+=== release 1.11.1 ===
+
+2017-01-12 Sebastian Dröge <slomo@coaxion.net>
+
+ * configure.ac:
+ releasing 1.11.1
+
+2017-01-12 14:36:22 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * po/el.po:
+ * po/hr.po:
+ * po/id.po:
+ * po/zh_CN.po:
+ po: Update translations
+
+2017-01-11 17:53:32 -0800 Andre McCurdy <armccurdy@gmail.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: free seqh after calling qtdemux_parse_svq3_stsd_data()
+ The seqh buffer allocated in qtdemux_parse_svq3_stsd_data() needs to
+ be freed by the caller after use.
+ https://bugzilla.gnome.org/show_bug.cgi?id=777157
+ Signed-off-by: Andre McCurdy <armccurdy@gmail.com>
+
+2017-01-10 16:01:35 +0100 Edward Hervey <edward@centricular.com>
+
+ * gst/isomp4/fourcc.h:
+ * gst/isomp4/qtdemux.c:
+ isomp4: Don't spam debug log with knonw/padding atoms
+ Only output WARNING messages for atoms we don't know how to handle
+ instead of for padding/known atoms we don't need to do any processing
+ on
+ https://bugzilla.gnome.org/show_bug.cgi?id=777095
+
+2017-01-09 19:05:10 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/rtp/gstrtph263depay.c:
+ * gst/rtp/gstrtpsbcdepay.c:
+ * gst/rtpmanager/rtpjitterbuffer.c:
+ * gst/rtsp/gstrtspsrc.c:
+ * sys/v4l2/gstv4l2bufferpool.c:
+ Fix indentation
+
+2017-01-09 19:04:04 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/check/elements/rtpjitterbuffer.c:
+ tests: rtpjitterbuffer: fix compiler warning due to c99-ism
+ rtpjitterbuffer.c:592:3: error: ‘for’ loop initial declarations are only allowed in C99 mode
+
+2016-11-11 14:31:03 +1100 Matthew Waters <matthew@centricular.com>
+
+ * gst/autodetect/gstautodetect.c:
+ autodetect: bring the element state down after success
+ Otherwise some messages that are emitted by the element on NULL->READY
+ will not reach the application.
+ https://bugzilla.gnome.org/show_bug.cgi?id=764947
+
+2017-01-08 01:13:32 +1100 Jan Schmidt <jan@centricular.com>
+
+ * gst/isomp4/atoms.c:
+ * gst/isomp4/atoms.h:
+ * gst/isomp4/gstqtmux.c:
+ qtmux: Write tfdt atom into fragmented files.
+ The DASH spec requires that tfdt atoms be present, so
+ write one out. ISO/IEC 23009-1:2014 6.3.4.2
+ https://bugzilla.gnome.org/show_bug.cgi?id=708221
+
+2017-01-07 23:55:42 +1100 Jan Schmidt <jan@centricular.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Don't reset output timestamps when no tfdt
+ If a fragmented stream doesn't have a tfdt, don't
+ reset the output timestamps at each fragment boundary
+ by erroneously using the default value of 0. Introduced
+ by commit 69fc48
+ https://bugzilla.gnome.org/show_bug.cgi?id=754230
+
+2016-12-16 16:51:48 -0300 Thibault Saunier <thibault.saunier@osg.samsung.com>
+
+ * ext/vpx/meson.build:
+ * gst/equalizer/meson.build:
+ * gst/isomp4/meson.build:
+ * meson.build:
+ meson: Install presets files
+
+2017-01-03 10:12:30 +0530 Garima Gaur <garima.g@samsung.com>
+
+ * gst/avi/gstavidemux.c:
+ avidemux: fix some caps leaks
+ https://bugzilla.gnome.org//show_bug.cgi?id=776789
+
+2016-12-22 17:34:08 +0200 Vivia Nikolaidou <vivia@ahiru.eu>
+
+ * gst/multifile/gstsplitmuxsink.c:
+ splitmuxsink: Return a bin with a "location" property as a sink
+ Splitmuxsink might be called with a custom bin as a sink. If it has a
+ "location" property, it can be used.
+
+2016-11-18 22:42:18 +1100 Jan Schmidt <jan@centricular.com>
+
+ * gst/multifile/gstsplitmuxsink.c:
+ * gst/multifile/gstsplitmuxsink.h:
+ splitmux: Rewrite buffer collection and scheduling
+ Majorly change the way that splitmuxsink collects
+ incoming data and sends it to the output, so that it
+ makes all decisions about when / where to split files
+ on the input side.
+ Use separate queues for each stream, so they can be
+ grown individually and kept as small as possible.
+ This removes raciness I observed where sometimes
+ some data would end up put in a different output file
+ over multiple runs with the same input.
+ Also fixes hangs with input queues getting full
+ and causing muxing to stall out.
+
+2016-11-17 23:40:27 +1100 Jan Schmidt <jan@centricular.com>
+
+ * gst/multifile/gstsplitmuxsink.c:
+ * gst/multifile/gstsplitmuxsink.h:
+ * tests/check/elements/splitmux.c:
+ splitmuxsink: Add format-location-full signal
+ Add a new signal for formatting the filename, which receives
+ a GstSample containing the first buffer from the reference
+ stream that will be muxed into that file.
+ Useful for creating filenames that are based on the
+ running time or other attributes of the buffer.
+ To make it work, opening of files and setting filenames is
+ now deferred until there is some data to write to it,
+ which also requires some changes to how async state changes
+ and gap events are handled.
+
+2016-12-31 01:54:01 +1100 Jan Schmidt <jan@centricular.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Always snap to the start of the keyframe
+ When performing a key-unit seek, always snap to the start ts
+ of the keyframe buffer we landed on so that the keyframe is
+ entirely within the resulting outgoing segment. That seems
+ the most sensible result, since the user requested snapping
+ to the keyframe position.
+
+2016-12-31 01:48:04 +1100 Jan Schmidt <jan@centricular.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Omit cslg_shift when snapping seeks
+ Segments times and seek requests are stored and handled
+ in raw 'PTS' time, without the cslg_shift - which only applies
+ to outgoing samples. Omit the cslg_shift portion when
+ extracting PTS to compare for internal seek snaps.
+ If the cslg_shift is included, then keyframe+snap-before seeks
+ generate a segment start/stop time that already includes the
+ cslg_shift, and it's then added a 2nd time, causing the
+ first buffer(s) to have timestamps that are out of segment.
+
+2016-12-30 22:31:38 +1100 Jan Schmidt <jan@centricular.com>
+
+ * gst/isomp4/atoms.c:
+ qtmux: Remove bogus check in atom_stsc_add_new_entry()
+ Remove an old check from atom_stsc_add_new_entry() that
+ extends the last entry in the STSC if the samples per chunk
+ matches, as the new interleave merging logic requires that
+ the final entry by updateable. There's already code
+ below which simply merges the final entry into the previous
+ one when needed, so rely on that instead.
+ Fixes asserts like:
+ ERROR:atoms.c:2940:atom_stsc_update_entry: assertion failed:
+ (atom_array_index (&stsc->entries, len - 1).first_chunk == first_chunk)
+
+2016-04-24 21:38:51 +0900 Seungha Yang <sh.yang@lge.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Fix key_time in gst_qtdemux_adjust_seek()
+ time in segment should be PTS based (not DTS).
+ https://bugzilla.gnome.org/show_bug.cgi?id=765498
+
+2016-12-28 22:49:27 +1100 Jan Schmidt <jan@centricular.com>
+
+ * gst/multifile/gstsplitmuxpartreader.c:
+ * gst/multifile/gstsplitmuxpartreader.h:
+ * gst/multifile/gstsplitmuxsrc.c:
+ splitmuxsrc: Pass seek flags when activating.
+ Pass all seek flags when activating a part
+ based on a seek, so that SNAP flags are preserved.
+
+2016-11-26 01:13:19 +1100 Jan Schmidt <jan@centricular.com>
+
+ * gst/multifile/gstsplitmuxpartreader.c:
+ splitmux: Fix a small race in the splitmuxsrc
+ Make sure the state of the parser is set to
+ collecting streams before chaining up to the
+ parent change_state() method, to close a
+ small window that can cause playback to
+ never commence.
+
+2017-01-02 15:06:33 +0100 Edward Hervey <edward@centricular.com>
+
+ * tests/check/elements/amrparse.c:
+ check: Remove dead code
+
+2016-12-31 09:52:25 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/multifile/gstmultifilesink.c:
+ * gst/multifile/gstmultifilesink.h:
+ multifilesink: refactor max_files handling a bit
+ Use GQueue instead of a GSList so we don't have to traverse
+ the whole list to append something every time. And it also
+ keeps track of the number of items in it for us.
+ Add a function to add filenames to the list of old files and
+ use it in more places, so that memory doesn't build up in
+ other modes either if no max_files limit is specified.
+ https://bugzilla.gnome.org/show_bug.cgi?id=766991
+
+2016-05-29 17:21:47 +0100 Ursula Maplehurst <ursula@kangatronix.co.uk>
+
+ * gst/multifile/gstmultifilesink.c:
+ multifilesink: don't leak memory when no max-files limit is set
+ Technically we weren't leaking the memory, just storing it internally
+ and never using it until the element is freed. But we'd still use more
+ and more memory over time, so this is not good over longer periods
+ of time. Only keep track of files if there's actually a limit set,
+ so that we will prune the list from time to time.
+ https://bugzilla.gnome.org/show_bug.cgi?id=766991
+
+2016-12-29 12:39:20 +0100 Mark Nauwelaerts <mnauw@users.sourceforge.net>
+
+ * gst/matroska/matroska-demux.c:
+ matroskademux: adjust segment stop for KEY_UNIT negative rate seeking
+
+2016-12-29 12:25:35 +0100 Mark Nauwelaerts <mnauw@users.sourceforge.net>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: implement pull mode SNAP flag seeking
+
+2016-12-29 11:26:33 +0100 Mark Nauwelaerts <mnauw@users.sourceforge.net>
+
+ * gst/avi/gstavidemux.c:
+ avidemux: tweak KEY_UNIT SNAP seek handling
+ Previously, seeking to position y where y is (strictly) within a keyframe
+ would seek to that keyframe both with SNAP_BEFORE and SNAP_AFTER,
+ where the latter is now adjusted to really snap to the next keyframe.
+
+2016-12-28 13:23:11 +0100 Mark Nauwelaerts <mnauw@users.sourceforge.net>
+
+ * gst/avi/gstavidemux.c:
+ avidemux: correctly perform pull mode KEY_UNIT seeking
+ Rather amazingly (and equally unnoticed), keyunit seeking resulted in segments
+ where start != time (which is bogus for simple avi timeline). So, properly
+ adjust the segment (start) rather than fiddling with segment time (only).
+
+2016-12-28 13:04:54 +0100 Mark Nauwelaerts <mnauw@users.sourceforge.net>
+
+ * gst/avi/gstavidemux.c:
+ avidemux: restore considering of pull mode KEY_UNIT seeking
+ ... by using the original seek event's flags rather than the corresponding
+ segment flags, which do not have such counterpart flags (and
+ do no longer have them covertly sneaking in nowadays).
+
+2015-05-08 12:44:01 +0200 Nicola Murino <nicola.murino@gmail.com>
+
+ * gst/matroska/matroska-mux.c:
+ matroskamux: only drop actual streamheader buffers with xiph codecs
+ With Xiph codecs the stream header buffers are both in the caps and are
+ usually also at the beginning of each input stream, but it's perfectly
+ possible that the input stream does not have the stream header buffers
+ inline in the data. Matroskamux would drop the first N buffers assuming
+ they're stream headers, but this meant it would drop actual payload data
+ when the stream didn't contain the stream headers inline. Fix this by
+ only dropping leading buffers if they're flagged as stream headers. This
+ fixes issues with streams that are being tapped into after streaming
+ has started.
+ https://bugzilla.gnome.org/show_bug.cgi?id=749098
+
+2016-12-21 17:43:58 +0100 Nicola Murino <nicola.murino@gmail.com>
+
+ * tests/check/elements/matroskamux.c:
+ matroskamux: adjust unit test to modified behaviour
+ Now matroskamux mark all packets of audio-only streams as keyframes so
+ in test_block_group after pushing the test audio data 4 buffers are produced
+ and not more 2. The last buffer is the original data and must match with what
+ pushed. The remaining ones are matroskamux headers
+ https://bugzilla.gnome.org/show_bug.cgi?id=754696
+
+2016-05-30 01:15:31 +0200 Nicola Murino <nicola.murino@gmail.com>
+
+ * gst/matroska/matroska-mux.c:
+ matroskamux: mark all packets of audio-only streams as keyframes
+ This helps with streaming audio-only streams via multifdsink,
+ tcpserversink and such.
+ https://bugzilla.gnome.org/show_bug.cgi?id=754696
+
+2015-03-28 18:15:36 +0100 Nicola Murino <nicola.murino@gmail.com>
+
+ * gst/matroska/matroska-mux.c:
+ matroskamux: add G722 audio support
+ https://bugzilla.gnome.org/show_bug.cgi?id=746574
+
+2016-12-13 11:11:07 +0900 Wonchul Lee <wonchul.lee@collabora.com>
+
+ * gst/udp/gstudpsrc.c:
+ updsrc: Add to join multiple multicast interfaces
+ https://bugzilla.gnome.org/show_bug.cgi?id=776030
+
+2015-03-25 13:51:30 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * gst/rtp/gstrtpklvdepay.c:
+ rtpklvdepay: add the SPARSE flag to the outgoing stream-start event
+
+2016-12-14 14:37:45 -0800 Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
+
+ * gst/rtpmanager/gstrtpbin.c:
+ * gst/rtpmanager/gstrtpsession.c:
+ rtpmanager: place content before Since-version API marker
+ Avoids confusing the parser
+
+2016-12-14 14:16:53 -0800 Reynaldo H. Verdejo Pinochet <reynaldo@osg.samsung.com>
+
+ * ext/shout2/gstshout2.c:
+ shout2: fix 404 in package origin
+
+2016-12-14 21:45:15 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Check if we have enough data available when parsing edit lists
+ Also consume the data entry by entry to get complicated indexing out of
+ the code.
+ https://bugzilla.gnome.org/show_bug.cgi?id=776107
+
+2016-12-14 19:15:03 +0100 Víctor Manuel Jáquez Leal <vjaquez@igalia.com>
+
+ * sys/v4l2/gstv4l2object.c:
+ v4l2object: Don't check size in a non-list value
+ After commit 1ea9735a I see these error while using the webcam
+ integrated in my laptop:
+ GStreamer-CRITICAL **: gst_value_list_get_size: assertion 'GST_VALUE_HOLDS_LIST (value)' failed
+ The issue is gst_v4l2src_value_simplify() was doing its job of
+ generating a single value, rather than the original list. That why,
+ when getting the list size, a critical warning was raised.
+ This patch takes advantage of the compiler optimizations to verify
+ first if the list was simplified, thus use it directly, otherwise,
+ if it is a list, verify its size.
+ https://bugzilla.gnome.org/show_bug.cgi?id=776106
+
+2016-12-14 10:39:12 +0100 Havard Graff <havard.graff@gmail.com>
+
+ * tests/check/elements/rtpjitterbuffer.c:
+ tests/jitterbuffer: Major refactoring and cleanups
+ * Changed PCMU->TEST for common macros
+ * Changed verify-functions (lost & rtx) into macros.
+ * Remove option to add marker-bit for test-buffers (not used anywhere)
+ * Add new push_test_buffer function that makes sure there are correlation
+ between dts and the time on the clock. (classic test-mistake)
+ * Established a generic starting-point for tests with the
+ construct_deterministic_initial_state function and use it where
+ applicable, which removes lots of "boilerplate" everywhere.
+ * Add basic lost-event test
+ * Remove as much "magic constants" as possible.
+ * Remove 3 tests that no longer are testing anything that others don't,
+ and was completely unmaintainable.
+ * Remove unnecessary use of the testclock
+ * Verify each test is testing what it actually says it does (and modify
+ where it doesn't)
+ In general, make the tests much smaller, better, more maintainable and
+ readable.
+ https://bugzilla.gnome.org/show_bug.cgi?id=774409
+
+2016-12-14 09:54:11 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * .gitignore:
+ * Makefile.am:
+ * configure.ac:
+ * gst-plugins-good.spec.in:
+ Remove generated .spec file
+ Likely extremely bitrotten, and we should not ship this anyway.
+
+2016-12-14 10:15:10 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Check that the XiTh size is big enough
+ https://bugzilla.gnome.org/show_bug.cgi?id=775794
+
+2016-12-09 20:27:53 +0900 Heekyoung Seo <heekyoung.seo@lge.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Check node length of video sample description
+ Add check for node length of video sample description and its fields and
+ for the XiTh atom.
+ Also unify the code a bit.
+ https://bugzilla.gnome.org/show_bug.cgi?id=775794
+
+2016-12-08 18:50:52 +0900 Heekyoung Seo <heekyoung.seo@lge.com>
+
+ * gst/isomp4/fourcc.h:
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Enable xvid/mp2 codec support
+ Add support for xvid video and mp2 audio, add m2v1 fourcc.
+ https://bugzilla.gnome.org/show_bug.cgi?id=775794
+
+2016-12-13 22:32:46 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtp/gstrtpvp9depay.c:
+ * tests/check/elements/rtpjitterbuffer.c:
+ * tests/check/elements/rtprtx.c:
+ * tests/check/elements/vp9enc.c:
+ gst: Don't declare variables inside the for loop header
+ This is a C99 feature.
+
+2016-12-11 13:27:27 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/audiofx/gstscaletempo.c:
+ scaletempo: Ensure to reinit buffers whenever they were not allocated yet
+ That is, whenever we go through start/stop we have to ensure that on the
+ next opportunity the buffers are reallocated again. Otherwise the
+ buffers might be NULL because the element was reused with the same
+ configuration as before (i.e. set_caps() wouldn't have reinited the
+ buffers).
+ https://bugzilla.gnome.org/show_bug.cgi?id=775898
+
+2016-12-10 12:52:18 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * docs/design/Makefile.am:
+ * docs/design/design-rtpauxiliary.txt:
+ * docs/design/design-rtpcollision.txt:
+ * docs/design/design-rtpretransmission.txt:
+ docs: design: remove, moved to gst-docs
+
+2016-12-09 17:17:35 -0300 Thibault Saunier <tsaunier@gnome.org>
+
+ * meson.build:
+ meson: Support building without Gst debug
+
+2016-12-09 17:55:39 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/flx/gstflxdec.c:
+ * gst/flx/gstflxdec.h:
+ flxdec: Only send SEGMENT events after CAPS
+ I.e., don't just forward the event but delay it if we don't have caps on
+ the srcpad yet.
+
+2016-12-09 17:49:40 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/flx/gstflxdec.c:
+ flxdec: Unref and unmap buffers in all code paths as needed
+ https://bugzilla.gnome.org/show_bug.cgi?id=775888
+
+2016-12-06 17:42:31 +0530 Arun Raghavan <arun@osg.samsung.com>
+
+ * sys/v4l2/gstv4l2object.c:
+ v4l2object: Don't set empty interlace-mode list
+ If for some reason we fail to probe formats (all try_fmt calls fail, for
+ example), this is not a critical error, but we end up with an empty list
+ of interlace modes. This causes all subsequent negotiation to fail.
+ This patch fixes interlace-mode setting to be skipped if we failed to
+ detect any.
+ https://bugzilla.gnome.org/show_bug.cgi?id=775702
+
+2016-12-07 17:22:22 +0530 Garima Gaur <garima.g@samsung.com>
+
+ * gst/monoscope/gstmonoscope.c:
+ monoscope: Unref allocation query after finished with it
+ https://bugzilla.gnome.org/show_bug.cgi?id=775752
+
+2016-12-06 07:48:47 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/flx/gstflxdec.c:
+ flxdec: Allocate 0-initialized memory for the decoded frame
+ Otherwise we might leak arbitrary information from the uninitialized
+ memory if not every pixel is written.
+ https://scarybeastsecurity.blogspot.gr/2016/12/1days-0days-pocs-more-gstreamer-flic.html
+
+2016-12-05 07:57:19 -0700 Matt Staples <staples255@gmail.com>
+
+ * gst/rtsp/gstrtspsrc.c:
+ rtspsrc: Fix session cleanup when handling redirect on PLAY
+ Redirect on PLAY wasn't doing the necessary session cleanup. Fixed by
+ removing code from gst_rtspsrc_send that changed the state varable upon
+ encountering a redirect. Better to let the redirect handlers in
+ gst_rtspsrc_retrieve_sdp and gst_rtspsrc_play do their own
+ state-dependent cleanup.
+ https://bugzilla.gnome.org/show_bug.cgi?id=775543
+
+2016-09-07 16:10:27 +0300 Aleix Conchillo Flaque <aleix@oblong.com>
+
+ * gst/rtsp/gstrtspsrc.c:
+ rtspsrc: always send teardown request
+ Allow CMD_CLOSE to cancel all commands not only CMD_PAUSE
+ and ignore CMD_WAIT while closing.
+ https://bugzilla.gnome.org/show_bug.cgi?id=748360
+
+2016-12-03 08:19:27 +0100 Edward Hervey <bilboed@bilboed.com>
+
+ * README:
+ * common:
+ Automatic update of common submodule
+ From f980fd9 to 39ac2f5
+
+2016-12-01 17:08:09 +0100 Edward Hervey <bilboed@bilboed.com>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ * gst/rtpmanager/rtpjitterbuffer.c:
+ jitterbuffer: Don't leak duplicate items
+ When providing items with a seqnum, there is a (very small) probability
+ that an element with the same seqnum already exists. Don't forget
+ to free that item if it wasn't inserted.
+ And avoid returning undefined values when dealing with duplicate items
+
+2016-12-01 11:23:02 +0100 Edward Hervey <edward@centricular.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Sanitize unknown codec caps
+ We might have non-printable characters in the unknown fourcc, replace
+ them with '_', in the same way we do it for unknown tags.
+
+2016-12-01 20:04:28 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/avi/gstavidemux.c:
+ avidemux: Free vprp chunk also if it existed but we made no use of it
+ https://bugzilla.gnome.org/show_bug.cgi?id=775479
+
+2016-12-01 17:38:33 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/matroska/matroska-read-common.c:
+ matroskademux: Fix memory leak when parsing attachments
+ gst_tag_image_data_to_image_sample() does not take ownership of the
+ passed memory, so don't set it to NULL to allow us to free it later.
+ https://bugzilla.gnome.org/show_bug.cgi?id=775472
+
+2016-12-01 14:56:18 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/matroska/matroska-read-common.c:
+ matroskademux: Unify zlib/bzip2 decompress loops with the ones from qtdemux
+ Especially, simplify the code a bit.
+
+2016-12-01 14:41:48 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Increase inflate buffer in bigger steps
+ 1024 bytes is quite small, let's do 4096 bytes (or one page).
+ Also remove redundant if, we're always in that case when getting here.
+
+2016-12-01 14:30:49 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Ensure that size of the pasp atom is as much as we need
+ https://bugzilla.gnome.org/show_bug.cgi?id=775455
+
+2016-12-01 14:30:10 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Free compressed moov node and it's corresponding decompressed data
+ https://bugzilla.gnome.org/show_bug.cgi?id=775455
+
+2016-12-01 14:29:21 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Check size of compressed MOOV header against available data
+ And actually read the size of the cmvd atom from the right position.
+ https://bugzilla.gnome.org/show_bug.cgi?id=775455
+
+2016-12-01 14:27:55 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Fix zlib inflate loop
+ Handle errors cleanly, deallocate all memory and return the actual size
+ of the inflated data.
+ https://bugzilla.gnome.org/show_bug.cgi?id=775455
+
+2016-12-01 13:38:16 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/audioparsers/gstaacparse.c:
+ aacparse: Make sure we have enough data in the codec_data to be able to parse it
+ Also error out cleanly if mapping the buffer failed.
+ https://bugzilla.gnome.org/show_bug.cgi?id=775450
+
+2016-12-01 13:32:22 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Fix out of bounds read in tag parsing code
+ We can't simply assume that the length of the tag value as given
+ inside the stream is correct but should also check against the amount of
+ data we have actually available.
+ https://bugzilla.gnome.org/show_bug.cgi?id=775451
+
+2016-12-01 15:06:06 +0530 Garima Gaur <garima.g@samsung.com>
+
+ * gst/rtp/gstrtph264depay.c:
+ * gst/rtp/gstrtpsbcdepay.c:
+ rtp: Fix some memory leaks in usage of gst_pad_get_current_caps()
+ https://bugzilla.gnome.org/show_bug.cgi?id=775071
+
+2016-11-30 17:56:02 +0200 Vivia Nikolaidou <vivia@ahiru.eu>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Read interlacing information from 'fiel' atom
+ Read interlacing and TFF/BFF information from the 'fiel' atom and pass it
+ into the caps
+ https://bugzilla.gnome.org/show_bug.cgi?id=775414
+
+2016-11-29 13:55:40 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Fix compiler warning
+ qtdemux.c: In function ‘qtdemux_parse_trak’:
+ qtdemux.c:10184:38: error: format ‘%lu’ expects argument of type ‘long unsigned int’, but argument 9 has type ‘gint {aka const int}’ [-Werror=format=]
+ GST_DEBUG_OBJECT (qtdemux, "Found jpeg: len %u, need %lu", len,
+ ^
+
+2016-11-28 13:45:24 -0800 Scott D Phillips <scott.d.phillips@intel.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Change off_t type to gint
+ off_t is a signed integer type provided by sys/types.h on posix systems.
+ Replace with gint for building on non-posix systems (like windows).
+ https://bugzilla.gnome.org/show_bug.cgi?id=775287
+
+2016-11-22 21:00:25 -0800 Scott D Phillips <scott.d.phillips@intel.com>
+
+ * meson.build:
+ meson: add libm to has_function checks
+ The functions from math.h may be implemented in libm.
+ https://bugzilla.gnome.org/show_bug.cgi?id=774876
+
+2016-10-27 23:02:37 +0530 Nirbheek Chauhan <nirbheek@centricular.com>
+
+ * ext/meson.build:
+ Revert "meson: dv plugin now works on MSVC"
+ This reverts commit 05a89613feff70cff416367f5aa807a1d5c68b63.
+ Let's not put in stuff that needs unreleased Meson. This can go in
+ for the next cycle.
+
+2016-11-28 13:51:41 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/avi/gstavidemux.c:
+ avidemux: Ensure that tags are valid UTF-8 before adding them to the taglist
+ https://bugzilla.gnome.org/show_bug.cgi?id=775219
+
+2016-11-28 12:22:49 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/multipart/multipartdemux.c:
+ multipartdemux: Post an error message on the bus if we got EOS without having added any pads
+
+2016-11-28 12:00:09 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * ext/soup/gstsouphttpsrc.c:
+ souphttpsrc: Handle non-UTF8 headers and error reasons more gracefully
+ Especially don't put them into GstStructures in one way or another, just
+ ignore them or error out cleanly depending on the importance of their
+ content.
+
+2016-11-28 09:30:25 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtp/gstrtpvrawpay.c:
+ vrawpay: Error out cleanly if mapping the video frame fails
+ Instead of later dereferencing NULL and crashing.
+
+2016-11-27 11:14:13 +0100 Edward Hervey <edward@centricular.com>
+
+ * gst/rtpmanager/gstrtprtxsend.c:
+ rtprtxsend: Update statistics before pushing
+ If an element queries the number of retransmission buffers pushed
+ *while* the push is still taking place (and before the object lock
+ is taken just after) it would end up with the wrong statistic
+ being reported.
+ Increment it just before the push, avoids races when getting statistics
+ https://bugzilla.gnome.org/show_bug.cgi?id=768723
+
+2016-11-26 11:20:51 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * .gitmodules:
+ common: use https protocol for common submodule
+ https://bugzilla.gnome.org/show_bug.cgi?id=775110
+
+2016-07-28 18:51:24 +0200 Philipp Zabel <p.zabel@pengutronix.de>
+
+ * sys/v4l2/gstv4l2bufferpool.c:
+ gstv4l2bufferpool: lock flush_stop against regular qbuf
+ These can be called from different threads and both manipulate the
+ pool->buffers array. Lock them properly and let flush_stop move the
+ array contents into a temporary array on the stack to avoid having
+ to call release_buffer under the object lock.
+ https://bugzilla.gnome.org/show_bug.cgi?id=775015
+
+2016-11-24 14:25:22 +0100 Philipp Zabel <p.zabel@pengutronix.de>
+
+ * sys/v4l2/gstv4l2bufferpool.c:
+ gstv4l2bufferpool: remove critical error message when process is called on an inactive pool
+ If the pool is inactive, it is guaranteed to also be flushing, so the
+ following check will return GST_FLOW_FLUSHING anyway.
+ This can happen if a v4l2src is blocking on DQBUF in create and is sent
+ an EOS event on another thread. In that case the pool is set to
+ flushing/inactive without locking, the v4l2src is unblocked, and may
+ call pool_process with a valid buffer on the already inactive pool.
+ https://bugzilla.gnome.org/show_bug.cgi?id=775014
+
+2016-11-24 14:41:52 +0100 Philipp Zabel <p.zabel@pengutronix.de>
+
+ * sys/v4l2/gstv4l2src.c:
+ v4l2src: release buffer if create fails
+ gst_base_src_get_range does not expect a buffer to be returned in
+ the error case, so we are leaking a reference here if create fails.
+ https://bugzilla.gnome.org/show_bug.cgi?id=775014
+
+2016-11-23 18:34:04 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/gstrtpbin.c:
+ rtpbin: Handle create_session() returning NULL in bundle code
+ CID 1394492.
+
+2016-11-22 16:42:55 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/isomp4/gstqtmux.c:
+ qtmux: Make sure to only change DTS of writable buffers
+ And trivial cleanup
+ https://bugzilla.gnome.org/show_bug.cgi?id=774840
+
+2016-11-22 16:42:26 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/isomp4/gstqtmux.c:
+ qtmux: Error out much earlier if we don't have a valid PTS
+ https://bugzilla.gnome.org/show_bug.cgi?id=774840
+
+2016-11-22 16:18:41 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/isomp4/gstqtmux.c:
+ qtmux: Only use buffer durations if they are actually valid
+ https://bugzilla.gnome.org/show_bug.cgi?id=774840
+
+2016-11-22 15:59:19 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/isomp4/gstqtmux.c:
+ qtmux: Revert commits that set DTS and duration on buffers unconditionally
+ 39f7e52266fde3b3c035e22cbcbb2bb1fa207b17 was setting the buffer duration
+ to 0 if is not valid, under the assumption that this is "the last"
+ buffer and no others are coming next. This is wrong, last_buf is the
+ previous buffer and not the very last one.
+ 4e3c13c87c258c9c95e2217d32ab314d12b5fffc was setting DTS to 0 if there
+ was none. This will set DTS to 0 for all e.g. audio streams, completely
+ messing up calculations if streams don't start at 0.
+ https://bugzilla.gnome.org/show_bug.cgi?id=774840
+
+2016-11-22 15:58:37 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/isomp4/gstqtmux.c:
+ qtmux: Only write "gap" edit list if there is a non-zero gap
+ https://bugzilla.gnome.org/show_bug.cgi?id=774840
+
+2016-11-23 07:09:06 +1100 Matthew Waters <matthew@centricular.com>
+
+ * gst/flx/flx_color.c:
+ * gst/flx/flx_fmt.h:
+ * gst/flx/gstflxdec.c:
+ * gst/flx/gstflxdec.h:
+ flxdec: rewrite logic based on GstByteReader/Writer
+ Solves overreading/writing the given arrays and will error out if the
+ streams asks to do that.
+ Also does more error checking that the stream is valid and won't
+ overrun any allocated arrays. Also mitigate integer overflow errors
+ calculating allocation sizes.
+ https://bugzilla.gnome.org/show_bug.cgi?id=774859
+
+2016-11-23 11:20:49 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/flx/gstflxdec.c:
+ flxdec: Don't unref() parent in the chain function
+ We don't own the reference here, it is owned by the caller and given to
+ us for the scope of this function. Leftover mistake from 0.10 porting.
+ https://bugzilla.gnome.org/show_bug.cgi?id=774897
+
+2016-11-22 20:33:29 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * ext/vpx/gstvpxdec.c:
+ vpxdec: libvpx's release buffer is sometimes called with fb->priv==NULL
+ Don't assert on this but just ignore these cases.
+
+2016-11-22 20:24:59 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/matroska/matroska-demux.c:
+ matroskademux: Fix cluster searching if we search multiple times in one chunk
+ After finding a cluster id in the byte reader, we skip ahead the reader
+ position by one further byte to be able to continue searching from there
+ inside the same chunk if the cluster candidate was a false positive.
+ We have to accomodate for that additional byte when resuming the search,
+ otherwise all following pulls are off-by-one for every resume and we run
+ into an assertion.
+
+2016-11-22 20:01:20 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/matroska/matroska-ids.c:
+ matroska: Add size checks to the parsing of FLAC headers
+
+2016-11-22 23:46:00 +1100 Matthew Waters <matthew@centricular.com>
+
+ * gst/flx/gstflxdec.c:
+ flxdec: fix some warnings comparing unsigned < 0
+ bf43f44fcfada5ec4a3ce60cb374340486fe9fac was comparing an unsigned
+ expression to be < 0 which was always false.
+ gstflxdec.c: In function ‘flx_decode_brun’:
+ gstflxdec.c:322:33: warning: comparison of unsigned expression < 0 is always false [-Wtype-limits]
+ if ((glong) row - count < 0) {
+ ^
+ gstflxdec.c:332:33: warning: comparison of unsigned expression < 0 is always false [-Wtype-limits]
+ if ((glong) row - count < 0) {
+ ^
+ https://bugzilla.gnome.org/show_bug.cgi?id=774834
+
+2016-11-21 16:17:31 +0200 Vivia Nikolaidou <vivia@ahiru.eu>
+
+ * gst/isomp4/gstqtmuxmap.c:
+ qtmux: Enable up to 16 unpositioned raw audio channels
+ https://bugzilla.gnome.org/show_bug.cgi?id=774789
+
+2016-11-22 19:05:00 +1100 Matthew Waters <matthew@centricular.com>
+
+ * gst/flx/gstflxdec.c:
+ flxdec: add some write bounds checking
+ Without checking the bounds of the frame we are writing into, we can
+ write off the end of the destination buffer.
+ https://scarybeastsecurity.blogspot.dk/2016/11/0day-exploit-advancing-exploitation.html
+ https://bugzilla.gnome.org/show_bug.cgi?id=774834
+
+2016-11-21 15:25:23 +0000 David Evans <bbcrddave@gmail.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Be sure not to read off end of FLAC dfLa box
+ https://bugzilla.gnome.org/show_bug.cgi?id=773712
+
+2016-11-21 11:48:58 +0100 Nicola Murino <nicola.murino@gmail.com>
+
+ * gst/matroska/matroska-demux.c:
+ matroskademux: add support for skipping invalid data in push mode
+ https://bugzilla.gnome.org/show_bug.cgi?id=774566
+
+2016-11-21 11:48:29 +0100 Nicola Murino <nicola.murino@gmail.com>
+
+ * gst/matroska/matroska-parse.c:
+ * gst/matroska/matroska-read-common.c:
+ * gst/matroska/matroska-read-common.h:
+ matroskaparse: add support for skipping invalid data
+ https://bugzilla.gnome.org/show_bug.cgi?id=774566
+
+2016-11-18 17:00:59 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtsp/gstrtspsrc.c:
+ rtspsrc: Move to new helper function to parse authentication responses
+ https://bugzilla.gnome.org/show_bug.cgi?id=774416
+
+2016-11-20 14:12:16 +0100 christophecvr <stefansat@telenet.be>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Fix wrong compiler warning with gcc 6.2
+ | ../../../git/gst/isomp4/qtdemux.c: In function 'qtdemux_parse_tree':
+ | ../../../git/gst/isomp4/qtdemux.c:10224:24: error: 'size' may be used uninitialized in this function [-Werror=maybe-uninitialized]
+ | offset += size;
+ | ^~
+ | ../../../git/gst/isomp4/qtdemux.c:10197:25: note: 'size' was declared here
+ | guint32 size, tag;
+ | ^~~~
+ https://bugzilla.gnome.org/show_bug.cgi?id=774747
+
+2016-11-20 16:15:07 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * Makefile.am:
+ * configure.ac:
+ * win32/MANIFEST:
+ * win32/common/config.h:
+ win32: remove copies of generated headers
+
+2016-11-20 13:14:08 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/avi/gstavidemux.c:
+ * gst/avi/gstavidemux.h:
+ avidemux: Ensure that raw video have properly aligned buffers
+ That is, aligned to to 32 bytes for video. Fixes crashes if the raw
+ buffers are passed to SIMD processing functions.
+ https://bugzilla.gnome.org/show_bug.cgi?id=774428
+
+2016-11-20 13:08:27 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Ensure that raw audio and video have properly aligned buffers
+ That is, aligned to the basic type for audio and to 32 bytes for video.
+ Fixes crashes if the raw buffers are passed to SIMD processing functions.
+ https://bugzilla.gnome.org/show_bug.cgi?id=774428
+
+2016-11-14 14:44:11 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/isomp4/gstqtmux.c:
+ qtmux: Always write edit lists for the tracks to give a more accurate duration
+ Always write an edit list for the whole track. In general this is not
+ necessary except for the case of having a gap or DTS adjustment but
+ it allows to give the whole track's duration in the usually more
+ accurate media timescale.
+ https://bugzilla.gnome.org/show_bug.cgi?id=774403
+
+2016-11-18 22:45:45 +0900 Seungha Yang <sh.yang@lge.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Remove useless return variable
+ qtdemux_expose_streams() returns flow error immediately, if there is an error.
+ So, the variable for the flow return is not needed.
+ https://bugzilla.gnome.org/show_bug.cgi?id=774674
+
+2016-11-17 13:59:48 +0000 David Evans <bbcrddave@gmail.com>
+
+ * gst/isomp4/fourcc.h:
+ * gst/isomp4/qtdemux.c:
+ * gst/isomp4/qtdemux_dump.c:
+ * gst/isomp4/qtdemux_dump.h:
+ * gst/isomp4/qtdemux_types.c:
+ qtdemux: Add support for FLAC encapsulated in ISOBMFF
+ As defined by
+ https://git.xiph.org/?p=flac.git;a=blob_plain;f=doc/isoflac.txt
+ https://bugzilla.gnome.org/show_bug.cgi?id=773712
+
+2016-11-17 19:59:53 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/rtpmanager/gstrtpmux.c:
+ rtpmux: Mark pad as needing reconfiguration again if it failed
+ And return FLUSHING instead of NOT_NEGOTIATED on flushing pads.
+ https://bugzilla.gnome.org/show_bug.cgi?id=774623
+
+2016-11-17 19:59:26 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/monoscope/gstmonoscope.c:
+ monoscope: Mark pad as needing reconfiguration again if it failed
+ And return FLUSHING instead of NOT_NEGOTIATED on flushing pads.
+ https://bugzilla.gnome.org/show_bug.cgi?id=774623
+
+2016-11-17 19:58:52 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/deinterlace/gstdeinterlace.c:
+ deinterlace: Mark pad as needing reconfiguration again if reconfiguration failed
+ And consider negotiation failures on flushing pads as FLUSHING, not as
+ NOT_NEGOTIATED.
+ https://bugzilla.gnome.org/show_bug.cgi?id=774623
+
+2016-11-17 19:56:23 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * ext/dv/gstdvdec.c:
+ dvdec: Fix handling of negotiation failures
+ Return NOT_NEGOTIATED if sending the caps event fails, or FLUSHING if
+ the pad was flushing at that point.
+ https://bugzilla.gnome.org/show_bug.cgi?id=774623
+
+2016-11-17 17:16:26 -0800 Scott D Phillips <scott.d.phillips@intel.com>
+
+ * meson.build:
+ meson: add_global_arguments -> add_project_arguments
+ https://bugzilla.gnome.org/show_bug.cgi?id=774656
+
+2016-11-16 10:53:51 +0530 Vinod Kesti <vinodkesti@yahoo.com>
+
+ * gst/multifile/gstsplitmuxsink.c:
+ splitmuxsink: pad request fails for flvmux
+ splitmuxsink requests pad from element using pad template like "video_%u", "audio_%u" and "sink_%d". This is true for most of the muxers.
+ But splitmuxsink not able to request pad to flvmux as flvmux has "audio" and "video" as pad templates.
+ fix: splitmuxsink should fallback to "audio" and "video" when template not found.
+ https://bugzilla.gnome.org/show_bug.cgi?id=774507
+
+2016-11-17 10:24:28 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/matroska/matroska-parse.c:
+ matroskaparse: Add remaining relevant parts from a3a55305 to the parser
+ https://bugzilla.gnome.org/show_bug.cgi?id=774566
+
+2016-11-16 22:39:01 +0100 Nicola Murino <nicola.murino@gmail.com>
+
+ * gst/matroska/matroska-parse.c:
+ matroskaparse: ignore parsing errors at the end of the file
+ This is the same change as a3a55305 for the parser.
+ https://bugzilla.gnome.org/show_bug.cgi?id=774566
+
+2016-11-16 08:56:34 +0100 Philippe Normand <philn@igalia.com>
+
+ * docs/plugins/gst-plugins-good-plugins.signals:
+ * gst/rtpmanager/gstrtpbin.c:
+ * gst/rtpmanager/gstrtpbin.h:
+ * tests/check/Makefile.am:
+ * tests/check/elements/.gitignore:
+ * tests/check/elements/rtpbundle.c:
+ * tests/check/meson.build:
+ * tests/examples/rtp/.gitignore:
+ * tests/examples/rtp/Makefile.am:
+ * tests/examples/rtp/client-rtpbundle.c:
+ * tests/examples/rtp/server-rtpbundle.c:
+ rtpbin: receive bundle support
+ A new signal named on-bundled-ssrc is provided and can be
+ used by the application to redirect a stream to a different
+ GstRtpSession or to keep the RTX stream grouped within the
+ GstRtpSession of the same media type.
+ https://bugzilla.gnome.org/show_bug.cgi?id=772740
+
+2016-11-15 16:52:39 +0530 Vinod Kesti <vinodkesti@yahoo.com>
+
+ * gst/audioparsers/gstaacparse.c:
+ aacparse: assertion while converting ADTS stream to RAW
+ aacparse resizes input buffer while converting ADTS stream to RAW,
+ During buffer resize buffer write permission is not checked.
+ This throws gst_buffer_is_writable assertion and leads to AV sync issue some times.
+ It is corrected by making buffer writeable using gst_buffer_make_writable
+ https://bugzilla.gnome.org/show_bug.cgi?id=774129
+
+2016-11-15 21:17:51 +0900 Seungha Yang <sh.yang@lge.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Don't modify upstream TIME segment
+ TIME segment implies that stream/running time is being handled by upstream.
+ So, we shouldn't override it without any clue.
+ This patch is for fixing seek in DASH streaming.
+ https://bugzilla.gnome.org/show_bug.cgi?id=774196
+
+2016-11-14 22:33:27 +0530 Arun Raghavan <arun@osg.samsung.com>
+
+ * config.h.meson:
+ meson: Add define for v4l2-probe config option
+
+2016-11-14 17:37:51 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/interleave/deinterleave.c:
+ deinterleave: Reset caps accumulator to ANY when resyncing the adapter, not EMPTY
+ The accumulator is filled by intersecting with all the pad caps, as such
+ it must be initialized with ANY (like it is before the iteration is
+ started) and not to EMPTY.
+ Fixes the CAPS query always returning EMPTY caps when resyncing happened
+ during the query, e.g. because pads were added/removed.
+
+2016-11-14 12:13:14 +0100 Petr Kulhavy <brain@jikos.cz>
+
+ * gst/udp/gstudpsrc.c:
+ udpsrc: remove redundant saddr unref
+ The g_object_unref (saddr) before receiving message seems to be redundant as it
+ is done just before jumping to retry
+ Though not directly related, part of
+ https://bugzilla.gnome.org/show_bug.cgi?id=772841
+
+2016-11-12 23:34:23 +0100 Petr Kulhavy <brain@jikos.cz>
+
+ * gst/udp/gstudpsrc.c:
+ udpsrc: receive control messages only in multicast
+ Control messages are used only in multicast mode - to detect if the destination
+ address is not ours and possibly drop the packet. However in non-multicast
+ modes the messages are still allocated and freed even if not used. Therefore
+ request control messages from g_socket_receive_message() only in multicast
+ mode.
+ https://bugzilla.gnome.org/show_bug.cgi?id=772841
+
+2016-11-11 10:45:01 -0800 Scott D Phillips <scott.d.phillips@intel.com>
+
+ * gst/matroska/matroska-mux.c:
+ Use intermediate guint when handling GstVideoMultiviewFlags
+ The underlying integer type of the enum GstVideoMultiviewFlags is
+ implementation defined and may not have the same size as guint.
+ https://bugzilla.gnome.org/show_bug.cgi?id=774293
+
+2016-11-11 10:44:18 -0800 Scott D Phillips <scott.d.phillips@intel.com>
+
+ * gst/multifile/gstsplitfilesrc.c:
+ splitfilesrc: update uri_get_type to match the prototype in GstURIHandlerInterface
+ https://bugzilla.gnome.org/show_bug.cgi?id=774293
+
+2016-10-26 22:37:34 -0700 Scott D Phillips <scott.d.phillips@intel.com>
+
+ * meson.build:
+ meson: don't add_global_arguments when being built as a subproject
+ https://bugzilla.gnome.org/show_bug.cgi?id=773568
+
+2016-10-21 15:49:36 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * gst/audioparsers/gstflacparse.c:
+ * gst/audioparsers/gstflacparse.h:
+ flacparse: fix header rewriting being ignored
+ https://bugzilla.gnome.org/show_bug.cgi?id=727802
+
+2016-11-09 06:25:27 +0000 Sean DuBois <sean@siobud.com>
+
+ * gst/flv/gstflvmux.c:
+ * gst/flv/gstflvmux.h:
+ flvmux: Add metadatacreator property
+ Allow users to set metadatacreator value in the meta packet
+ https://bugzilla.gnome.org/show_bug.cgi?id=774131
+
+2016-11-01 19:56:36 +0200 Vivia Nikolaidou <vivia@toolsonair.com>
+
+ * gst/multifile/gstsplitmuxsink.c:
+ * gst/multifile/gstsplitmuxsink.h:
+ splitmuxsink: Use first buffer TS as mux start time
+ Do not use last buffer TS + buffer duration because buffer duration
+ might be inaccurate, especially for frame rates like 30fps where a
+ rounding error is observed.
+ https://bugzilla.gnome.org/show_bug.cgi?id=773785
+
+2016-11-03 15:03:59 +0100 Havard Graff <havard.graff@gmail.com>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ * tests/check/elements/rtpjitterbuffer.c:
+ rtpjitterbuffer: fix timer-reuse bug
+ When doing rtx, the jitterbuffer will always add an rtx-timer for the next
+ sequence number.
+ In the case of the packet corresponding to that sequence number arriving,
+ that same timer will be reused, and simply moved on to wait for the
+ following sequence number etc.
+ Once an rtx-timer expires (after all retries), it will be rescheduled as
+ a lost-timer instead for the same sequence number.
+ Now, if this particular sequence-number now arrives (after the timer has
+ become a lost-timer), the reuse mechanism *should* now set a new
+ rtx-timer for the next sequence number, but the bug is that it does
+ not change the timer-type, and hence schedules a lost-timer for that
+ following sequence number, with the result that you will have a very
+ early lost-event for a packet that might still arrive, and you will
+ never be able to send any rtx for this packet.
+ Found by Erlend Graff - erlend@pexip.com
+ https://bugzilla.gnome.org/show_bug.cgi?id=773891
+
+2016-10-09 15:59:05 +0200 Havard Graff <havard.graff@gmail.com>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ * gst/rtpmanager/rtpjitterbuffer.c:
+ * gst/rtpmanager/rtpjitterbuffer.h:
+ * tests/check/elements/rtpjitterbuffer.c:
+ rtpjitterbuffer: fix lost-event using dts instead of pts
+ The lost-event was using a different time-domain (dts) than the outgoing
+ buffers (pts). Given certain network-conditions these two would become
+ sufficiently different and the lost-event contained timestamp/duration
+ that was really wrong. As an example GstAudioDecoder could produce
+ a stream that jumps back and forth in time after receiving a lost-event.
+ The previous behavior calculated the pts (based on the rtptime) inside the
+ rtp_jitter_buffer_insert function, but now this functionality has been
+ refactored into a new function rtp_jitter_buffer_calculate_pts that is
+ called much earlier in the _chain function to make pts available to
+ various calculations that wrongly used dts previously
+ (like the lost-event).
+ There are however two calculations where using dts is the right thing to
+ do: calculating the receive-jitter and the rtx-round-trip-time, where the
+ arrival time of the buffer from the network is the right metric
+ (and is what dts in fact is today).
+ The patch also adds two tests regarding B-frames or the
+ “rtptime-going-backwards”-scenario, as there were some concerns that this
+ patch might break this behavior (which the tests shows it does not).
+
+2016-11-03 16:33:53 +0100 Havard Graff <havard.graff@gmail.com>
+
+ * gst/rtpmanager/gstrtpjitterbuffer.c:
+ * tests/check/elements/rtpjitterbuffer.c:
+ rtpjitterbuffer: fix bug in reschedule_timer
+ The new timeout is always going to be (timeout + delay), however, the
+ old behavior compared the current timeout to just (timeout), basically
+ being (delay) off.
+ This would happen if rtx-delay == rtx-retry-timeout, with the result that
+ a second rtx attempt for any buffers would be scheduled immediately instead
+ of after rtx-delay ms.
+ Simply calculate (new_timeout = timeout + delay) and then use that instead.
+ https://bugzilla.gnome.org/show_bug.cgi?id=773905
+
+2016-11-03 13:27:51 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * tests/check/elements/wavparse.c:
+ * tests/files/Makefile.am:
+ * tests/files/audiotestsrc.wav:
+ tests: wavparse: add test for processing an actual .wav file
+ https://bugzilla.gnome.org/show_bug.cgi?id=773861
+
+2016-11-03 12:34:51 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/wavparse/gstwavparse.c:
+ wavparse: Don't set caps to NULL after setting them on the srcpad
+ We would like to check later on EOS if we found a known stream type or
+ not, to possibly post an error message.
+ https://bugzilla.gnome.org/show_bug.cgi?id=773861
+
+2016-11-02 14:33:28 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/isomp4/gstqtmux.c:
+ qtmux: Don't deref NULL pads in debug output
+ That tends to crash.
+
+2016-11-02 11:46:07 +1100 Jan Schmidt <jan@centricular.com>
+
+ * gst/isomp4/qtdemux.c:
+ isomp4: Don't use gst_video_colorimetry_to_string_full()
+ The API was reverted. Just use the plain
+ gst_video_colorimetry_to_string() function.
+
+2016-11-02 11:00:13 +1100 Jan Schmidt <jan@centricular.com>
+
+ * gst/multifile/gstsplitmuxsink.c:
+ splitmuxsink: Fix GObject warnings on shutdown.
+ Commit 83e718 added a pad template to splitmux request
+ pads, which means that GstElement now releases the pads on
+ dispose, but after having removed all elements in the bin
+ and unlinked them. Make sure we can handle cleanup in that case
+ without throwing assertions.
+ https://bugzilla.gnome.org/show_bug.cgi?id=773784
+
+2016-11-02 02:25:51 +1100 Jan Schmidt <jan@centricular.com>
+
+ * gst/multifile/gstsplitmuxsrc.c:
+ * gst/multifile/gstsplitmuxsrc.h:
+ splitmuxsrc: Store seek seqnum and send it on EOS / segment events.
+ GES relies on the EOS event having the seqnum of the seek that
+ caused it.
+
+2016-11-02 02:25:00 +1100 Jan Schmidt <jan@centricular.com>
+
+ * gst/multifile/gstsplitmuxsrc.c:
+ splitmuxsrc: Forward a not-linked error on the bus
+ Handle not-linked as for other fatal errors and post it
+ onto the bus so the app knows
+
+2016-11-01 21:00:15 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Fix compiler warning
+ qtdemux.c: In function ‘qtdemux_parse_tree’:
+ qtdemux.c:10139:16: error: ‘color_table_id’ may be used uninitialized in this function [-Werror=maybe-uninitialized]
+ if (color_table_id != 0) {
+ ^
+ qtdemux.c:10121:19: note: ‘color_table_id’ was declared here
+ guint16 color_table_id;
+ ^~~~~~~~~~~~~~
+
+2016-10-20 17:40:59 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/isomp4/gstqtmux.c:
+ qtmux: Use a default interleave of 250ms for all codecs
+ https://bugzilla.gnome.org/show_bug.cgi?id=773217
+
+2016-10-19 14:33:33 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/isomp4/gstqtmux.c:
+ qtmux: Use a default interleave when ProRes is used
+ The ProRes guidelines suggest an interleave of 0.5s is common, but
+ specifies that for ProRes at most 2MB (for SD) and 4MB (for HD) should
+ be used per chunk.
+ It might also make sense to use similar numbers in general.
+ https://bugzilla.gnome.org/show_bug.cgi?id=773217
+
+2016-10-19 14:25:28 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/isomp4/atoms.c:
+ * gst/isomp4/gstqtmux.c:
+ * gst/isomp4/gstqtmux.h:
+ qtmux: Allow configuring the interleave size in bytes/time
+ Previously we were switching from one chunk to another on every single
+ buffer. This wastes some space in the headers and, depending on the
+ software, might depend in more reads (e.g. if the software is reading
+ multiple samples in one go if they're in the same chunk).
+ The ProRes guidelines suggest an interleave of 0.5s is common, but
+ specifies that for ProRes at most 2MB (for SD) and 4MB (for HD) should
+ be used per chunk. This will be handled in a follow-up commit.
+ https://bugzilla.gnome.org/show_bug.cgi?id=773217
+
+2016-09-30 18:22:27 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/isomp4/gstqtmux.c:
+ qtmux: Set compressor name, horizontal/vertical resolution and depth for ProRes
+ This is also required by some software to handle ProRes files.
+ https://bugzilla.gnome.org/show_bug.cgi?id=769048
+
+2016-09-30 18:05:38 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/isomp4/fourcc.h:
+ * gst/isomp4/gstqtmux.c:
+ * gst/isomp4/qtdemux.c:
+ qt: Add support for ProRes 4444 XQ
+ And also 4444 in the muxer.
+ https://bugzilla.gnome.org/show_bug.cgi?id=769048
+
+2016-09-30 17:58:37 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/isomp4/atoms.c:
+ * gst/isomp4/atoms.h:
+ * gst/isomp4/fourcc.h:
+ * gst/isomp4/gstqtmux.c:
+ * gst/isomp4/qtdemux_types.c:
+ qtmux: Write 'clap' atom for ProRes
+ It's required for ProRes to work with other software.
+ It is also in the MP4 standard, but inventing values here seems a bit
+ tricky for the general case and it does not really give any extra
+ information.
+ https://bugzilla.gnome.org/show_bug.cgi?id=769048
+
+2016-09-30 09:55:58 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Read colorimetry information from colr atom if available
+ https://bugzilla.gnome.org/show_bug.cgi?id=772181
+
+2016-09-29 21:56:18 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/isomp4/atoms.c:
+ * gst/isomp4/atoms.h:
+ * gst/isomp4/fourcc.h:
+ * gst/isomp4/gstqtmux.c:
+ qtmux: Always write colr atom with the colorimetry information
+ https://bugzilla.gnome.org/show_bug.cgi?id=772181
+
+2016-09-29 18:16:18 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/isomp4/atoms.c:
+ * gst/isomp4/atoms.h:
+ * gst/isomp4/gstqtmux.c:
+ qtmux: Fix writing of the 'fiel' extension atom
+ This was also wrong for JPEG2000. Also write it for all MOV files and
+ JPEG2000, not only for ProRes.
+ https://bugzilla.gnome.org/show_bug.cgi?id=769048
+
+2016-09-29 17:40:23 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/isomp4/atoms.c:
+ qtmux: Write 4 bytes of zeroes at the end of the sample description extensions
+ This is working around some broken software.
+ https://bugzilla.gnome.org/show_bug.cgi?id=769048
+
+2016-09-28 20:55:24 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/isomp4/atoms.c:
+ atoms: 'pasp' atom is also part of MP4, write it always
+ https://bugzilla.gnome.org/show_bug.cgi?id=769048
+
+2016-07-11 19:30:12 +0300 Vivia Nikolaidou <vivia@ahiru.eu>
+
+ * gst/isomp4/atoms.c:
+ * gst/isomp4/atoms.h:
+ * gst/isomp4/fourcc.h:
+ * gst/isomp4/gstqtmux.c:
+ qtmux: Write additional atoms for prores video
+ These required atoms are: colorimetry, field information, spatial/temporal
+ quality, and vendor.
+ https://bugzilla.gnome.org/show_bug.cgi?id=769048
+
+2014-06-16 17:20:32 +0200 Stian Selnes <stian.selnes@gmail.com>
+
+ * gst/rtp/gstrtph263depay.c:
+ rtph263depay: Don't drop mode b packets with picture start code
+ Some buggy payloaders, e.g. rtph263pay, may use mode B for packets
+ that starts with a picture (or GOB) start code although it's not
+ allowed. Let's be nice and not drop these packets/frames.
+ https://bugzilla.gnome.org/show_bug.cgi?id=773516
+
+2016-06-22 13:59:35 +0200 Havard Graff <havard.graff@gmail.com>
+
+ * gst/rtp/gstrtph263ppay.c:
+ * tests/check/elements/rtph263.c:
+ rtph263ppay: Fix caps leak
+ Fix leaking caps when downstream has not-fixed caps.
+ https://bugzilla.gnome.org/show_bug.cgi?id=773515
+
+2016-10-26 16:42:19 +0200 Stian Selnes <stian@pexip.com>
+
+ * gst/rtp/gstrtph263pay.c:
+ rtph263pay: Fix indentation
+ https://bugzilla.gnome.org/show_bug.cgi?id=773514
+
+2016-10-18 11:35:58 +0200 Stian Selnes <stian@pexip.com>
+
+ * gst/rtp/gstrtph263pay.c:
+ rtph263pay: Use GST_TRACE_OBJECT for logging bitstream parsing
+ Bump the bitstream parsing to TRACE log level so it doesn't flood the
+ output when trying to read the more useful DEBUG and LOG messages.
+ Also use GST_DEBUG_OBJECT instead of GST_DEBUG in various places
+ https://bugzilla.gnome.org/show_bug.cgi?id=773514
+
+2016-10-18 11:09:10 +0200 Stian Selnes <stian@pexip.com>
+
+ * gst/rtp/gstrtph263pay.c:
+ rtph263pay: Fix leak for B-fragments
+ Altough commits 6a16be7, 64f9d08 and 0c7e3a8 fixed some issues they
+ introduced others. This patch fixes the leak of one macroblock for every
+ B fragment.
+ Macroblock structures must not be freed immediately after finding the
+ boundaries as they are stored and used later. However the inital dummy
+ structure (used for finding the first boundary) must be freed.
+ CID #1212156
+ https://bugzilla.gnome.org/show_bug.cgi?id=773512
+
+2016-10-20 13:14:13 +0200 Alejandro G. Castro <alex@igalia.com>
+
+ * gst/rtpmanager/rtpsession.c:
+ rtpbin: avoid generating errors when rtcp messages are empty and check the queue is not empty
+ Add a check to verify all the output buffers were empty for the
+ session in a timout and log an error.
+ https://bugzilla.gnome.org/show_bug.cgi?id=773269
+
+2016-10-26 13:21:29 +0200 Alejandro G. Castro <alex@igalia.com>
+
+ * gst/rtpmanager/gstrtpsession.c:
+ * gst/rtpmanager/rtpsession.c:
+ * gst/rtpmanager/rtpsession.h:
+ rtpbin: pipeline gets an EOS when any rtpsources byes
+ Instead of sending EOS when a source byes we have to wait for
+ all the sources to be gone, which means they already sent BYE and
+ were removed from the session. We now handle the EOS in the rtcp
+ loop checking the amount of sources in the session.
+ https://bugzilla.gnome.org/show_bug.cgi?id=773218
+
+2016-10-21 17:31:00 +0000 Matt Staples <staples255@gmail.com>
+
+ * gst/rtsp/gstrtspsrc.c:
+ rtspsrc: Also handle redirect on PLAY
+ https://bugzilla.gnome.org/show_bug.cgi?id=772610
+
+2016-08-30 10:24:43 +0200 Petr Kulhavy <brain@jikos.cz>
+
+ * gst/rtsp/gstrtspsrc.c:
+ rtspsrc: allow missing control attribute in case of a single stream
+ Improve RFC2326 - chapter C.3 compatibility:
+ In case just a single stream is specified in SDP and the control attribute
+ is missing do not drop the stream but rather assume "a=control:*"
+ https://bugzilla.gnome.org/show_bug.cgi?id=770568
+
+2016-10-08 18:11:17 +0200 William Manley <will@williammanley.net>
+
+ * sys/v4l2/gstv4l2allocator.c:
+ v4l2: Warn, don't assert if v4l gives us a buffer with a too large size
+ I've seen problems where the `bytesused` field of `v4l2_buffer` would be
+ a silly number causing the later call to:
+ gst_memory_resize (group->mem[i], 0, group->planes[i].bytesused);
+ to result in this error to be printed:
+ (pulsevideo:11): GStreamer-CRITICAL **: gst_memory_resize: assertion 'size + mem->offset + offset <= mem->maxsize' failed
+ besides causing who-knows what other problems.
+ We make the assumption that this buffer has still been dequeued correctly
+ so just clamp to a valid size so downstream elements won't end up in
+ undefined behaviour.
+ The invalid `v4l2_buffer` I saw from my capture device was:
+ buffer = {
+ index = 0,
+ type = 1,
+ bytesused = 534748928, // <- Invalid
+ flags = 8260, // V4L2_BUF_FLAG_TIMESTAMP_MONOTONIC | V4L2_BUF_FLAG_ERROR | V4L2_BUF_FLAG_DONE
+ field = 01330, // <- Invalid
+ timestamp = {
+ tv_sec = 0,
+ tv_usec = 0
+ },
+ timecode = {
+ type = 0,
+ flags = 0,
+ frames = 0 '\000',
+ seconds = 0 '\000',
+ minutes = 0 '\000',
+ hours = 0 '\000',
+ userbits = "\000\000\000"
+ },
+ sequence = 0,
+ memory = 2,
+ m = {
+ offset = 3537219584,
+ userptr = 140706665836544, // Could be nonsense, not sure
+ planes = 0x7ff8d2d5b000,
+ fd = -757747712
+ },
+ length = 2764800,
+ reserved2 = 0,
+ reserved = 0
+ }
+ This is from gdb with my own annotations added.
+ This was with gst-plugins-good 1.8.1, a Magewell XI100DUSB-HDMI video
+ capture device and kernel 3.13 using a dodgy HDMI cable which is great at
+ breaking HDMI capture devices. I'm using io-mode=userptr and have built
+ gst-plugins-good without libv4l.
+ https://bugzilla.gnome.org/show_bug.cgi?id=769765
+
+2016-10-20 20:41:07 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/isomp4/gstqtmux.c:
+ qtmux: Use a better default value for the movie header timescale
+ Take the maximum video timescale, or if no video track is present the
+ previous value of 1800.
+ https://bugzilla.gnome.org/show_bug.cgi?id=769041
+
+2016-10-20 20:07:19 +0300 Sebastian Dröge <sebastian@centricular.com>
+
+ * gst/isomp4/gstqtmux.c:
+ qtmux: Be more clever with the default video track timescale
+ Use the number of milliframes per second for integral and drop-frame
+ framerates, as suggested by the QT file format specification and other
+ places. We already did that for integral framerates before, but not for
+ drop-frame framerates. This now keeps precision better.
+ For all other framerates, check if it's close to a well-known framerate
+ and use that instead.
+ https://bugzilla.gnome.org/show_bug.cgi?id=769041
+
+2016-10-10 13:00:01 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: extract interlaced information from jpeg video
+ This information is hidden in a small chunk of data.
+ Format found at https://developer.apple.com/standards/qtff-2001.pdf,
+ page 92, "Video Sample Description", under table 3.1.
+ https://bugzilla.gnome.org/show_bug.cgi?id=767771
+
+2016-10-26 12:46:28 +0530 Jagadish <jagadishkamathk@gmail.com>
+
+ * ext/gdk_pixbuf/gstgdkpixbufoverlay.c:
+ gdkpixbufoverlay: Fixing x and y offset computation
+ While computing the x and y offsets, it's the video resolution and
+ resized overlay resolution to be used instead of actual overlay image
+ resoltuion. Due to this, the overlay image used to get wrongly overlayed
+ in undesired location
+ https://bugzilla.gnome.org/show_bug.cgi?id=757292
+
+2016-11-01 18:09:00 +0000 Tim-Philipp Müller <tim@centricular.com>
+
+ * meson.build:
+ meson: update version
+
+2016-10-24 16:56:31 +0000 Enrique Ocaña González <eocanha@igalia.com>
+
+ * gst/isomp4/qtdemux.c:
+ qtdemux: Use the tfdt decode time on byte streams when it's significantly different than the time in the last sample
+ We consider there's a sifnificant difference when it's larger than on second
+ or than half the duration of the last processed fragment in case the latter is
+ larger.
+ https://bugzilla.gnome.org/show_bug.cgi?id=754230
+
+=== release 1.11.0 ===
+
+2016-11-01 18:53:15 +0200 Sebastian Dröge <sebastian@centricular.com>
+
+ * configure.ac:
+ Back to development
+
=== release 1.10.0 ===
-2016-11-01 Sebastian Dröge <slomo@coaxion.net>
+2016-11-01 17:57:44 +0200 Sebastian Dröge <sebastian@centricular.com>
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
* configure.ac:
- releasing 1.10.0
+ * docs/plugins/gst-plugins-good-plugins.args:
+ * 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 1.10.0
+
+2016-11-01 17:47:31 +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:
+ Update .po files
2016-11-01 17:41:51 +0200 Sebastian Dröge <sebastian@centricular.com>
-# GStreamer 1.10 Release Notes
-
-**GStreamer 1.10.0 was released on 1st November 2016.**
-
-The GStreamer team is proud to announce a new major feature release in the
-stable 1.x API series of your favourite cross-platform multimedia framework!
-
-As always, this release is again packed with new features, bug fixes and other
-improvements.
-
-See [https://gstreamer.freedesktop.org/releases/1.10/][latest] for the latest
-version of this document.
-
-*Last updated: Tuesday 1 Nov 2016, 15:00 UTC [(log)][gitlog]*
-
-[latest]: https://gstreamer.freedesktop.org/releases/1.10/
-[gitlog]: https://cgit.freedesktop.org/gstreamer/www/log/src/htdocs/releases/1.10/release-notes-1.10.md
-
-## Introduction
-
-The GStreamer team is proud to announce a new major feature release in the
-stable 1.x API series of your favourite cross-platform multimedia framework!
-
-As always, this release is again packed with new features, bug fixes and other
-improvements.
-
-## Highlights
-
-- Several convenience APIs have been added to make developers' lives easier
-- A new `GstStream` API provides applications a more meaningful view of the
- structure of streams, simplifying the process of dealing with media in
- complex container formats
-- Experimental `decodebin3` and `playbin3` elements which bring a number of
- improvements which were hard to implement within `decodebin` and `playbin`
-- A new `parsebin` element to automatically unpack and parse a stream, stopping
- just short of decoding
-- Experimental new `meson`-based build system, bringing faster build and much
- better Windows support (including for building with Visual Studio)
-- A new `gst-docs` module has been created, and we are in the process of moving
- our documentation to a markdown-based format for easier maintenance and
- updates
-- A new `gst-examples` module has been create, which contains example
- GStreamer applications and is expected to grow with many more examples in
- the future
-- Various OpenGL and OpenGL|ES-related fixes and improvements for greater
- efficiency on desktop and mobile platforms, and Vulkan support on Wayland was
- also added
-- Extensive improvements to the VAAPI plugins for improved robustness and
- efficiency
-- Lots of fixes and improvements across the board, spanning RTP/RTSP, V4L2,
- Bluetooth, audio conversion, echo cancellation, and more!
-
-## Major new features and changes
-
-### Noteworthy new API, features and other changes
-
-#### Core API additions
-
-##### Receive property change notifications via bus messages
-
-New API was added to receive element property change notifications via
-bus messages. So far, applications had to connect a callback to an element's
-`notify::property-name` signal via the GObject API, which was inconvenient for
-at least two reasons: one had to implement a signal callback function, and that
-callback function would usually be called from one of the streaming threads, so
-one had to marshal (send) any information gathered or pending requests to the
-main application thread which was tedious and error-prone.
-
-Enter [`gst_element_add_property_notify_watch()`][notify-watch] and
-[`gst_element_add_property_deep_notify_watch()`][deep-notify-watch] which will
-watch for changes of a property on the specified element, either only for this
-element or recursively for a whole bin or pipeline. Whenever such a
-property change happens, a `GST_MESSAGE_PROPERTY_NOTIFY` message will be posted
-on the pipeline bus with details of the element, the property and the new
-property value, all of which can be retrieved later from the message in the
-application via [`gst_message_parse_property_notify()`][parse-notify]. Unlike
-the GstBus watch functions, this API does not rely on a running GLib main loop.
-
-The above can be used to be notified asynchronously of caps changes in the
-pipeline, or volume changes on an audio sink element, for example.
-
-[notify-watch]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstElement.html#gst-element-add-property-notify-watch
-[deep-notify-watch]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstElement.html#gst-element-add-property-deep-notify-watch
-[parse-notify]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstMessage.html#gst-message-parse-property-notify
-
-##### GstBin "deep" element-added and element-removed signals
-
-GstBin has gained `"deep-element-added"` and `"deep-element-removed"` signals
-which makes it easier for applications and higher-level plugins to track when
-elements are added or removed from a complex pipeline with multiple sub-bins.
-
-`playbin` makes use of this to implement the new `"element-setup"` signal which
-can be used to configure elements as they are added to `playbin`, just like the
-existing `"source-setup"` signal which can be used to configure the source
-element created.
-
-##### Error messages can contain additional structured details
-
-It is often useful to provide additional, structured information in error,
-warning or info messages for applications (or higher-level elements) to make
-intelligent decisions based on them. To allow this, error, warning and info
-messages now have API for adding arbitrary additional information to them
-using a `GstStructure`:
-[`GST_ELEMENT_ERROR_WITH_DETAILS`][element-error-with-details] and
-corresponding API for the other message types.
-
-This is now used e.g. by the new [`GST_ELEMENT_FLOW_ERROR`][element-flow-error]
-API to include the actual flow error in the error message, and the
-[souphttpsrc element][souphttpsrc-detailed-errors] to provide the HTTP
-status code, and the URL (if any) to which a redirection has happened.
-
-[element-error-with-details]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstElement.html#GST-ELEMENT-ERROR-WITH-DETAILS:CAPS
-[element-flow-error]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstElement.html#GST-ELEMENT-FLOW-ERROR:CAPS
-[souphttpsrc-detailed-errors]: https://cgit.freedesktop.org/gstreamer/gst-plugins-good/tree/ext/soup/gstsouphttpsrc.c?id=60d30db912a1aedd743e66b9dcd2e21d71fbb24f#n1318
-
-##### Redirect messages have official API now
-
-Sometimes, elements need to redirect the current stream URL and tell the
-application to proceed with this new URL, possibly using a different
-protocol too (thus changing the pipeline configuration). Until now, this was
-informally implemented using `ELEMENT` messages on the bus.
-
-Now this has been formalized in the form of a new `GST_MESSAGE_REDIRECT` message.
-A new redirect message can be created using [`gst_message_new_redirect()`][new-redirect].
-If needed, multiple redirect locations can be specified by calling
-[`gst_message_add_redirect_entry()`][add-redirect] to add further redirect
-entries, all with metadata, so the application can decide which is
-most suitable (e.g. depending on the bitrate tags).
-
-[new-redirect]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstMessage.html#gst-message-new-redirect
-[add-redirect]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstMessage.html#gst-message-add-redirect-entry
-
-##### New pad linking convenience functions that automatically create ghost pads
-
-New pad linking convenience functions were added:
-[`gst_pad_link_maybe_ghosting()`][pad-maybe-ghost] and
-[`gst_pad_link_maybe_ghosting_full()`][pad-maybe-ghost-full] which were
-previously internal to GStreamer have now been exposed for general use.
-
-The existing pad link functions will refuse to link pads or elements at
-different levels in the pipeline hierarchy, requiring the developer to
-create ghost pads where necessary. These new utility functions will
-automatically create ghostpads as needed when linking pads at different
-levels of the hierarchy (e.g. from an element inside a bin to one that's at
-the same level in the hierarchy as the bin, or in another bin).
-
-[pad-maybe-ghost]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstPad.html#gst-pad-link-maybe-ghosting
-[pad-maybe-ghost-full]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstPad.html#gst-pad-link-maybe-ghosting-full
-
-##### Miscellaneous
-
-Pad probes: IDLE and BLOCK probes now work slightly differently in pull mode,
-so that push and pull mode have opposite scenarios for idle and blocking probes.
-In push mode, it will block with some data type and IDLE won't have any data.
-In pull mode, it will block _before_ getting a buffer and will be IDLE once some
-data has been obtained. ([commit][commit-pad-probes], [bug][bug-pad-probes])
-
-[commit-pad-probes]: https://cgit.freedesktop.org/gstreamer/gstreamer/commit/gst/gstpad.c?id=368ee8a336d0c868d81fdace54b24431a8b48cbf
-[bug-pad-probes]: https://bugzilla.gnome.org/show_bug.cgi?id=761211
-
-[`gst_parse_launch_full()`][parse-launch-full] can now be made to return a
-`GstBin` instead of a top-level pipeline by passing the new
-`GST_PARSE_FLAG_PLACE_IN_BIN` flag.
-
-[parse-launch-full]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/gstreamer-GstParse.html#gst-parse-launch-full
-
-The default GStreamer debug log handler can now be removed before
-calling `gst_init()`, so that it will never get installed and won't be active
-during initialization.
-
-A new [`STREAM_GROUP_DONE` event][stream-group-done-event] was added. In some
-ways it works similar to the `EOS` event in that it can be used to unblock
-downstream elements which may be waiting for further data, such as for example
-`input-selector`. Unlike `EOS`, further data flow may happen after the
-`STREAM_GROUP_DONE` event though (and without the need to flush the pipeline).
-This is used to unblock input-selector when switching between streams in
-adaptive streaming scenarios (e.g. HLS).
-
-[stream-group-done-event]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstEvent.html#gst-event-new-stream-group-done
-
-The `gst-launch-1.0` command line tool will now print unescaped caps in verbose
-mode (enabled by the -v switch).
-
-[`gst_element_call_async()`][call-async] has been added as convenience API for
-plugin developers. It is useful for one-shot operations that need to be done
-from a thread other than the current streaming thread. It is backed by a
-thread-pool that is shared by all elements.
-
-[call-async]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstElement.html#gst-element-call-async
-
-Various race conditions have been fixed around the `GstPoll` API used by e.g.
-`GstBus` and `GstBufferPool`. Some of these manifested themselves primarily
-on Windows.
-
-`GstAdapter` can now keep track of discontinuities signalled via the `DISCONT`
-buffer flag, and has gained [new API][new-adapter-api] to track PTS, DTS and
-offset at the last discont. This is useful for plugins implementing advanced
-trick mode scenarios.
-
-[new-adapter-api]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer-libs/html/GstAdapter.html#gst-adapter-pts-at-discont
-
-`GstTestClock` gained a new [`"clock-type"` property][clock-type-prop].
-
-[clock-type-prop]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer-libs/html/GstTestClock.html#GstTestClock--clock-type
-
-#### GstStream API for stream announcement and stream selection
-
-New stream listing and stream selection API: new API has been added to
-provide high-level abstractions for streams ([`GstStream`][stream-api])
-and collections of streams ([`GstStreamCollections`][stream-collection-api]).
-
-##### Stream listing
-
-A [`GstStream`][stream-api] contains all the information pertinent to a stream,
-such as stream id, caps, tags, flags and stream type(s); it can represent a
-single elementary stream (e.g. audio, video, subtitles, etc.) or a container
-stream. This will depend on the context. In a decodebin3/playbin3 one
-it will typically be elementary streams that can be selected and unselected.
-
-A [`GstStreamCollection`][stream-collection-api] represents a group of streams
-and is used to announce or publish all available streams. A GstStreamCollection
-is immutable - once created it won't change. If the available streams change,
-e.g. because a new stream appeared or some streams disappeared, a new stream
-collection will be published. This new stream collection may contain streams
-from the previous collection if those streams persist, or completely new ones.
-Stream collections do not yet list all theoretically available streams,
-e.g. other available DVD angles or alternative resolutions/bitrate of the same
-stream in case of adaptive streaming.
-
-New events and messages have been added to notify or update other elements and
-the application about which streams are currently available and/or selected.
-This way, we can easily and seamlessly let the application know whenever the
-available streams change, as happens frequently with digital television streams
-for example. The new system is also more flexible. For example, it is now also
-possible for the application to select multiple streams of the same type
-(e.g. in a transcoding/transmuxing scenario).
-
-A [`STREAM_COLLECTION` message][stream-collection-msg] is posted on the bus
-to inform the parent bin (e.g. `playbin3`, `decodebin3`) and/or the application
-about what streams are available, so you no longer have to hunt for this
-information at different places. The available information includes number of
-streams of each type, caps, tags etc. Bins and/or the application can intercept
-the message synchronously to select and deselect streams before any data is
-produced - for the case where elements such as the demuxers support the new
-stream API, not necessarily in the parsebin compatibility fallback case.
-
-Similarly, there is also a [`STREAM_COLLECTION` event][stream-collection-event]
-to inform downstream elements of the available streams. This event can be used
-by elements to aggregate streams from multiple inputs into one single collection.
-
-The `STREAM_START` event was extended so that it can also contain a GstStream
-object with all information about the current stream, see
-[`gst_event_set_stream()`][event-set-stream] and
-[`gst_event_parse_stream()`][event-parse-stream].
-[`gst_pad_get_stream()`][pad-get-stream] is a new utility function that can be
-used to look up the GstStream from the `STREAM_START` sticky event on a pad.
-
-[stream-api]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/gstreamer-GstStream.html
-[stream-collection-api]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/gstreamer-GstStreamCollection.html
-[stream-collection-msg]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstMessage.html#gst-message-new-stream-collection
-[stream-collection-event]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstEvent.html#gst-event-new-stream-collection
-[event-set-stream]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstEvent.html#gst-event-set-stream
-[event-parse-stream]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstEvent.html#gst-event-parse-stream
-[pad-get-stream]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstPad.html#gst-pad-get-stream
-
-##### Stream selection
-
-Once the available streams have been published, streams can be selected via
-their stream ID using the new `SELECT_STREAMS` event, which can be created
-with [`gst_event_new_select_streams()`][event-select-streams]. The new API
-supports selecting multiple streams per stream type. In the future, we may also
-implement explicit deselection of streams that will never be used, so
-elements can skip these and never expose them or output data for them in the
-first place.
-
-The application is then notified of the currently selected streams via the
-new `STREAMS_SELECTED` message on the pipeline bus, containing both the current
-stream collection as well as the selected streams. This might be posted in
-response to the application sending a `SELECT_STREAMS` event or when
-`decodebin3` or `playbin3` decide on the streams to be initially selected without
-application input.
-
-[event-select-streams]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gstreamer/html/GstEvent.html#gst-event-new-select-streams
-
-##### Further reading
-
-See further below for some notes on the new elements supporting this new
-stream API, namely: `decodebin3`, `playbin3` and `parsebin`.
-
-More information about the new API and the new elements can also be found here:
-
-- GStreamer [stream selection design docs][streams-design]
-- Edward Hervey's talk ["The new streams API: Design and usage"][streams-talk] ([slides][streams-slides])
-- Edward Hervey's talk ["Decodebin3: Dealing with modern playback use cases"][db3-talk] ([slides][db3-slides])
-
-[streams-design]: https://cgit.freedesktop.org/gstreamer/gstreamer/tree/docs/design/part-stream-selection.txt
-[streams-talk]: https://gstconf.ubicast.tv/videos/the-new-gststream-api-design-and-usage/
-[streams-slides]: https://gstreamer.freedesktop.org/data/events/gstreamer-conference/2016/Edward%20Hervey%20-%20The%20New%20Streams%20API%20Design%20and%20Usage.pdf
-[db3-talk]: https://gstconf.ubicast.tv/videos/decodebin3-or-dealing-with-modern-playback-use-cases/
-[db3-slides]: https://gstreamer.freedesktop.org/data/events/gstreamer-conference/2015/Edward%20Hervey%20-%20decodebin3.pdf
-
-#### Audio conversion and resampling API
-
-The audio conversion library received a completely new and rewritten audio
-resampler, complementing the audio conversion routines moved into the audio
-library in the [previous release][release-notes-1.8]. Integrating the resampler
-with the other audio conversion library allows us to implement generic
-conversion much more efficiently, as format conversion and resampling can now
-be done in the same processing loop instead of having to do it in separate
-steps (our element implementations do not make use of this yet though).
-
-The new audio resampler library is a combination of some of the best features
-of other samplers such as ffmpeg, speex and SRC. It natively supports S16, S32,
-F32 and F64 formats and uses optimized x86 and neon assembly for most of its
-processing. It also has support for dynamically changing sample rates by incrementally
-updating the filter tables using linear or cubic interpolation. According to
-some benchmarks, it's one of the fastest and most accurate resamplers around.
-
-The `audioresample` plugin has been ported to the new audio library functions
-to make use of the new resampler.
-
-[release-notes-1.8]: https://gstreamer.freedesktop.org/releases/1.8/
-
-#### Support for SMPTE timecodes
-
-Support for SMPTE timecodes was added to the GStreamer video library. This
-comes with an abstraction for timecodes, [`GstVideoTimeCode`][video-timecode]
-and a [`GstMeta`][video-timecode-meta] that can be placed on video buffers for
-carrying the timecode information for each frame. Additionally there is
-various API for making handling of timecodes easy and to do various
-calculations with them.
-
-A new plugin called [`timecode`][timecode-plugin] was added, that contains an
-element called `timecodestamper` for putting the timecode meta on video frames
-based on counting the frames and another element called `timecodewait` that
-drops all video (and audio) until a specific timecode is reached.
-
-Additionally support was added to the Decklink plugin for including the
-timecode information when sending video out or capturing it via SDI, the
-`qtmux` element is able to write timecode information into the MOV container,
-and the `timeoverlay` element can overlay timecodes on top of the video.
-
-More information can be found in the [talk about timecodes][timecode-talk] at
-the GStreamer Conference 2016.
-
-[video-timecode]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstvideo.html#GstVideoTimeCode
-[video-timecode-meta]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-libs/html/gst-plugins-base-libs-gstvideometa.html#gst-buffer-add-video-time-code-meta
-[timecode-plugin]: https://cgit.freedesktop.org/gstreamer/gst-plugins-bad/tree/gst/timecode
-[timecode-talk]: https://gstconf.ubicast.tv/videos/smpte-timecodes-in-gstreamer/
-
-#### GStreamer OpenMAX IL plugin
-
-The last gst-omx release, 1.2.0, was in July 2014. It was about time to get
-a new one out with all the improvements that have happened in the meantime.
-From now on, we will try to release gst-omx together with all other modules.
-
-This release features a lot of bugfixes, improved support for the Raspberry Pi
-and in general improved support for zerocopy rendering via EGL and a few minor
-new features.
-
-At this point, gst-omx is known to work best on the Raspberry Pi platform but
-it is also known to work on various other platforms. Unfortunately, we are
-not including configurations for any other platforms, so if you happen to use
-gst-omx: please send us patches with your configuration and code changes!
-
-### New Elements
-
-#### decodebin3, playbin3, parsebin (experimental)
-
-This release features new decoding and playback elements as experimental
-technology previews: `decodebin3` and `playbin3` will soon supersede the
-existing `decodebin` and `playbin` elements. We skipped the number 2 because
-it was already used back in the 0.10 days, which might cause confusion.
-Experimental technology preview means that everything should work fine already,
-but we can't guarantee there won't be minor behavioural changes in the
-next cycle. In any case, please test and report any problems back.
-
-Before we go into detail about what these new elements improve, let's look at
-the new [`parsebin`][parsebin] element. It works similarly to `decodebin` and
-`decodebin3`, only that it stops one step short and does not plug any actual
-decoder elements. It will only plug parsers, tag readers, demuxers and
-depayloaders. Also note that parsebin does not contain any queueing element.
-
-[`decodebin3`'s][decodebin3] internal architecture is slightly different from
-the existing `decodebin` element and fixes many long-standing issues with our
-decoding engine. For one, data is now fed into the internal `multiqueue` element
-*after* it has been parsed and timestamped, which means that the `multiqueue`
-element now has more knowledge and is able to calculate the interleaving of the
-various streams, thus minimizing memory requirements and doing away with magic
-values for buffering limits that were conceived when videos were 240p or 360p.
-Anyone who has tried to play back 4k video streams with decodebin2
-will have noticed the limitations of that approach. The improved timestamp
-tracking also enables `multiqueue` to keep streams of the same type (audio,
-video) aligned better, making sure switching between streams of the same type
-is very fast.
-
-Another major improvement in `decodebin3` is that it will no longer decode
-streams that are not being used. With the old `decodebin` and `playbin`, when
-there were 8 audio streams we would always decode all 8 streams even
-if 7 were not actually used. This caused a lot of CPU overhead, which was
-particularly problematic on embedded devices. When switching between streams
-`decodebin3` will try hard to re-use existing decoders. This is useful when
-switching between multiple streams of the same type if they are encoded in the
-same format.
-
-Re-using decoders is also useful when the available streams change on the fly,
-as might happen with radio streams (chained Oggs), digital television
-broadcasts, when adaptive streaming streams change bitrate, or when switching
-gaplessly to the next title. In order to guarantee a seamless transition, the
-old `decodebin2` would plug a second decoder for the new stream while finishing
-up the old stream. With `decodebin3`, this is no longer needed - at least not
-when the new and old format are the same. This will be particularly useful
-on embedded systems where it is often not possible to run multiple decoders
-at the same time, or when tearing down and setting up decoders is fairly
-expensive.
-
-`decodebin3` also allows for multiple input streams, not just a single one.
-This will be useful, in the future, for gapless playback, or for feeding
-multiple external subtitle streams to decodebin/playbin.
-
-`playbin3` uses `decodebin3` internally, and will supercede `playbin`.
-It was decided that it would be too risky to make the old `playbin` use the
-new `decodebin3` in a backwards-compatible way. The new architecture
-makes it awkward, if not impossible, to maintain perfect backwards compatibility
-in some aspects, hence `playbin3` was born, and developers can migrate to the
-new element and new API at their own pace.
-
-All of these new elements make use of the new `GstStream` API for listing and
-selecting streams, as described above. `parsebin` provides backwards
-compatibility for demuxers and parsers which do not advertise their streams
-using the new API yet (which is most).
-
-The new elements are not entirely feature-complete yet: `playbin3` does not
-support so-called decodersinks yet where the data is not decoded inside
-GStreamer but passed directly for decoding to the sink. `decodebin3` is missing
-the various `autoplug-*` signals to influence which decoders get autoplugged
-in which order. We're looking to add back this functionality, but it will probably
-be in a different way, with a single unified signal and using GstStream perhaps.
-
-For more information on these new elements, check out Edward Hervey's talk
-[*decodebin3 - dealing with modern playback use cases*][db3-talk]
-
-[parsebin]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-plugins/html/gst-plugins-base-plugins-parsebin.html
-[decodebin3]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-base-plugins/html/gst-plugins-base-plugins-decodebin3.html
-[db3-talk]: https://gstconf.ubicast.tv/videos/decodebin3-or-dealing-with-modern-playback-use-cases/
-
-#### LV2 ported from 0.10 and switched from slv2 to lilv2
-
-The LV2 wrapper plugin has been ported to 1.0 and moved from using the
-deprecated slv2 library to its replacement liblv2. We support sources and
-filter elements. lv2 is short for *Linux Audio Developer's Simple Plugin API
-(LADSPA) version 2* and is an open standard for audio plugins which includes
-support for audio synthesis (generation), digital signal processing of digital
-audio, and MIDI. The new lv2 plugin supersedes the existing LADSPA plugin.
-
-#### WebRTC DSP Plugin for echo-cancellation, gain control and noise suppression
-
-A set of new elements ([webrtcdsp][webrtcdsp], [webrtcechoprobe][webrtcechoprobe])
-based on the WebRTC DSP software stack can now be used to improve your audio
-voice communication pipelines. They support echo cancellation, gain control,
-noise suppression and more. For more details you may read
-[Nicolas' blog post][webrtc-blog-post].
-
-[webrtcdsp]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-bad-plugins/html/gst-plugins-bad-plugins-webrtcdsp.html
-[webrtcechoprobe]: https://gstreamer.freedesktop.org/data/doc/gstreamer/head/gst-plugins-bad-plugins/html/gst-plugins-bad-plugins-webrtcechoprobe.html
-[webrtc-blog-post]: https://ndufresne.ca/2016/06/gstreamer-echo-canceller/
-
-#### Fraunhofer FDK AAC encoder and decoder
-
-New encoder and decoder elements wrapping the Fraunhofer FDK AAC library have
-been added (`fdkaacdec`, `fdkaacdec`). The Fraunhofer FDK AAC encoder is
-generally considered to be a very high-quality AAC encoder, but unfortunately
-it comes under a non-free license with the option to obtain a paid, commercial
-license.
-
-### Noteworthy element features and additions
-
-#### Major RTP and RTSP improvements
-
-- The RTSP server and source element, as well as the RTP jitterbuffer now support
- remote clock synchronization according to [RFC7273][https://tools.ietf.org/html/rfc7273].
-- Support for application and profile specific RTCP packets was added.
-- The H265/HEVC payloader/depayloader is again in sync with the final RFC.
-- Seeking stability of the RTSP source and server was improved a lot and
- runs stably now, even when doing scrub-seeking.
-- The RTSP server received various major bugfixes, including for regressions that
- caused the IP/port address pool to not be considered, or NAT hole punching
- to not work anymore. [Bugzilla #766612][https://bugzilla.gnome.org/show_bug.cgi?id=766612]
-- Various other bugfixes that improve the stability of RTP and RTSP, including
- many new unit / integration tests.
-
-#### Improvements to splitmuxsrc and splitmuxsink
-
-- The splitmux element received reliability and error handling improvements,
- removing at least one deadlock case. `splitmuxsrc` now stops cleanly at the end
- of the segment when handling seeks with a stop time. We fixed a bug with large
- amounts of downstream buffering causing incorrect out-of-sequence playback.
-
-- `splitmuxsrc` now has a `"format-location"` signal to directly specify the list
- of files to play from.
-
-- `splitmuxsink` can now optionally send force-keyunit events to upstream
- elements to allow splitting files more accurately instead of having to wait
- for upstream to provide a new keyframe by itself.
-
-#### OpenGL/GLES improvements
-
-##### iOS and macOS (OS/X)
-
-- We now create OpenGL|ES 3.x contexts on iOS by default with a fallback to
- OpenGL|ES 2.x if that fails.
-- Various zerocopy decoding fixes and enhancements with the
- encoding/decoding/capturing elements.
-- libdispatch is now used on all Apple platforms instead of GMainLoop, removing
- the expensive poll()/pthread_*() overhead.
-
-##### New API
-
-- `GstGLFramebuffer` - for wrapping OpenGL frame buffer objects. It provides
- facilities for attaching `GstGLMemory` objects to the necessary attachment
- points, binding and unbinding and running a user-supplied function with the
- framebuffer bound.
-- `GstGLRenderbuffer` (a `GstGLBaseMemory` subclass) - for wrapping OpenGL
- render buffer objects that are typically used for depth/stencil buffers or
- for color buffers where we don't care about the output.
-- `GstGLMemoryEGL` (a `GstGLMemory` subclass) - for combining `EGLImage`s with a GL
- texture that replaces `GstEGLImageMemory` bringing the improvements made to the
- other `GstGLMemory` implementations. This fixes a performance regression in
- zerocopy decoding on the Raspberry Pi when used with an updated gst-omx.
-
-##### Miscellaneous improvements
-
-- `gltestsrc` is now usable on devices/platforms with OpenGL 3.x and OpenGL|ES
- and has completed or gained support for new patterns in line with the
- existing ones in `videotestsrc`.
-- `gldeinterlace` is now available on devices/platforms with OpenGL|ES
- implementations.
-- The dispmanx backend (used on the Raspberry Pi) now supports the
- `gst_video_overlay_set_window_handle()` and
- `gst_video_overlay_set_render_rectangle()` functions.
-- The `gltransformation` element now correctly transforms mouse coordinates (in
- window space) to stream coordinates for both perspective and orthographic
- projections.
-- The `gltransformation` element now detects if the
- `GstVideoAffineTransformationMeta` is supported downstream and will efficiently
- pass its transformation downstream. This is a performance improvement as it
- results in less processing being required.
-- The wayland implementation now uses the multi-threaded safe event-loop API
- allowing correct usage in applications that call wayland functions from
- multiple threads.
-- Support for native 90 degree rotations and horizontal/vertical flips
- in `glimagesink`.
-
-#### Vulkan
-
-- The Vulkan elements now work under Wayland and have received numerous
- bugfixes.
-
-#### QML elements
-
-- `qmlglsink` video sink now works on more platforms, notably, Windows, Wayland,
- and Qt's eglfs (for embedded devices with an OpenGL implementation) including
- the Raspberry Pi.
-- New element `qmlglsrc` to record a QML scene into a GStreamer pipeline.
-
-#### KMS video sink
-
-- New element `kmssink` to render video using Direct Rendering Manager
- (DRM) and Kernel Mode Setting (KMS) subsystems in the Linux
- kernel. It is oriented to be used mostly in embedded systems.
-
-#### Wayland video sink
-
-- `waylandsink` now supports the wl_viewporter extension allowing
- video scaling and cropping to be delegated to the Wayland
- compositor. This extension is also been made optional, so that it can
- also work on current compositors that don't support it. It also now has
- support for the video meta, allowing zero-copy operations in more
- cases.
-
-#### DVB improvements
-
-- `dvbsrc` now has better delivery-system autodetection and several
- new parameter sanity-checks to improve its resilience to configuration
- omissions and errors. Superfluous polling continues to be trimmed down,
- and the debugging output has been made more consistent and precise.
- Additionally, the channel-configuration parser now supports the new dvbv5
- format, enabling `dvbbasebin` to automatically playback content transmitted
- on delivery systems that previously required manual description, like ISDB-T.
-
-#### DASH, HLS and adaptivedemux
-
-- HLS now has support for Alternate Rendition audio and video tracks. Full
- support for Alternate Rendition subtitle tracks will be in an upcoming release.
-- DASH received support for keyframe-only trick modes if the
- `GST_SEEK_FLAG_TRICKMODE_KEY_UNITS` flag is given when seeking. It will
- only download keyframes then, which should help with high-speed playback.
- Changes to skip over multiple frames based on bandwidth and other metrics
- will be added in the near future.
-- Lots of reliability fixes around seek handling and bitrate switching.
-
-#### Bluetooth improvements
-
-- The `avdtpsrc` element now supports metadata such as track title, artist
- name, and more, which devices can send via AVRCP. These are published as
- tags on the pipeline.
-- The `a2dpsink` element received some love and was cleaned up so that it
- actually works after the initial GStreamer 1.0 port.
-
-#### GStreamer VAAPI
-
-- All the decoders have been split, one plugin feature per codec. So
- far, the available ones, depending on the driver, are:
- `vaapimpeg2dec`, `vaapih264dec`, `vaapih265dec`, `vaapivc1dec`, `vaapivp8dec`,
- `vaapivp9dec` and `vaapijpegdec` (which already was split).
-- Improvements when mapping VA surfaces into memory. It now differentiates
- between negotiation caps and allocations caps, since the allocation
- memory for surfaces may be bigger than one that is going to be
- mapped.
-- `vaapih265enc` now supports constant bitrate mode (CBR).
-- Since several VA drivers are unmaintained, we decide to keep a whitelist
- with the va drivers we actually test, which is mostly the i915 and to a lesser
- degree gallium from the mesa project. Exporting the environment variable
- `GST_VAAPI_ALL_DRIVERS` disables the whitelist.
-- Plugin features are registered at run-time, according to their support by
- the loaded VA driver. So only the decoders and encoder supported by the
- system are registered. Since the driver can change, some dependencies are
- tracked to invalidate the GStreamer registry and reload the plugin.
-- `dmabuf` importation from upstream has been improved, gaining performance.
-- `vaapipostproc` now can negotiate buffer transformations via caps.
-- Decoders now can do I-frame only reverse playback. This decodes I-frames
- only because the surface pool is smaller than the required by the GOP to show all the
- frames.
-- The upload of frames onto native GL textures has been optimized too, keeping
- a cache of the internal structures for the offered textures by the sink.
-
-#### V4L2 changes
-
-- More pixels formats are now supported
-- Decoder is now using `G_SELECTION` instead of the deprecated `G_CROP`
-- Decoder now uses the `STOP` command to handle EOS
-- Transform element can now scale the pixel aspect ratio
-- Colorimetry support has been improved even more
-- We now support the `OUTPUT_OVERLAY` type of video node in v4l2sink
-
-#### Miscellaneous
-
-- `multiqueue`'s input pads gained a new `"group-id"` property which
- can be used to group input streams. Typically one will assign
- different id numbers to audio, video and subtitle streams for
- example. This way `multiqueue` can make sure streams of the same
- type advance in lockstep if some of the streams are unlinked and the
- `"sync-by-running-time"` property is set. This is used in
- decodebin3/playbin3 to implement almost-instantaneous stream
- switching. The grouping is required because different downstream
- paths (audio, video, etc.) may have different buffering/latency
- etc. so might be consuming data from multiqueue with a slightly
- different phase, and if we track different stream groups separately
- we minimize stream switching delays and buffering inside the
- `multiqueue`.
-- `alsasrc` now supports ALSA drivers without a position for each
- channel, this is common in some professional or industrial hardware.
-- `libvpx` based decoders (`vp8dec` and `vp9dec`) now create multiple threads on
- computers with multiple CPUs automatically.
-- `rfbsrc` - used for capturing from a VNC server - has seen a lot of
- debugging. It now supports the latest version of the RFB
- protocol and uses GIO everywhere.
-- `tsdemux` can now read ATSC E-AC-3 streams.
-- New `GstVideoDirection` video orientation interface for rotating, flipping
- and mirroring video in 90° steps. It is implemented by the `videoflip` and
- `glvideoflip` elements currently.
-- It is now possible to give `appsrc` a duration in time, and there is now a
- non-blocking try-pull API for `appsink` that returns NULL if nothing is
- available right now.
-- `x264enc` has support now for chroma-site and colorimetry settings
-- A new JPEG2000 parser element was added, and the JPEG2000 caps were cleaned
- up and gained more information needed in combination with RTP and various
- container formats.
-- Reverse playback support for `videorate` and `deinterlace` was implemented
-- Various improvements everywhere for reverse playback and `KEY_UNITS` trick mode
-- New cleaned up `rawaudioparse` and `rawvideoparse` elements that replace the
- old `audioparse` and `videoparse` elements. There are compatibility element
- factories registered with the old names to allow existing code to continue
- to work.
-- The Decklink plugin gained support for 10 bit video SMPTE timecodes, and
- generally got many bugfixes for various issues.
-- New API in `GstPlayer` for setting the multiview mode for stereoscopic
- video, setting an HTTP/RTSP user agent and a time offset between audio and
- video. In addition to that, there were various bugfixes and the new
- gst-examples module contains Android, iOS, GTK+ and Qt example applications.
-- `GstBin` has new API for suppressing various `GstElement` or `GstObject`
- flags that would otherwise be affected by added/removed child elements. This
- new API allows `GstBin` subclasses to handle for themselves if they
- should be considered a sink or source element, for example.
-- The `subparse` element can handle WebVTT streams now.
-- A new `sdpsrc` element was added that can read an SDP from a file, or get it
- as a string as property and then sets up an RTP pipeline accordingly.
-
-### Plugin moves
-
-No plugins were moved this cycle. We'll make up for it next cycle, promise!
-
-### Rewritten memory leak tracer
-
-GStreamer has had basic functionality to trace allocation and freeing of
-both mini-objects (buffers, events, caps, etc.) and objects in the form of the
-internal `GstAllocTrace` tracing system. This API was never exposed in the
-1.x API series though. When requested, this would dump a list of objects and
-mini-objects at exit time which had still not been freed at that point,
-enabled with an environment variable. This subsystem has now been removed
-in favour of a new implementation based on the recently-added tracing framework.
-
-Tracing hooks have been added to trace the creation and destruction of
-GstObjects and mini-objects, and a new tracer plugin has been written using
-those new hooks to track which objects are still live and which are not. If
-GStreamer has been compiled against the libunwind library, the new leaks tracer
-will remember where objects were allocated from as well. By default the leaks
-tracer will simply output a warning if leaks have been detected on `gst_deinit()`.
-
-If the `GST_LEAKS_TRACER_SIG` environment variable is set, the leaks tracer
-will also handle the following UNIX signals:
-
- - `SIGUSR1`: log alive objects
- - `SIGUSR2`: create a checkpoint and print a list of objects created and
- destroyed since the previous checkpoint.
-
-Unfortunately this will not work on Windows due to no signals, however.
-
-If the `GST_LEAKS_TRACER_STACK_TRACE` environment variable is set, the leaks
-tracer will also log the creation stack trace of leaked objects. This may
-significantly increase memory consumption however.
-
-New `MAY_BE_LEAKED` flags have been added to GstObject and GstMiniObject, so
-that objects and mini-objects that are likely to stay around forever can be
-flagged and blacklisted from the leak output.
-
-To give the new leak tracer a spin, simply call any GStreamer application such
-as `gst-launch-1.0` or `gst-play-1.0` like this:
-
- GST_TRACERS=leaks gst-launch-1.0 videotestsrc num-buffers=10 ! fakesink
-
-If there are any leaks, a warning will be raised at the end.
-
-It is also possible to trace only certain types of objects or mini-objects:
-
- GST_TRACERS="leaks(GstEvent,GstMessage)" gst-launch-1.0 videotestsrc num-buffers=10 ! fakesink
-
-This dedicated leaks tracer is much much faster than valgrind since all code is
-executed natively instead of being instrumented. This makes it very suitable
-for use on slow machines or embedded devices. It is however limited to certain
-types of leaks and won't catch memory leaks when the allocation has been made
-via plain old `malloc()` or `g_malloc()` or other means. It will also not trace
-non-GstObject GObjects.
-
-The goal is to enable leak tracing on GStreamer's Continuous-Integration and
-testing system, both for the regular unit tests (make check) and media tests
-(gst-validate), so that accidental leaks in common code paths can be detected
-and fixed quickly.
-
-For more information about the new tracer, check out Guillaume Desmottes's
-["Tracking Memory Leaks"][leaks-talk] talk or his [blog post][leaks-blog] about
-the topic.
-
-[leaks-talk]: https://gstconf.ubicast.tv/videos/tracking-memory-leaks/
-[leaks-blog]: https://blog.desmottes.be/?post/2016/06/20/GStreamer-leaks-tracer
-
-### GES and NLE changes
-
-- Clip priorities are now handled by the layers, and the GESTimelineElement
- priority property is now deprecated and unused
-- Enhanced (de)interlacing support to always use the `deinterlace` element
- and expose needed properties to users
-- Allow reusing clips children after removing the clip from a layer
-- We are now testing many more rendering formats in the gst-validate
- test suite, and failures have been fixed.
-- Also many bugs have been fixed in this cycle!
-
-### GStreamer validate changes
-
-This cycle has been focused on making GstValidate more than just a validating
-tool, but also a tool to help developers debug their GStreamer issues. When
-reporting issues, we try to gather as much information as possible and expose
-it to end users in a useful way. For an example of such enhancements, check out
-Thibault Saunier's [blog post](improving-debugging-gstreamer-validate) about
-the new Not Negotiated Error reporting mechanism.
-
-Playbin3 support has been added so we can run validate tests with `playbin3`
-instead of playbin.
-
-We are now able to properly communicate between `gst-validate-launcher` and
-launched subprocesses with actual IPC between them. That has enabled the test
-launcher to handle failing tests specifying the exact expected issue(s).
-
-[improving-debugging-gstreamer-validate]: https://blogs.s-osg.org/improving-debugging-gstreamer-validate/
-
-### gst-libav changes
-
-gst-libav uses the recently released ffmpeg 3.2 now, which brings a lot of
-improvements and bugfixes from the ffmpeg team in addition to various new
-codec mappings on the GStreamer side and quite a few bugfixes to the GStreamer
-integration to make it more robust.
-
-## Build and Dependencies
-
-### Experimental support for Meson as build system
-
-#### Overview
-
-We have have added support for building GStreamer using the
-[Meson build system][meson]. This is currently experimental, but should work
-fine at least on Linux using the gcc or clang toolchains and on Windows using
-the MingW or MSVC toolchains.
-
-Autotools remains the primary build system for the time being, but we hope to
-someday replace it and will steadily work towards that goal.
-
-More information about the background and implications of all this and where
-we're hoping to go in future with this can be found in [Tim's mail][meson-mail]
-to the gstreamer-devel mailing list.
-
-For more information on Meson check out [these videos][meson-videos] and also
-the [Meson talk][meson-gstconf] at the GStreamer Conference.
-
-Immediate benefits for Linux users are faster builds and rebuilds. At the time
-of writing the Meson build of GStreamer is used by default in GNOME's jhbuild
-system.
-
-The Meson build currently still lacks many of the fine-grained configuration
-options to enable/disable specific plugins. These will be added back in due
-course.
-
-Note: The meson build files are not distributed in the source tarballs, you will
-need to get GStreamer from git if you want try it out.
-
-[meson]: http://mesonbuild.com/
-[meson-mail]: https://lists.freedesktop.org/archives/gstreamer-devel/2016-September/060231.html
-[meson-videos]: http://mesonbuild.com/videos.html
-[meson-gstconf]: https://gstconf.ubicast.tv/videos/gstreamer-development-on-windows-ans-faster-builds-everywhere-with-meson/
-
-#### Windows Visual Studio toolchain support
-
-Windows users might appreciate being able to build GStreamer using the MSVC
-toolchain, which is not possible using autotools. This means that it will be
-possible to debug GStreamer and applications in Visual Studio, for example.
-We require VS2015 or newer for this at the moment.
-
-There are two ways to build GStreamer using the MSVC toolchain:
-
-1. Using the MSVC command-line tools (`cl.exe` etc.) via Meson's "ninja" backend.
-2. Letting Meson's "vs2015" backend generate Visual Studio project files that
- can be opened in Visual Studio and compiled from there.
-
-This is currently only for adventurous souls though. All the bits are in place,
-but support for all of this has not been merged into GStreamer's cerbero build
-tool yet at the time of writing. This will hopefully happen in the next cycle,
-but for now this means that those wishing to compile GStreamer with MSVC will
-have to get their hands dirty.
-
-There are also no binary SDK builds using the MSVC toolchain yet.
-
-For more information on GStreamer builds using Meson and the Windows toolchain
-check out Nirbheek Chauhan's blog post ["Building and developing GStreamer using Visual Studio"][msvc-blog].
-
-[msvc-blog]: http://blog.nirbheek.in/2016/07/building-and-developing-gstreamer-using.html
-
-### Dependencies
-
-#### gstreamer
-
-libunwind was added as an optional dependency. It is used only for debugging
-and tracing purposes.
-
-The `opencv` plugin in gst-plugins-bad can now be built against OpenCV
-version 3.1, previously only 2.3-2.5 were supported.
-
-#### gst-plugins-ugly
-
-- `mpeg2dec` now requires at least libmpeg2 0.5.1 (from 2008).
-
-#### gst-plugins-bad
-
-- `gltransformation` now requires at least graphene 1.4.0.
-
-- `lv2` now plugin requires at least lilv 0.16 instead of slv2.
-
-### Packaging notes
-
-Packagers please note that the `gst/gstconfig.h` public header file in the
-GStreamer core library moved back from being an architecture dependent include
-to being architecture independent, and thus it is no longer installed into
-`$(libdir)/gstreamer-1.0/include/gst` but into the normal include directory
-where it lives happily ever after with all the other public header files. The
-reason for this is that we now check whether the target supports unaligned
-memory access based on predefined compiler macros at compile time instead of
-checking it at configure time.
-
-## Platform-specific improvements
-
-### Android
-
-#### New universal binaries for all supported ABIs
-
-We now provide a "universal" tarball to allow building apps against all the
-architectures currently supported (x86, x86-64, armeabi, armeabi-v7a,
-armeabi-v8a). This is needed for building with recent versions of the Android
-NDK which defaults to building against all supported ABIs. Use [the Android
-player example][android-player-example-build] as a reference for the required
-changes.
-
-[android-player-example-build]: https://cgit.freedesktop.org/gstreamer/gst-examples/commit/playback/player/android?id=a5cdde9119f038a1eb365aca20faa9741a38e788
-
-#### Miscellaneous
-
-- New `ahssrc` element that allows reading the hardware sensors, e.g. compass
- or accelerometer.
-
-### macOS (OS/X) and iOS
-
-- Support for querying available devices on OS/X via the GstDeviceProvider
- API was added.
-- It is now possible to create OpenGL|ES 3.x contexts on iOS and use them in
- combination with the VideoToolbox based decoder element.
-- many OpenGL/GLES improvements, see OpenGL section above
-
-### Windows
-
-- gstconfig.h: Always use dllexport/import on Windows with MSVC
-- Miscellaneous fixes to make libs and plugins compile with the MVSC toolchain
-- MSVC toolchain support (see Meson section above for more details)
-
-## New Modules for Documentation, Examples, Meson Build
-
-Three new git modules have been added recently:
-
-### gst-docs
-
-This is a new module where we will maintain documentation in the markdown
-format.
-
-It contains the former gstreamer.com SDK tutorials which have kindly been made
-available by Fluendo under a Creative Commons license. The tutorials have been
-reviewed and updated for GStreamer 1.x and will be available as part of the
-[official GStreamer documentation][doc] going forward. The old gstreamer.com
-site will then be shut down with redirects pointing to the updated tutorials.
-
-Some of the existing docbook XML-formatted documentation from the GStreamer
-core module such as the *Application Development Manual* and the *Plugin
-Writer's Guide* have been converted to markdown as well and will be maintained
-in the gst-docs module in future. They will be removed from the GStreamer core
-module in the next cycle.
-
-This is just the beginning. Our goal is to provide a more cohesive documentation
-experience for our users going forward, and easier to create and maintain
-documentation for developers. There is a lot more work to do, get in touch if
-you want to help out.
-
-If you encounter any problems or spot any omissions or outdated content in the
-new documentation, please [file a bug in bugzilla][doc-bug] to let us know.
-
-We will probably release gst-docs as a separate tarball for distributions to
-package in the next cycle.
-
-[doc]: http://gstreamer.freedesktop.org/documentation/
-[doc-bug]: https://bugzilla.gnome.org/enter_bug.cgi?product=GStreamer&component=documentation
-
-### gst-examples
-
-A new [module][examples-git] has been added for examples. It does not contain
-much yet, currently it only contains a small [http-launch][http-launch] utility
-that serves a pipeline over http as well as various [GstPlayer playback frontends][puis]
-for Android, iOS, Gtk+ and Qt.
-
-More examples will be added over time. The examples in this repository should
-be more useful and more substantial than most of the examples we ship as part
-of our other modules, and also written in a way that makes them good example
-code. If you have ideas for examples, let us know.
-
-No decision has been made yet if this module will be released and/or packaged.
-It probably makes sense to do so though.
-
-[examples-git]: https://cgit.freedesktop.org/gstreamer/gst-examples/tree/
-[http-launch]: https://cgit.freedesktop.org/gstreamer/gst-examples/tree/network/http-launch/
-[puis]: https://cgit.freedesktop.org/gstreamer/gst-examples/tree/playback/player
-
-### gst-build
-
-[gst-build][gst-build-git] is a new meta module to build GStreamer using the
-new Meson build system. This module is not required to build GStreamer with
-Meson, it is merely for convenience and aims to provide a development setup
-similar to the existing `gst-uninstalled` setup.
-
-gst-build makes use of Meson's [subproject feature][meson-subprojects] and sets
-up the various GStreamer modules as subprojects, so they can all be updated and
-built in parallel.
-
-This module is still very new and highly experimental. It should work at least
-on Linux and Windows (OS/X needs some build fixes). Let us know of any issues
-you encounter by popping into the `#gstreamer` IRC channel or by
-[filing a bug][gst-build-bug].
-
-This module will probably not be released or packaged (does not really make sense).
-
-[gst-build-git]: https://cgit.freedesktop.org/gstreamer/gst-build/tree/
-[gst-build-bug]: https://bugzilla.gnome.org/enter_bug.cgi?product=GStreamer&component=gst-build
-[meson-subprojects]: https://github.com/mesonbuild/meson/wiki/Subprojects
-
-## Contributors
-
-Aaron Boxer, Aleix Conchillo Flaqué, Alessandro Decina, Alexandru Băluț, Alex
-Ashley, Alex-P. Natsios, Alistair Buxton, Allen Zhang, Andreas Naumann, Andrew
-Eikum, Andy Devar, Anthony G. Basile, Arjen Veenhuizen, Arnaud Vrac, Artem
-Martynovich, Arun Raghavan, Aurélien Zanelli, Barun Kumar Singh, Bernhard
-Miller, Brad Lackey, Branko Subasic, Carlos Garcia Campos, Carlos Rafael
-Giani, Christoffer Stengren, Daiki Ueno, Damian Ziobro, Danilo Cesar Lemes de
-Paula, David Buchmann, Dimitrios Katsaros, Duncan Palmer, Edward Hervey,
-Emmanuel Poitier, Enrico Jorns, Enrique Ocaña González, Fabrice Bellet,
-Florian Zwoch, Florin Apostol, Francisco Velazquez, Frédéric Bertolus, Fredrik
-Fornwall, Gaurav Gupta, George Kiagiadakis, Georg Lippitsch, Göran Jönsson,
-Graham Leggett, Gregoire Gentil, Guillaume Desmottes, Gwang Yoon Hwang, Haakon
-Sporsheim, Haihua Hu, Havard Graff, Heinrich Fink, Hoonhee Lee, Hyunjun Ko,
-Iain Lane, Ian, Ian Jamison, Jagyum Koo, Jake Foytik, Jakub Adam, Jan
-Alexander Steffens (heftig), Jan Schmidt, Javier Martinez Canillas, Jerome
-Laheurte, Jesper Larsen, Jie Jiang, Jihae Yi, Jimmy Ohn, Jinwoo Ahn, Joakim
-Johansson, Joan Pau Beltran, Jonas Holmberg, Jonathan Matthew, Jonathan Roy,
-Josep Torra, Julien Isorce, Jun Ji, Jürgen Slowack, Justin Kim, Kazunori
-Kobayashi, Kieran Bingham, Kipp Cannon, Koop Mast, Kouhei Sutou, Kseniia, Kyle
-Schwarz, Kyungyong Kim, Linus Svensson, Luis de Bethencourt, Marcin Kolny,
-Marcin Lewandowski, Marianna Smidth Buschle, Mario Sanchez Prada, Mark
-Combellack, Mark Nauwelaerts, Martin Kelly, Matej Knopp, Mathieu Duponchelle,
-Mats Lindestam, Matthew Gruenke, Matthew Waters, Michael Olbrich, Michal Lazo,
-Miguel París Díaz, Mikhail Fludkov, Minjae Kim, Mohan R, Munez, Nicola Murino,
-Nicolas Dufresne, Nicolas Huet, Nikita Bobkov, Nirbheek Chauhan, Olivier
-Crête, Paolo Pettinato, Patricia Muscalu, Paulo Neves, Peng Liu, Peter
-Seiderer, Philippe Normand, Philippe Renon, Philipp Zabel, Pierre Lamot, Piotr
-Drąg, Prashant Gotarne, Raffaele Rossi, Ray Strode, Reynaldo H. Verdejo
-Pinochet, Santiago Carot-Nemesio, Scott D Phillips, Sebastian Dröge, Sebastian
-Rasmussen, Sergei Saveliev, Sergey Borovkov, Sergey Mamonov, Sergio Torres
-Soldado, Seungha Yang, sezero, Song Bing, Sreerenj Balachandran, Stefan Sauer,
-Stephen, Steven Hoving, Stian Selnes, Thiago Santos, Thibault Saunier, Thijs
-Vermeir, Thomas Bluemel, Thomas Jones, Thomas Klausner, Thomas Scheuermann,
-Tim-Philipp Müller, Ting-Wei Lan, Tom Schoonjans, Ursula Maplehurst, Vanessa
-Chipirras Navalon, Víctor Manuel Jáquez Leal, Vincent Penquerc'h, Vineeth TM,
-Vivia Nikolaidou, Vootele Vesterblom, Wang Xin-yu (王昕宇), William Manley,
-Wim Taymans, Wonchul Lee, Xabier Rodriguez Calvar, Xavier Claessens, xlazom00,
-Yann Jouanin, Zaheer Abbas Merali
-
-... and many others who have contributed bug reports, translations, sent
-suggestions or helped testing.
-
-## Bugs fixed in 1.10
-
-More than [750 bugs][bugs-fixed-in-1.10] have been fixed during
-the development of 1.10.
-
-This list does not include issues that have been cherry-picked into the
-stable 1.8 branch and fixed there as well, all fixes that ended up in the
-1.8 branch are also included in 1.10.
-
-This list also does not include issues that have been fixed without a bug
-report in bugzilla, so the actual number of fixes is much higher.
-
-[bugs-fixed-in-1.10]: https://bugzilla.gnome.org/buglist.cgi?bug_status=RESOLVED&bug_status=VERIFIED&classification=Platform&limit=0&list_id=164074&order=bug_id&product=GStreamer&query_format=advanced&resolution=FIXED&target_milestone=1.8.1&target_milestone=1.8.2&target_milestone=1.8.3&target_milestone=1.8.4&target_milestone=1.9.1&target_milestone=1.9.2&target_milestone=1.9.90&target_milestone=1.10.0
-
-## Stable 1.10 branch
-
-After the 1.10.0 release there will be several 1.10.x bug-fix releases which
-will contain bug fixes which have been deemed suitable for a stable branch,
-but no new features or intrusive changes will be added to a bug-fix release
-usually. The 1.10.x bug-fix releases will be made from the git 1.10 branch,
-which is a stable branch.
-
-### 1.10.0
-
-1.10.0 was released on 1st November 2016.
-
-## Known Issues
-
-- iOS builds with iOS 6 SDK and old C++ STL. You need to select iOS 6 instead
- of 7 or 8 in your projects settings to be able to link applications.
- [Bug #766366](https://bugzilla.gnome.org/show_bug.cgi?id=766366)
-- Code signing for Apple platforms has some problems currently, requiring
- manual work to get your application signed. [Bug #771860](https://bugzilla.gnome.org/show_bug.cgi?id=771860)
-- Building applications with Android NDK r13 on Windows does not work. Other
- platforms and earlier/later versions of the NDK are not affected.
- [Bug #772842](https://bugzilla.gnome.org/show_bug.cgi?id=772842)
-- The new leaks tracer may deadlock the application (or exhibit other undefined
- behaviour) when `SIGUSR` handling is enabled via the `GST_LEAKS_TRACER_SIG`
- environment variable. [Bug #770373](https://bugzilla.gnome.org/show_bug.cgi?id=770373)
-- vp8enc crashes on 32 bit Windows, but was working fine in 1.6. 64 bit Windows is unaffected.
- [Bug #763663](https://bugzilla.gnome.org/show_bug.cgi?id=763663)
-
-## Schedule for 1.12
-
-Our next major feature release will be 1.12, and 1.11 will be the unstable
-development version leading up to the stable 1.12 release. The development
-of 1.11/1.12 will happen in the git master branch.
-
-The plan for the 1.12 development cycle is yet to be confirmed, but it is
-expected that feature freeze will be around early/mid-January,
-followed by several 1.11 pre-releases and the new 1.12 stable release
-in March.
-
-1.12 will be backwards-compatible to the stable 1.10, 1.8, 1.6, 1.4, 1.2 and
-1.0 release series.
-
-- - -
-
-*These release notes have been prepared by Olivier Crête, Sebastian Dröge,
-Nicolas Dufresne, Edward Hervey, Víctor Manuel Jáquez Leal, Tim-Philipp
-Müller, Reynaldo H. Verdejo Pinochet, Arun Raghavan, Thibault Saunier,
-Jan Schmidt, Wim Taymans, Matthew Waters*
-
-*License: [CC BY-SA 4.0](http://creativecommons.org/licenses/by-sa/4.0/)*
-
+This is GStreamer 1.11.1.
-Release notes for GStreamer Good Plugins 1.10.0
+Release notes for GStreamer Good Plugins 1.11.1
-The GStreamer team is pleased to announce the first release of the new stable
-1.10 release series. The 1.10 release series is adding new features on top of
-the 1.0, 1.2, 1.4, 1.6 and 1.8 series and is part of the API and ABI-stable 1.x
-release series of the GStreamer multimedia framework.
+The GStreamer team is pleased to announce the first release of the unstable
+1.11 release series. The 1.11 release series is adding new features on top of
+the 1.0, 1.2, 1.4, 1.6, 1.8 and 1.10 series and is part of the API and ABI-stable 1.x release
+series of the GStreamer multimedia framework. The unstable 1.11 release series
+will lead to the stable 1.12 release series in the next weeks. Any newly added
+API can still change until that point.
-Binaries for Android, iOS, Mac OS X and Windows will be provided shortly after
-the source release by the GStreamer project during the stable 1.10 release
-series.
+Full release notes will be provided at some point during the 1.11 release
+cycle, highlighting all the new features, bugfixes, performance optimizations
+and other important changes.
+
+
+Binaries for Android, iOS, Mac OS X and Windows will be provided in the next days.
"Such ingratitude. After all the times I've saved your life."
Bugs fixed in this release
- * 762207 : flvmux: Ensure we fallback to DTS when clipping
- * 772496 : tests: Fix memory leak by gst_caps_to_string()
- * 772497 : waveform : Fix Memory leak by gst_caps_to_string
- * 772644 : Fix level test in CK_FORK=no mode
- * 772656 : Fix souphttpsrc tests without CK_FORK=no
- * 773509 : souphttpsrc: connection loss / reconnect issues
- * 773580 : v4l2object: fix extra-controls leak
- * 773582 : matroskamux does not allow resolutions above 4096x4096
- * 773643 : wavparse: crashes on invalid wav file
+ * 708221 : mp4dashmux: add the tfdt atom to the moof
+ * 746574 : matroskamux: add G722 audio support
+ * 748360 : rtspsrc: teardown usually never happens
+ * 749098 : matroskamux: drop streamheader buffers only if they really are headers
+ * 754696 : matroskamux: audio-only streams have all buffers flagged as delta units, causing problems with tcpserversink/multifdsink
+ * 757631 : progressreport format=bytes will not send msg
+ * 766991 : multifilesink: leaks memory when max-files property == 0
+ * 767771 : qtdemux/jpegdec: Interlaced content detected as progressive
+ * 768723 : rtprtx: test is sometimes failing
+ * 769041 : qtmux: Downscaling time value loses precision
+ * 769048 : qtmux: prores-related fixes
+ * 772181 : isomp4: Parse/store colorimetry, chroma-site and interlaced-mode/field-order
+ * 772740 : rtpbin: receiving RTP bundle support
+ * 773217 : qtmux: Allow configuring the maximum interleave size in bytes/time
+ * 773514 : rtph263pay: Use GST_TRACE for logging bitsream parsing
+ * 773712 : isomp4: Add support for FLAC
+ * 773785 : splitmuxsink: Use first buffer TS as mux start time
+ * 773828 : qtmux: Crash on EOS with GST_DEBUG enabled
+ * 774129 : 'gst_buffer_is_writable' assertion in aacparse
+ * 774131 : flvmux: Add metadatacreator property
+ * 774403 : qtmux: Always write edit lists for the tracks to give a more accurate duration
+ * 774409 : tests/jitterbuffer: Major refactoring and cleanups
+ * 774566 : matroskaparse: error out on last buffer
+ * 774674 : qtdemux: Remove useless return variable
+ * 774747 : qtdemux: compiler warning with gcc 6.2
+ * 774789 : qtmux: Enable up to 16 unpositioned raw audio channels
+ * 774840 : qtmux: Fix various timestamp and duration related issues
+ * 774876 : meson: add libm to has_function checks
+ * 775287 : qtdemux: change off_t type to gint
+ * 775414 : qtdemux: Correctly read interlacing information
+ * 775702 : v4l2object: Don't set empty interlace-mode list
+ * 775752 : monoscope: Leaks allocation query
+ * 776030 : udpsrc: Add to join multiple multicast interfaces
+ * 776106 : v4l2object: Don't check size in a non-list value
+ * 776789 : avidemux: fix memory leak in usage of gst_pad_template_new() API
+ * 777095 : isomp4: Don't spam debug log with knonw/padding atoms
+ * 777157 : qtdemux: seqh buffer not freed after calling qtdemux_parse_svq3_stsd_data()
==== Download ====
Contributors to this release
- * Branko Subasic
- * Gaurav Gupta
- * Jan Alexander Steffens (heftig)
+ * Aleix Conchillo Flaque
+ * Alejandro G. Castro
+ * Andre McCurdy
+ * Arun Raghavan
+ * David Evans
+ * Edward Hervey
+ * Enrique Ocaña González
+ * Garima Gaur
+ * Havard Graff
+ * Heekyoung Seo
+ * Jagadish
* Jan Schmidt
* Mark Nauwelaerts
- * Michael Olbrich
- * Nicolas Dufresne
+ * Matt Staples
+ * Matthew Waters
+ * Nicola Murino
* Nirbheek Chauhan
+ * Petr Kulhavy
+ * Philipp Zabel
+ * Philippe Normand
+ * Reynaldo H. Verdejo Pinochet
* Scott D Phillips
+ * Sean DuBois
* Sebastian Dröge
+ * Seungha Yang
+ * Stian Selnes
* Thibault Saunier
* Tim-Philipp Müller
- * Tobias Schneider
+ * Ursula Maplehurst
+ * Vincent Penquerc'h
+ * Vinod Kesti
+ * Vivia Nikolaidou
+ * Víctor Manuel Jáquez Leal
+ * William Manley
+ * Wonchul Lee
+ * christophecvr
\ No newline at end of file
dnl initialize autoconf
dnl releases only do -Wall, git and prerelease does -Werror too
dnl use a three digit version number for releases, and four for git/pre
-AC_INIT([GStreamer Good Plug-ins],[1.11.0.1],[http://bugzilla.gnome.org/enter_bug.cgi?product=GStreamer],[gst-plugins-good])
+AC_INIT([GStreamer Good Plug-ins],[1.11.1],[http://bugzilla.gnome.org/enter_bug.cgi?product=GStreamer],[gst-plugins-good])
AG_GST_INIT
[GStreamer API Version])
AG_GST_LIBTOOL_PREPARE
-AS_LIBTOOL(GST, 1100, 0, 1100)
+AS_LIBTOOL(GST, 1101, 0, 1101)
dnl *** required versions of GStreamer stuff ***
-GST_REQ=1.11.0.1
-GSTPB_REQ=1.11.0.1
+GST_REQ=1.11.1
+GSTPB_REQ=1.11.1
dnl *** autotools stuff ****
<RANGE></RANGE>
<FLAGS>rw</FLAGS>
<NICK>Multicast Interface</NICK>
-<BLURB>The network interface on which to join the multicast group.</BLURB>
+<BLURB>The network interface on which to join the multicast group.This allows multiple interfaces seperated by comma. ("eth0,eth1").</BLURB>
<DEFAULT>NULL</DEFAULT>
</ARG>
<FLAGS>rw</FLAGS>
<NICK>User Agent</NICK>
<BLURB>The User-Agent string to send to the server.</BLURB>
-<DEFAULT>"GStreamer/1.10.0"</DEFAULT>
+<DEFAULT>"GStreamer/1.11.1"</DEFAULT>
</ARG>
<ARG>
<FLAGS>rw</FLAGS>
<NICK>Client Name</NICK>
<BLURB>The PulseAudio client name to use.</BLURB>
-<DEFAULT>"lt-gst-plugins-good-plugins-scan"</DEFAULT>
+<DEFAULT>"gst-plugins-good-plugins-scan"</DEFAULT>
</ARG>
<ARG>
<FLAGS>rw</FLAGS>
<NICK>Client Name</NICK>
<BLURB>The PulseAudio client_name_to_use.</BLURB>
-<DEFAULT>"lt-gst-plugins-good-plugins-scan"</DEFAULT>
+<DEFAULT>"gst-plugins-good-plugins-scan"</DEFAULT>
</ARG>
<ARG>
</ARG>
<ARG>
+<NAME>GstFlvMux::metadatacreator</NAME>
+<TYPE>gchar*</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>metadatacreator</NICK>
+<BLURB>The value of metadatacreator in the meta packet.</BLURB>
+<DEFAULT>NULL</DEFAULT>
+</ARG>
+
+<ARG>
<NAME>GstCapsSetter::caps</NAME>
<TYPE>GstCaps*</TYPE>
<RANGE></RANGE>
<ARG>
<NAME>GstQTMux::movie-timescale</NAME>
<TYPE>guint</TYPE>
-<RANGE>>= 1</RANGE>
+<RANGE></RANGE>
<FLAGS>rwx</FLAGS>
<NICK>Movie timescale</NICK>
-<BLURB>Timescale to use in the movie (units per second).</BLURB>
-<DEFAULT>1800</DEFAULT>
+<BLURB>Timescale to use in the movie (units per second, 0 == default).</BLURB>
+<DEFAULT>0</DEFAULT>
</ARG>
<ARG>
</ARG>
<ARG>
+<NAME>GstQTMux::interleave-bytes</NAME>
+<TYPE>guint64</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Interleave (bytes)</NICK>
+<BLURB>Interleave between streams in bytes.</BLURB>
+<DEFAULT>0</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstQTMux::interleave-time</NAME>
+<TYPE>guint64</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Interleave (time)</NICK>
+<BLURB>Interleave between streams in nanoseconds.</BLURB>
+<DEFAULT>250000000</DEFAULT>
+</ARG>
+
+<ARG>
<NAME>GstQTMoovRecover::broken-input</NAME>
<TYPE>gchar*</TYPE>
<RANGE></RANGE>
<ARG>
<NAME>GstMP4Mux::movie-timescale</NAME>
<TYPE>guint</TYPE>
-<RANGE>>= 1</RANGE>
+<RANGE></RANGE>
<FLAGS>rwx</FLAGS>
<NICK>Movie timescale</NICK>
-<BLURB>Timescale to use in the movie (units per second).</BLURB>
-<DEFAULT>1800</DEFAULT>
+<BLURB>Timescale to use in the movie (units per second, 0 == default).</BLURB>
+<DEFAULT>0</DEFAULT>
</ARG>
<ARG>
</ARG>
<ARG>
+<NAME>GstMP4Mux::interleave-bytes</NAME>
+<TYPE>guint64</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Interleave (bytes)</NICK>
+<BLURB>Interleave between streams in bytes.</BLURB>
+<DEFAULT>0</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstMP4Mux::interleave-time</NAME>
+<TYPE>guint64</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Interleave (time)</NICK>
+<BLURB>Interleave between streams in nanoseconds.</BLURB>
+<DEFAULT>250000000</DEFAULT>
+</ARG>
+
+<ARG>
<NAME>GstMJ2Mux::dts-method</NAME>
<TYPE>GstQTMuxDtsMethods</TYPE>
<RANGE></RANGE>
<ARG>
<NAME>GstMJ2Mux::movie-timescale</NAME>
<TYPE>guint</TYPE>
-<RANGE>>= 1</RANGE>
+<RANGE></RANGE>
<FLAGS>rwx</FLAGS>
<NICK>Movie timescale</NICK>
-<BLURB>Timescale to use in the movie (units per second).</BLURB>
-<DEFAULT>1800</DEFAULT>
+<BLURB>Timescale to use in the movie (units per second, 0 == default).</BLURB>
+<DEFAULT>0</DEFAULT>
</ARG>
<ARG>
</ARG>
<ARG>
+<NAME>GstMJ2Mux::interleave-bytes</NAME>
+<TYPE>guint64</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Interleave (bytes)</NICK>
+<BLURB>Interleave between streams in bytes.</BLURB>
+<DEFAULT>0</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstMJ2Mux::interleave-time</NAME>
+<TYPE>guint64</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Interleave (time)</NICK>
+<BLURB>Interleave between streams in nanoseconds.</BLURB>
+<DEFAULT>250000000</DEFAULT>
+</ARG>
+
+<ARG>
<NAME>GstISMLMux::dts-method</NAME>
<TYPE>GstQTMuxDtsMethods</TYPE>
<RANGE></RANGE>
<ARG>
<NAME>GstISMLMux::movie-timescale</NAME>
<TYPE>guint</TYPE>
-<RANGE>>= 1</RANGE>
+<RANGE></RANGE>
<FLAGS>rwx</FLAGS>
<NICK>Movie timescale</NICK>
-<BLURB>Timescale to use in the movie (units per second).</BLURB>
-<DEFAULT>1800</DEFAULT>
+<BLURB>Timescale to use in the movie (units per second, 0 == default).</BLURB>
+<DEFAULT>0</DEFAULT>
</ARG>
<ARG>
</ARG>
<ARG>
+<NAME>GstISMLMux::interleave-bytes</NAME>
+<TYPE>guint64</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Interleave (bytes)</NICK>
+<BLURB>Interleave between streams in bytes.</BLURB>
+<DEFAULT>0</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>GstISMLMux::interleave-time</NAME>
+<TYPE>guint64</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Interleave (time)</NICK>
+<BLURB>Interleave between streams in nanoseconds.</BLURB>
+<DEFAULT>250000000</DEFAULT>
+</ARG>
+
+<ARG>
<NAME>Gst3GPPMux::dts-method</NAME>
<TYPE>GstQTMuxDtsMethods</TYPE>
<RANGE></RANGE>
<ARG>
<NAME>Gst3GPPMux::movie-timescale</NAME>
<TYPE>guint</TYPE>
-<RANGE>>= 1</RANGE>
+<RANGE></RANGE>
<FLAGS>rwx</FLAGS>
<NICK>Movie timescale</NICK>
-<BLURB>Timescale to use in the movie (units per second).</BLURB>
-<DEFAULT>1800</DEFAULT>
+<BLURB>Timescale to use in the movie (units per second, 0 == default).</BLURB>
+<DEFAULT>0</DEFAULT>
</ARG>
<ARG>
</ARG>
<ARG>
+<NAME>Gst3GPPMux::interleave-bytes</NAME>
+<TYPE>guint64</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Interleave (bytes)</NICK>
+<BLURB>Interleave between streams in bytes.</BLURB>
+<DEFAULT>0</DEFAULT>
+</ARG>
+
+<ARG>
+<NAME>Gst3GPPMux::interleave-time</NAME>
+<TYPE>guint64</TYPE>
+<RANGE></RANGE>
+<FLAGS>rw</FLAGS>
+<NICK>Interleave (time)</NICK>
+<BLURB>Interleave between streams in nanoseconds.</BLURB>
+<DEFAULT>250000000</DEFAULT>
+</ARG>
+
+<ARG>
<NAME>GstSplitFileSrc::location</NAME>
<TYPE>gchar*</TYPE>
<RANGE></RANGE>
GstPlugin
GstPluginFeature
GstDeviceProviderFactory
+ GstDynamicTypeFactory
GstElementFactory
GstTracerFactory
GstTypeFindFactory
</SIGNAL>
<SIGNAL>
+<NAME>GstSplitMuxSink::format-location-full</NAME>
+<RETURNS>gchar*</RETURNS>
+<FLAGS>l</FLAGS>
+GstSplitMuxSink *gstsplitmuxsink
+guint arg1
+GstSample *arg2
+</SIGNAL>
+
+<SIGNAL>
<NAME>GstSplitMuxSrc::format-location</NAME>
<RETURNS>GStrv</RETURNS>
<FLAGS>l</FLAGS>
<description>Source for video data via IEEE1394 interface</description>
<filename>../../ext/raw1394/.libs/libgst1394.so</filename>
<basename>libgst1394.so</basename>
- <version>1.10.0</version>
+ <version>1.11.1</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>ASCII Art video sink</description>
<filename>../../ext/aalib/.libs/libgstaasink.so</filename>
<basename>libgstaasink.so</basename>
- <version>1.10.0</version>
+ <version>1.11.1</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>ALaw audio conversion routines</description>
<filename>../../gst/law/.libs/libgstalaw.so</filename>
<basename>libgstalaw.so</basename>
- <version>1.10.0</version>
+ <version>1.11.1</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>adds an alpha channel to video - constant or via chroma-keying</description>
<filename>../../gst/alpha/.libs/libgstalpha.so</filename>
<basename>libgstalpha.so</basename>
- <version>1.10.0</version>
+ <version>1.11.1</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>RGBA from/to AYUV colorspace conversion preserving the alpha channel</description>
<filename>../../gst/alpha/.libs/libgstalphacolor.so</filename>
<basename>libgstalphacolor.so</basename>
- <version>1.10.0</version>
+ <version>1.11.1</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>APEv1/2 tag reader</description>
<filename>../../gst/apetag/.libs/libgstapetag.so</filename>
<basename>libgstapetag.so</basename>
- <version>1.10.0</version>
+ <version>1.11.1</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>Audio effects plugin</description>
<filename>../../gst/audiofx/.libs/libgstaudiofx.so</filename>
<basename>libgstaudiofx.so</basename>
- <version>1.10.0</version>
+ <version>1.11.1</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>Parsers for various audio formats</description>
<filename>../../gst/audioparsers/.libs/libgstaudioparsers.so</filename>
<basename>libgstaudioparsers.so</basename>
- <version>1.10.0</version>
+ <version>1.11.1</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>parses au streams</description>
<filename>../../gst/auparse/.libs/libgstauparse.so</filename>
<basename>libgstauparse.so</basename>
- <version>1.10.0</version>
+ <version>1.11.1</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>Plugin contains auto-detection plugins for video/audio in- and outputs</description>
<filename>../../gst/autodetect/.libs/libgstautodetect.so</filename>
<basename>libgstautodetect.so</basename>
- <version>1.10.0</version>
+ <version>1.11.1</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>AVI stream handling</description>
<filename>../../gst/avi/.libs/libgstavi.so</filename>
<basename>libgstavi.so</basename>
- <version>1.10.0</version>
+ <version>1.11.1</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>Colored ASCII Art video sink</description>
<filename>../../ext/libcaca/.libs/libgstcacasink.so</filename>
<basename>libgstcacasink.so</basename>
- <version>1.10.0</version>
+ <version>1.11.1</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>Cairo-based elements</description>
<filename>../../ext/cairo/.libs/libgstcairo.so</filename>
<basename>libgstcairo.so</basename>
- <version>1.10.0</version>
+ <version>1.11.1</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>Audio Cutter to split audio into non-silent bits</description>
<filename>../../gst/cutter/.libs/libgstcutter.so</filename>
<basename>libgstcutter.so</basename>
- <version>1.10.0</version>
+ <version>1.11.1</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>elements for testing and debugging</description>
<filename>../../gst/debugutils/.libs/libgstdebug.so</filename>
<basename>libgstdebug.so</basename>
- <version>1.10.0</version>
+ <version>1.11.1</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>Deinterlacer</description>
<filename>../../gst/deinterlace/.libs/libgstdeinterlace.so</filename>
<basename>libgstdeinterlace.so</basename>
- <version>1.10.0</version>
+ <version>1.11.1</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<name>sink</name>
<direction>sink</direction>
<presence>always</presence>
- <details>video/x-raw, format=(string){ AYUV, ARGB, ABGR, RGBA, BGRA, Y444, xRGB, xBGR, RGBx, BGRx, RGB, BGR, YUY2, YVYU, UYVY, Y42B, I420, YV12, Y41B, NV12, NV21 }, width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ], framerate=(fraction)[ 0/1, 2147483647/1 ]; video/x-raw(ANY), format=(string){ I420, YV12, YUY2, UYVY, AYUV, RGBx, BGRx, xRGB, xBGR, RGBA, BGRA, ARGB, ABGR, RGB, BGR, Y41B, Y42B, YVYU, Y444, v210, v216, NV12, NV21, NV16, NV61, NV24, GRAY8, GRAY16_BE, GRAY16_LE, v308, IYU2, RGB16, BGR16, RGB15, BGR15, UYVP, A420, RGB8P, YUV9, YVU9, IYU1, ARGB64, AYUV64, r210, I420_10LE, I420_10BE, I422_10LE, I422_10BE, Y444_10LE, Y444_10BE, GBR, GBR_10LE, GBR_10BE, NV12_64Z32, A420_10LE, A420_10BE, A422_10LE, A422_10BE, A444_10LE, A444_10BE, P010_10LE, P010_10BE }, width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ], framerate=(fraction)[ 0/1, 2147483647/1 ]</details>
+ <details>video/x-raw, format=(string){ AYUV, ARGB, ABGR, RGBA, BGRA, Y444, xRGB, xBGR, RGBx, BGRx, RGB, BGR, YUY2, YVYU, UYVY, Y42B, I420, YV12, Y41B, NV12, NV21 }, width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ], framerate=(fraction)[ 0/1, 2147483647/1 ]; video/x-raw(ANY), format=(string){ I420, YV12, YUY2, UYVY, VYUY, AYUV, RGBx, BGRx, xRGB, xBGR, RGBA, BGRA, ARGB, ABGR, RGB, BGR, Y41B, Y42B, YVYU, Y444, v210, v216, NV12, NV21, NV16, NV61, NV24, GRAY8, GRAY16_BE, GRAY16_LE, v308, IYU2, RGB16, BGR16, RGB15, BGR15, UYVP, A420, RGB8P, YUV9, YVU9, IYU1, ARGB64, AYUV64, r210, I420_10LE, I420_10BE, I422_10LE, I422_10BE, Y444_10LE, Y444_10BE, GBR, GBR_10LE, GBR_10BE, NV12_64Z32, A420_10LE, A420_10BE, A422_10LE, A422_10BE, A444_10LE, A444_10BE, P010_10LE, P010_10BE }, width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ], framerate=(fraction)[ 0/1, 2147483647/1 ]</details>
</caps>
<caps>
<name>src</name>
<direction>source</direction>
<presence>always</presence>
- <details>video/x-raw, format=(string){ AYUV, ARGB, ABGR, RGBA, BGRA, Y444, xRGB, xBGR, RGBx, BGRx, RGB, BGR, YUY2, YVYU, UYVY, Y42B, I420, YV12, Y41B, NV12, NV21 }, width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ], framerate=(fraction)[ 0/1, 2147483647/1 ]; video/x-raw(ANY), format=(string){ I420, YV12, YUY2, UYVY, AYUV, RGBx, BGRx, xRGB, xBGR, RGBA, BGRA, ARGB, ABGR, RGB, BGR, Y41B, Y42B, YVYU, Y444, v210, v216, NV12, NV21, NV16, NV61, NV24, GRAY8, GRAY16_BE, GRAY16_LE, v308, IYU2, RGB16, BGR16, RGB15, BGR15, UYVP, A420, RGB8P, YUV9, YVU9, IYU1, ARGB64, AYUV64, r210, I420_10LE, I420_10BE, I422_10LE, I422_10BE, Y444_10LE, Y444_10BE, GBR, GBR_10LE, GBR_10BE, NV12_64Z32, A420_10LE, A420_10BE, A422_10LE, A422_10BE, A444_10LE, A444_10BE, P010_10LE, P010_10BE }, width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ], framerate=(fraction)[ 0/1, 2147483647/1 ]</details>
+ <details>video/x-raw, format=(string){ AYUV, ARGB, ABGR, RGBA, BGRA, Y444, xRGB, xBGR, RGBx, BGRx, RGB, BGR, YUY2, YVYU, UYVY, Y42B, I420, YV12, Y41B, NV12, NV21 }, width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ], framerate=(fraction)[ 0/1, 2147483647/1 ]; video/x-raw(ANY), format=(string){ I420, YV12, YUY2, UYVY, VYUY, AYUV, RGBx, BGRx, xRGB, xBGR, RGBA, BGRA, ARGB, ABGR, RGB, BGR, Y41B, Y42B, YVYU, Y444, v210, v216, NV12, NV21, NV16, NV61, NV24, GRAY8, GRAY16_BE, GRAY16_LE, v308, IYU2, RGB16, BGR16, RGB15, BGR15, UYVP, A420, RGB8P, YUV9, YVU9, IYU1, ARGB64, AYUV64, r210, I420_10LE, I420_10BE, I422_10LE, I422_10BE, Y444_10LE, Y444_10BE, GBR, GBR_10LE, GBR_10BE, NV12_64Z32, A420_10LE, A420_10BE, A422_10LE, A422_10BE, A444_10LE, A444_10BE, P010_10LE, P010_10BE }, width=(int)[ 1, 2147483647 ], height=(int)[ 1, 2147483647 ], framerate=(fraction)[ 0/1, 2147483647/1 ]</details>
</caps>
</pads>
</element>
<description>DTMF plugins</description>
<filename>../../gst/dtmf/.libs/libgstdtmf.so</filename>
<basename>libgstdtmf.so</basename>
- <version>1.10.0</version>
+ <version>1.11.1</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>DV demuxer and decoder based on libdv (libdv.sf.net)</description>
<filename>../../ext/dv/.libs/libgstdv.so</filename>
<basename>libgstdv.so</basename>
- <version>1.10.0</version>
+ <version>1.11.1</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>effect plugins from the effectv project</description>
<filename>../../gst/effectv/.libs/libgsteffectv.so</filename>
<basename>libgsteffectv.so</basename>
- <version>1.10.0</version>
+ <version>1.11.1</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>GStreamer audio equalizers</description>
<filename>../../gst/equalizer/.libs/libgstequalizer.so</filename>
<basename>libgstequalizer.so</basename>
- <version>1.10.0</version>
+ <version>1.11.1</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>The FLAC Lossless compressor Codec</description>
<filename>../../ext/flac/.libs/libgstflac.so</filename>
<basename>libgstflac.so</basename>
- <version>1.10.0</version>
+ <version>1.11.1</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>FLV muxing and demuxing plugin</description>
<filename>../../gst/flv/.libs/libgstflv.so</filename>
<basename>libgstflv.so</basename>
- <version>1.10.0</version>
+ <version>1.11.1</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>FLC/FLI/FLX video decoder</description>
<filename>../../gst/flx/.libs/libgstflxdec.so</filename>
<basename>libgstflxdec.so</basename>
- <version>1.10.0</version>
+ <version>1.11.1</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>GdkPixbuf-based image decoder, overlay and sink</description>
<filename>../../ext/gdk_pixbuf/.libs/libgstgdkpixbuf.so</filename>
<basename>libgstgdkpixbuf.so</basename>
- <version>1.10.0</version>
+ <version>1.11.1</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>GOOM visualization filter</description>
<filename>../../gst/goom/.libs/libgstgoom.so</filename>
<basename>libgstgoom.so</basename>
- <version>1.10.0</version>
+ <version>1.11.1</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>GOOM 2k1 visualization filter</description>
<filename>../../gst/goom2k1/.libs/libgstgoom2k1.so</filename>
<basename>libgstgoom2k1.so</basename>
- <version>1.10.0</version>
+ <version>1.11.1</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>Demux ICY tags from a stream</description>
<filename>../../gst/icydemux/.libs/libgsticydemux.so</filename>
<basename>libgsticydemux.so</basename>
- <version>1.10.0</version>
+ <version>1.11.1</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>Demux ID3v1 and ID3v2 tags from a file</description>
<filename>../../gst/id3demux/.libs/libgstid3demux.so</filename>
<basename>libgstid3demux.so</basename>
- <version>1.10.0</version>
+ <version>1.11.1</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>Still frame stream generator</description>
<filename>../../gst/imagefreeze/.libs/libgstimagefreeze.so</filename>
<basename>libgstimagefreeze.so</basename>
- <version>1.10.0</version>
+ <version>1.11.1</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>Audio interleaver/deinterleaver</description>
<filename>../../gst/interleave/.libs/libgstinterleave.so</filename>
<basename>libgstinterleave.so</basename>
- <version>1.10.0</version>
+ <version>1.11.1</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>ISO base media file format support (mp4, 3gpp, qt, mj2)</description>
<filename>../../gst/isomp4/.libs/libgstisomp4.so</filename>
<basename>libgstisomp4.so</basename>
- <version>1.10.0</version>
+ <version>1.11.1</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<name>audio_%u</name>
<direction>sink</direction>
<presence>request</presence>
- <details>audio/x-raw, format=(string){ S32LE, S32BE, S24LE, S24BE, S16LE, S16BE, S8, U8 }, layout=(string)interleaved, channels=(int)[ 1, 2 ], rate=(int)[ 1, 2147483647 ]; audio/mpeg, mpegversion=(int)1, layer=(int)3, channels=(int)[ 1, 2 ], rate=(int)[ 1, 2147483647 ]; audio/mpeg, mpegversion=(int)4, stream-format=(string)raw, channels=(int)[ 1, 8 ], rate=(int)[ 1, 2147483647 ]; audio/x-adpcm, layout=(string)dvi, block_align=(int)[ 64, 8096 ], channels=(int)[ 1, 2 ], rate=(int)[ 1, 2147483647 ]; audio/x-alaw, channels=(int)[ 1, 2 ], rate=(int)[ 1, 2147483647 ]; audio/x-mulaw, channels=(int)[ 1, 2 ], rate=(int)[ 1, 2147483647 ]; audio/AMR, rate=(int)8000, channels=(int)[ 1, 2 ]; audio/AMR-WB, rate=(int)16000, channels=(int)[ 1, 2 ]; audio/x-alac, channels=(int)[ 1, 2 ], rate=(int)[ 1, 2147483647 ]</details>
+ <details>audio/x-raw, format=(string){ S32LE, S32BE, S24LE, S24BE, S16LE, S16BE, S8, U8 }, layout=(string)interleaved, channels=(int)[ 1, 2 ], rate=(int)[ 1, 2147483647 ]; audio/x-raw, format=(string){ S32LE, S32BE, S24LE, S24BE, S16LE, S16BE, S8, U8 }, layout=(string)interleaved, channel-mask=(bitmask)0x0000000000000000, channels=(int)[ 1, 16 ], rate=(int)[ 1, 2147483647 ]; audio/mpeg, mpegversion=(int)1, layer=(int)3, channels=(int)[ 1, 2 ], rate=(int)[ 1, 2147483647 ]; audio/mpeg, mpegversion=(int)4, stream-format=(string)raw, channels=(int)[ 1, 8 ], rate=(int)[ 1, 2147483647 ]; audio/x-adpcm, layout=(string)dvi, block_align=(int)[ 64, 8096 ], channels=(int)[ 1, 2 ], rate=(int)[ 1, 2147483647 ]; audio/x-alaw, channels=(int)[ 1, 2 ], rate=(int)[ 1, 2147483647 ]; audio/x-mulaw, channels=(int)[ 1, 2 ], rate=(int)[ 1, 2147483647 ]; audio/AMR, rate=(int)8000, channels=(int)[ 1, 2 ]; audio/AMR-WB, rate=(int)16000, channels=(int)[ 1, 2 ]; audio/x-alac, channels=(int)[ 1, 2 ], rate=(int)[ 1, 2147483647 ]</details>
</caps>
<caps>
<name>subtitle_%u</name>
<description>JACK audio elements</description>
<filename>../../ext/jack/.libs/libgstjack.so</filename>
<basename>libgstjack.so</basename>
- <version>1.10.0</version>
+ <version>1.11.1</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>JPeg plugin library</description>
<filename>../../ext/jpeg/.libs/libgstjpeg.so</filename>
<basename>libgstjpeg.so</basename>
- <version>1.10.0</version>
+ <version>1.11.1</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>Audio level plugin</description>
<filename>../../gst/level/.libs/libgstlevel.so</filename>
<basename>libgstlevel.so</basename>
- <version>1.10.0</version>
+ <version>1.11.1</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>Matroska and WebM stream handling</description>
<filename>../../gst/matroska/.libs/libgstmatroska.so</filename>
<basename>libgstmatroska.so</basename>
- <version>1.10.0</version>
+ <version>1.11.1</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<name>audio_%u</name>
<direction>sink</direction>
<presence>request</presence>
- <details>audio/mpeg, mpegversion=(int)1, layer=(int)[ 1, 3 ], channels=(int)[ 1, 2147483647 ], rate=(int)[ 1, 2147483647 ]; audio/mpeg, mpegversion=(int){ 2, 4 }, stream-format=(string)raw, channels=(int)[ 1, 2147483647 ], rate=(int)[ 1, 2147483647 ]; audio/x-ac3, channels=(int)[ 1, 2147483647 ], rate=(int)[ 1, 2147483647 ]; audio/x-eac3, channels=(int)[ 1, 2147483647 ], rate=(int)[ 1, 2147483647 ]; audio/x-dts, channels=(int)[ 1, 2147483647 ], rate=(int)[ 1, 2147483647 ]; audio/x-vorbis, channels=(int)[ 1, 2147483647 ], rate=(int)[ 1, 2147483647 ]; audio/x-flac, channels=(int)[ 1, 2147483647 ], rate=(int)[ 1, 2147483647 ]; audio/x-opus; audio/x-speex, channels=(int)[ 1, 2147483647 ], rate=(int)[ 1, 2147483647 ]; audio/x-raw, format=(string){ U8, S16BE, S16LE, S24BE, S24LE, S32BE, S32LE, F32LE, F64LE }, layout=(string)interleaved, channels=(int)[ 1, 2147483647 ], rate=(int)[ 1, 2147483647 ]; audio/x-tta, width=(int){ 8, 16, 24 }, channels=(int){ 1, 2 }, rate=(int)[ 8000, 96000 ]; audio/x-pn-realaudio, raversion=(int){ 1, 2, 8 }, channels=(int)[ 1, 2147483647 ], rate=(int)[ 1, 2147483647 ]; audio/x-wma, wmaversion=(int)[ 1, 3 ], block_align=(int)[ 0, 65535 ], bitrate=(int)[ 0, 524288 ], channels=(int)[ 1, 2147483647 ], rate=(int)[ 1, 2147483647 ]; audio/x-alaw, channels=(int){ 1, 2 }, rate=(int)[ 8000, 192000 ]; audio/x-mulaw, channels=(int){ 1, 2 }, rate=(int)[ 8000, 192000 ]; audio/x-adpcm, layout=(string)dvi, block_align=(int)[ 64, 8192 ], channels=(int){ 1, 2 }, rate=(int)[ 8000, 96000 ]; audio/x-adpcm, layout=(string)g726, channels=(int)1, rate=(int)8000</details>
+ <details>audio/mpeg, mpegversion=(int)1, layer=(int)[ 1, 3 ], channels=(int)[ 1, 2147483647 ], rate=(int)[ 1, 2147483647 ]; audio/mpeg, mpegversion=(int){ 2, 4 }, stream-format=(string)raw, channels=(int)[ 1, 2147483647 ], rate=(int)[ 1, 2147483647 ]; audio/x-ac3, channels=(int)[ 1, 2147483647 ], rate=(int)[ 1, 2147483647 ]; audio/x-eac3, channels=(int)[ 1, 2147483647 ], rate=(int)[ 1, 2147483647 ]; audio/x-dts, channels=(int)[ 1, 2147483647 ], rate=(int)[ 1, 2147483647 ]; audio/x-vorbis, channels=(int)[ 1, 2147483647 ], rate=(int)[ 1, 2147483647 ]; audio/x-flac, channels=(int)[ 1, 2147483647 ], rate=(int)[ 1, 2147483647 ]; audio/x-opus; audio/x-speex, channels=(int)[ 1, 2147483647 ], rate=(int)[ 1, 2147483647 ]; audio/x-raw, format=(string){ U8, S16BE, S16LE, S24BE, S24LE, S32BE, S32LE, F32LE, F64LE }, layout=(string)interleaved, channels=(int)[ 1, 2147483647 ], rate=(int)[ 1, 2147483647 ]; audio/x-tta, width=(int){ 8, 16, 24 }, channels=(int){ 1, 2 }, rate=(int)[ 8000, 96000 ]; audio/x-pn-realaudio, raversion=(int){ 1, 2, 8 }, channels=(int)[ 1, 2147483647 ], rate=(int)[ 1, 2147483647 ]; audio/x-wma, wmaversion=(int)[ 1, 3 ], block_align=(int)[ 0, 65535 ], bitrate=(int)[ 0, 524288 ], channels=(int)[ 1, 2147483647 ], rate=(int)[ 1, 2147483647 ]; audio/x-alaw, channels=(int){ 1, 2 }, rate=(int)[ 8000, 192000 ]; audio/x-mulaw, channels=(int){ 1, 2 }, rate=(int)[ 8000, 192000 ]; audio/x-adpcm, layout=(string)dvi, block_align=(int)[ 64, 8192 ], channels=(int){ 1, 2 }, rate=(int)[ 8000, 96000 ]; audio/G722, channels=(int)1, rate=(int)16000; audio/x-adpcm, layout=(string)g726, channels=(int)1, rate=(int)8000</details>
</caps>
<caps>
<name>subtitle_%u</name>
<description>MuLaw audio conversion routines</description>
<filename>../../gst/law/.libs/libgstmulaw.so</filename>
<basename>libgstmulaw.so</basename>
- <version>1.10.0</version>
+ <version>1.11.1</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>Reads/Writes buffers from/to sequentially named files</description>
<filename>../../gst/multifile/.libs/libgstmultifile.so</filename>
<basename>libgstmultifile.so</basename>
- <version>1.10.0</version>
+ <version>1.11.1</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>multipart stream manipulation</description>
<filename>../../gst/multipart/.libs/libgstmultipart.so</filename>
<basename>libgstmultipart.so</basename>
- <version>1.10.0</version>
+ <version>1.11.1</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>Template for a video filter</description>
<filename>../../gst/debugutils/.libs/libgstnavigationtest.so</filename>
<basename>libgstnavigationtest.so</basename>
- <version>1.10.0</version>
+ <version>1.11.1</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>Open Sound System (OSS) version 4 support for GStreamer</description>
<filename>../../sys/oss4/.libs/libgstoss4audio.so</filename>
<basename>libgstoss4audio.so</basename>
- <version>1.10.0</version>
+ <version>1.11.1</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>OSS (Open Sound System) support for GStreamer</description>
<filename>../../sys/oss/.libs/libgstossaudio.so</filename>
<basename>libgstossaudio.so</basename>
- <version>1.10.0</version>
+ <version>1.11.1</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>PNG plugin library</description>
<filename>../../ext/libpng/.libs/libgstpng.so</filename>
<basename>libgstpng.so</basename>
- <version>1.10.0</version>
+ <version>1.11.1</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>PulseAudio plugin library</description>
<filename>../../ext/pulse/.libs/libgstpulse.so</filename>
<basename>libgstpulse.so</basename>
- <version>1.10.0</version>
+ <version>1.11.1</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>ReplayGain volume normalization</description>
<filename>../../gst/replaygain/.libs/libgstreplaygain.so</filename>
<basename>libgstreplaygain.so</basename>
- <version>1.10.0</version>
+ <version>1.11.1</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>Real-time protocol plugins</description>
<filename>../../gst/rtp/.libs/libgstrtp.so</filename>
<basename>libgstrtp.so</basename>
- <version>1.10.0</version>
+ <version>1.11.1</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>RTP session management plugin library</description>
<filename>../../gst/rtpmanager/.libs/libgstrtpmanager.so</filename>
<basename>libgstrtpmanager.so</basename>
- <version>1.10.0</version>
+ <version>1.11.1</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>transfer data via RTSP</description>
<filename>../../gst/rtsp/.libs/libgstrtsp.so</filename>
<basename>libgstrtsp.so</basename>
- <version>1.10.0</version>
+ <version>1.11.1</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>Shape Wipe transition filter</description>
<filename>../../gst/shapewipe/.libs/libgstshapewipe.so</filename>
<basename>libgstshapewipe.so</basename>
- <version>1.10.0</version>
+ <version>1.11.1</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>Sends data to an icecast server using libshout2</description>
<filename>../../ext/shout2/.libs/libgstshout2.so</filename>
<basename>libgstshout2.so</basename>
- <version>1.10.0</version>
+ <version>1.11.1</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>libshout2</package>
- <origin>http://www.icecast.org/download.html</origin>
+ <origin>http://www.icecast.org/download/</origin>
<elements>
<element>
<name>shout2send</name>
<description>Apply the standard SMPTE transitions on video images</description>
<filename>../../gst/smpte/.libs/libgstsmpte.so</filename>
<basename>libgstsmpte.so</basename>
- <version>1.10.0</version>
+ <version>1.11.1</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>libsoup HTTP client src/sink</description>
<filename>../../ext/soup/.libs/libgstsouphttpsrc.so</filename>
<basename>libgstsouphttpsrc.so</basename>
- <version>1.10.0</version>
+ <version>1.11.1</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>Run an FFT on the audio signal, output spectrum data</description>
<filename>../../gst/spectrum/.libs/libgstspectrum.so</filename>
<basename>libgstspectrum.so</basename>
- <version>1.10.0</version>
+ <version>1.11.1</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>Speex plugin library</description>
<filename>../../ext/speex/.libs/libgstspeex.so</filename>
<basename>libgstspeex.so</basename>
- <version>1.10.0</version>
+ <version>1.11.1</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>Tag writing plug-in based on taglib</description>
<filename>../../ext/taglib/.libs/libgsttaglib.so</filename>
<basename>libgsttaglib.so</basename>
- <version>1.10.0</version>
+ <version>1.11.1</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>transfer data via UDP</description>
<filename>../../gst/udp/.libs/libgstudp.so</filename>
<basename>libgstudp.so</basename>
- <version>1.10.0</version>
+ <version>1.11.1</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>elements for Video 4 Linux</description>
<filename>../../sys/v4l2/.libs/libgstvideo4linux2.so</filename>
<basename>libgstvideo4linux2.so</basename>
- <version>1.10.0</version>
+ <version>1.11.1</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>resizes a video by adding borders or cropping</description>
<filename>../../gst/videobox/.libs/libgstvideobox.so</filename>
<basename>libgstvideobox.so</basename>
- <version>1.10.0</version>
+ <version>1.11.1</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>Crops video into a user-defined region</description>
<filename>../../gst/videocrop/.libs/libgstvideocrop.so</filename>
<basename>libgstvideocrop.so</basename>
- <version>1.10.0</version>
+ <version>1.11.1</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>Video filters plugin</description>
<filename>../../gst/videofilter/.libs/libgstvideofilter.so</filename>
<basename>libgstvideofilter.so</basename>
- <version>1.10.0</version>
+ <version>1.11.1</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>Video mixer</description>
<filename>../../gst/videomixer/.libs/libgstvideomixer.so</filename>
<basename>libgstvideomixer.so</basename>
- <version>1.10.0</version>
+ <version>1.11.1</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>VP8 plugin</description>
<filename>../../ext/vpx/.libs/libgstvpx.so</filename>
<basename>libgstvpx.so</basename>
- <version>1.10.0</version>
+ <version>1.11.1</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>Encode raw audio into WAV</description>
<filename>../../gst/wavenc/.libs/libgstwavenc.so</filename>
<basename>libgstwavenc.so</basename>
- <version>1.10.0</version>
+ <version>1.11.1</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>Wavpack lossless/lossy audio format handling</description>
<filename>../../ext/wavpack/.libs/libgstwavpack.so</filename>
<basename>libgstwavpack.so</basename>
- <version>1.10.0</version>
+ <version>1.11.1</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>Parse a .wav file into raw audio</description>
<filename>../../gst/wavparse/.libs/libgstwavparse.so</filename>
<basename>libgstwavparse.so</basename>
- <version>1.10.0</version>
+ <version>1.11.1</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>X11 video input plugin using standard Xlib calls</description>
<filename>../../sys/ximage/.libs/libgstximagesrc.so</filename>
<basename>libgstximagesrc.so</basename>
- <version>1.10.0</version>
+ <version>1.11.1</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<description>Encodes a YUV frame into the yuv4mpeg format (mjpegtools)</description>
<filename>../../gst/y4m/.libs/libgsty4menc.so</filename>
<basename>libgsty4menc.so</basename>
- <version>1.10.0</version>
+ <version>1.11.1</version>
<license>LGPL</license>
<source>gst-plugins-good</source>
<package>GStreamer Good Plug-ins source release</package>
<release>
<Version>
+ <revision>1.11.1</revision>
+ <branch>master</branch>
+ <name></name>
+ <created>2017-01-12</created>
+ <file-release rdf:resource="http://gstreamer.freedesktop.org/src/gst-plugins-good/gst-plugins-good-1.11.1.tar.xz" />
+ </Version>
+ </release>
+
+ <release>
+ <Version>
<revision>1.10.0</revision>
<branch>master</branch>
<name></name>
projects / platform / upstream / gst-plugins-good.git / commitdiff