libnice 0.1.21 (2023-01-07)
===========================
Only use `ifr_ifindex` if OS supports it, fixes build on iOS and FreeBSD
libnice 0.1.20 (2023-01-06)
===========================
Remove support for GStreamer 0.10 builds
Add macro to check LIBNICE version
Added utility function to get the STUN server from a candidate
Support additional header in built-in HTTP proxy client
Add boxed type for NiceAddress for bindings
Add API to get the interface index for a local address
Explicitly bind to a specific interface when creating UDP sockets
Limit the number of stored incoming checks based on a property
Do asynchronous DNS resolution for STUN and TURN servers
Add introspection friendly API to get an allocated string from a NiceCandidate
Enable gst-full to link in a single element
libnice 0.1.19 (2022-05-03)
===========================
Allow incoming connchecks before remote candidates are set, allows for connection based on received bind requests
Implement RFC 7675 for Consent Freshness
Use a single server reflexive and local relay candidate, reduces useless duplicated local candidates
Improved ICE restart implementation
Use Windows native crypto API, removing the need for OpenSSL
Add bytestream ICE-TCP and improve ICE-TCP
Add API to know if a NiceAddress is link-local
Add API to extact the relay address from a relayed NiceAddress
Improve support for detection addresses on Android, iOS, macOS
A number of bug fixes
libnice 0.1.18 (2020-10-20)
===========================
Remove the autotools build system, now only meson is available
On Windows, use crypto library instead of CryptGenRandom() which is deprecated
On Windows, use GetBestInterfaceEx() for UWP compatibility
On Windows, fix the listing of interfaces to use the correct APIs
On Windows, implement ignoring interfaces
Accept receiving messages in multiple steps over TCP
Accept duplicated ports as last option instead of spinning forever
Use sendmmsg if possible to send multiple packets in one call
Fail gathering if no port is available
Hide the implementation of NiceCandidate, this hides some parts that were previously visible
Enable TURN server connects where both TCP and UDP use the same port number
Don't count rejected STUN messages as keepalive packets
libnice 0.1.17 (2020-05-22)
===========================
Add API to retrieve the underlying BSD sockets
Support systems with multiple loopback devices
Ignore non-running network interfaces
Ignore multiple interface prefixes
Now tries to nominate matching pairs across components and streams
Retry TURN deallocation on timeout, requires not destoying the NiceAgent right after the stream
Use different port for every host candidate
Make timeouts and retransmissions more in line with the RFCs
Find OpenSSL without pkg-config, for Windows
Complete meson support
GLib required version update to 2.54
Removed deprecated GLib APIs
Many ICE compatibility and performance improvements
Many bug fixes
libnice 0.1.16 (2019-05-09)
===========================
Add API to make it easier to implement ICE trickle
Add async closing of agent, to cleanly close TURN allocations
Add Google non-standard NOMINATION STUN attribute
Fix tests on Windows
Fix some racy tests
Highlights
new msdk plugin for Intel's Media SDK for hardware-accelerated
video encoding and decoding on Intel graphics hardware on Windows
or Linux.
x264enc can now use multiple x264 library versions compiled for
different bit depths at runtime, to transparently provide support
for multiple bit depths.
videoscale and videoconvert now support multi-threaded scaling
and conversion, which is particularly useful with higher
resolution video.
h264parse will now automatically insert AU delimiters if needed
when outputting byte-stream format, which improves standard
compliance and is needed in particular for HLS playback on
iOS/macOS.
rtpbin has acquired bundle support for incoming streams
Major new features and changes Noteworthy new API
The video library gained support for a number of new video
formats:
GBR_12LE, GBR_12BE, GBRA_12LE, GBRA_12BE (planar 4:4:4
RGB/RGBA, 12 bits per channel) GBRA_10LE, GBRA_10BE (planar
4:4:4:4 RGBA, 10 bits per channel) GBRA (planar 4:4:4:4
ARGB, 8 bits per channel) I420_12BE, I420_12LE (planar 4:2:0
YUV, 12 bits per channel) I422_12BE,I422_12LE (planar 4:2:2
YUV, 12 bits per channel) Y444_12BE, Y444_12LE (planar 4:4:4
YUV, 12 bits per channel) VYUY (another packed 4:2:2 YUV
format)
The high-level GstPlayer API was extended with functions for
taking video snapshots and enabling accurate seeking. It can
optionally also use the still-experimental playbin3 element
now.
New Elements
msdk: new plugin for Intel's Media SDK for hardware-accelerated
video encoding and decoding on Intel graphics hardware on Windows
or Linux. This includes an H.264 encoder/decoder (msdkh264dec,
msdkh264enc), an H.265 encoder/decoder (msdkh265dec, msdkh265enc),
an MJPEG encoder/encoder (msdkmjpegdec, msdkmjpegenc), an MPEG-2
video encoder (msdkmpeg2enc) and a VP8 encoder (msdkvp8enc).
iqa is a new Image Quality Assessment plugin based on DSSIM,
similar to the old (unported) videomeasure element.
The faceoverlay element, which allows you to overlay SVG graphics
over a detected face in a video stream, has been ported from
0.10.
our ffmpeg wrapper plugin now exposes/maps the ffmpeg Opus audio
decoder (avdec_opus) as well as the GoPro CineForm HD / CFHD
decoder (avdec_cfhd), and also a parser/writer for the IVF
format (avdemux_ivf and avmux_ivf).
audiobuffersplit is a new element that splits raw audio buffers
into equal-sized buffers
audiomixmatrix is a new element that mixes N:M audio channels
according to a configured mix matrix.
The timecodewait element got renamed to avwait and can operate
in different modes now.
The opencv video processing plugin has gained a new dewarp
element that dewarps fisheye images.
ttml is a new plugin for parsing and rendering subtitles in
Timed Text Markup Language (TTML) format. For the time being
these elements will not be autoplugged during media playback
however, unless the GST_TTML_AUTOPLUG=1 environment variable
is set. Only the EBU-TT-D profile is supported at this point.
New element features and additions
x264enc can now use multiple x264 library versions compiled for
different bit depths at runtime, to transparently provide support
for multiple bit depths. A new configure parameter
--with-x264-libraries has been added to specify additional paths
to look for additional x264 libraries to load. Background is
that the libx264 library is always compile for one specific bit
depth and the x264enc element would simply support the depth
supported by the underlying library. Now we can support multiple
depths.
x264enc also picks up the interlacing mode automatically from
the input caps now and passed interlacing/TFF information
correctly to the library.
videoscale and videoconvert now support multi-threaded scaling
and conversion, which is particularly useful with higher
resolution video. This has to be enabled explicitly via the
"n-threads" property.
videorate's new "rate" property lets you set a speed factor on
the output stream
splitmuxsink's buffer collection and scheduling was rewritten
to make processing and splitting deterministic; before it was
possible for a buffer to end up in a different file chunk in
different runs. splitmuxsink also gained a new "format-location-full"
signal that works just like the existing "format-location"
signal only that it is also passed the primary stream's first
buffer as argument, so that it is possible to construct the
file name based on metadata such as the buffer timestamp or any
GstMeta attached to the buffer. The new "max-size-timecode"
property allows for timecode-based splitting. splitmuxsink will
now also automatically start a new file if the input caps change
in an incompatible way.
fakesink has a new "drop-out-of-segment" property to not drop
out-of-segment buffers, which is useful for debugging purposes.
identity gained a "ts-offset" property.
both fakesink and identity now also print what kind of metas
are attached to buffers when printing buffer details via the
"last-message" property used by gst-launch-1.0 -v.
multiqueue: made "min-interleave-time" a configurable property.
video nerds will be thrilled to know that videotestsrc's snow
is now deterministic. videotestsrc also gained some new properties
to make the ball pattern based on system time, and invert colours
each second ("animation-mode", "motion", and "flip" properties).
oggdemux reverse playback should work again now. You're welcome.
playbin3 and urisourcebin now have buffering enabled by default,
and buffering message aggregation was fixed.
tcpclientsrc now has a "timeout" property
appsink has gained support for buffer lists. For backwards
compatibility reasons users need to enable this explicitly with
gst_app_sink_set_buffer_list_support(), however. Once activated,
a pulled GstSample can contain either a buffer list or a single
buffer.
splitmuxsrc reverse playback was fixed and handling of sparse
streams, such as subtitle tracks or metadata tracks, was improved.
matroskamux has acquired support for muxing G722 audio; it also
marks all buffers as keyframes now when streaming only audio,
so that tcpserversink will behave properly with audio-only
streams.
qtmux gained support for ProRes 4444 XQ, HEVC/H.265 and CineForm
(GoPro) formats, and generally writes more video stream-related
metadata into the track headers. It is also allows configuration
of the maximum interleave size in bytes and time now. For
fragmented mp4 we always write the tfdt atom now as required
by the DASH spec.
qtdemux supports FLAC, xvid, mp2, S16L and CineForm (GoPro)
tracks now, and generally tries harder to extract more video-related
information from track headers, such as colorimetry or interlacing
details. It also received a couple of fixes for the scenario
where upstream operates in TIME format and feeds chunks to
qtdemux (e.g. DASH or MSE).
audioecho has two new properties to apply a delay only to certain
channels to create a surround effect, rather than an echo on
all channels. This is useful when upmixing from stereo, for
example. The "surround-delay" property enables this, and the
"surround-mask" property controls which channels are considered
surround sound channels in this case.
webrtcdsp gained various new properties for gain control and
also exposes voice activity detection now, in which case it
will post "voice-activity" messages on the bus whenever the
voice detection status changes.
The decklink capture elements for Blackmagic Decklink cards
have seen a number of improvements:
decklinkvideosrc will post a warning message on "no signal"
and an info message when the signal lock has been (re)acquired.
There is also a new read-only "signal" property that can
be used to query the signal lock status. The GAP flag will
be set on buffers that are captured without a signal lock.
The new drop-no-signal-frames will make decklinkvideosrc
drop all buffers that have been captured without an input
signal. The "skip-first-time" property will make the source
drop the first few buffers, which is handy since some devices
will at first output buffers with the wrong resolution
before they manage to figure out the right input format and
decide on the actual output caps.
decklinkaudiosrc supports more than just 2 audio channels
now.
The capture sources no longer use the "hardware" timestamps
which turn out to be useless and instead just use the
pipeline clock directly.
srtpdec now also has a readonly "stats" property, just like
srtpenc.
rtpbin gained RTP bundle support, as used by e.g. WebRTC. The
first rtpsession will have a rtpssrcdemux element inside splitting
the streams based on their SSRC and potentially dispatch to a
different rtpsession. Because retransmission SSRCs need to be
merged with the corresponding media stream the ::on-bundled-ssrc
signal is emitted on rtpbin so that the application can find
out to which session the SSRC belongs.
rtprtxqueue gained two new properties exposing retransmission
statistics ("requests" and "fulfilled-requests")
kmssink will now use the preferred mode for the monitor and
render to the base plane if nothing else has set a mode yet.
This can also be done forcibly in any case via the new
"force-modesetting" property. Furthermore, kmssink now allows
only the supported connector resolutions as input caps in order
to avoid scaling or positioning of the input stream, as kmssink
can't know whether scaling or positioning would be more appropriate
for the use case at hand.
waylandsink can now take DMAbuf buffers as input in the presence
of a compatible Wayland compositor. This enables zero-copy
transfer from a decoder or source that outputs DMAbuf. It will
also set surface opacity hint to allow better rendering
optimization in the compositor.
udpsrc can be bound to more than one interface when joining a
multicast group, this is done by giving a comma separate list
of interfaces such as multicast-iface="eth0,eth1".
Plugin moves
dataurisrc moved from gst-plugins-bad to core
The rawparse plugin containing the rawaudioparse and rawvideoparse
elements moved from gst-plugins-bad to gst-plugins-base. These
elements supersede the old videoparse and audioparse elements.
They work the same, with just some minor API changes. The old
legacy elements still exist in gst-plugins-bad, but may be
removed at some point in the future.
timecodestamper is an element that attaches time codes to video
buffers in form of GstVideoTimeCodeMetas. It had a "clock-source"
property which has now been removed because it was fairly useless
in practice. It gained some new properties however: the
"first-timecode" property can be used to set the inital timecode;
alternatively "first-timecode-to-now" can be set, and then the
current system time at the time the first buffer arrives is
used as base time for the time codes.
Plugin removals
The mad mp1/mp2/mp3 decoder plugin was removed from gst-plugins-ugly,
as libmad is GPL licensed, has been unmaintained for a very
long time, and there are better alternatives available. Use the
mpg123audiodec element from the mpg123 plugin in gst-plugins-ugly
instead, or avdec_mp3 from the gst-libav module which wraps the
ffmpeg library. We expect that we will be able to move mp3
decoding to gst-plugins-good in the next cycle seeing that most
patents around mp3 have expired recently or are about to expire.
The mimic plugin was removed from gst-plugins-bad. It contained
a decoder and encoder for a video codec used by MSN messenger
many many years ago (in a galaxy far far away). The underlying
library is unmaintained and no one really needs to use this
codec any more. Recorded videos can still be played back with
the MIMIC decoder in gst-libav.
Miscellaneous API additions
Request pad name templates passed to gst_element_request_pad()
may now contain multiple specifiers, such as e.g. src_%u_%u.
gst_buffer_iterate_meta_filtered() is a variant of
gst_buffer_iterate_meta() that only returns metas of the requested
type and skips all other metas.
gst_pad_task_get_state() gets the current state of a task in a
thread-safe way.
gst_uri_get_media_fragment_table() provides the media fragments
of an URI as a table of key=value pairs.
gst_print(), gst_println(), gst_printerr(), and gst_printerrln()
can be used to print to stdout or stderr. These functions are
similar to g_print() and g_printerr() but they also support all
the additional format specifiers provided by the GStreamer
logging system, such as e.g. GST_PTR_FORMAT.
a GstParamSpecArray has been added, for elements who want to
have array type properties, such as the audiomixmatrix element
for example. There are also two new functions to set and get
properties of this type from bindings:
gst_util_set_object_array() gst_util_get_object_array()
various helper functions have been added to make it easier to
set or get GstStructure fields containing caps-style array or
list fields from language bindings (which usually support
GValueArray but don't know about the GStreamer specific fundamental
types):
gst_structure_get_array() gst_structure_set_array()
gst_structure_get_list() gst_structure_set_list()
a new 'dynamic type' registry factory type was added to register
dynamically loadable GType types. This is useful for automatically
loading enum/flags types that are used in caps, such as for
example the GstVideoMultiviewFlagsSet type used in multiview
video caps.
there is a new GstProxyControlBinding for use with GstController.
This allows proxying the control interface from one property
on one GstObject to another property (of the same type) in
another GstObject. So e.g. in parent-child relationship, one
may need to call gst_object_sync_values() on the child and have
a binding (set elsewhere) on the parent update the value. This
is used in glvideomixer and glsinkbin for example, where
sync_values() on the child pad or element will call sync_values()
on the exposed bin pad or element.
Note that this doesn't solve GObject property forwarding, that
must be taken care of by the implementation manually or using
GBinding.
gst_base_parse_drain() has been made public for subclasses to
use.
`gst_base_sink_set_drop_out_of_segment()' can be used by
subclasses to prevent GstBaseSink from dropping buffers that
fall outside of the segment.
gst_calculate_linear_regression() is a new utility function to
calculate a linear regression.
gst_debug_get_stack_trace is an easy way to retrieve a stack
trace, which can be useful in tracer plugins.
allocators: the dmabuf allocator is now sub-classable, and there
is a new GST_CAPS_FEATURE_MEMORY_DMABUF define.
video decoder subclasses can use the newly-added function
gst_video_decoder_allocate_output_frame_with_params() to pass
a GstBufferPoolAcquireParams to the buffer pool for each buffer
allocation.
the video time code API has gained a dedicated GstVideoTimeCodeInterval
type plus related API, including functions to add intervals to
timecodes.
There is a new libgstbadallocators-1.0 library in gst-plugins-bad,
which may go away again in future releases once the
GstPhysMemoryAllocator interface API has been validated by more
users and was moved to libgstallocators-1.0 from gst-plugins-base.
GstPlayer
New API has been added to:
get the number of audio/video/subtitle streams:
gst_player_media_info_get_number_of_streams()
gst_player_media_info_get_number_of_video_streams()
gst_player_media_info_get_number_of_audio_streams()
gst_player_media_info_get_number_of_subtitle_streams()
enable accurate seeking: gst_player_config_set_seek_accurate()
and gst_player_config_get_seek_accurate()
get a snapshot image of the video in RGBx, BGRx, JPEG, PNG or
native format: gst_player_get_video_snapshot()
selecting use of a specific video sink element
(gst_player_video_overlay_video_renderer_new_with_sink())
If the environment variable GST_PLAYER_USE_PLAYBIN3 is set,
GstPlayer will use the still-experimental playbin3 element and
the GstStreams API for playback.
Miscellaneous changes
video caps for interlaced video may contain an optional
"field-order" field now in the case of interlaced-mode=interleaved
to signal that the field order is always the same throughout
the stream. This is useful to signal to muxers such as mp4mux.
The new field is parsed from/to GstVideoInfo of course.
video decoder and video encoder base classes try harder to proxy
interlacing, colorimetry and chroma-site related fields in caps
properly.
The buffer stored in the PROTECTION events is now left unchanged.
This is a change of behaviour since 1.8, especially for the
mssdemux element which used to decode the base64 parsed data
wrapped in the protection events emitted by the demuxer.
PROTECTION events can now be injected into the pipeline from
the application; source elements deriving from GstBaseSrc will
forward those downstream now.
The DASH demuxer is now correctly parsing the MSPR-2.0
ContentProtection nodes and emits Protection events accordingly.
Applications relying on those events might need to decode the
base64 data stored in the event buffer before using it.
The registry can now also be disabled by setting the environment
variable GST_REGISTRY_DISABLE=yes, with similar effect as the
GST_DISABLE_REGISTRY compile time switch.
Seeking performance with gstreamer-vaapi based decoders was
improved. It would recreate the decoder and surfaces on every
seek which can be quite slow.
more robust handling of input caps changes in videoaggregator-based
elements such as compositor.
Lots of adaptive streaming-related fixes across the board (DASH,
MSS, HLS). Also:
mssdemux, the Microsoft Smooth Streaming demuxer, has seen
various fixes for live streams, duration reporting and
seeking.
The DASH manifest parser now extracts MS PlayReady
ContentProtection objects from manifests and sends them
downstream as PROTECTION events. It also supports multiple
Period elements in external xml now.
gst-libav was updated to ffmpeg 3.3 but should still work with
any 3.x version.
GstEncodingProfile has been generally enhanced so it can, for
example, be used to get possible profiles for a given file
extension. It is now possible to define profiles based on element
factory names or using a path to a .gep file containing a
serialized profile.
audioconvert can now do endianness conversion in-place. All
other conversions still require a copy, but e.g. sign conversion
and a few others could also be implemented in-place now.
The new, experimental playbin3 and urisourcebin elements got
many bugfixes and improvements and should generally be closer
to a full replacement of the old elements.
interleave now supports > 64 channels.
OpenCV elements, grabcut and retinex has been ported to use
GstOpencvVideoFilter base class, increasing code reuse and
fixing buffer map/unmap issues. Redundant copie of images has
been removed in edgedetect, cvlaplace and cvsobel. This comes
with various cleanup and Meson support.
OpenGL integration
As usual the GStreamer OpenGL integration library has seen
numerous fixes and performance improvements all over the place,
and is hopefully ready now to become API stable and be moved
to gst-plugins-base during the 1.14 release cycle.
The GStreamer OpenGL integration layer has also gained support
for the Vivante EGL FB windowing system, which improves performance
on platforms such as Freescale iMX.6 for those who are stuck
with the proprietary driver. The qmlglsink element also supports
this now if Qt is used with eglfs or wayland backend, and it
works in conjunction with gstreamer-imx of course.
various qmlglsrc improvements
Tracing framework and debugging improvements
New tracing hooks have been added to track GstMiniObject and
GstObject ref/unref operations.
The memory leaks tracer can optionally use this to retrieve
stack traces if enabled with e.g.
GST_TRACERS=leaks(filters="GstEvent,GstMessage",stack-traces-flags=full)
The GST_DEBUG_FILE environment variable, which can be used to
write the debug log output to a file instead of printing it to
stderr, can now contain a name pattern, which is useful for
automated testing and continuous integration systems. The
following format specifiers are supported:
%p: will be replaced with the PID %r: will be replaced with
a random number, which is useful for instance when running
two processes with the same PID but in different containers.
Tools
gst-inspect-1.0 can now list elements by type with the new
--types command-line option, e.g. gst-inspect-1.0 --types=Audio/Encoder
will show a list of audio encoders.
gst-launch-1.0 and gst_parse_launch() have gained a new operator
(:) that allows linking all pads between two elements. This is
useful in cases where the exact number of pads or type of pads
is not known beforehand, such as in the uridecodebin : encodebin
scenario, for example. In this case, multiple links will be
created if the encodebin has multiple profiles compatible with
the output of uridecodebin.
gst-device-monitor-1.0 now shows a gst-launch-1.0 snippet for
each device that shows how to make use of it in a gst-launch-1.0
pipeline string.
GStreamer RTSP server
The RTSP server now also supports Digest authentication in
addition to Basic authentication.
The GstRTSPClient class has gained a pre-*-request signal and
virtual method for each client request type, emitted in the
beginning of each rtsp request. These signals or virtual methods
let the application validate the requests, configure the
media/stream in a certain way and also generate error status
codes in case of an error or a bad request.
GStreamer VAAPI
GstVaapiDisplay now inherits from GstObject, thus the VA display
logging messages are better and tracing the context sharing is
more readable.
When uploading raw images into a VA surfaces now VADeriveImages
are tried fist, improving the upload performance, if it is
possible.
The decoders and the post-processor now can push dmabuf-based
buffers to downstream under certain conditions. For example:
GST_GL_PLATFORM=egl gst-play-1.0 video-sample.mkv
--videosink=glimagesink
Refactored the wrapping of VA surface into gstreamer memory,
adding lock when mapping and unmapping, and many other fixes.
Now vaapidecodebin loads vaapipostproc dynamically. It is
possible to avoid it usage with the environment variable
GST_VAAPI_DISABLE_VPP=1.
Regarding encoders: they have primary rank again, since they
can discover, in run-time, the color formats they can use for
upstream raw buffers and caps renegotiation is now possible.
Also the encoders push encoding info downstream via tags.
About specific encoders: added constant bit-rate encoding mode
for VP8 and H265 encoder handles P010_10LE color format.
Regarding decoders, flush operation has been improved, now the
internal VA encoder is not recreated at each flush. Also there
are several improvements in the handling of H264 and H265
streams.
VAAPI plugins try to create their on GstGL context (when
available) if they cannot find it in the pipeline, to figure
out what type of VA Display they should create.
Regarding vaapisink for X11, if the backend reports that it is
unable to render correctly the current color format, an internal
VA post-processor, is instantiated (if available) and converts
the color format.
GStreamer Editing Services and NLE
Enhanced auto transition behaviour
Fix some races in nlecomposition
Allow building with msvc
Added a UNIX manpage for ges-launch
API changes:
Added ges_deinit (allowing the leak tracer to work properly)
Added ges_layer_get_clips_in_interval Finally hide internal
symbols that should never have been exposed
GStreamer validate
Port gst-validate-launcher to python 3
gst-validate-launcher now checks if blacklisted bugs have been
fixed on bugzilla and errors out if it is the case
Allow building with msvc
Add ability for the launcher to run GStreamer unit tests
Added a way to activate the leaks tracer on our tests and fix
leaks
Make the http server multithreaded
New testsuite for running various test scenarios on the DASH-IF
test vectors
GStreamer Python Bindings
Overrides has been added for IntRange, Int64Range, DoubleRange,
FractionRange, Array and List. This finally enables Python
programmers to fully read and write GstCaps objects.
Build and Dependencies
Meson build files are now disted in tarballs, for jhbuild and
so distro packagers can start using it. Note that the Meson-based
build system is not 100% feature-equivalent with the autotools-based
one yet.
Some plugin filenames have been changed to match the plugin
names: for example the file name of the encoding plugin in
gst-plugins-base containing the encodebin element was
libgstencodebin.so and has been changed to libgstencoding.so.
This affects only a handful of plugins across modules.
Developers who install GStreamer from source and just do make
install after updating the source code, without doing make
uninstall first, will have to manually remove the old installed
plugin files from the installation prefix, or they will get
'Cannot register existing type' critical warnings.
Most of the docbook-based documentation (FAQ, Application
Development Manual, Plugin Writer's Guide, design documents)
has been converted to markdown and moved into a new gst-docs
module. The gtk-doc library API references and the plugins
documentation are still built as part of the source modules
though.
GStreamer core now optionally uses libunwind and libdw to
generate backtraces. This is useful for tracer plugins used
during debugging and development.
There is a new libgstbadallocators-1.0 library in gst-plugins-bad
(which may go away again in future releases once the
GstPhysMemoryAllocator interface API has been validated by more
users).
gst-omx and gstreamer-vaapi modules can now also be built using
the Meson build system.
The qtkitvideosrc element for macOS was removed. The API is
deprecated since 10.9 and it wasn't shipped in the binaries
since a few releases.
Platform-specific improvements Android
androidmedia: add support for VP9 video decoding/encoding and
Opus audio decoding (where supported)
OS/X and iOS
avfvideosrc, which represents an iPhone camera or, on a Mac, a
screencapture session, so far allowed you to select an input
device by device index only. New API adds the ability to select
the position (front or back facing) and device-type (wide angle,
telephoto, etc.). Furthermore, you can now also specify the
orientation (portrait, landscape, etc.) of the videostream.
Bugs fixed in 1.12
More than 635 bugs have been fixed during the development of 1.12.
This list does not include issues that have been cherry-picked into
the stable 1.10 branch and fixed there as well, all fixes that ended
up in the 1.10 branch are also included in 1.12.
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. Stable 1.12 branch
After the 1.12.0 release there will be several 1.12.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 1.12.x bug-fix releases
will be made from the git 1.12 branch, which is a stable branch.
1.12.0
1.12.0 was released on 4th May 2017.
1.12.1
The first 1.12 bug-fix release (1.12.1) was released on 20 June
2017. This release only contains bugfixes and it should be safe to
update from 1.12.x. Major bugfixes in 1.12.1
Various fixes for crashes, assertions, deadlocks and memory
leaks Fix for regression when seeking to the end of ASF files
Fix for regression in (raw)videoparse that caused it to omit
video metadata Fix for regression in discoverer that made it
show more streams than actually available Numerous bugfixes to
the adaptive demuxer base class and the DASH demuxer Various
playbin3/urisourcebin related bugfixes Vivante DirectVIV (imx6)
texture uploader works with single-plane (e.g. RGB) video formats
now Intel Media SDK encoder now outputs valid PTS and keyframe
flags OpenJPEG2000 plugin can be loaded again on MacOS and
correctly displays 8 bit RGB images now Fixes to DirectSound
source/sink for high CPU usage and wrong latency/buffer size
calculations gst-libav was updated to ffmpeg n3.3.2 ... and
many, many more!
1.12.2
The second 1.12 bug-fix release (1.12.2) was released on 14 July
2017. This release only contains bugfixes and it should be safe to
update from 1.12.x. Major bugfixes in 1.12.2
Various fixes for crashes, assertions, deadlocks and memory
leaks Regression fix for playback of live HLS streams Regression
fix for crash when playing back a tunneled RTSP stream Regression
fix for playback of RLE animations in MOV containers Regression
fix for RTP GSM payloading producing corrupted output Major
bugfixes to the MXF demuxer, mostly related to seeking and fixes
to the frame reordering handling in the MXF muxer and demuxer
Fix for playback of mono streams on MacOS More fixes for index
handling of ASF containers Various fixes to adaptivedemux, DASH
and HLS demuxers Fix deadlock in gstreamer-editing-services
during class initialization ... and many, many more!
# 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/)*
libnice 0.1.10 (2015-01-28)
===========================
Fix bug on component change on the sink
libnice 0.1.9 (2015-01-28)
==========================
Make it possible to statically build the GStreamer plugins
Bug fixes, in particular fix compatibility with coTurn servers
Documentation fixes
libnice 0.1.6 (2014-04-28)
==========================
API: nice_agent_restart_stream() to do a ICE restart on a single strema
API: nice_component_state_to_string() to get a printable name for a component
state
API: nice_agent_forget_relays() to forget the relays set for a
specific component, along with nice_agent_restart_stream(), it allows
changing the current relay without dropping the connection.
It is now possible to add relays after the initial candidate gathering.
Many bug fixes
wiz
adam
ryoon
rillig
leot
maya
prlw1
joerg
obache
richard
drochner