.c:
+ souphttpclientsink: Fix lifetime of stream headers and queued buffers
+ Stream headers are updated whenever ::set_caps is called, so we can't assume
+ they'll be valid before the message body is written out. We *can* assume that
+ for queued buffers, but SOUP_MEMORY_STATIC is still wrong for those.
+ Also, add some debug logging for stream header interactions.
+ https://bugzilla.gnome.org/show_bug.cgi?id=737771
+
+2014-10-02 03:26:22 +0200 Matej Knopp <matej.knopp@gmail.com>
+
+ * gst/audioparsers/gstaacparse.c:
+ aacparse: fix memory leak when prepending ADTS headers
+ https://bugzilla.gnome.org/show_bug.cgi?id=737761
+
+2014-09-23 10:48:09 +0200 Antonio Ospite <ao2@ao2.it>
+
+ * gst/interleave/interleave.c:
+ * gst/interleave/interleave.h:
+ interleave: interleave samples following the Default Channel Ordering
+ In order to have a full mapping between channel positions in the audio
+ stream and loudspeaker positions, the channel-mask alone is not enough:
+ the channels must be interleaved following some Default Channel Ordering
+ as mentioned in the WAVEFORMATEXTENSIBLE[1] specification.
+ As a Default Channel Ordering use the one implied by
+ GstAudioChannelPosition which follows the ordering defined in SMPTE
+ 2036-2-2008[2].
+ NOTE that the relative order in the Top Layer is not exactly the same as
+ the one from the WAVEFORMATEXTENSIBLE[1] specification; let's hope users
+ using so may channels are already aware of such discrepancies.
+ [1] http://msdn.microsoft.com/en-us/library/windows/hardware/dn653308%28v=vs.85%29.aspx
+ [2] http://www.itu.int/dms_pub/itu-r/opb/rep/R-REP-BS.2159-2-2011-PDF-E.pdf
+ Fixes: https://bugzilla.gnome.org/show_bug.cgi?id=737127
+
+2014-10-02 10:10:11 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/wavenc/gstwavenc.c:
+ wavenc: Send CAPS event after the pad was activated
+ Otherwise the CAPS event will be dropped and we never configure any caps at
+ all, leading to weird behaviour in many situations. Especially header
+ rewriting is not going to work if a capsfilter is after wavenc.
+ https://bugzilla.gnome.org/show_bug.cgi?id=737735
+
+2014-10-01 23:12:30 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
+
+ * ext/soup/gstsouphttpclientsink.c:
+ souphttpclientsink: Add some more useful debug logging
+
+2014-10-01 23:05:03 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
+
+ * ext/soup/gstsouphttpclientsink.c:
+ souphttpclientsink: Free queued buffers in ::reset
+ ::render sets a new callback for writing out new buffers only if there aren't
+ already buffers queued for writing with a previously-scheduled callback.
+ However, if the previously-scheduled callback is interrupted by a state change
+ (either manually or due to an error) and there are still buffers in the queue,
+ restarting the pipeline will result in buffers being queued forever, and no
+ callbacks will ever be scheduled, and no buffers will be written out.
+ https://bugzilla.gnome.org/show_bug.cgi?id=737739
+
+2014-10-01 17:29:29 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/videomixer/videomixer2.c:
+ videomixer: Actually use the correct GstVideoInfo for conversion
+
+2014-10-01 17:24:59 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/videomixer/videomixer2.c:
+ videomixer: Revert the last commit and handle resolutions differences properly
+ This is about converting the format, not about converting any widths and
+ heights. Subclasses are expected to handler different resolutions themselves,
+ like the videomixers already do properly.
+
+2014-10-01 17:12:59 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/videomixer/videomixer2.c:
+ videomixer: GstVideoConverter currently can't rescale and will assert
+ Leads to ugly assertions instead of properly erroring out:
+ CRITICAL **: gst_video_converter_new: assertion 'in_info->width == out_info->width' failed
+
+2014-09-30 11:35:12 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * ext/vpx/gstvp8enc.c:
+ * ext/vpx/gstvp9enc.c:
+ vp8enc/vp9enc: Protect the encoder with a mutex in all situations
+
+2014-09-30 11:31:43 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * ext/vpx/gstvp9enc.c:
+ vp9enc: Allow caps renegotiation
+ https://bugzilla.gnome.org/show_bug.cgi?id=726329
+
+2014-09-30 11:28:39 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * ext/vpx/gstvp8enc.c:
+ vp8enc: finish() and drain() should return a GstFlowReturn
+
+2014-03-14 12:59:02 +0100 Jose Antonio Santos Cadenas <santoscadenas@gmail.com>
+
+ * ext/vpx/gstvp8enc.c:
+ vp8enc: Allow caps renegotiation
+ https://bugzilla.gnome.org/show_bug.cgi?id=726329
+
+2014-09-29 11:49:45 +0200 Aurélien Zanelli <aurelien.zanelli@parrot.com>
+
+ * sys/v4l2/gstv4l2object.c:
+ v4l2object: set colorspace for output devices
+ When the v4l2 device is an output device, the application shall set the
+ colorspace. So map GStreamer colorimetry info to V4L2 colorspace and set
+ on set_format. In case we have no colorimetry information, we try to
+ guess it according to pixel format and video size.
+ https://bugzilla.gnome.org/show_bug.cgi?id=737579
+
+2014-09-29 22:48:16 +0530 Arun Raghavan <arun@accosted.net>
+
+ * ext/pulse/pulsesink.c:
+ * ext/pulse/pulsesrc.c:
+ pulse: Add some documentation about threading and synchronisation
+ This gives a quick introduction to how the pulsesink/pulsesrc code
+ interacts with the pa_threaded_mainloop that we start up to communicate
+ with the server.
+
+2014-09-29 20:18:08 +0530 Arun Raghavan <arun@accosted.net>
+
+ * ext/pulse/pulsesink.c:
+ pulsesink: Make emitting stream status messages synchronous
+ The stream status messages are emitted in the PA mainloop thread, which
+ means the mainloop lock is taken, followed by the Gst object lock (by
+ gst_element_post_message()). In all other locations, the order of
+ locking is reversed (this is unavoidable in a bunch of cases where the
+ object lock is taken by GstBaseSink or GstAudioBaseSink, and then we get
+ control to take the mainloop lock).
+ The only way to guarantee that the defer callback for stream status
+ messages doesn't deadlock is to either stop posting those messages, or
+ make sure that the message emission is completed before we proceed to
+ any point that might take the object lock before the mainloop lock
+ (which is what we do after this patch).
+ https://bugzilla.gnome.org/show_bug.cgi?id=736071
+
+2014-09-16 12:12:49 +0200 Antonio Ospite <ao2@ao2.it>
+
+ * gst/wavenc/gstwavenc.c:
+ wavenc: print channel masks in hexadecimal
+
+2014-09-27 16:01:21 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * sys/v4l2/gstv4l2deviceprovider.h:
+ v4l2: remove redundant struct declaration
+
+2014-09-26 13:46:16 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp/gstrtspsrc.c:
+ rtspsrc: Fix compiler warnings
+ gstrtspsrc.c:7939:11: error: implicit conversion from enumeration type 'GstSDPResult' to different enumeration type
+ 'GstRTSPResult' [-Werror,-Wenum-conversion]
+ res = gst_sdp_message_new (&sdp);
+ ~ ^~~~~~~~~~~~~~~~~~~~~~~~~~
+ gstrtspsrc.c:7944:11: error: implicit conversion from enumeration type 'GstSDPResult' to different enumeration type
+ 'GstRTSPResult' [-Werror,-Wenum-conversion]
+ res = gst_sdp_message_parse_uri (uri, sdp);
+ ~ ^~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~
+
+2014-09-25 15:01:14 +0200 Jonas Holmberg <jonashg@axis.com>
+
+ * gst/matroska/matroska-demux.c:
+ matroskademux: make demuxer reusable
+ Remove pads from flow combiner and reset last
+ flow return to FLOW_OK by resetting the flow combiner.
+ This prevents FLOW_FLUSHING when trying to re-use the
+ demuxer after setting it back to NULL/READY state.
+ https://bugzilla.gnome.org/show_bug.cgi?id=737359
+
+2014-09-24 16:46:36 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/videomixer/Makefile.am:
+ * gst/videomixer/gstcms.c:
+ * gst/videomixer/gstcms.h:
+ * gst/videomixer/videoconvert.c:
+ * gst/videomixer/videoconvert.h:
+ * gst/videomixer/videomixer2.c:
+ * gst/videomixer/videomixer2pad.h:
+ * gst/videomixer/videomixerorc-dist.c:
+ * gst/videomixer/videomixerorc-dist.h:
+ * gst/videomixer/videomixerorc.orc:
+ videomixer: use video library code instead of copy
+
+2014-09-18 16:39:19 +0530 Sanjay NM <sanjay.nm@samsung.com>
+
+ * gst/audioparsers/gstmpegaudioparse.c:
+ audioparsers: Added index check before using the index
+ https://bugzilla.gnome.org/show_bug.cgi?id=736878
+
+2014-09-23 23:33:37 +0200 Matej Knopp <matej.knopp@gmail.com>
+
+ * gst/isomp4/gstqtmux.c:
+ qtmux: Do not infer DTS on buffers from sparse streams.
+ DTS delta is used to calculate sample duration. If buffer has missing DTS, we take either segment start or previous buffer end time, whichever is later.
+ This must only be done for non sparse streams, sparse streams can have gaps between buffers (which is handled later by adding extra empty buffer with duration that fills the gap)
+ https://bugzilla.gnome.org/show_bug.cgi?id=737095
+
+2014-09-18 17:08:37 +0530 Sanjay NM <sanjay.nm@samsung.com>
+
+ * gst/goom/ifs.c:
+ goom: Clarified precedence between % and ?
+ https://bugzilla.gnome.org/show_bug.cgi?id=736887
+
+2014-09-18 17:59:31 +0530 Sanjay NM <sanjay.nm@samsung.com>
+
+ * gst/rtsp/gstrtspsrc.c:
+ rtsp: clarify expression so operator precedence is clear
+ https://bugzilla.gnome.org/show_bug.cgi?id=736903
+
+2014-09-18 16:04:03 +0530 Sanjay NM <sanjay.nm@samsung.com>
+
+ * ext/libpng/gstpngdec.c:
+ * gst/alpha/gstalpha.c:
+ * gst/audiofx/audiodynamic.c:
+ * gst/audiofx/audiofxbasefirfilter.c:
+ * gst/audiofx/gstscaletempo.c:
+ * gst/avi/gstavidemux.c:
+ * gst/avi/gstavimux.c:
+ * gst/deinterlace/gstdeinterlace.c:
+ * gst/isomp4/qtdemux.c:
+ * gst/matroska/matroska-mux.c:
+ * gst/rtpmanager/gstrtpmux.c:
+ * gst/rtpmanager/gstrtprtxreceive.c:
+ * gst/rtpmanager/rtpsession.c:
+ Miscellaneous minor cleanups
+ Fix redundant variables and assignments,
+ and unreachable breaks.
+ https://bugzilla.gnome.org/show_bug.cgi?id=736875
+ https://bugzilla.gnome.org/show_bug.cgi?id=736876
+ https://bugzilla.gnome.org/show_bug.cgi?id=736879
+ https://bugzilla.gnome.org/show_bug.cgi?id=736880
+ https://bugzilla.gnome.org/show_bug.cgi?id=736881
+ https://bugzilla.gnome.org/show_bug.cgi?id=736888
+ https://bugzilla.gnome.org/show_bug.cgi?id=736890
+ https://bugzilla.gnome.org/show_bug.cgi?id=736892
+ https://bugzilla.gnome.org/show_bug.cgi?id=736893
+ https://bugzilla.gnome.org/show_bug.cgi?id=736894
+
+2014-09-24 00:12:14 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/videobox/gstvideobox.c:
+ videobox: remove duplicate assignments
+ https://bugzilla.gnome.org/show_bug.cgi?id=736897
+
+2014-09-23 22:55:48 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/audioparsers/gstflacparse.c:
+ flacparse: Only calculate with durations != -1
+
+2014-09-23 19:08:48 +0200 Matej Knopp <matej.knopp@gmail.com>
+
+ * gst/isomp4/gstqtmux.c:
+ qtmux: collect pad for sparse stream should be created with lock set to false
+ Avoids waiting for buffers from sparse streams
+ https://bugzilla.gnome.org/show_bug.cgi?id=737095
+
+2014-09-23 19:07:25 +0200 Matej Knopp <matej.knopp@gmail.com>
+
+ * gst/isomp4/gstqtmux.c:
+ qtmux: fix subtitle buffer duration and strip null termination
+ Strip the \0 off the subtitle as we already know the size and also remember
+ to set the duration as buffer copying doesn't do it.
+ https://bugzilla.gnome.org/show_bug.cgi?id=737095
+
+2014-09-23 19:06:18 +0200 Matej Knopp <matej.knopp@gmail.com>
+
+ * gst/isomp4/atoms.c:
+ qtmux: move subtitle layer above video and set alternate group
+ layer -1 is above video, that is 0
+ And having all subtitles in alternate group 2 means that only one
+ should be selected at a time.
+ https://bugzilla.gnome.org/show_bug.cgi?id=737095
+
+2014-09-23 09:47:31 +0200 Edward Hervey <bilboed@bilboed.com>
+
+ * tests/check/elements/souphttpsrc.c:
+ check/soup: Temporarily disable G_ENABLE_DIAGNOSTIC
+ The SOUP_SERVER_PORT property has been deprecated in recent libsoup
+ versions.
+
+2014-09-23 09:43:05 +0200 Edward Hervey <bilboed@bilboed.com>
+
+ * tests/check/elements/souphttpsrc.c:
+ check/soup: Define minimum version required
+ To avoid deprecation warnings
+
+2014-09-19 19:14:28 +0200 Matej Knopp <matej.knopp@gmail.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Handle mp4a without ESDS atom
+ https://bugzilla.gnome.org/show_bug.cgi?id=736986
+
+2014-09-22 16:15:27 +0200 Linus Svensson <linussn@axis.com>
+
+ * sys/ximage/gstximagesrc.c:
+ ximagesrc: Fix build problem without XFIXES
+
+2014-09-19 14:34:13 +0530 Sanjay NM <sanjay.nm@samsung.com>
+
+ * gst/dtmf/gstrtpdtmfdepay.c:
+ dtmf: Removed unused structure members
+ https://bugzilla.gnome.org/show_bug.cgi?id=736883
+
+2014-09-11 13:48:44 -0300 Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
+
+ * gst/isomp4/atoms.c:
+ isomp4: fix wrong DAR calculation for PAR <= 1
+ CID #1226452
+ https://bugzilla.gnome.org/show_bug.cgi?id=736396
+
+2014-09-18 16:59:52 +0530 Sanjay NM <sanjay.nm@samsung.com>
+
+ * gst/flv/gstflvdemux.c:
+ flv: Removed unreachable break statements
+ https://bugzilla.gnome.org/show_bug.cgi?id=736884
+
+2014-09-17 16:37:11 +0200 Ognyan Tonchev <ognyan@axis.com>
+
+ * gst/rtpmanager/gstrtpbin.c:
+ rtpbin: do not leak encsink pad in error case
+ https://bugzilla.gnome.org/show_bug.cgi?id=736807
+
+2014-09-17 16:23:21 +0200 Ognyan Tonchev <ognyan@axis.com>
+
+ * gst/multipart/multipartdemux.c:
+ multipartdemux: do not leak new stream event
+ https://bugzilla.gnome.org/show_bug.cgi?id=736805
+
+2014-09-15 09:08:18 +0530 Ravi Kiran K N <ravi.kiran@samsung.com>
+
+ * gst/y4m/gsty4mencode.c:
+ * gst/y4m/gsty4mencode.h:
+ y4menc: port y4menc to use GstVideoEncoder base class
+ https://bugzilla.gnome.org/show_bug.cgi?id=735085
+
+2014-09-17 13:55:18 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * sys/osxaudio/gstosxcoreaudio.c:
+ * sys/osxaudio/gstosxcoreaudiocommon.c:
+ * sys/osxaudio/gstosxcoreaudiohal.c:
+ * sys/osxaudio/gstosxcoreaudioremoteio.c:
+ osxaudio: OSStatus is not a fourcc, so don't print it as one...
+
+2014-09-16 14:26:08 +0200 Ognyan Tonchev <ognyan@axis.com>
+
+ * gst/audioparsers/gstflacparse.c:
+ flacparse: do not leak uid after parsing TOC event
+ https://bugzilla.gnome.org/show_bug.cgi?id=736739
+
+2014-09-16 22:47:13 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtp/gstrtpvrawdepay.c:
+ rtpvrawdepay: Declare some more required caps fields in the sink template caps
+ Now only missing are width and height, which are expressed as strings
+ for RTP... so we can't put them into the template caps.
+
+2014-09-16 16:46:07 +0530 Vineeth T M <vineeth.tm@samsung.com>
+
+ * ext/gdk_pixbuf/gstgdkpixbufdec.c:
+ * ext/gdk_pixbuf/gstgdkpixbufdec.h:
+ gdkpixbufdec: modify wrong packetized mode logic
+ packetized mode is being set when framerate is being set
+ which is not correct. Changing the same by checking the
+ input segement format. If input segment is in TIME it is
+ Packetized, and if it is in BYTES it is not.
+ https://bugzilla.gnome.org/show_bug.cgi?id=736252
+
+2014-09-16 11:26:22 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * ext/jpeg/gstjpegdec.c:
+ jpegdec: Remove unused variable and use correct decoder variable name
+
+2014-09-16 11:25:42 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * ext/libpng/gstpngdec.c:
+ pngdec: Remove unused variable
+
+2014-09-16 13:24:15 +0530 Vineeth T M <vineeth.tm@samsung.com>
+
+ * ext/jpeg/gstjpegdec.c:
+ jpeggdec: modify wrong packetized mode logic
+ packetized mode is being set when framerate is being set
+ which is not correct. Changing the same by checking the
+ input segement format. If input segment is in TIME it is
+ Packetized, and if it is in BYTES it is not.
+ https://bugzilla.gnome.org/show_bug.cgi?id=736252
+
+2014-09-16 13:23:16 +0530 Vineeth T M <vineeth.tm@samsung.com>
+
+ * ext/libpng/gstpngdec.c:
+ pngdec: modify wrong packetized mode logic
+ packetized mode is being set when framerate is being set
+ which is not correct. Changing the same by checking the
+ input segement format. If input segment is in TIME it is
+ Packetized, and if it is in BYTES it is not.
+ https://bugzilla.gnome.org/show_bug.cgi?id=736252
+
+2014-09-15 14:39:41 +0200 Antonio Ospite <ao2@ao2.it>
+
+ * sys/ximage/gstximagesrc.c:
+ * sys/ximage/gstximagesrc.h:
+ * sys/ximage/ximageutil.c:
+ * sys/ximage/ximageutil.h:
+ ximagesrc: Remove unused screen-num property
+ The screen number can be still specified as part of the display-name
+ property (e.g. for screen 1 of display 0 use display-name=":0.1").
+ https://bugzilla.gnome.org/show_bug.cgi?id=736122
+
+2014-09-04 16:10:51 +0200 Antonio Ospite <ao2@ao2.it>
+
+ * sys/ximage/gstximagesrc.c:
+ ximagesrc: Draw the cursor only when it is active in the capturing region
+ Use XQueryPointer to check that the pointer is actually active inside
+ the capturing region.
+ This prevents drawing the cursor when the pointer is partially outside
+ of the captured region but not active inside the region; in particular
+ this avoids drawing the "window resize" cursor shapes to the captured
+ image when the mouse pointer crosses a window border.
+ NOTE that this is not only an optimization, this also happen to fix
+ a serious problem in multi-screen setups.
+ Because XFixes gives no information of what screen the pointer is on,
+ ximagesrc was always drawing the cursor on the captured screen even if
+ the mouse pointer was on another screen.
+ For example, when capturing from screen 1 (i.e. display-name=":0.1") the
+ cursor was drawn in the captured image even when the mouse pointer was
+ actually on screen 0, which is wrong and visually confusing.
+ https://bugzilla.gnome.org/show_bug.cgi?id=690646
+
+2014-09-05 11:33:31 +0200 Antonio Ospite <ao2@ao2.it>
+
+ * sys/ximage/gstximagesrc.c:
+ ximagesrc: Fix drawing the cursor when it is outside the capturing region
+ When the cursor is partially or totally out of the capturing region on
+ the top side or on the left side, it gets drawn fully inside of the
+ region with its coordinates rounded up to the left or to the top border.
+ This is immediately noticeable when using the xid property to capture
+ a specific window.
+ To fix the issue, allow negative cx and cx coordinates when checking the
+ boundaries before drawing the cursor.
+ NOTE that the boundaries checking calculations still allows the cursor
+ to be drawn when it is only partially outside of the capturing region,
+ but this makes sense and gives a more pleasing visual behaviour.
+ https://bugzilla.gnome.org/show_bug.cgi?id=690646
+
+2014-09-05 00:15:30 +0200 Antonio Ospite <ao2@ao2.it>
+
+ * sys/ximage/gstximagesrc.c:
+ * sys/ximage/gstximagesrc.h:
+ ximagesrc: Fix the destination coordinates of the cursor
+ XFixes provides the cursor coordinates relative to the root window, this
+ is not taken into account when using the xid property to capture
+ a specific window, the result is that the cursor gets drawn at the wrong
+ position.
+ In order to fix this consider the window location when calculating the
+ cursor position in the destination image.
+ https://bugzilla.gnome.org/show_bug.cgi?id=690646
+
+2014-09-15 14:51:24 +0200 Peter Korsgaard <peter@korsgaard.com>
+
+ * sys/v4l2/gstv4l2allocator.c:
+ v4l2allocator: O_CLOEXEC needs _GNU_SOURCE
+ Similar to 94f3d6fc / bz 709423
+ On some systems (E.G. uClibc and older Glibc versions), O_CLOEXEC is only
+ defined when _GNU_SOURCE is specified, so do so.
+ https://bugzilla.gnome.org/show_bug.cgi?id=736670
+
+2014-09-15 18:11:37 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * gst/debugutils/gstcapssetter.c:
+ capssetter: update to 1.0 transform_caps sematics
+ In 1.0, we pass the complete caps to transform_caps to allow for better
+ optimizations. Make this function actually work on non-simple caps
+ instead of just ignoring the configured filter caps.
+
+2014-09-08 14:06:00 +0200 Peter G. Baum <peter@dr-baum.net>
+
+ * gst/wavenc/gstwavenc.c:
+ * gst/wavenc/gstwavenc.h:
+ wavenc: use WAVE_FORMAT_EXTENSIBLE for more than 2 channels
+ https://bugzilla.gnome.org/show_bug.cgi?id=733444
+
+2014-09-12 15:06:50 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/wavparse/gstwavparse.c:
+ wavparse: Fix parsing of adtl chunks
+ We have to skip 12 bytes of data for the chunk, and the data size
+ passed to the sub-chunk parsing functions should have 4 bytes less
+ than the data size.
+ Also when parsing the sub-chunks, check if we actually have enough
+ data to read instead of just crashing.
+ https://bugzilla.gnome.org/show_bug.cgi?id=736266
+
+2014-09-12 10:55:23 +0530 Sanjay NM <sanjay.nm@samsung.com>
+
+ * gst/udp/gstudpsrc.c:
+ udp: include string.h for memcmp and memset
+ https://bugzilla.gnome.org//show_bug.cgi?id=736528
+
+2014-09-12 13:36:18 +0530 Anuj Jaiswal <anuj.jaiswal@samsung.com>
+
+ * gst/matroska/matroska-mux.c:
+ matroskamux: don't bitwise OR the same flag twice
+ https://bugzilla.gnome.org//show_bug.cgi?id=736543
+
+2014-09-12 10:35:36 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/matroska/matroska-demux.c:
+ matroskademux: handle real audio 28_8
+ Fixes duplicate check for 14_4.
+ https://bugzilla.gnome.org//show_bug.cgi?id=736543
+
+2014-09-11 14:46:09 +0530 Anuj Jaiswal <anuj.jaiswal@samsung.com>
+
+ * gst/multifile/gstmultifilesink.c:
+ multifilesink: don't OR the same flag twice
+ https://bugzilla.gnome.org/show_bug.cgi?id=736462
+
+2014-09-11 12:52:11 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * ext/soup/gstsouphttpsrc.c:
+ souphttpsrc: If the server reports "Accept-Ranges: none" don't try range requests
+
+2014-09-10 09:50:45 +0200 Ognyan Tonchev <ognyan@axis.com>
+
+ * sys/v4l2/gstv4l2sink.c:
+ v4l2sink: Unref pool after usage
+ https://bugzilla.gnome.org/show_bug.cgi?id=736384
+
+2014-09-09 19:03:50 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
+
+ * sys/v4l2/gstv4l2transform.c:
+ v4l2transform: Don't rank it for now
+ This will prevent the converter to be picked automatically in case
+ someone implement dynamic converter selection support. I'd like this
+ to be ranked only for known device, as it's hard to be sure a device is
+ a converter suited for general purpose. Re-negotiation is also needed
+ before we can rank it.
+ https://bugzilla.gnome.org/show_bug.cgi?id=733607
+
+2014-09-05 08:29:20 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
+
+ * sys/v4l2/gstv4l2src.c:
+ * sys/v4l2/gstv4l2src.h:
+ v4l2: Detect bad drivers timestamps
+ Even though the UVC driver do a great deal of effort to prevent bad
+ timestamp to be sent to userspace, there still exist UVC hardware that
+ are so buggy that the timestamp endup nearly random. This code detect
+ and ignore timestamp from these drivers, making these camera usable.
+ This has been tested on both invalid and valid cameras, making sure it
+ does not trigger for valid cameras.
+ https://bugzilla.gnome.org/show_bug.cgi?id=732910
+
+2014-08-29 17:09:30 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * sys/v4l2/gstv4l2allocator.c:
+ v4l2allocator: Workaround driver that don't support REQBUFS(0)
+ There is still around 18 drivers not yet ported to videobuf2. These driver
+ don't support freeing buffetrs through REQBUFS(0) hence for these the
+ memory type probing fails. In order to gain back our previous behaviour in
+ presence of these, we implement a workaround that assuming MMAP is
+ supported. Note that an allocator is only created for device with
+ STREAMING support in the device capabilities. In such case one of MMAP,
+ USERPTR and DMABUF is required. Though DMABUF came afterward, so is
+ not an option and in practice none of these drivers will only do USERPTR.
+ https://bugzilla.gnome.org/show_bug.cgi?id=735660
+ Also-by: Hans de Goede <hdegoede@redhat.com>
+
+2014-09-04 15:11:40 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
+
+ * sys/v4l2/gstv4l2bufferpool.c:
+ * sys/v4l2/gstv4l2object.c:
+ * sys/v4l2/gstv4l2object.h:
+ * sys/v4l2/gstv4l2videodec.c:
+ v4l2: Merge min_buffers_for* variable into one
+ Reuse the same min_buffers variable for both capture and output, this
+ reduce the length of lines and make the code more readable.
+ https://bugzilla.gnome.org/show_bug.cgi?id=736072
+
+2014-09-04 18:35:46 +0200 Aurélien Zanelli <aurelien.zanelli@parrot.com>
+
+ * sys/v4l2/gstv4l2bufferpool.c:
+ * sys/v4l2/gstv4l2object.c:
+ * sys/v4l2/gstv4l2object.h:
+ v4l2: set min_latency for output device according to required minimum number of buffers
+ Since we can get the minimum number of buffers needed by an output
+ device to work, use it to set min_latency which will determine how many
+ buffers are queued.
+ https://bugzilla.gnome.org/show_bug.cgi?id=736072
+
+2014-09-09 16:10:56 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/check/elements/udpsrc.c:
+ tests: udpsrc: add check to make sure multiple memory chunks are used
+
+2014-09-09 15:55:18 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/check/elements/udpsrc.c:
+ tests: udpsrc: wait for buffers with GCond instead of sleeping
+ Avoids half-second sleep for no reason.
+
+2014-09-09 15:31:32 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/check/elements/udpsrc.c:
+ tests: udpsrc: split out socket setup
+
+2014-09-09 13:46:56 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/udp/gstudpsrc.c:
+ udpsrc: more efficient memory handling
+ Drop use of g_socket_get_available_bytes() which is
+ not useful on all systems (where it returns the size
+ of the entire buffer not that of the next pending
+ packet), and is yet another syscall and apparently
+ very inefficient on Windows in the UDP case.
+ Instead, when reading UDP packets, use the more featureful
+ g_socket_receive_message() call that allows to read into
+ scattered memory, and allocate one memory chunk which is
+ likely to be large enough for a packet, while also providing
+ a larger allocated memory chunk just in case the packet
+ is larger than expected. If the received data fits into the
+ first chunk, we'll just add that to the buffer we return
+ and re-use the fallback buffer for next time, otherwise we
+ add both chunks to the buffer.
+ This reduces memory waste more reliably on systems where
+ get_available_bytes() doesn't work properly.
+ In a multimedia streaming scenario, incoming UDP packets
+ are almost never fragmented and thus almost always smaller
+ than the MTU size, which is also why we don't try to do
+ something smarter with more fallback memory chunks of
+ different sizes. The fallback scenario is just for when
+ someone built a broken sender pipeline (not using a
+ payloader or somesuch)
+ https://bugzilla.gnome.org/show_bug.cgi?id=610364
+
+2014-09-09 12:15:43 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/udp/gstudpsrc.c:
+ * gst/udp/gstudpsrc.h:
+ udpsrc: rework memory allocation bits and ensure we always have two chunks of memories to read into
+ First chunk is the likely/expected buffer size, second is as
+ fallback in case the packet is larger in the end.
+ Next step: actually use these.
+
+2014-09-09 09:42:15 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/udp/gstudpsrc.c:
+ * gst/udp/gstudpsrc.h:
+ udpsrc: track max packet size and save allocator negotiated by GstBaseSrc
+
+2014-09-08 16:15:05 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/audiofx/audioecho.c:
+ audioecho: fix example command line
+
+2014-09-07 12:46:08 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/avi/gstavidemux.c:
+ avidemux: fix crash with certain videos
+ This is a regression from 1.2 caused by the port
+ to the pad flow combiner.
+ https://bugzilla.gnome.org/show_bug.cgi?id=736192
+
+2014-09-04 16:21:20 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/matroska/matroska-demux.c:
+ * gst/matroska/matroska-read-common.h:
+ matroska-demux: Don't handle parse errors at the end of file as an error
+ But only if they happen after the Matroska segment.
+ https://bugzilla.gnome.org/show_bug.cgi?id=735833
+
+2014-09-04 12:14:11 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * ext/soup/gstsouphttpsrc.c:
+ souphttpsrc: Include redirection target in error messages
+ Just giving the original URI can give the false impression that e.g.
+ that one failed host name resolution, while actually the redirection target
+ did.
+
+2014-09-02 11:13:44 +0400 Andrei Sarakeev <sarakusha@gmail.com>
+
+ * gst/videomixer/videomixer2.c:
+ videomixer: Fix synchronization if dynamically changing the FPS
+ https://bugzilla.gnome.org/show_bug.cgi?id=735859
+
+2014-09-02 13:52:43 +0530 Ravi Kiran K N <ravi.kiran@samsung.com>
+
+ * gst/smpte/gstsmpte.c:
+ smpte: Check if input caps are the same and create output caps from video info
+ This makes sure that also properties like the pixel-aspect-ratio are the same
+ between both streams and that the output caps contain all fields necessary for
+ complete video caps.
+ https://bugzilla.gnome.org/show_bug.cgi?id=735804
+
+2014-09-02 17:22:07 +0530 Vineeth T M <vineeth.tm@samsung.com>
+
+ * gst/imagefreeze/gstimagefreeze.c:
+ imagefreeze: replace with gst_buffer_copy
+ gst_buffer_ref and gst_buffer_writable is being used to create a writable copy of source buffer.
+ replacing the same with gst_buffer_copy as the functionality is same.
+ https://bugzilla.gnome.org/show_bug.cgi?id=735880
+
+2014-09-03 23:06:53 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: mark jpeg and png as parsed so avdec_mjpeg can be used too
+ https://bugzilla.gnome.org/show_bug.cgi?id=735971
+
+2014-09-03 11:46:13 +0530 Vineeth T M <vineeth.tm@samsung.com>
+
+ * ext/gdk_pixbuf/gstgdkpixbufdec.c:
+ gdkpixbufdec: free query after use
+ In gst_gdk_pixbuf_dec_setup_pool(), query is being allocated using
+ gst_query_new_allocation(), but the same is not unreferenced
+ hence calling gst_query_unref() after usage of query.
+ https://bugzilla.gnome.org/show_bug.cgi?id=735950
+
+2014-09-03 23:46:34 +1000 Jan Schmidt <jan@centricular.com>
+
+ * gst/isomp4/qtdemux.c:
+ * gst/isomp4/qtdemux_types.c:
+ qtdemux: Silence some warnings for normal file contents
+
+2014-09-01 09:56:02 +0200 Nicolas Huet <nicolas.huet@parrot.com>
+
+ * gst/audioparsers/gstaacparse.c:
+ aacparse: Fix parsing issue when the buffer does not have a complete ADTS/LOAS frame
+ https://bugzilla.gnome.org/show_bug.cgi?id=735520
+
+2014-09-02 09:09:49 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * ext/vpx/gstvp9dec.c:
+ vp9dec: Get input width/height from the codec instead of the input caps
+ They are reported properly by libvpx if the correct struct members are used.
+ This also fixes handling of resolution changes without input caps changes.
+ https://bugzilla.gnome.org/show_bug.cgi?id=719359
+
+2013-10-22 18:49:22 +0100 Tom Greenwood <tcdgreenwood@hotmail.com>
+
+ * ext/vpx/gstvp8dec.c:
+ vp8dec: Fix for handling resolution changes when decoding VP8
+ If the resolution changes in the bitstream without the input caps changing we
+ would previously output corrupted video or crash.
+ https://bugzilla.gnome.org/show_bug.cgi?id=719359
+
+2014-09-02 00:55:17 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * ext/vpx/gstvp9dec.c:
+ vp9dec: Fix segfault when a new caps is received
+ Remember to unref the output caps when a new caps event is received
+ as it should generate a new one based on the new caps.
+ https://bugzilla.gnome.org/show_bug.cgi?id=734266
+
+2014-09-02 00:54:35 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * tests/check/elements/vp8dec.c:
+ tests: vp8dec: add test for caps renegotiation
+ Check that vp8dec can properly accept a new caps when upstream
+ changes it
+ https://bugzilla.gnome.org/show_bug.cgi?id=734266
+
+2014-08-05 10:34:39 +0200 Jose Antonio Santos Cadenas <santoscadenas@gmail.com>
+
+ * ext/vpx/gstvp8dec.c:
+ vp8dec: Reset output and input states when changing format
+ https://bugzilla.gnome.org/show_bug.cgi?id=734266
+
+2014-09-01 16:39:23 +0530 Vineeth T M <vineeth.tm@samsung.com>
+
+ * gst/imagefreeze/gstimagefreeze.c:
+ imagefreeze: Don't call gst_caps_unref() on template caps when already unreferenced
+ Adding an extra condition while calling gst_caps_unref (templ)
+ and replacing gst_caps_make_writable (gst_caps_ref (caps)) with
+ gst_caps_copy (caps) in line 177, since the functionality is same.
+ https://bugzilla.gnome.org/show_bug.cgi?id=735795
+
+2014-08-29 12:01:27 +0200 Hans de Goede <hdegoede@redhat.com>
+
+ * sys/v4l2/gstv4l2object.c:
+ v4l2: get_nearest_size: Fix "Unsupported field type" errors
+ Most V4L2 ioctls like try_fmt will adjust input fields to match what the
+ hardware can do rather then returning -EINVAL. As is docmented here:
+ http://linuxtv.org/downloads/v4l-dvb-apis/vidioc-g-fmt.html
+ EINVAL is only returned if the buffer type field is invalid or not supported.
+ So upon requesting V4L2_FIELD_NONE devices which can only do interlaced
+ mode will change the field value to e.g. V4L2_FIELD_BOTTOM as only returning
+ half the lines is the closest they can do to progressive modes.
+ In essence this means that we've failed to get a (usable) progessive mode
+ and should fall back to interlaced mode.
+ This commit adds a check for having gotten a usable field value after the first
+ try_fmt, to force fallback to interlaced mode even if the try_fmt succeeded,
+ thereby fixing get_nearest_size failing on these devices.
+ https://bugzilla.gnome.org/show_bug.cgi?id=735660
+
+2014-08-29 10:57:20 +0200 Hans de Goede <hdegoede@redhat.com>
+
+ * sys/v4l2/gstv4l2object.c:
+ v4l2: get_nearest_size: Always reinit all struct fields on retry
+ They may have been modified by the ioctl even if it failed. This also makes
+ the S_FMT fallback path try progressive first, making it consistent with the
+ preferred TRY_FMT path.
+ https://bugzilla.gnome.org/show_bug.cgi?id=735660
+
+2014-08-29 11:55:26 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/wavparse/gstwavparse.c:
+ wavparse: Store size of data tag in a 64 bit integer locally too
+ Otherwise we will clip the DS64 value of RF64 files to 32 bits again.
+
+2014-08-29 11:53:23 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/wavparse/gstwavparse.c:
+ wavparse: Use 64 bit scaling functions now that fact is a 64 bit integer
+
+2014-08-27 18:55:18 +0200 Peter G. Baum <peter@dr-baum.net>
+
+ * gst/wavparse/gstwavparse.c:
+ * gst/wavparse/gstwavparse.h:
+ wavparse: support rf64 format
+ https://bugzilla.gnome.org/show_bug.cgi?id=735627
+
+2014-08-28 13:48:50 -0600 Jason Litzinger <jlitzinger@control4.com>
+
+ * gst/multipart/multipartdemux.c:
+ multipartdemux: Ensure caps before pad added.
+ This stores the stream-start, sets caps, and then adds the pad,
+ which ensures that the caps are set for the "pad-added" callback.
+ https://bugzilla.gnome.org/show_bug.cgi?id=735626
+
+2014-08-28 15:03:50 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
+
+ * gst/flv/gstflvmux.c:
+ flvmux: Fallback to PTS if DTS is missing
+ Fixing a regression introduce when fixing:
+ https://bugzilla.gnome.org/show_bug.cgi?id=731352
+
+2014-08-28 16:13:29 +0530 Vineeth T M <vineeth.tm@samsung.com>
+
+ * gst/imagefreeze/gstimagefreeze.c:
+ imagefreeze: Remove impossible error condition
+ We return EOS after the first buffer, and GstPad will make sure now that we
+ won't get any other buffer afterwards until a flush happens. No need to check
+ for it ourselves.
+ https://bugzilla.gnome.org/show_bug.cgi?id=735581
+
+2014-08-28 13:53:23 +0530 Vineeth T M <vineeth.tm@samsung.com>
+
+ * ext/gdk_pixbuf/gstgdkpixbufdec.c:
+ gdkpixbufdec: EOS and NOT_LINKED are no errors in general
+ Don't post an error message for them but let upstream handle
+ anything accordingly.
+ https://bugzilla.gnome.org/show_bug.cgi?id=735564
+
+2014-08-27 21:07:26 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * gst/flv/gstflvmux.c:
+ * gst/flv/gstflvmux.h:
+ flvmux: Correctly offset timestamp
+ The previous method would break AV sync in the case audio or video
+ didn't start at the same point in running time.
+ https://bugzilla.gnome.org/show_bug.cgi?id=731352
+
+2014-08-27 20:56:12 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * gst/flv/gstflvmux.c:
+ flvmux: Save dts from buffer
+ We no longer set dts in muxed buffer. This would lead to encoding tags
+ with timestamp 0 instead of the timestamp of previous buffer.
+ https://bugzilla.gnome.org/show_bug.cgi?id=731352
+
+2014-07-28 20:58:59 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
+
+ * gst/flv/gstflvmux.c:
+ * gst/flv/gstflvmux.h:
+ flvmux: Ensure Timestamp starts at 0
+ FLV documentation stipulates that timestamp must start at zero.
+ In order to respect this rule, keep the first timestamp around
+ and offset the timestamp from this value. This allow for longer
+ recording time in presence of timestamp that does not start
+ at 0 already.
+ https://bugzilla.gnome.org/show_bug.cgi?id=731352
+
+2014-06-06 23:17:52 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
+
+ * gst/flv/gstflvdemux.c:
+ * gst/flv/gstflvdemux.h:
+ * gst/flv/gstflvmux.c:
+ flv: Tag timestamp are DTS not PTS
+ The tags in FLV are DTS. In audio cases, and for many video format this makes
+ no difference, but for AVC with B-Frames, PTS need to be computed from
+ composition timestamp CTS, with PTS = DTS + CTS.
+ https://bugzilla.gnome.org/show_bug.cgi?id=731352
+
+2014-08-07 21:58:14 -0400 Youness Alaoui <kakaroto@kakaroto.homelinux.net>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ jitterbuffer: Allow rtp caps without clock-rate
+ The jitterbuffer shouldn't force clock-rate on its sink pad, this will cause a negotiation issue since rtpssrcdemux doesn't have the clock-rate and doesn't add it to the caps. The documentation states that the clock-rate can either be specified through the caps or through the request-pt-map signal, so we must remove clock-rate from the pad templates and we must accept the GST_EVENT_CAPS if the caps don't have the clock-rate.
+ https://bugzilla.gnome.org/show_bug.cgi?id=734322
+
+2014-08-18 14:05:52 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: avoid crashing on dash streams
+ DASH/fragmented moov might have no samples as those are carried
+ in moof fragments. Avoid crashing or failing the stream because
+ of that.
+
+2014-08-18 10:33:48 +0530 Ravi Kiran K N <ravi.kiran@samsung.com>
+
+ * tests/examples/equalizer/demo.c:
+ * tests/examples/spectrum/demo-audiotest.c:
+ * tests/examples/spectrum/demo-osssrc.c:
+ examples: use 'post-messages' property instead of deprecated 'message' property
+ https://bugzilla.gnome.org/show_bug.cgi?id=734979
+
+2014-08-18 11:45:54 +0200 Víctor Manuel Jáquez Leal <vjaquez@igalia.com>
+
+ * gst/udp/gstudpsrc.c:
+ udp: fix udpsrc documentation
+ udpsrc gtk-doc documentation refers to sockfd and closefd properties which has
+ been removed. This patch replaces those references to socket and close-socket
+ respectively.
+ https://bugzilla.gnome.org/show_bug.cgi?id=734987
+
+2014-08-15 10:09:56 +1000 Jan Schmidt <jan@centricular.com>
+
+ * gst/isomp4/gstqtmux.c:
+ qtmux: Make the default timescale 1/1800 second
+ The old default timescale of 1 millisecond produces irrational
+ numbers for a lot of framerate/audio-packet-duration multiples.
+ 1/1800 is a nicer number, as it tends to produce better fractions
+ and therefore slightly higher accuracy overall
+
+2014-08-15 01:17:27 +1000 Jan Schmidt <jan@centricular.com>
+
+ * gst/matroska/matroska-demux.c:
+ matroska: Use gst_video_guess_framerate() function
+ Remove local framerate guessing function in favour of
+ the new gst_video_guess_framerate() function.
+
+2014-08-15 01:12:20 +1000 Jan Schmidt <jan@centricular.com>
+
+ * gst/isomp4/Makefile.am:
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Improve framerate calculation/guessing
+ Change the way the output framerate is calculated
+ to ignore the first sample (which is sometimes truncated
+ in my testing) and use the new gst_video_guess_framerate()
+ function to recognise common standard framerates better.
+ Remove the code that was sorting the first 20 sample
+ durations and then ignoring the result.
+
+2014-08-14 16:36:44 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/videomixer/videomixer2.c:
+ videomixer: Use the best width/height/etc if downstream can handle that
+ Before it was always using whatever downstream preferred, while
+ the code and documentation claimed something different.
+ https://bugzilla.gnome.org/show_bug.cgi?id=727180
+
+2014-08-14 11:29:00 +0530 Ravi Kiran K N <ravi.kiran@samsung.com>
+
+ * gst/videomixer/videomixer2.c:
+ videomixer: Avoid double free of VideoConvert
+ https://bugzilla.gnome.org/show_bug.cgi?id=734764
+
+2014-08-13 11:58:35 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/flv/gstflvdemux.c:
+ flvdemux: fix indentation
+
+2014-08-13 11:54:26 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/flv/gstflvdemux.c:
+ flvdemux: un-break duration querying
+ Commit 2b9493b5 broke this in two ways: a) we should only
+ pass duration queries in TIME format upstream (or at least
+ not those in DEFAULT or BYTE format), and b) we mustn't
+ overwrite the default value of 'res' from TRUE to FALSE
+ and not set it again later. This led to bogus durations
+ being reported for FLV playback from file, because TIME
+ queries would fail (as 'res' had been set to FALSE) and
+ parsers then do a BYTE query as fallback and try to
+ guesstimate something in return, which of course goes
+ horribly wrong since the BYTE size returned is for the
+ muxed file.
+
+2014-08-13 13:23:10 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/videofilter/gstvideobalance.c:
+ videobalance: Allow any raw caps in passthrough mode, not just the ones we handle
+
+2014-08-13 13:04:21 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/videofilter/gstvideobalance.c:
+ videobalance: Allow ANY capsfeatures, but only in passthrough mode
+ When changing the properties to not be in passthrough mode anymore,
+ we will only accept caps we can process ourselves, potentially causing
+ a not-negotiated error.
+ https://bugzilla.gnome.org/show_bug.cgi?id=720345
+
+2014-08-12 11:34:30 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * docs/plugins/inspect/plugin-1394.xml:
+ * docs/plugins/inspect/plugin-aasink.xml:
+ * docs/plugins/inspect/plugin-alaw.xml:
+ * docs/plugins/inspect/plugin-alpha.xml:
+ * docs/plugins/inspect/plugin-alphacolor.xml:
+ * docs/plugins/inspect/plugin-apetag.xml:
+ * docs/plugins/inspect/plugin-audiofx.xml:
+ * docs/plugins/inspect/plugin-audioparsers.xml:
+ * docs/plugins/inspect/plugin-auparse.xml:
+ * docs/plugins/inspect/plugin-autodetect.xml:
+ * docs/plugins/inspect/plugin-avi.xml:
+ * docs/plugins/inspect/plugin-cacasink.xml:
+ * docs/plugins/inspect/plugin-cairo.xml:
+ * docs/plugins/inspect/plugin-cutter.xml:
+ * docs/plugins/inspect/plugin-debug.xml:
+ * docs/plugins/inspect/plugin-deinterlace.xml:
+ * docs/plugins/inspect/plugin-dtmf.xml:
+ * docs/plugins/inspect/plugin-dv.xml:
+ * docs/plugins/inspect/plugin-effectv.xml:
+ * docs/plugins/inspect/plugin-equalizer.xml:
+ * docs/plugins/inspect/plugin-flac.xml:
+ * docs/plugins/inspect/plugin-flv.xml:
+ * docs/plugins/inspect/plugin-flxdec.xml:
+ * docs/plugins/inspect/plugin-gdkpixbuf.xml:
+ * docs/plugins/inspect/plugin-goom.xml:
+ * docs/plugins/inspect/plugin-goom2k1.xml:
+ * docs/plugins/inspect/plugin-icydemux.xml:
+ * docs/plugins/inspect/plugin-id3demux.xml:
+ * docs/plugins/inspect/plugin-imagefreeze.xml:
+ * docs/plugins/inspect/plugin-interleave.xml:
+ * docs/plugins/inspect/plugin-isomp4.xml:
+ * docs/plugins/inspect/plugin-jack.xml:
+ * docs/plugins/inspect/plugin-jpeg.xml:
+ * docs/plugins/inspect/plugin-level.xml:
+ * docs/plugins/inspect/plugin-matroska.xml:
+ * docs/plugins/inspect/plugin-mulaw.xml:
+ * docs/plugins/inspect/plugin-multifile.xml:
+ * docs/plugins/inspect/plugin-multipart.xml:
+ * docs/plugins/inspect/plugin-navigationtest.xml:
+ * docs/plugins/inspect/plugin-oss4.xml:
+ * docs/plugins/inspect/plugin-ossaudio.xml:
+ * docs/plugins/inspect/plugin-png.xml:
+ * docs/plugins/inspect/plugin-pulseaudio.xml:
+ * docs/plugins/inspect/plugin-replaygain.xml:
+ * docs/plugins/inspect/plugin-rtp.xml:
+ * docs/plugins/inspect/plugin-rtpmanager.xml:
+ * docs/plugins/inspect/plugin-rtsp.xml:
+ * docs/plugins/inspect/plugin-shapewipe.xml:
+ * docs/plugins/inspect/plugin-shout2send.xml:
+ * docs/plugins/inspect/plugin-smpte.xml:
+ * docs/plugins/inspect/plugin-soup.xml:
+ * docs/plugins/inspect/plugin-spectrum.xml:
+ * docs/plugins/inspect/plugin-speex.xml:
+ * docs/plugins/inspect/plugin-taglib.xml:
+ * docs/plugins/inspect/plugin-udp.xml:
+ * docs/plugins/inspect/plugin-video4linux2.xml:
+ * docs/plugins/inspect/plugin-videobox.xml:
+ * docs/plugins/inspect/plugin-videocrop.xml:
+ * docs/plugins/inspect/plugin-videofilter.xml:
+ * docs/plugins/inspect/plugin-videomixer.xml:
+ * docs/plugins/inspect/plugin-vpx.xml:
+ * docs/plugins/inspect/plugin-wavenc.xml:
+ * docs/plugins/inspect/plugin-wavpack.xml:
+ * docs/plugins/inspect/plugin-wavparse.xml:
+ * docs/plugins/inspect/plugin-ximagesrc.xml:
+ * docs/plugins/inspect/plugin-y4menc.xml:
+ docs: update for git
+
+2014-08-12 11:33:56 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * configure.ac:
+ configure: build ximagesrc again when checks succeed
+ Third time lucky, hopefully.
+
+2014-08-11 09:26:17 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * configure.ac:
+ configure: fix x11 checks to be non-fatal again
+ Must pass an action-if-not-found argument to
+ PKG_CHECK_MODULES or it will error out when
+ it can't find the module requested. Also fix
+ AC_CHECK_LIB usage, extra libs argument was
+ in the wrong place.
+
+2014-08-07 17:12:38 +0300 George Kiagiadakis <george.kiagiadakis@collabora.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: forward DISCONT from upstream to the output streams
+ This makes sense in DASH reverse playback, where the upstream dashdemux
+ will download DASH segments in reverse order, but push their buffers
+ forward to qtdemux and mark each segment start as DISCONT. This needs
+ to be forwarded downstream to the parser/decoder, otherwise it won't work.
+ https://bugzilla.gnome.org/show_bug.cgi?id=734443
+
+2014-08-10 18:55:07 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * configure.ac:
+ configure: use pkg-config to detect x11 and simplify checks
+ AC_PATH_XTRA macro unnecessarily pulls in libSM and libICE.
+ https://bugzilla.gnome.org/show_bug.cgi?id=731047
+
+2014-08-10 12:30:07 +0200 Mark Nauwelaerts <mnauw@users.sourceforge.net>
+
+ * tests/check/elements/rtp-payloading.c:
+ tests: rtp-payloading: adjust test data to avoid NAL chopping
+ ... and correspondingly unexpected buffer sizes.
+
+2014-08-09 14:22:42 +0200 Sebastian Rasmussen <sebras@hotmail.com>
+
+ * ext/speex/gstspeexenc.c:
+ speexenc: Improve annotation of internal function
+ https://bugzilla.gnome.org/show_bug.cgi?id=734542
+
+2014-08-08 12:54:30 +0200 Sebastian Rasmussen <sebras@hotmail.com>
+
+ * gst/shapewipe/gstshapewipe.c:
+ * tests/examples/shapewipe/shapewipe-example.c:
+ shapewipe: Unref caps and element after usage
+ https://bugzilla.gnome.org/show_bug.cgi?id=734478
+
+2014-08-09 20:47:30 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: improve debug logging of fourccs
+ If we can't show ASCII, at least show them
+ in big endian order.
+
+2014-08-09 20:46:04 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: add support for 'wma ' mapping as found in some ismv files
+ e.g. To_The_Limit_720_2962.ismv
+
+2014-08-09 18:31:20 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: add support for 'vc-1' mapping as found in some ismv files
+ e.g. To_The_Limit_720_2962.ismv
+
+2014-08-07 16:34:36 +0200 Sebastian Rasmussen <sebras@hotmail.com>
+
+ * gst/rtp/gstrtph263ppay.c:
+ rtph263ppay: Unref pad template caps after use
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=734435
+
+2014-08-08 12:36:01 +0200 Sebastian Rasmussen <sebras@hotmail.com>
+
+ * gst/videomixer/videomixer2.c:
+ videomixer: Unref allowed caps after usage
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=734474
+
+2014-08-08 12:40:49 +0200 Sebastian Rasmussen <sebras@hotmail.com>
+
+ * gst/imagefreeze/gstimagefreeze.c:
+ imagefreeze: Unref pad template caps after usage
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=734475
+
+2014-08-08 12:44:09 +0200 Sebastian Rasmussen <sebras@hotmail.com>
+
+ * gst/debugutils/gstnavseek.c:
+ navseek: Unref peer pad after usage
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=734476
+
+2014-08-08 12:29:52 +0200 Sebastian Rasmussen <sebras@hotmail.com>
+
+ * gst/rtpmanager/gstrtpmux.c:
+ rtpmux: Unref pad template caps after usage
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=734473
+
+2014-08-05 11:47:39 +0200 Srimanta Panda <srimanta@axis.com>
+
+ * gst/rtp/gstrtph264pay.c:
+ rtph264pay: append packetization mode parameter to SDP
+ Append packetization-mode parameter to SDP description.
+ Packetization mode signals the properties of an RTP payload type.
+ https://bugzilla.gnome.org/show_bug.cgi?id=733556
+
+2014-08-08 03:58:14 +1000 Jan Schmidt <jan@centricular.com>
+
+ * gst/isomp4/gstqtmux.c:
+ * gst/isomp4/qtdemux.c:
+ isomp4/qtmux: Write correct file duration when gaps exist.
+ When writing out a trak with an edit list, make sure the
+ overall file duration is also updated to reflect the
+ lengthening of the stream.
+ Add some more debug to qtdemux to warn about streams that
+ are longer than the file and get truncated.
+
+2014-08-04 15:39:17 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp/gstrtspsrc.c:
+ rtspsrc: Push the correct segment in TCP mode when seeking
+
+2014-08-03 12:33:32 +0200 Mark Nauwelaerts <mnauw@users.sourceforge.net>
+
+ * gst/rtp/gstrtph264pay.c:
+ rtph264pay: unbreak au aligned byte-stream payloading
+
+2014-07-22 13:24:09 +0200 Srimanta Panda <srimanta@axis.com>
+
+ * gst/rtp/gstrtph264pay.c:
+ rtph264pay: append profile-level-id to SDP
+ Append profile-level-id to SDP if available.
+ https://bugzilla.gnome.org/show_bug.cgi?id=733539
+
+2014-07-31 18:47:49 +0200 Edward Hervey <edward@collabora.com>
+
+ * Makefile.am:
+ * common:
+ Makefile: Add usage of build-checks step
+ Allows building checks without running them
+
+2014-07-31 09:53:53 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
+
+ * sys/ximage/ximageutil.c:
+ ximagesrc: Fix warning about missing return value
+
+2014-07-24 15:28:09 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * sys/ximage/gstximagesrc.c:
+ * sys/ximage/ximageutil.c:
+ * sys/ximage/ximageutil.h:
+ ximagesrc: Add missing return value to Buffer dispose function
+ Depending ont he build, the method could return FALSE, hence never
+ free the buffers, or already TRUE and lead to a crash:
+ Fixes: https://bugzilla.gnome.org/show_bug.cgi?id=733695
+
+2014-07-28 16:49:16 +0200 Philippe Normand <philn@igalia.com>
+
+ * gst/interleave/interleave.c:
+ * tests/check/elements/interleave.c:
+ interleave: set output caps layout to interleaved
+ Set output caps layout independently from input caps layout which can
+ be either non-interleaved or interleaved.
+ https://bugzilla.gnome.org/show_bug.cgi?id=733866
+
+2014-07-26 12:06:39 -0300 Thiago Santos <ts.santos@osg.sisa.samsung.com>
+
+ * sys/v4l2/gstv4l2bufferpool.c:
+ v4l2bufferpool: clear gcond
+
+2014-07-25 14:30:33 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
+
+ * sys/v4l2/gstv4l2bufferpool.c:
+ Revert "v4l2bufferpool: Workaround elements not requesting any buffers"
+ This was a tempory workaround, we should fix the encoders that do not
+ negotatiate the amount of buffers they need.
+ This reverts commit d03bcba3db15d06dbdea6b776a6f28ed2f03272a.
+
+2014-07-08 14:31:59 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * sys/v4l2/gstv4l2object.c:
+ v4l2object: Don't share own pool if min exceed V4L2 capacity
+ If the minimum required buffer exceed V4L2 capacity, don't share down
+ pool. This allow support very high latency, like with x264enc default
+ encoding settings.
+ https://bugzilla.gnome.org/show_bug.cgi?id=732288
+
+2014-07-25 17:42:20 +0200 Aurélien Zanelli <aurelien.zanelli@parrot.com>
+
+ * sys/v4l2/gstv4l2object.c:
+ v4l2object: query minimum required buffers for output
+ Some v4l2 devices could require a minimum buffers different from default
+ values. Rather than blindly propose a pool with min-buffers set to the
+ default value, it ask the device using control ioctl.
+ https://bugzilla.gnome.org/show_bug.cgi?id=733750
+
+2014-07-23 18:40:10 +0200 Aurélien Zanelli <aurelien.zanelli@parrot.com>
+
+ * sys/v4l2/gstv4l2sink.c:
+ v4l2sink: use directly 'obj' instead of 'v4l2sink->v4l2object'
+ https://bugzilla.gnome.org/show_bug.cgi?id=733616
+
+2014-07-23 18:39:50 +0200 Aurélien Zanelli <aurelien.zanelli@parrot.com>
+
+ * sys/v4l2/gstv4l2object.c:
+ * sys/v4l2/gstv4l2sink.c:
+ v4l2: set debug messages according to device type and IO mode
+ https://bugzilla.gnome.org/show_bug.cgi?id=733616
+
+2014-05-24 19:02:59 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.co.uk>
+
+ * sys/v4l2/gstv4l2object.c:
+ v4l2object: Remove is_active checks
+ These checks are no longer required with recent change to the bufferpool. This
+ should allow changing the configuartion, hence the way forward renegotiation
+ support.
+ https://bugzilla.gnome.org/show_bug.cgi?id=728268
+
+2014-07-21 18:11:16 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/isomp4/qtdemux.c:
+ * gst/isomp4/qtdemux_lang.c:
+ qtdemux: fix language code parsing for 3-letter codes starting with 'a'
+ And handle special value for 'unspecified' explicitly.
+ https://developer.apple.com/library/mac/documentation/QuickTime/QTFF/QTFFChap4/qtff4.html
+
+2014-07-08 02:18:27 +0200 Nicola Murino <nicola.murino@gmail.com>
+
+ * ext/jpeg/gstjpegenc.c:
+ jpegenc: Add support for encoding from NV21 and NV12
+ https://bugzilla.gnome.org/show_bug.cgi?id=732870
+
+2014-07-19 18:04:38 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * configure.ac:
+ Back to development
+
projects / platform / upstream / gstreamer.git / commitdiff