+=== release 1.5.90 ===
+
+2015-08-19 Sebastian Dröge <slomo@coaxion.net>
+
+ * configure.ac:
+ releasing 1.5.90
+
+2015-08-19 11:29:55 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * po/el.po:
+ * po/zh_CN.po:
+ po: Update translations
+
+2015-08-13 17:29:58 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/multifile/gstmultifilesrc.c:
+ multifilesrc: fix regression with starting from index set via index property
+ When we haven't started yet, set the start_index when we set the index property,
+ so that we start at the right index position after the initial seek. The index
+ property was never really meant to be for writing, but it used to work, so let's
+ support it for backwards compatibility.
+ https://bugzilla.gnome.org/show_bug.cgi?id=739472
+
+2015-08-18 10:52:11 +0100 Alex Ashley <bugzilla@ashley-family.net>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: fix offset calculation when parsing CENC aux info
+ Commit 7d7e54ce6863ff53e188d0276d2651b65082ffdb added support for
+ DASH common encryption, however commit
+ bb336840c0b0b02fa18dc4437ce0ded3d9142801 that went onto master
+ shortly before the CENC commit caused the calculation of the CENC
+ aux info offset to be incorrect.
+ The base_offset was being added if present, but if the base_offset
+ is relative to the start of the moof, the offset was being added twice.
+ The correct approach is to calculate the offset from the start of the
+ moof and use that offset when parsing the CENC aux info.
+
+2015-08-17 14:28:24 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * ext/flac/gstflacenc.c:
+ flacenc: actually return true for accept-caps query handling
+
+2015-08-17 14:07:10 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
+
+ * gst/rtp/gstrtpg723pay.c:
+ * gst/rtp/gstrtpgsmpay.c:
+ * gst/rtp/gstrtpklvpay.c:
+ rtp: copy metadata in the (de)payloaders which is missed before
+ https://bugzilla.gnome.org/show_bug.cgi?id=753706
+
+2015-08-16 15:21:51 -0400 Dustin Spicuzza <dustin@virtualroadside.com>
+
+ * configure.ac:
+ * sys/directsound/gstdirectsoundsink.c:
+ * sys/directsound/gstdirectsoundsink.h:
+ directsoundsink: allow specifying audio playback device
+ https://bugzilla.gnome.org/show_bug.cgi?id=753670
+
+2015-08-16 13:51:47 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * ext/flac/gstflacenc.c:
+ flacenc: remove single entry if from loop
+ Iterate from the 2nd channel on and create the 1 channel struct
+ outside to make loop structure simpler and only slightly faster.
+
+2015-08-16 13:21:41 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * ext/flac/gstflacenc.c:
+ flacenc: implement proper accept-caps
+ Should just compare with what can be immediatelly accepted by
+ the element. flacenc can't renegotiate so if it has a caps already
+ it should only accept if it is that caps otherwise just use the
+ template caps
+
+2015-08-16 13:03:36 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * ext/flac/gstflacenc.c:
+ flacenc: improve sink pad template caps
+ Removes the need for custom caps query handling and makes it more
+ correct from the beginning on the template. It is a bit uglier
+ to read because there is 1 entry per channel but makes code easier
+ to maintain.
+
+2015-08-16 12:41:56 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/y4m/gsty4mencode.c:
+ y4mencode: fix gst-launch version in documentation
+
+2015-08-15 22:32:21 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * ext/speex/gstspeexenc.c:
+ * ext/wavpack/gstwavpackenc.c:
+ * gst/law/alaw-encode.c:
+ * gst/law/mulaw-encode.c:
+ audioencoders: use template subset check for accept-caps
+ It is faster than doing a query that propagates downstream and
+ should be enough
+ Elements: speexenc, wavpackenc, mulawenc, alawenc
+
+2015-08-15 22:29:41 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * ext/jpeg/gstjpegenc.c:
+ * ext/libpng/gstpngenc.c:
+ * ext/vpx/gstvp8enc.c:
+ * ext/vpx/gstvp9enc.c:
+ * gst/y4m/gsty4mencode.c:
+ videoencoders: use template subset check for accept-caps
+ It is faster than doing a query that propagates downstream and
+ should be enough
+ Elements: jpegenc, pngenc, vp8enc, vp9enc, y4menc
+
+2015-08-16 17:21:24 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/audioparsers/gstmpegaudioparse.c:
+ mpegaudioparse: use new baseparse API to fix tag handling
+ https://bugzilla.gnome.org/show_bug.cgi?id=679768
+
+2015-03-17 17:50:37 -0400 Olivier Crête <olivier.crete@collabora.com>
+
+ * gst/audioparsers/gstaacparse.c:
+ * gst/audioparsers/gstac3parse.c:
+ * gst/audioparsers/gstamrparse.c:
+ * gst/audioparsers/gstdcaparse.c:
+ * gst/audioparsers/gstsbcparse.c:
+ * gst/audioparsers/gstwavpackparse.c:
+ audioparsers: use new base parse API to fix tag handling
+ https://bugzilla.gnome.org/show_bug.cgi?id=679768
+
+2015-08-16 14:37:53 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/audioparsers/gstflacparse.c:
+ flacparse: use new baseparse API and fix tag handling
+ https://bugzilla.gnome.org/show_bug.cgi?id=679768
+
+2015-08-16 13:04:02 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Use signed integer type to be able to check for negative subtraction results
+ CID 1315829
+
+2015-08-16 11:50:34 +0100 Luis de Bethencourt <luis@debethencourt.com>
+
+ * gst/rtp/gstrtpvorbisdepay.c:
+ rtpvorbisdepay: remove dead code
+ payload_buffer must be NULL in ignore_reserved. Check will always be false.
+ Introduced by b1089fb5207697ba26edb4ff66ed0f465c6df3cf
+ CID #1316476
+
+2015-08-15 22:45:53 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/law/alaw-encode.c:
+ * gst/law/alaw-encode.h:
+ alawenc: port to AudioEncoder base class
+
+2015-08-15 09:16:23 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * ext/flac/gstflacdec.c:
+ * ext/speex/gstspeexdec.c:
+ * ext/wavpack/gstwavpackdec.c:
+ * gst/law/alaw-decode.c:
+ * gst/law/mulaw-decode.c:
+ audiodecoders: use default pad accept-caps handling
+ Avoids useless check of downstream caps when handling an
+ accept-caps query
+ Elements: flacdec, speexdec, wavpackdec, mulawdec, alawdec
+
+2015-08-15 08:49:57 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * ext/jpeg/gstjpegdec.c:
+ * ext/libpng/gstpngdec.c:
+ * ext/vpx/gstvp8dec.c:
+ * ext/vpx/gstvp9dec.c:
+ videodecoders: use default pad accept-caps handling
+ Avoids useless check of downstream caps when handling an
+ accept-caps query
+ Elements: jpegdec, pngdec, vp8dec, vp9dec
+
+2015-08-15 11:31:04 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/law/alaw-decode.c:
+ alawdec: make error handling a bit nicer
+ Print the element along with the debug to make it easier to trace
+ the failures
+
+2015-08-15 11:04:16 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/law/alaw-decode.c:
+ * gst/law/alaw-decode.h:
+ alawdec: port to audiodecoder base class
+ mulawdec was already ported, alawdec was left behind.
+
+2015-08-15 10:34:14 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: only look for more samples in moofs in pull-mode
+ For playback of some fragmented formats with qtdemux it will
+ try to look for the next moof after finishing one but it is only
+ possible for pull-mode. For playback of streaming fragmented formats
+ such as DASH it should just not try to look for another moof but
+ instead wait for more data.
+ https://bugzilla.gnome.org/show_bug.cgi?id=752602
+ https://bugzilla.gnome.org/show_bug.cgi?id=752603
+
+2015-08-15 12:58:50 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/audioparsers/gstdcaparse.c:
+ dcaparse: Don't look for a second syncword
+ There are streams out there that consistently contain garbage between
+ every frame so we never ever find a second consecutive syncword.
+ See https://bugzilla.gnome.org/show_bug.cgi?id=738237
+
+2015-08-15 11:12:05 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * ext/vpx/gstvp8enc.c:
+ * ext/vpx/gstvp9enc.c:
+ vp8enc, vp9enc: reset multipass file index when stopping encoder
+ Fixes multipass encoding when re-using the same element/pipeline
+ for subsequent encoding runs.
+ https://bugzilla.gnome.org/show_bug.cgi?id=747728
+
+2015-08-15 11:09:42 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * ext/vpx/gstvp9enc.c:
+ * ext/vpx/gstvp9enc.h:
+ vp9enc: provide support for multiple pass cache files
+ Some files may provide different caps insight of one stream. Since
+ vp9enc support caps reinit, we should support cache reinit too.
+ If more then file cache file will be created, the naming will be:
+ cache cache.1 cache.2 ...
+ Based on patch by: Oleksij Rempel <linux@rempel-privat.de>
+ https://bugzilla.gnome.org/show_bug.cgi?id=747728
+
+2015-08-14 11:41:42 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * tests/check/elements/aacparse.c:
+ tests: aacparse: use caps query instead of accept-caps
+ The accept-caps query just does a shallow check at the current
+ element while at this test we want it to also look at downstream.
+ So use caps query there.
+ https://bugzilla.gnome.org/show_bug.cgi?id=753623
+
+2015-08-14 11:40:22 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/audioparsers/gstaacparse.c:
+ * gst/audioparsers/gstac3parse.c:
+ * gst/audioparsers/gstamrparse.c:
+ * gst/audioparsers/gstdcaparse.c:
+ * gst/audioparsers/gstflacparse.c:
+ * gst/audioparsers/gstmpegaudioparse.c:
+ * gst/audioparsers/gstsbcparse.c:
+ * gst/audioparsers/gstwavpackparse.c:
+ audioparsers: enable accept-template flag
+ Do a quick check with the pad template caps as it is enough. Users
+ should have figured the appropriate full caps on a previous caps query
+ https://bugzilla.gnome.org/show_bug.cgi?id=753623
+
+2015-08-14 15:46:53 +0200 George Kiagiadakis <george.kiagiadakis@collabora.com>
+
+ * gst/rtsp/gstrtspsrc.c:
+ * gst/rtsp/gstrtspsrc.h:
+ rtspsrc: send the User-Agent header
+ Sometimes it is useful to know this information on the
+ server side. Other popular implementations (vlc, ffmpeg, ...)
+ also send this header on every message.
+ This includes a new "user-agent" property that the user
+ can set to use a custom User-Agent string. The default
+ is "GStreamer/<version>"
+ https://bugzilla.gnome.org/show_bug.cgi?id=750101
+
+2015-08-14 15:42:42 +0200 George Kiagiadakis <george.kiagiadakis@collabora.com>
+
+ * gst/rtsp/gstrtspsrc.c:
+ rtspsrc: wrap gst_rtsp_message_init_request in a local function
+ This will allow adding common request initialization, like the
+ user agent string, in just one place.
+
+2015-08-14 09:36:09 +0530 Prashant Gotarne <ps.gotarne@samsung.com>
+
+ * gst/audiofx/audioecho.c:
+ audioecho: make sure buffer gets reallocated if max_delay changes
+ https://bugzilla.gnome.org/show_bug.cgi?id=753490
+
+2015-07-09 09:51:26 +0200 Oleksij Rempel <linux@rempel-privat.de>
+
+ * ext/vpx/gstvp8enc.c:
+ * ext/vpx/gstvp8enc.h:
+ vp8enc: provide support for multiple pass cache files
+ Some files may provide different caps insight of one stream. Since vp8enc
+ support caps reinit, we should support cache reinit too.
+ If more then file cache file will be created, the naming will be:
+ cache
+ cache.1
+ cache.2
+ ...
+ https://bugzilla.gnome.org/show_bug.cgi?id=747728
+
+2015-04-15 22:51:51 +0200 Ramiro Polla <ramiro.polla@collabora.co.uk>
+
+ * gst/rtp/gstrtpmp4gdepay.c:
+ rtpmp4gdepay: fix timestamps for RTP packets with multiple AUs
+ Use constantDuration to calculate the timestamp of non-first AU in the
+ RTP packet.
+ If constantDuration is not present in the MIME parameters, its value
+ must be calculated based on the timing information from two consecutive
+ RTP packets with AU-Index equal to 0.
+ https://bugzilla.gnome.org/show_bug.cgi?id=747881
+
+2015-08-14 06:43:13 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * ext/soup/gstsouphttpsrc.c:
+ souphttpsrc: remove unnecessary if, g_free is null safe
+
+2015-08-14 08:33:56 +0100 Alex Ashley <bugzilla@ashley-family.net>
+
+ * ext/soup/gstsouphttpsrc.c:
+ * ext/soup/gstsouphttpsrc.h:
+ souphttpsrc: add property to set HTTP method
+ To allow souphttpsrc to be use HTTP methods other than GET
+ (e.g. HEAD), add a "method" property that is a string. If this
+ property is not set, GET is used.
+ https://bugzilla.gnome.org/show_bug.cgi?id=752413
+
+2015-08-14 11:13:01 +0200 Edward Hervey <bilboed@bilboed.com>
+
+ * tests/check/generic/states.c:
+ check: Rename states unit test
+ Makes it easier to differentiate from other modules states unit test
+
+2015-08-14 09:21:25 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/goom/gstaudiovisualizer.c:
+ * gst/goom/gstaudiovisualizer.h:
+ * gst/goom2k1/gstaudiovisualizer.c:
+ * gst/goom2k1/gstaudiovisualizer.h:
+ goom: Rename get_type() function of base class to prevent symbol conflicts
+ This is a problem when statically linking.
+
+2015-08-13 16:32:55 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ rtpjitterbuffer: Keep the DTS estimate if we got no DTS after a jitterbuffer reset
+ Otherwise we will just output buffers without timestamps after a reset if no
+ timestamps are provided by upstream, e.g. when using RTSP over TCP.
+ https://bugzilla.gnome.org/show_bug.cgi?id=749536
+
+2015-08-12 17:16:01 +0530 Ravi Kiran K N <ravi.kiran@samsung.com>
+
+ * gst/matroska/matroska-demux.h:
+ * gst/matroska/matroska-parse.h:
+ matroska: Remove unused variable
+ https://bugzilla.gnome.org/show_bug.cgi?id=753556
+
+2015-08-04 20:59:17 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtp/Makefile.am:
+ * gst/rtp/gstrtpL16depay.c:
+ * gst/rtp/gstrtpL24depay.c:
+ * gst/rtp/gstrtpac3depay.c:
+ * gst/rtp/gstrtpac3pay.c:
+ * gst/rtp/gstrtpamrdepay.c:
+ * gst/rtp/gstrtpamrpay.c:
+ * gst/rtp/gstrtpbvdepay.c:
+ * gst/rtp/gstrtpceltdepay.c:
+ * gst/rtp/gstrtpceltpay.c:
+ * gst/rtp/gstrtpdvdepay.c:
+ * gst/rtp/gstrtpdvpay.c:
+ * gst/rtp/gstrtpg722depay.c:
+ * gst/rtp/gstrtpg723pay.c:
+ * gst/rtp/gstrtpg726depay.c:
+ * gst/rtp/gstrtpg729depay.c:
+ * gst/rtp/gstrtpg729pay.c:
+ * gst/rtp/gstrtpgsmdepay.c:
+ * gst/rtp/gstrtpgsmpay.c:
+ * gst/rtp/gstrtpgstdepay.c:
+ * gst/rtp/gstrtpgstpay.c:
+ * gst/rtp/gstrtph261depay.c:
+ * gst/rtp/gstrtph261pay.c:
+ * gst/rtp/gstrtph263depay.c:
+ * gst/rtp/gstrtph263pay.c:
+ * gst/rtp/gstrtph263pdepay.c:
+ * gst/rtp/gstrtph263ppay.c:
+ * gst/rtp/gstrtph264depay.c:
+ * gst/rtp/gstrtph264pay.c:
+ * gst/rtp/gstrtpilbcdepay.c:
+ * gst/rtp/gstrtpj2kdepay.c:
+ * gst/rtp/gstrtpj2kpay.c:
+ * gst/rtp/gstrtpjpegdepay.c:
+ * gst/rtp/gstrtpjpegpay.c:
+ * gst/rtp/gstrtpmp1sdepay.c:
+ * gst/rtp/gstrtpmp2tdepay.c:
+ * gst/rtp/gstrtpmp2tpay.c:
+ * gst/rtp/gstrtpmp4adepay.c:
+ * gst/rtp/gstrtpmp4apay.c:
+ * gst/rtp/gstrtpmp4gdepay.c:
+ * gst/rtp/gstrtpmp4gpay.c:
+ * gst/rtp/gstrtpmp4vdepay.c:
+ * gst/rtp/gstrtpmp4vpay.c:
+ * gst/rtp/gstrtpmpadepay.c:
+ * gst/rtp/gstrtpmpapay.c:
+ * gst/rtp/gstrtpmpvdepay.c:
+ * gst/rtp/gstrtpmpvpay.c:
+ * gst/rtp/gstrtppcmadepay.c:
+ * gst/rtp/gstrtppcmudepay.c:
+ * gst/rtp/gstrtpqcelpdepay.c:
+ * gst/rtp/gstrtpqdmdepay.c:
+ * gst/rtp/gstrtpsbcdepay.c:
+ * gst/rtp/gstrtpsbcpay.c:
+ * gst/rtp/gstrtpsirendepay.c:
+ * gst/rtp/gstrtpspeexdepay.c:
+ * gst/rtp/gstrtpspeexpay.c:
+ * gst/rtp/gstrtpsv3vdepay.c:
+ * gst/rtp/gstrtptheoradepay.c:
+ * gst/rtp/gstrtptheorapay.c:
+ * gst/rtp/gstrtptheorapay.h:
+ * gst/rtp/gstrtputils.c:
+ * gst/rtp/gstrtputils.h:
+ * gst/rtp/gstrtpvorbisdepay.c:
+ * gst/rtp/gstrtpvorbispay.c:
+ * gst/rtp/gstrtpvorbispay.h:
+ * gst/rtp/gstrtpvp8depay.c:
+ * gst/rtp/gstrtpvp8pay.c:
+ * gst/rtp/gstrtpvrawdepay.c:
+ * gst/rtp/gstrtpvrawpay.c:
+ rtp: Copy metadata in the (de)payloader, but only the relevant ones
+ The payloader didn't copy anything so far, the depayloader copied every
+ possible meta. Let's make it consistent and just copy all metas without
+ tags or with only the video tag.
+ https://bugzilla.gnome.org/show_bug.cgi?id=751774
+
+2015-08-10 18:20:15 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: fix small typo in comment
+
+2015-08-10 16:19:18 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * gst/goom2k1/gstgoom.c:
+ goom2k1/doc: Fixup previous commit
+
+2015-08-10 15:55:19 -0400 Nicolas Dufresne <nicolas.dufresne@collabora.com>
+
+ * docs/plugins/gst-plugins-good-plugins-sections.txt:
+ * gst/goom2k1/gstgoom.c:
+ * gst/goom2k1/gstgoom.h:
+ goom2k1/doc: Use GstGoom2k1 namespace
+ The doc generator isn't happy when we have class name clash. Simply
+ use it's own namespace.
+
+2015-08-10 17:10:42 +0530 Prashant Gotarne <ps.gotarne@samsung.com>
+
+ * gst/audiofx/audioecho.c:
+ audioecho: removed unused variable in set_property
+ unused local variable 'delay' is removed.
+ https://bugzilla.gnome.org/show_bug.cgi?id=753450
+
+2015-08-10 12:45:27 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: fix suboptimal queue iteration code
+
+2015-08-09 17:25:45 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: don't use glib 2.44-only API
+
+2015-07-29 14:14:50 +0100 Alex Ashley <bugzilla@ashley-family.net>
+
+ * gst/isomp4/fourcc.h:
+ * gst/isomp4/qtdemux.c:
+ * gst/isomp4/qtdemux.h:
+ * gst/isomp4/qtdemux_types.c:
+ qtdemux: add support for ISOBMFF Common Encryption
+ This commit adds support for ISOBMFF Common Encryption (cenc), as
+ defined in ISO/IEC 23001-7. It uses a GstProtection event to
+ pass the contents of PSSH boxes to downstream decryptor elements
+ and attached GstProtectionMeta to each sample.
+ https://bugzilla.gnome.org/show_bug.cgi?id=705991
+
+2015-08-10 14:13:50 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
+
+ * gst/rtp/gstrtph264depay.c:
+ rtph264depay: checking if depay has sps/pps nals before insertion
+ https://bugzilla.gnome.org/show_bug.cgi?id=753430
+
+2015-08-08 16:44:49 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/matroska/matroska-mux.c:
+ matroskamux: fix outdated comment
+ The default behaviour was changed in the 0.10 -> 1.x
+ transition, but the comment was not updated.
+
+2015-08-08 17:42:22 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtp/gstrtptheorapay.c:
+ rtptheorapay: If flushing a packet failed, go out of the loop immediately
+
+2015-08-08 17:41:02 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtp/gstrtpvorbispay.c:
+ rtpvorbispay: If flushing a packet failed, go out of the loop immediately
+
+2015-08-08 17:34:50 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtp/gstrtptheorapay.c:
+ * gst/rtp/gstrtptheorapay.h:
+ rtptheorapay: Extract pixel format from the ident header to put it into the sampling field of the caps
+ We always put 4:2:0 into the caps before, which obviously is wrong for 4:2:2
+ and 4:4:4 formats.
+
+2015-08-06 17:46:13 +0200 George Kiagiadakis <george.kiagiadakis@collabora.com>
+
+ * gst/rtp/gstrtpklvdepay.c:
+ * gst/rtp/gstrtpklvpay.c:
+ rtpklv(de)pay: add "RTP" in the klass string
+ GstRTSPMedia uses this classification to detect the real payloader
+ inside a dynpay bin and asserts if it doesn't find it, therefore
+ it is required
+ https://bugzilla.gnome.org/show_bug.cgi?id=753325
+
+2015-08-05 11:13:09 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * tests/check/elements/rtpaux.c:
+ tests: rtpaux: use a dynamic pt in the test
+ 1) Tests that using dynamic PT instead of the default ones work
+ 2) If we ever decide to change the codec here we don't need to
+ worry about change the PT for the default one of the new codec
+ in the test
+ https://bugzilla.gnome.org/show_bug.cgi?id=746445
+
+2015-08-05 10:53:15 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
+
+ * gst/rtpmanager/gstrtprtxsend.c:
+ rtprtxsend: print valid type where guint32 is expected
+ https://bugzilla.gnome.org/show_bug.cgi?id=746445
+
+2015-08-06 11:33:37 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
+
+ * gst/rtp/gstrtpL16pay.c:
+ * gst/rtp/gstrtpg722pay.c:
+ * gst/rtp/gstrtpg723pay.c:
+ * gst/rtp/gstrtpg729pay.c:
+ * gst/rtp/gstrtpgsmpay.c:
+ * gst/rtp/gstrtph261pay.c:
+ * gst/rtp/gstrtph263pay.c:
+ * gst/rtp/gstrtpjpegpay.c:
+ * gst/rtp/gstrtpmp2tpay.c:
+ * gst/rtp/gstrtpmpapay.c:
+ * gst/rtp/gstrtpmpvpay.c:
+ * gst/rtp/gstrtppcmapay.c:
+ * gst/rtp/gstrtppcmupay.c:
+ rtppayload: set standard payload type as default
+ Initialize the PT to the default value of the codec and check if
+ it is still the default before declaring the pt to be dynamic or
+ not when setting the caps.
+ Also use the PT constants from the rtp lib when possible
+ https://bugzilla.gnome.org/show_bug.cgi?id=747965
+
+2015-07-26 12:07:56 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: store the moof-offset also for push mode
+ It will be used in some cases for getting the correct offsets
+ from trun atoms.
+ https://bugzilla.gnome.org/show_bug.cgi?id=752603
+
+2015-07-26 02:09:24 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/isomp4/atoms.h:
+ * gst/isomp4/qtdemux.c:
+ * gst/isomp4/qtdemux_types.h:
+ qtdemux: handle default-base-is-moof flag
+ Handle the flag from the tfhd that signals the base offset to
+ start from the moof atom
+ https://bugzilla.gnome.org/show_bug.cgi?id=752603
+
+2015-07-29 18:54:35 -0600 Glen Diener <grd@loganmill.net>
+
+ * gst/matroska/matroska-demux.c:
+ * gst/matroska/matroska-read-common.c:
+ * gst/matroska/matroska-read-common.h:
+ matroskademux: Preserve forward referenced track tags
+ https://bugzilla.gnome.org/show_bug.cgi?id=752850
+
+2015-08-04 18:07:35 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * tests/check/elements/rtpaux.c:
+ tests: rtpaux: fix test failure
+ The RTP PT for alaw is 8.
+ Less than 50 packets are received in the length of this test so it
+ would never drop a buffer or would drop only the last buffer and
+ it would fail sometimes when the received wouldn't receive the
+ retransmission packet in time.
+ https://bugzilla.gnome.org/show_bug.cgi?id=746445
+
+2015-08-04 20:59:17 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtp/gstrtpstreamdepay.c:
+ rtpstreamdepay: Only allow activation in push mode
+ We need a proper caps event from upstream with the full RTP caps as we can't
+ create caps ourselves from thin air. Fixes usage of rtpstreamdepay after e.g.
+ a filesrc or any other element that supports pull mode.
+ https://bugzilla.gnome.org/show_bug.cgi?id=753066
+
+2015-08-04 16:28:17 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * ext/soup/gstsouphttpsrc.c:
+ soup: fix typo in translated string
+ https://bugzilla.gnome.org/show_bug.cgi?id=753240
+
+2015-08-04 12:25:46 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtp/gstrtph264depay.c:
+ rtph264depay: Put the profile and level into the caps
+
+2015-08-04 12:09:12 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtp/gstrtph264depay.c:
+ rtph264depay: Only update the srcpad caps if something else than the codec_data changed
+ h264parse does the same, let's keep the behaviour consistent. As we now
+ include the codec_data inside the stream too here, this causes less caps
+ renegotiation.
+
+2015-08-04 11:48:27 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtp/gstrtph264depay.c:
+ rtph264depay: PPS replaces and old PPS if it has the same id, independent of SPS id
+ The spec says:
+ When a picture parameter set NAL unit with a particular value of
+ pic_parameter_set_id is received, its content replaces the content of the
+ previous picture parameter set NAL unit, in decoding order, with the same
+ value of pic_parameter_set_id (when a previous picture parameter set NAL unit
+ with the same value of pic_parameter_set_id was present in the bitstream).
+
+2015-08-03 13:45:59 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/multifile/gstsplitmuxsink.c:
+ splitmuxsink: remove extra \n at debug message
+
+2015-08-03 13:42:20 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/multifile/gstsplitmuxsink.c:
+ splitmuxsink: prevent deadlock when states change too fast
+ If the GOP is completed, pads have to start gathering for the
+ next one but it is possible that the the state might go to
+ COLLECTING_GOP_START and back to WAITING_GOP_COMPLETE before the
+ thread has a chance to wake up and proceed, leaving it trapped in
+ the check_completed_gop loop and deadlocking the other threads
+ waiting for it to advance.
+ To solve it, this patch also checks that tha input running time
+ hasn't changed to prevent this scenario.
+
+2015-08-03 17:55:01 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtp/gstrtph264depay.c:
+ rtph264depay: Insert SPS/PPS NALs into the stream
+ h264parse does the same and this fixes decoding of some streams with 32 SPS
+ (or 256 PPS). It is allowed to have SPS ID 0 to 31 (or PPS ID 0 to 255), but
+ the field in the codec_data for the number of SPS or PPS is only 5 (or 8) bit.
+ As such, 32 SPS (or 256 PPS) are interpreted as 0 everywhere.
+ This looks like a mistake in the part of the spec about the codec_data.
+
+2015-07-30 11:29:27 +0900 Eunhae Choi <eunhae1.choi@samsung.com>
+
+ * ext/soup/gstsouphttpsrc.c:
+ souphttpsrc: handle empty http proxy string
+ 1) If the system http_proxy environment variable is not set
+ or set to an empty string, we must not set proxy to avoid
+ http connection error.
+ 2) In case of proxy property setting, if user want to clear
+ the proxy setting, they should be able to set it to NULL or
+ an empty string again, so this is fixed too.
+ 3) Check if the proxy string was parsed correctly.
+ https://bugzilla.gnome.org/show_bug.cgi?id=752866
+
+2015-07-29 15:46:20 +0530 Ravi Kiran K N <ravi.kiran@samsung.com>
+
+ * ext/dv/gstdvdemux.c:
+ * ext/dv/gstdvdemux.h:
+ dvdemux: remove unused variable
+ Remove unused variable 'framecount' from dvdemux
+ https://bugzilla.gnome.org/show_bug.cgi?id=753008
+
+2015-07-30 15:32:09 +0900 Vineeth TM <vineeth.tm@samsung.com>
+
+ * gst/rtsp/gstrtspsrc.c:
+ rtspsrc: assertion error due to wrong condition check
+ In media to caps function, reserved_keys array is being used for variable i,
+ leading to GLib-CRITICAL **: g_ascii_strcasecmp: assertion 's1 != NULL' failed
+ changed it to variable j
+ https://bugzilla.gnome.org/show_bug.cgi?id=753009
+
+2015-07-30 15:21:20 +0900 Vineeth TM <vineeth.tm@samsung.com>
+
+ * gst/rtp/gstrtpmp4vdepay.c:
+ rtpmp4vdepay: rtpbuffer is being unref'ed twice
+ process_rtp_packet doesn't transfer the rtp buffer to mp4v_process_depay
+ the refernce should not be removed here
+ https://bugzilla.gnome.org/show_bug.cgi?id=753042
+
+2015-07-29 11:26:46 +0100 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp/gstrtspsrc.c:
+ rtspsrc: Strip keys from the fmtp that we use internally in our caps
+ Skip keys from the fmtp, which we already use ourselves for the
+ caps. Some software is adding random things like clock-rate into
+ the fmtp, and we would otherwise here set a string-typed clock-rate
+ in the caps... and thus fail to create valid RTP caps
+ https://bugzilla.gnome.org/show_bug.cgi?id=753009
+
+2015-07-29 19:28:33 +1000 Jan Schmidt <jan@centricular.com>
+
+ * gst/multifile/gstsplitmuxsink.c:
+ splitmuxsink: Support mpegtsmux as a muxer.
+ As a fallback, look for a pad template sink_%d on
+ the muxer when requesting pads, to support mpegtsmux
+ https://bugzilla.gnome.org/show_bug.cgi?id=752999
+
+2015-06-25 01:35:27 +1000 Jan Schmidt <jan@centricular.com>
+
+ * gst/multifile/gstsplitmuxpartreader.c:
+ * gst/multifile/gstsplitmuxpartreader.h:
+ splitmuxsrc: Use a separate lock to delay typefind.
+ Don't hold the main splitmux part lock over
+ the parent state change function, as it prevents
+ posting error messages that happen. Since the purpose
+ is to prevent typefinding from proceeding, use a
+ separate mutex just for that.
+
+2015-07-29 13:43:50 +0900 Vineeth TM <vineeth.tm@samsung.com>
+
+ * gst/matroska/matroska-read-common.c:
+ matroska: fix memory leak
+ After adding to tag list, key_val is not being free'd
+ resulting in memory leak
+ https://bugzilla.gnome.org/show_bug.cgi?id=752992
+
+2015-07-27 13:34:14 +0900 Manasa Athreya <manasa.athreya@lge.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: fix 16-bit PCM audio advertised with 'raw ' fourcc
+ 'NONE' and 'raw ' fourcc don't always contain U8 audio, it can
+ be more bits as well, in which case it's just like 'twos'.
+ https://bugzilla.gnome.org/show_bug.cgi?id=752613
+
+2015-07-24 15:10:05 +0200 Dimitrios Katsaros <patcherwork@gmail.com>
+
+ * sys/v4l2/gstv4l2object.c:
+ * sys/v4l2/gstv4l2src.c:
+ v4l2: Allow framerate to be large then 100pfs
+ This limit was arbitrary. We still fixate near 100pfs for compatibility.
+ https://bugzilla.gnome.org/show_bug.cgi?id=752825
+
+2015-07-25 03:25:28 -0400 Olivier Crête <olivier.crete@ocrete.ca>
+
+ * gst/avi/gstavidemux.c:
+ avidemux: Stop without posting error on flushing
+ This could just be a normal pipeline shutdown.
+
+2015-07-23 15:00:08 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
+
+ * sys/v4l2/gstv4l2bufferpool.c:
+ v4l2bufferpool: set GST_BUFFER_COPY_FLAGS to copy flags also
+ https://bugzilla.gnome.org/show_bug.cgi?id=752618
+
+2015-07-16 18:09:30 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/check/Makefile.am:
+ * tests/check/elements/.gitignore:
+ * tests/check/elements/matroskademux.c:
+ tests: add minmal matroskademux test for subtitle output
+ Some of the subtitle chunks will have embedded
+ NUL-terminators (last three), some don't (first three),
+ some will have markup, some won't, some will be valid
+ UTF-8 (all but last), some won't (last stanza).
+ https://bugzilla.gnome.org/show_bug.cgi?id=752421
+
+2015-07-16 18:49:26 +0300 Dimitrios Christidis <dchristidis@mykolab.com>
+
+ * gst/matroska/matroska-demux.c:
+ matroskademux: fix for subtitle buffers with NUL terminators
+ Commit 45892ec8 created a regression where g_utf8_validate() would fail
+ if the subtitle buffer had a NUL terminator as part of the data.
+ https://bugzilla.gnome.org/show_bug.cgi?id=752421
+
+2015-07-21 13:31:05 +0200 Stian Selnes <stian@pexip.com>
+
+ * gst/rtp/gstrtpvp8depay.c:
+ rtpvp8depay: Check available bytes before copy
+ Need to check that the number of bytes we want to copy from the adapter
+ actually is available and handle the error case gracefully. This error
+ may happen if malformed packets are received and we don't have a
+ complete frame.
+ https://bugzilla.gnome.org/show_bug.cgi?id=752663
+
+2015-07-16 09:32:36 +0900 Paul Hyunil <paul.hyunil@lge.com>
+
+ * gst/isomp4/fourcc.h:
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Support subtitle when track subtype is fourcc_subt
+ https://bugzilla.gnome.org/show_bug.cgi?id=752655
+
+2015-07-20 16:59:40 +0800 Song Bing <b06498@freescale.com>
+
+ * sys/v4l2/gstv4l2bufferpool.c:
+ v4l2bufferpool: Set timestamp when queue buffer.
+ Should set timestamp when queue buffer.
+ https://bugzilla.gnome.org/show_bug.cgi?id=752618
+
+2015-07-16 15:12:17 +0200 Havard Graff <havard.graff@gmail.com>
+
+ * gst/rtpmanager/gstrtpmux.c:
+ * tests/check/elements/rtpmux.c:
+ rtpmux: handle different ssrc's on sinkpads
+ Do this by not putting the ssrc from the src pads in the caps used to
+ probe other sinkpads, and then intersecting with it later.
+ https://bugzilla.gnome.org/show_bug.cgi?id=752491
+
+2015-07-16 17:19:03 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/avi/gstavimux.c:
+ * gst/matroska/matroska-demux.c:
+ * gst/matroska/matroska-mux.c:
+ * gst/matroska/matroska-parse.c:
+ * gst/matroska/webm-mux.c:
+ Update mailing list address from sourceforge to freedesktop
+
+2015-07-15 13:44:52 +0300 Dimitrios Christidis <dchristidis@mykolab.com>
+
+ * gst/matroska/matroska-demux.c:
+ matroskademux: fix trailing '*' displayed with some text subtitles
+ The subtitle buffer we push out should not include a NUL terminator
+ as part of the data, we just add such a terminator for safety, but
+ it should not be included in the buffer size.
+ A NUL terminator is not valid UTF-8, so checks will fail if it's
+ included in the size, and the NUL will be replaced by the fallback
+ character specified when converting, i.e. '*'.
+ https://bugzilla.gnome.org/show_bug.cgi?id=752421
+
+2015-07-15 18:23:05 +0200 Wim Taymans <wtaymans@redhat.com>
+
+ * ext/pulse/pulsedeviceprovider.c:
+ * ext/pulse/pulseutil.c:
+ * ext/pulse/pulseutil.h:
+ pulse: add properties to GstDevice
+ Add the extra properties we get from pulse to the GstDevice we expose
+ with the device monitor
+
+2015-07-15 17:20:20 +0530 Ravi Kiran K N <ravi.kiran@samsung.com>
+
+ * gst/audiofx/audioinvert.c:
+ * gst/audiofx/audiowsincband.c:
+ audiofx: Fix typo in example pipelines
+ Fix typo in example pipelines of audiowsincband and audioinvert.
+ https://bugzilla.gnome.org/show_bug.cgi?id=752416
+
+2015-04-15 18:27:04 +0200 George Kiagiadakis <george.kiagiadakis@collabora.com>
+
+ * gst/multifile/gstsplitmuxsink.c:
+ splitmuxsink: add a "format-location" signal that allows better control over filenames
+ In certain applications, splitting into files named after a base
+ location template and an incremental sequence number is not enough.
+ This signal gives more fine-grained control to the application to
+ decide how to name the files.
+ https://bugzilla.gnome.org/show_bug.cgi?id=750106
+
+2015-04-15 20:13:27 +0300 Ilya Konstantinov <ilya.konstantinov@gmail.com>
+
+ * sys/osxaudio/gstosxcoreaudio.c:
+ osxaudiosrc: no resampling on OS X
+ Unlike Remote IO, AUHAL doesn't have built-in resampling
+ for sources -- confirmed by Core Audio engineer Doug Wyatt:
+ http://lists.apple.com/archives/coreaudio-api/2006/Sep/msg00088.html
+ https://bugzilla.gnome.org/show_bug.cgi?id=743758
+
+2015-04-15 18:29:14 +0300 Ilya Konstantinov <ilya.konstantinov@gmail.com>
+
+ * sys/osxaudio/gstosxcoreaudio.c:
+ osxaudiosrc: avoid get_channel_layout
+ This only produces a warning and serves no purpose.
+ https://bugzilla.gnome.org/show_bug.cgi?id=743758
+
+2015-04-07 15:40:14 +0530 Arun Raghavan <arun@centricular.com>
+
+ * sys/osxaudio/gstosxcoreaudio.c:
+ osxaudio: Avoid making a duplicate structure in caps for mono/stereo case
+ For 1ch or 2ch devices, we just need to set the caps to allow both
+ options since CoreAudio will up/downmix appropriately.
+ Also fixes the condition for the 2ch case to be exact, rather than at
+ least 2 channels since the downmix will not take place in the >stereo
+ case.
+
+2015-04-06 16:22:34 +0530 Arun Raghavan <arun@centricular.com>
+
+ * sys/osxaudio/gstosxcoreaudio.c:
+ * sys/osxaudio/gstosxcoreaudiocommon.c:
+ * sys/osxaudio/gstosxcoreaudiohal.c:
+ * sys/osxaudio/gstosxcoreaudioremoteio.c:
+ osxaudio: Don't set the format on an initialized AudioUnit
+ We need to initialize the AudioUnit early to be able to probe the
+ underlying device, but according to the AudioUnitInitialize() and
+ AudioUnitUninitialize() documentation, format changes should be done
+ while the AudioUnit is uninitialized. So we explicitly uninitialize the
+ AudioUnit during a format change and reinitialize it when we're done.
+
+2015-04-06 15:55:59 +0530 Arun Raghavan <arun@centricular.com>
+
+ * sys/osxaudio/gstosxaudioringbuffer.c:
+ * sys/osxaudio/gstosxcoreaudio.c:
+ * sys/osxaudio/gstosxcoreaudio.h:
+ osxaudio: Minor spelling fix (unitialize -> uninitialize)
+
+2015-03-21 20:34:25 +0200 Ilya Konstantinov <ilya.konstantinov@gmail.com>
+
+ * sys/osxaudio/gstosxaudiosink.c:
+ * sys/osxaudio/gstosxaudiosrc.c:
+ * sys/osxaudio/gstosxcoreaudio.c:
+ * sys/osxaudio/gstosxcoreaudio.h:
+ osxaudio: Fix lockup in _audio_unit_property_listener
+ _audio_unit_property_listener is called either from a Core Audio thread
+ or as a result of a Core Audio API (e.g. AudioUnitInitialize)
+ from our own thread. In the latter case, osxbuf can be already locked
+ (GStreamer's mutex is not recursive).
+ We introduce the flag cached_caps_valid and use it instead of nullifying
+ cached_caps when we cannot lock on osxbuf.
+ https://bugzilla.gnome.org/show_bug.cgi?id=743758
+
+2015-03-12 12:15:12 +0200 Ilya Konstantinov <ilya.konstantinov@gmail.com>
+
+ * sys/osxaudio/gstosxcoreaudio.c:
+ osxaudio: Invalidate cached caps on format change
+ Listen for changes in hardware stream format and channel layout, and
+ invalidate cached caps (since they contain the preferred caps).
+ https://bugzilla.gnome.org/show_bug.cgi?id=743758
+
+2015-03-09 23:34:06 +0200 Ilya Konstantinov <ilya.konstantinov@gmail.com>
+
+ * sys/osxaudio/gstosxaudioringbuffer.c:
+ * sys/osxaudio/gstosxaudiosink.c:
+ * sys/osxaudio/gstosxaudiosink.h:
+ * sys/osxaudio/gstosxaudiosrc.c:
+ * sys/osxaudio/gstosxaudiosrc.h:
+ * sys/osxaudio/gstosxcoreaudio.c:
+ * sys/osxaudio/gstosxcoreaudio.h:
+ * sys/osxaudio/gstosxcoreaudiocommon.c:
+ * sys/osxaudio/gstosxcoreaudiocommon.h:
+ * sys/osxaudio/gstosxcoreaudiohal.c:
+ * sys/osxaudio/gstosxcoreaudioremoteio.c:
+ osxaudio: Overhaul of probing caps
+ - Probing caps is unified between source and sink
+ - Hardware stream format is now reported as preferred capabilities
+ (dynamically updated when hardware configuration changes)
+ - Get hardware channel layout from Remote IO just like from HAL
+ - More comprehensive mapping between AudioChannelLabel and
+ GstAudioChannelPosition
+ - Support for unpositioned channel layouts
+ - Announce stereo-mono upmixing/downmixing in caps
+ https://bugzilla.gnome.org/show_bug.cgi?id=743758
+
+2015-03-09 23:15:56 +0200 Ilya Konstantinov <ilya.konstantinov@gmail.com>
+
+ * sys/osxaudio/gstosxcoreaudio.c:
+ osxaudio: AudioUnitInitialize on open
+ Call AudioUnitInitialize upon open. Otherwise, we cannot get
+ (hardware) stream format nor channel layout from the outer scope.
+
+2015-07-12 14:27:15 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/rtp/gstrtpL16depay.c:
+ * gst/rtp/gstrtpL24depay.c:
+ * gst/rtp/gstrtpac3depay.c:
+ * gst/rtp/gstrtpamrdepay.c:
+ * gst/rtp/gstrtpbvdepay.c:
+ * gst/rtp/gstrtpceltdepay.c:
+ * gst/rtp/gstrtpdvdepay.c:
+ * gst/rtp/gstrtpg722depay.c:
+ * gst/rtp/gstrtpg723depay.c:
+ * gst/rtp/gstrtpg726depay.c:
+ * gst/rtp/gstrtpg729depay.c:
+ * gst/rtp/gstrtpgsmdepay.c:
+ * gst/rtp/gstrtpgstdepay.c:
+ * gst/rtp/gstrtph261depay.c:
+ * gst/rtp/gstrtph263depay.c:
+ * gst/rtp/gstrtph263pdepay.c:
+ * gst/rtp/gstrtph264depay.c:
+ * gst/rtp/gstrtpilbcdepay.c:
+ * gst/rtp/gstrtpj2kdepay.c:
+ * gst/rtp/gstrtpjpegdepay.c:
+ * gst/rtp/gstrtpklvdepay.c:
+ * gst/rtp/gstrtpmp1sdepay.c:
+ * gst/rtp/gstrtpmp2tdepay.c:
+ * gst/rtp/gstrtpmp4adepay.c:
+ * gst/rtp/gstrtpmp4gdepay.c:
+ * gst/rtp/gstrtpmp4vdepay.c:
+ * gst/rtp/gstrtpmpadepay.c:
+ * gst/rtp/gstrtpmparobustdepay.c:
+ * gst/rtp/gstrtpmpvdepay.c:
+ * gst/rtp/gstrtppcmadepay.c:
+ * gst/rtp/gstrtppcmudepay.c:
+ * gst/rtp/gstrtpqcelpdepay.c:
+ * gst/rtp/gstrtpqdmdepay.c:
+ * gst/rtp/gstrtpsbcdepay.c:
+ * gst/rtp/gstrtpsirendepay.c:
+ * gst/rtp/gstrtpspeexdepay.c:
+ * gst/rtp/gstrtpsv3vdepay.c:
+ * gst/rtp/gstrtptheoradepay.c:
+ * gst/rtp/gstrtpvorbisdepay.c:
+ * gst/rtp/gstrtpvp8depay.c:
+ rtp: depayloaders: implement process_rtp_packet() vfunc
+ For more optimised RTP packet handling: means we don't
+ need to map the input buffer again but can just re-use
+ the mapping the base class has already done.
+ https://bugzilla.gnome.org/show_bug.cgi?id=750235
+
+2015-05-27 19:19:27 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/rtp/gstrtpvrawdepay.c:
+ rtpvrawdepay: implement process_rtp_packet() vfunc
+ For more optimised RTP packet handling: means we don't
+ need to map the input buffer again but can just re-use
+ the map the base class has already done.
+ https://bugzilla.gnome.org/show_bug.cgi?id=750235
+
+2015-07-10 00:13:32 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ rtpjitterbuffer: Fix indention
+
+2015-07-09 23:59:10 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ rtpjitterbuffer: Always estimate DTS from the current clock time
+ Estimating it from the RTP time will give us the PTS, so in cases of PTS!=DTS
+ we would produce wrong DTS. As now the estimated DTS is based on the clock,
+ don't store it in the jitterbuffer items as it would otherwise be used in the
+ skew calculations and would influence the results. We only really need the DTS
+ for timer calculations.
+ https://bugzilla.gnome.org/show_bug.cgi?id=749536
+
+2015-07-09 09:26:09 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * tests/check/elements/.gitignore:
+ gitignore: ignore rtph263 test
+
+2015-07-08 23:47:44 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * tests/check/elements/rtpjitterbuffer.c:
+ rtpjitterbuffer: fix build error with gcc (Debian 4.9.2-21) 4.9.2
+ Replace static constants with macros to make gcc happy
+ CC elements/elements_rtpjitterbuffer-rtpjitterbuffer.o
+ elements/rtpjitterbuffer.c:387:1: error: initializer element is not constant
+ static const GstClockTime PCMU_BUF_DURATION = PCMU_BUF_MS * GST_MSECOND;
+ ^
+ elements/rtpjitterbuffer.c:388:1: error: initializer element is not constant
+ static const guint PCMU_BUF_SIZE = 64000 * PCMU_BUF_MS / 1000;
+ ^
+ elements/rtpjitterbuffer.c:390:5: error: initializer element is not constant
+ PCMU_BUF_CLOCK_RATE * PCMU_BUF_MS / 1000;
+
+2015-07-08 23:40:45 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * tests/check/elements/rtpjitterbuffer.c:
+ rtpjitterbuffer: run indent and fix some comments
+ Fix indent on this file and break some comment lines into two to make
+ it fit 80 chars per line
+
+2015-07-08 15:02:24 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: rework segment event handling for adaptive streaming
+ When a new time segment is received upstream is going to restart
+ with a new atom. Make the neededbytes and todrop variables
+ reflect that to avoid waiting too much or dropping the
+ initial bytes that contain the header.
+
+2015-07-08 12:35:55 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: push data from adapter before starting new segment
+ The adapter might have data remaining from the previous segment,
+ push it all before clearing the adapter and starting a new segment.
+ It can accumulate data if it had pushed and got not-linked, returning
+ immediately without processing all the data. Before starting a new
+ segment this data should be handled.
+
+2015-07-08 19:59:13 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ rtpjitterbuffer: Calculate DTS from the clock if we had none for the first packet after a reset
+ https://bugzilla.gnome.org/show_bug.cgi?id=749536
+
+2015-07-08 21:08:36 +0200 Havard Graff <havard.graff@gmail.com>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ * tests/check/elements/rtpjitterbuffer.c:
+ rtpjitterbuffer: fix gap-time calculation and remove "late"
+ The amount of time that is completely expired and not worth waiting for,
+ is the duration of the packets in the gap (gap * duration) - the
+ latency (size) of the jitterbuffer (priv->latency_ns). This is the duration
+ that we make a "multi-lost" packet for.
+ The "late" concept made some sense in 0.10 as it reflected that a buffer
+ coming in had not been waited for at all, but had a timestamp that was
+ outside the jitterbuffer to wait for. With the rewrite of the waiting
+ (timeout) mechanism in 1.0, this no longer makes any sense, and the
+ variable no longer reflects anything meaningful (num > 0 is useless,
+ the duration is what matters)
+ Fixed up the tests that had been slightly modified in 1.0 to allow faulty
+ behavior to sneak in, and port some of them to use GstHarness.
+ https://bugzilla.gnome.org/show_bug.cgi?id=738363
+
+2015-06-30 11:21:31 +0200 Stian Selnes <stian@pexip.com>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ Revert "rtpjitterbuffer: Fix expected_dts calc in calculate_expected"
+ This reverts commit 05bd708fc5e881390fe839803b53144393d95ab0.
+ The reverted patch is wrong and introduces a regression because there
+ may still be time to receive some of the packets included in the gap
+ if they are reordered.
+
+2015-07-07 23:53:02 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: flush samples before adding more from moof
+ Avoids accumulating all samples from a fragmented stream that could
+ lead to a 'index-too-big' error once it goes over 50MB of data. It
+ could reach that before 2h of playback so it doesn't take that long.
+ As upstream elements are providing data in time format they should
+ be the ones that have more information about the full media index
+ and should be able to seek if possible.
+
+2015-07-07 23:56:12 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/isomp4/qtdemux.c:
+ * gst/isomp4/qtdemux.h:
+ qtdemux: rename upstream_newsegment to upstream_format_is_time
+ upstream_newsegment isn't really clear on what it means, it is set
+ to TRUE when the upstream element sends a segment in TIME format, so
+ rename it to be more clear about it.
+ It is important to know this because it means that upstream has
+ a notion of time and qtdemux is likely being driven by an upstream
+ element that is reading from a higher level abstraction than a file,
+ such as a DASH, MSS or DLNA element.
+
+2015-07-07 21:31:08 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: fix leak by flushing previous sample info from trak
+ In fragmented streaming, multiple moov/moof will be parsed and their
+ previously stored samples array might leak when new values are parsed.
+ The parse_trak and callees won't free the previously stored values
+ before parsing the new ones.
+ In step-by-step, this is what happens:
+ 1) initial moov is parsed, traks as well, streams are created. The
+ trak doesn't contain samples because they are in the moof's trun
+ boxes. n_samples is set to 0 while parsing the trak and the samples
+ array is still NULL.
+ 2) moofs are parsed, and their trun boxes will increase n_samples and
+ create/extend the samples array
+ 3) At some point a new moov might be sent (bitrate switching, for example)
+ and parsing the trak will overwrite n_samples with the values from
+ this trak. If the n_samples is set to 0 qtdemux will assume that
+ the samples array is NULL and will leak it when a new one is
+ created for the subsequent moofs.
+ This patch makes qtdemux properly free previous sample data before
+ creating new ones and adds an assert to catch future occurrences of
+ this issue when the code changes.
+
+2015-07-07 16:46:33 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: fix index size check and debug message
+ It is allocating samples_count + n_samples, not only n_samples
+
+2015-07-08 17:02:05 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ rtpjitterbuffer: Calculate receive time if we don't have any
+ This is required to properly schedule packet loss timers and make
+ sure all our calculations work properly.
+ https://bugzilla.gnome.org/show_bug.cgi?id=749536
+
+2015-07-08 15:13:17 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ rtpjitterbuffer: Handle seqnum gaps in TCP streams without erroring out or overflowing calculations
+ That is, handle DTS==GST_CLOCK_TIME_NONE correctly.
+ https://bugzilla.gnome.org/show_bug.cgi?id=749536
+
+2015-07-08 20:31:42 +0900 Vineeth T M <vineeth.tm@samsung.com>
+
+ * gst/avi/gstavidemux.c:
+ avidemux: fix event leak
+ when seek fails in avidemux, event is not being freed.
+ https://bugzilla.gnome.org/show_bug.cgi?id=752117
+
+2015-07-08 12:02:22 +0200 Stian Selnes <stian@pexip.com>
+
+ * gst/rtp/gstrtph263depay.c:
+ * tests/check/Makefile.am:
+ * tests/check/elements/rtph263.c:
+ rtph263depay: Make sure payload is large enough
+ Plus new unit test.
+ https://bugzilla.gnome.org/show_bug.cgi?id=752112
+
+2015-07-08 08:59:49 +0900 Vineeth TM <vineeth.tm@samsung.com>
+
+ * gst/rtp/gstrtpklvdepay.c:
+ rtpklvdepay: fix printf format compiler warning
+ v_len is of type guint64, but while print the value(16 + len_size + v_len)
+ G_GSIZE_FORMAT is being used instead of G_GUINT64_FORMAT
+ https://bugzilla.gnome.org/show_bug.cgi?id=752100
+
+2015-07-07 20:25:47 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * docs/plugins/gst-plugins-good-plugins-docs.sgml:
+ * docs/plugins/gst-plugins-good-plugins-sections.txt:
+ * docs/plugins/gst-plugins-good-plugins.args:
+ * docs/plugins/gst-plugins-good-plugins.hierarchy:
+ * docs/plugins/inspect/plugin-rtp.xml:
+ docs: add new RTP elements to docs
+
+2015-07-07 20:07:31 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/check/elements/rtp-payloading.c:
+ tests: rtp-payloading: add basic unit test for KLV payloading
+ Also make it so that the mtu is always set if specified, not
+ only in case of the rather weird bufferlist test code path.
+ This allows us to easily make the payloader fragment a payload
+ across multiple output packets by setting a small MTU on it.
+
+2015-07-07 19:58:42 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/rtp/gstrtpklvdepay.c:
+ * gst/rtp/gstrtpklvdepay.h:
+ rtpklvdepay: improve start detection and handle fragmented KLV units
+
+2015-07-05 20:25:10 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/rtp/Makefile.am:
+ * gst/rtp/gstrtp.c:
+ * gst/rtp/gstrtpklvdepay.c:
+ * gst/rtp/gstrtpklvdepay.h:
+ rtp: add SMPTE 336M KLV metadata depayloader
+ http://tools.ietf.org/html/rfc6597
+
+2014-08-09 10:08:42 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/rtp/Makefile.am:
+ * gst/rtp/gstrtp.c:
+ * gst/rtp/gstrtpklvpay.c:
+ * gst/rtp/gstrtpklvpay.h:
+ rtp: add SMPTE 336M KLV metadata payloader
+ http://tools.ietf.org/html/rfc6597
+
+2015-07-07 16:59:20 +0200 Stefan Sauer <ensonic@users.sf.net>
+
+ * gst/isomp4/atoms.c:
+ * gst/isomp4/atoms.h:
+ * gst/isomp4/atomsrecovery.c:
+ * gst/isomp4/properties.h:
+ * gst/matroska/matroska-mux.c:
+ * gst/rtpmanager/rtpsource.c:
+ docs: fix "Symbol name not found at the start of the comment block"
+ Add symbols or change comment into a regular comment.
+
+2015-07-07 16:58:53 +0200 Stefan Sauer <ensonic@users.sf.net>
+
+ * gst/audioparsers/gstamrparse.h:
+ docs: remove outdated doc strings
+
+2015-07-03 23:10:40 +0200 Stefan Sauer <ensonic@users.sf.net>
+
+ * docs/plugins/gst-plugins-good-plugins-docs.sgml:
+ docs: add missing plugins and ensure master doc is sorted
+
+2015-07-07 15:54:41 +0100 Luis de Bethencourt <luis@debethencourt.com>
+
+ * gst/imagefreeze/gstimagefreeze.c:
+ Revert "imagefreeze: Remove impossible error condition"
+ This reverts commit d46631c5c7312ad613397f8238c7a9714ae3ae94.
+ pad only handle EOS events but not EOS flow, and will push the buffer again
+ resulting in an assertion error. So we should not handle the buffer
+ and return EOS flow.
+
+2015-07-07 15:50:50 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/rtp/gstrtpg729depay.c:
+ rtpg729depay: unmap rtp buffer in error path
+
+2015-07-07 15:48:40 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/rtp/gstrtpg729pay.c:
+ rtpg729pay: fix buffer leak
+ The handle_buffer vfunc takes ownership of the input buffer.
+ Fixes elements/rtp-payloading under valgrind.
+
+2015-07-02 08:52:43 +0200 Tobias Mueller <muelli@cryptobitch.de>
+
+ * gst/goom/goom_core.c:
+ goom: Initialised variables to remove compiler warnings
+ goom_core.c: In function 'goom_update':
+ goom_core.c:685:5: error: 'param2' may be used uninitialized in this function [-Werror=maybe-uninitialized]
+ goom_lines_switch_to (goomInfo->gmline2, mode, param2, amplitude, couleur);
+ ^
+ goom_core.c:684:5: error: 'param1' may be used uninitialized in this function [-Werror=maybe-uninitialized]
+ goom_lines_switch_to (goomInfo->gmline1, mode, param1, amplitude, couleur);
+ ^
+ https://bugzilla.gnome.org/show_bug.cgi?id=752053
+
+2015-07-07 09:18:39 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/rtp/gstrtph261pay.c:
+ rtph261pay: fix indentation
+
+2015-07-06 19:11:00 +0900 Jimmy Ohn <yongjin.ohn@lge.com>
+
+ * gst/rtp/gstrtph261pay.c:
+ rtph261pay: Fix uninitialized variable compiler error
+ endpos variable does not correctly understand in the
+ 4.6.3 GCC version. So compile error appears when we do
+ compile rtph261pay using jhbuild.
+ This patch is fixed the compile error in 4.6.3 GCC version.
+ https://bugzilla.gnome.org/show_bug.cgi?id=751985
+
+2014-11-12 12:08:58 +0100 Jan Alexander Steffens (heftig) <jsteffens@make.tv>
+
+ * gst/flv/gstflvdemux.c:
+ flvdemux: Handle seek flags properly
+ Allows for non-keyframe seeks.
+ https://bugzilla.gnome.org/show_bug.cgi?id=738570
+
+2015-02-24 10:50:52 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: avoid looping reading the 'moof' atom forever
+ It gets stuck if it only finds a moof and no mfra/mfro or moov
+ atoms. Skip the moof to continue the parsing to have it either
+ play or error out.
+ https://bugzilla.gnome.org/show_bug.cgi?id=745089
+
+2015-06-26 13:24:17 +0900 Vineeth TM <vineeth.tm@samsung.com>
+
+ * ext/flac/gstflacdec.c:
+ flacdec: improve error handling
+ for files which have corrupted header, libflac is not able to
+ process the metadata properly. We just try to ignore the error
+ and continue with the processing, since metadata parsing is not
+ making much of a difference to libflac
+ https://bugzilla.gnome.org/show_bug.cgi?id=751334
+
+2015-07-06 20:16:38 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
+
+ * sys/ximage/ximageutil.c:
+ ximagesrc: add meta transform function
+ ximage metadata can't be transformed or copied, but provide an empty
+ transformation function instead of NULL to allow unconditional calling
+ of metas' transform functions.
+ https://bugzilla.gnome.org/show_bug.cgi?id=751778
+
+2014-06-16 16:14:28 +0200 Stian Selnes <stian.selnes@gmail.com>
+
+ * gst/rtp/gstrtph263pdepay.c:
+ rtph263pdepay: init debug category
+ https://bugzilla.gnome.org/show_bug.cgi?id=752012
+
+2014-06-20 10:59:14 +0200 Stian Selnes <stian@pexip.com>
+
+ * gst/rtp/gstrtpvp8depay.c:
+ rtpv8depay: ignore reserved bit in payload descriptor
+ Draft 16 of "RTP Payload Format for VP8" states in section 4.2 that:
+ R: Bit reserved for future use. MUST be set to zero and MUST be
+ ignored by the receiver.
+ https://bugzilla.gnome.org/show_bug.cgi?id=751929
+
+2015-07-04 20:56:42 +0200 Stian Selnes <stian@pexip.com>
+
+ * docs/plugins/gst-plugins-good-plugins-docs.sgml:
+ * docs/plugins/gst-plugins-good-plugins-sections.txt:
+ * gst/rtp/gstrtph261depay.c:
+ * gst/rtp/gstrtph261pay.c:
+ rtph261pay: rtph261depay: Add documentation
+ https://bugzilla.gnome.org/show_bug.cgi?id=751982
+
+2015-07-03 21:58:14 +0200 Stefan Sauer <ensonic@users.sf.net>
+
+ * common:
+ Automatic update of common submodule
+ From f74b2df to 9aed1d7
+
+2015-07-03 14:29:16 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtp/gstrtph261pay.c:
+ rtph261pay: Fix compiler warning
+ gstrtph261pay.c: In function 'gst_rtp_h261_pay_class_init':
+ gstrtph261pay.c:1003:17: error: variable 'gobject_class' set but not used [-Werror=unused-but-set-variable]
+ GObjectClass *gobject_class;
+
+2015-07-03 14:03:05 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtp/gstrtph261depay.c:
+ rtph261depay: Let the base class push the buffer so it can deal with the flow return
+
+2015-07-03 14:11:35 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtp/gstrtph261pay.c:
+ rtph261pay: Remove unused adapter
+
+2015-07-03 13:17:24 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtp/gstrtpspeexpay.c:
+ speexpay: Directly attach payload to the output buffer instead of copying it
+
+2015-07-03 13:07:20 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtp/gstrtpsbcpay.c:
+ sbcpay: Attach payload directly to the output instead of copying
+
+2014-12-01 14:18:40 +0100 Stian Selnes <stian@pexip.com>
+
+ * gst/rtp/Makefile.am:
+ * gst/rtp/gstrtp.c:
+ * gst/rtp/gstrtph261depay.c:
+ * gst/rtp/gstrtph261depay.h:
+ * gst/rtp/gstrtph261pay.c:
+ * gst/rtp/gstrtph261pay.h:
+ * tests/check/elements/rtp-payloading.c:
+ rtp: add H.261 RTP payloader and depayloader
+ Implementation according to RFC 4587.
+ Payloader create fragments on MB boundaries in order to match MTU size
+ the best it can. Some decoders/depayloaders in the wild are very strict
+ about receiving a continuous bit-stream (e.g. no no-op bits between
+ frames), so the payloader will shift the compressed bit-stream of a
+ frame to align with the last significant bit of the previous frame.
+ Depayloader does not try to be fancy in case of packet loss. It simply
+ drops all packets for a frame if there is a loss, keeping it simple.
+ https://bugzilla.gnome.org/show_bug.cgi?id=751886
+
+2015-07-03 12:18:52 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtp/gstrtpmpvdepay.c:
+ rtpmpvdepay: Don't forget to unmap the input buffer
+
+2015-07-03 12:14:47 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtp/gstrtpmpvpay.c:
+ rtpmpvpay: Create buffer lists instead of pushing each buffer individually
+
+2015-07-03 12:03:59 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtp/gstrtpmpapay.c:
+ rtpmpapay: Use buffer lists instead of pushing each fragment individually
+
+2015-07-03 10:51:57 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtp/gstrtpmp4apay.c:
+ rtpmp4apay: Create buffer lists and don't copy payload memory
+
+2015-06-29 16:14:18 +0200 Miguel París Díaz <mparisdiaz@gmail.com>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ rtpjitterbuffer: Consider timers len to compare with RTP_MAX_DROPOUT
+ When there are a lot of small gaps, we can consider that there is
+ a big gap (too losses) to reset the buffer.
+ https://bugzilla.gnome.org/show_bug.cgi?id=751636
+
+2015-06-29 15:53:52 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ * tests/check/elements/rtpjitterbuffer.c:
+ rtpjitterbuffer: If possible, always update the current time before looping over all timers
+ If we have a clock, update "now" now with the very latest running time we have.
+ If timers are unscheduled below we otherwise wouldn't update now (it's only updated
+ when timers expire), and also for the very first loop iteration now would otherwise
+ always be 0.
+ Also the time is used for the timeout functions, e.g. to calculate any times
+ for the next timeouts and we would otherwise pass too old times there.
+ https://bugzilla.gnome.org/show_bug.cgi?id=751636
+
+2015-07-02 14:34:57 +0100 Luis de Bethencourt <luis.bg@samsung.com>
+
+ * sys/v4l2/gstv4l2transform.c:
+ v4l2transform: fix memory leak
+ tmp needs to be freed before going out of scope in 'done'.
+ CID #1308954
+
+2015-07-02 12:23:45 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtp/gstrtph263ppay.c:
+ rtph263ppay: Generate buffer lists and attach the payload directly instead of copying it
+
+2015-07-02 09:48:02 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtp/gstrtph263pdepay.c:
+ rtph263pdepay: Simplify code a bit and do less direct memcpy and let GstBuffer do that for us
+
+2015-07-02 09:17:59 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtp/gstrtph263pay.c:
+ * gst/rtp/gstrtph263pay.h:
+ rtph263pay: Stop using an adapter and directly use the buffer
+ We always pushed one buffer into the adapter, then handled exactly that one
+ buffer and flushed it from the adapter. Now also don't memcpy() the actual
+ payload but just attach the input buffer's data to the output buffer.
+ This code still needs some serious refactoring/rewriting.
+
+2015-07-01 21:57:28 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtp/gstrtpgsmpay.c:
+ rtpgsmpay: Remove non-existing includes for now
+ git add -p mistake.
+
+2015-07-01 19:29:07 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtp/gstrtpgstpay.c:
+ rtpgstpay: Use the return value of gst_buffer_append()
+
+2015-07-01 19:19:13 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtp/gstrtpgsmpay.c:
+ rtpgsmpay: Attach payload to the output buffer instead of copying it
+
+2015-07-01 17:58:56 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtp/gstrtpg729pay.c:
+ rtpg729pay: Attach payload directly to output buffers instead of copying
+
+2015-07-01 17:43:51 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtp/gstrtpg723pay.c:
+ rtpg723pay: Attach payload buffer to the output instead of copying
+
+2015-07-01 17:30:39 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtp/gstrtpdvdepay.c:
+ rtpdvdepay: Map the output buffer once instead of once every 80 bytes
+
+2015-07-01 21:46:46 +0900 Jimmy Ohn <yongjin.ohn@lge.com>
+
+ * gst/avi/gstavidemux.c:
+ avidemux: fix return type of index_entry_offset_search()
+ It's a compare function and may return a negative value,
+ so should for correctness and consistency return a signed
+ integer.
+ https://bugzilla.gnome.org/show_bug.cgi?id=751780
+
+2015-07-01 14:12:57 +0200 Miguel París Díaz <mparisdiaz@gmail.com>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ rtpjitterbuffer: refactor handle_next_buffer
+ The goal of this patch is making handle_next_buffer function
+ more readable avoiding unnecesary gotos and adding other
+ cosmetic changes.
+
+2015-07-01 15:40:25 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtp/gstrtpac3pay.c:
+ rtpac3pay: Attach the payload to the output buffer instead of copying it
+ Might also want to produce buffer lists here if needed.
+
+2015-07-01 15:38:47 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtp/gstrtpilbcdepay.c:
+ * gst/rtp/gstrtpsirendepay.c:
+ rtp: Fix indention
+
+2015-07-01 12:37:11 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * tests/examples/rtp/Makefile.am:
+ * tests/examples/rtp/client-VP8-OPUS.sh:
+ * tests/examples/rtp/server-VTS-VP8-ATS-OPUS.sh:
+ rtp: Add examples with VTS/ATS for VP8/OPUS
+ Let's have an example with modern codecs.
+
+2015-06-30 18:11:33 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtp/gstrtph264pay.c:
+ rtph264pay: Use GST_WARNING_OBJECT() instead of GST_WARNING()
+
+2015-06-30 14:06:20 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtp/gstrtpvp8depay.c:
+ vp8depay: Don't lock/map every non-keyframe buffer twice
+ Just copy the complete header instead of first looking at the first byte
+ and then at the remaining 10 bytes.
+
+2015-06-29 16:05:44 +0100 Luis de Bethencourt <luis@debethencourt.com>
+
+ * sys/v4l2/gstv4l2object.c:
+ v4l2: document fallthrough cases
+ Pacify coverity and document fallthrough cases in switch statements.
+ CID #1308948, #1308947, #1308946
+
+2015-06-29 10:36:58 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ Revert "rtpjitterbuffer: If we have an immediate timeout, don't try to find an earlier timeout"
+ This reverts commit 0c21cd7177ea883c710999147ddcedb19004d182.
+ If we have multiple immediate timers, we want to first handle the one with the
+ lowest sequence number... which would be broken now.
+ Instead of this we should just use a GSequence for the timers, and have them
+ sorted first by timestamp, and for equal timestamps by sequence number. Then
+ we would always only have to take the very first timer from the list and never
+ have to look at any others.
+
+2015-06-29 10:14:05 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ rtpjitterbuffer: If we have an immediate timeout, don't try to find an earlier timeout
+ If we have lots of such immediate timeouts, we would otherwise have quadratic
+ runtime in the number of timeouts.
+
+2015-06-19 18:01:03 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/multifile/gstsplitmuxsrc.c:
+ splitmuxsrc: sticky events are sent automatically from the pad
+ No need to send them explicitly from the element
+ https://bugzilla.gnome.org/show_bug.cgi?id=751240
+
+2015-06-19 18:00:40 -0300 Thiago Santos <thiagoss@osg.samsung.com>
+
+ * gst/multifile/gstsplitmuxsrc.c:
+ splitmuxsrc: make sure to push sticky events before adding pad
+ It allows the caps to be set on the pad before being added for
+ dynamic autoplugging to work.
+ https://bugzilla.gnome.org/show_bug.cgi?id=751240
+
+2015-06-26 00:05:29 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
+
+ * gst/rtsp/gstrtspsrc.c:
+ * gst/rtsp/gstrtspsrc.h:
+ rtspsrc: Add new ntp-time-source property and deprecate use-pipeline-clock property
+ Enable to use new ntp-time-source property of rtpbin
+ https://bugzilla.gnome.org/show_bug.cgi?id=751496
+
+2015-06-25 23:19:58 +0900 Hyunjun Ko <zzoon.ko@samsung.com>
+
+ * gst/rtpmanager/gstrtpbin.c:
+ * gst/rtpmanager/gstrtpsession.c:
+ rtpbin/session: fix description
+ https://bugzilla.gnome.org/show_bug.cgi?id=751496
+
+2015-06-25 10:57:25 +0100 Luis de Bethencourt <luisbg@osg.samsung.com>
+
+ * gst/imagefreeze/gstimagefreeze.c:
+ * gst/matroska/matroska-demux.c:
+ * tests/examples/shapewipe/shapewipe-example.c:
+ docs: decodebin2 -> decodebin
+
+2015-06-25 10:47:06 +0100 Luis de Bethencourt <luisbg@osg.samsung.com>
+
+ * gst/deinterlace/gstdeinterlace.c:
+ deinterlace: update example pipeline
+ Update reference to decodebin2 to decodebin
+
+2015-06-25 10:45:35 +0100 Luis de Bethencourt <luisbg@osg.samsung.com>
+
+ * gst/deinterlace/gstdeinterlace.c:
+ deinterlace: remove dead assignments
+ Values in fields_required and same_buffer are overwritten before used. Removing
+ assignment
+
+2015-06-25 10:06:07 +0100 Tim-Philipp Müller <tim@centricular.com>
+
+ * ext/Makefile.am:
+ * ext/mikmod/Makefile.am:
+ * ext/mikmod/README:
+ * ext/mikmod/drv_gst.c:
+ * ext/mikmod/gstmikmod.c:
+ * ext/mikmod/gstmikmod.h:
+ * ext/mikmod/mikmod_reader.c:
+ * ext/mikmod/mikmod_types.c:
+ * ext/mikmod/mikmod_types.h:
+ * m4/Makefile.am:
+ * m4/libmikmod.m4:
+ * win32/MANIFEST:
+ * win32/vs8/libgstmikmod.vcproj:
+ mikmod: remove ancient unported plugin
+ This hasn't been touched in 11 years, and
+ clearly no one's been missing it.
+
+2015-06-23 20:15:13 +0900 Gilbok Lee <gilbok.lee@samsung.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: does not detect orientation
+ Most files don't contain the values for transposing the coordinates
+ back to the positive quadrant so qtdemux was ignoring the rotation
+ tag. To be able to properly handle those files qtdemux will also ignore
+ the transposing values to only detect the rotation using the values
+ abde from the transformation matrix:
+ [a b c]
+ [d e f]
+ [g h i]
+ https://bugzilla.gnome.org/show_bug.cgi?id=738681
+
+2015-06-25 00:04:16 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * configure.ac:
+ Back to development
+
=== release 1.5.2 ===
-2015-06-24 Sebastian Dröge <slomo@coaxion.net>
+2015-06-24 23:30:41 +0200 Sebastian Dröge <sebastian@centricular.com>
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
* configure.ac:
- releasing 1.5.2
+ * docs/plugins/gst-plugins-good-plugins.args:
+ * docs/plugins/gst-plugins-good-plugins.hierarchy:
+ * 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:
+ * gst-plugins-good.doap:
+ * win32/common/config.h:
+ Release 1.5.2
2015-06-24 22:56:12 +0200 Sebastian Dröge <sebastian@centricular.com>
projects / platform / upstream / gst-plugins-good.git / commitdiff