GStreamer 1.14 Release Notes
GStreamer 1.14.0 was originally released on 19 March 2018.
The latest bug-fix release in the 1.14 series is 1.14.1 and was
released on 17 May 2018.
See https://gstreamer.freedesktop.org/releases/1.14/ for the latest
version of this document.
Last updated: Thursday 17 May 2018, 12:00 UTC (log)
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
WebRTC support: real-time audio/video streaming to and from web browsers
Experimental support for the next-gen royalty-free AV1 video codec
Video4Linux: encoding support, stable element names and faster device probing
Support for the Secure Reliable Transport (SRT) video streaming protocol
RTP Forward Error Correction (FEC) support (ULPFEC)
RTSP 2.0 support in
rtspsrc and gst-rtsp-server
ONVIF audio backchannel support in gst-rtsp-server and
playbin3 gapless playback and pre-buffering support
tee, our stream splitter/duplication element, now does allocation query
aggregation which is important for efficient data handling and zero-copy
QuickTime muxer has a new prefill recording mode that allows file import
in Adobe Premiere and FinalCut Pro while the file is still being written.
rtpjitterbuffer fast-start mode and timestamp offset adjustment smoothing
souphttpsrc connection sharing, which allows for connection reuse, cookie sharing, etc.
nvdec: new plugin for hardware-accelerated video decoding using the NVIDIA NVDEC API
Adaptive DASH trick play support
ipcpipeline: new plugin that allows splitting a pipeline across
gobject-introspection annotation improvements for large parts of the library API
GStreamer C# bindings have been revived and seen many updates and fixes
The externally maintained GStreamer Rust bindings had many
usability improvements and cover most of the API now. Coinciding with the
1.14 release, a new release with the 1.14 API additions
Major new features and changes
There is now basic support for WebRTC in GStreamer in form of a new
webrtcbin element and a webrtc support library.
This allows you to build applications that set up connections with and stream
to and from other WebRTC peers, whilst leveraging all of the usual GStreamer
features such as hardware-accelerated encoding and decoding, OpenGL integration,
zero-copy and embedded platform support. And it's easy to build and integrate
into your application too!
WebRTC enables real-time communication of audio, video and data with web
browsers and native apps, and it is supported or about to be support by recent
versions of all major browsers and operating systems.
GStreamer's new WebRTC implementation uses libnice for
Interactive Connectivity Establishment (ICE) to figure out the best way to
communicate with other peers, punch holes into firewalls, and traverse NATs.
The implementation is not complete, but all the basics are there, and the
code sticks fairly close to the PeerConnection API.
Where functionality is missing it should be fairly obvious where it needs to
For more details, background and example code, check out Nirbheek's blog post
GStreamer has grown a WebRTC implementation, as well
as Matthew's GStreamer WebRTC talk from last year's
GStreamer Conference in Prague.
webrtcbin handles the transport aspects of webrtc connections
(see WebRTC section above for more details)
srtsrc elements for the Secure Reliable Transport (SRT)
video streaming protocol, which aims to be easy to use whilst striking a new
balance between reliability and latency for low latency video streaming use
cases. More details about SRT and the implementation in GStreamer in Olivier's
blog post SRT in GStreamer.
av1dec elements providing experimental support for the
next-generation royalty free video AV1 codec, alongside Matroska
support for it.
hlssink2 is a rewrite of the existing
hlssink element, but
unlike its predecessor
hlssink2 takes elementary streams as input and
handles the muxing to MPEG-TS internally. It also leverages
internally to do the splitting. This allows more control over the chunk
splitting and sizing process and relies less on the co-operation of an
upstream muxer. Different to the old
hlssink it also works with
pre-encoded streams and does not require close interaction with an upstream
audiolatency is a new element for measuring audio latency
end-to-end and is useful to measure roundtrip latency including both the
GStreamer-internal latency as well as latency added by external components
'fakevideosink is basically a null sink for video data and
very similar to
fakesink, only that it will answer allocation queries and
will advertise support for various video-specific things such
GstVideoOverlayCompositionMeta like a normal video
sink would. This is useful for throughput testing and testing the zero-copy
path when creating a new pipeline.
ipcpipeline: new plugin that allows the splitting of a pipeline into
multiple processes. Usually a GStreamer pipeline runs in a single process
and parallelism is achieved by distributing workloads using multiple threads.
This means that all elements in the pipeline have access to all the other
elements' memory space however, including that of any libraries used. For
security reasons one might therefore want to put sensitive parts of a
pipeline such as DRM and decryption handling into a separate process to
isolate it from the rest of the pipeline. This can now be achieved with the
ipcpipeline plugin. Check out George's blog post
ipcpipeline: Splitting a GStreamer pipeline into multiple processes
or his lightning talk from last year's
GStreamer Conference in Prague for all the gory details.
proxysrc are new elements to
pass data from one pipeline to another within the same process, very similar
to the existing
inter elements, but not limited to raw audio and video
data. These new proxy elements are very special in how they work under the
hood, which makes them extremely powerful, but also dangerous if not used
with care. The reason for this is that it's not just data that's passed
from sink to src, but these elements basically establish a two-way wormhole
that passes through queries and events in both directions, which means caps
negotiation and allocation query driven zero-copy can work through this
wormhole. There are scheduling considerations as well:
everything into the
proxysrc pipeline directly from the
streaming thread. There is a
queue element inside
proxysrc to decouple
the source thread from the sink thread, but that queue is not unlimited, so
it is entirely possible that the
proxysink pipeline thread gets stuck in
proxysrc pipeline, e.g. when that pipeline is paused or stops consuming
data for some other reason. This means that one should always shut down down
proxysrc pipeline before shutting down the
proxysink pipeline, for
example. Or at least take care when shutting down pipelines. Usually this is
not a problem though, especially not in live pipelines. For more information
see Nirbheek's blog post Decoupling GStreamer Pipelines,
and also check out out the new
ipcpipeline plugin for
sending data from one process to another process (see above).
lcms is a new LCMS-based ICC color profile correction element
openmptdec is a new OpenMPT-based decoder for module music
formats, such as S3M, MOD, XM, IT. It is built on top of a new
GstNonstreamAudioDecoder base class which aims to unify handling of
files which do not operate a streaming model. The
wildmidi plugin has also
been revived and is also implemented on top of this new base class.
The curl plugin has gained a new
element, which is useful for testing HTTP protocol version 2.0 amongst
msdk plugin has gained a MPEG-2 video decoder(msdkmpeg2dec),
VP8 decoder(msdkvp8dec) and a VC1/WMV decoder(msdkvc1dec)
Noteworthy new API
GstPromise provides future/promise-like functionality. This is
used in the GStreamer WebRTC implementation.
GstReferenceTimestampMeta is a new meta that
allows you to attach additional reference timestamps to a buffer. These
timestamps don't have to relate to the pipeline clock in any way. Examples
of this could be an NTP timestamp when the media was captured, a frame
counter on the capture side or the (local) UNIX timestamp when the media
was captured. The decklink elements make use of this.
GstVideoRegionOfInterestMeta: it's now possible to attach
generic free-form element-specific parameters to a region of interest meta,
for example to tell a downstream encoder to use certain codec parameters for
a certain region.
gst_bus_get_pollfd can be used to obtain a file
descriptor for the bus that can be
poll()-ed on for new messages. This
is useful for integration with non-GLib event loops.
gst_get_main_executable_path() can be used by wrapper plugins that need to
find things in the directory where the application executable is located. In
the same vein,
GST_PLUGIN_DEPENDENCY_FLAG_PATHS_ARE_RELATIVE_TO_EXE can be
used to signal that plugin dependency paths are relative to the main executable.
pad templates can be told about the
GType of the pad subclass of the pad
GstPadTemplate API API or the
gst-inspect-1.0 will use this information to print
new convenience functions to iterate over element pads without using the
appsrc have gained support for buffer lists:
GstBaseSrc subclasses can use
and applications can use
to push a buffer list into
GstHarness unit test harness has a couple of new convenience
functions to retrieve all pending data in the harness in form of a single
chunk of memory.
GstAudioStreamAlign is a new helper object for audio
elements that handles discontinuity detection and sample alignment. It will
align samples after the previous buffer's samples, but keep track of the
divergence between buffer timestamps and sample position (jitter). If it
exceeds a configurable threshold the alignment will be reset. This simply
factors out code that was duplicated in a number of elements into a common
GstVideoEncoder base class implements Quality of Service
(QoS) now. This is disabled by default and must be opted in by setting the
"qos" property, which will make the base class gather statistics about the
real-time performance of the pipeline from downstream elements (usually sinks
that sync the pipeline clock). Subclasses can then make use of this by checking
whether input frames are late already using
If late, they can just drop them and skip encoding in the hope that the pipeline
will catch up.
GstVideoOverlay interface gained a few helper functions
for installing and handling a
"render-rectangle" property on elements that
implement this interface, so that this functionality can also be used from
the command line for testing and debugging purposes. The property wasn't added
to the interface itself as that would require all implementors to provide
it which would not be backwards-compatible.
A new base class,
non-stream audio decoders was added to gst-plugins-bad. This base-class is
meant to be used for audio decoders that require the whole stream to be
loaded first before decoding can start. Examples of this are module formats
(MOD/S3M/XM/IT/etc), C64 SID tunes, video console music files (GYM/VGM/etc),
MIDI files and others. The new
openmptdec element is based on this.
Full list of API new in 1.14:
New RTP features and improvements
rtpulpfecdec are new
elements that implement Generic Forward Error Correction (FEC) using Uneven
Level Protection (ULP) as described in RFC 5109. This can be
used to protect against certain types of (non-bursty) packet loss, and
important packets such as those containing codec configuration data or
key frames can be protected with higher redundancy. Equally, packets that
are not particularly important can be given low priority or not be protected
at all. If packets are lost, the receiver can then hopefully restore the lost
packet(s) from the surrounding packets which were received. This is an
alternative to, or rather complementary to, dealing with packet loss using
retransmission (rtx). GStreamer has had retransmission support for a long
time, but Forward Error Correction allows for different trade-offs: The
advantage of Forward Error Correction is that it doesn't add latency, whereas
retransmission requires at least one more roundtrip to request and hopefully
receive lost packets; Forward Error Correction increases the required
bandwidth however, even in situations where there is no packet loss at all,
so one will typically want to fine-tune the overhead and mechanisms used
based on the characteristics of the link at the time.
New Redundant Audio Data (RED) encoders and decoders for RTP as per
RFC 2198 are also provided (
for chrome webrtc compatibility, as chrome will wrap ULPFEC-protected streams
in RED packets, and such streams need to be wrapped and unwrapped in order
to use ULPFEC with chrome.
a few new buffer flags for FEC support:
be used to mark important buffers, e.g. to flag RTP packets carrying keyframes
or codec setup data for RTP Forward Error Correction purposes, or to prevent
still video frames from being dropped by elements due to QoS. There already
GST_RTP_BUFFER_FLAG_REDUNDANT is used to
signal internally that a packet represents a redundant RTP packet and used
rtpstorage to hold back the packet and use it only for recovery from
packet loss. Further work is still needed in payloaders to make use of these.
rtpbin now has an option for increasing timestamp offsets gradually:
Sudden large changes to the internal
ts_offset may cause timestamps to
move backwards and may also cause visible glitches in media playback. The new
"max-ts-offset" properties let the
application control the rate to apply changes to
ts_offset. There have
also been some
BYE handling improvements in
rtpjitterbuffer has a new fast start mode: in many scenarios the jitter
buffer will have to wait for the full configured latency before it can start
outputting packets. The reason for that is that it often can't know what
the sequence number of the first expected RTP packet is, so it can't know
whether a packet earlier than the earliest packet received will still arrive
in future. This behaviour can now be bypassed by setting the
property to the number of consecutive packets needed to start, and the
jitter buffer will start output packets as soon as it has N consecutive
packets queued internally. This is particularly useful to get a first video
frame decoded and rendered as quickly as possible.
rtpL8depay provide RTP payloading and depayloading for
8-bit raw audio
New element features
playbin3 has gained support or gapless playback via the
signal where users can set the uri for the next item to play. For non-live
streams this will be emitted as soon as the first uri has finished
downloading, so with sufficiently large buffers it is now possible to
pre-buffer the next item well ahead of time (unlike
playbin where there
would not be a lot of time between
"about-to-finish" emission and the end
of the stream). If the stream format of the next stream is the same as that
of the previous stream, the data will be concatenated via the
element. Whether this will result in true gaplessness depends on the
container format and codecs used, there might still be codec-related gaps
between streams with some codecs.
tee now does allocation query aggregation, which is important for
zero-copy and efficient data handling, especially for video. Those
who want to drop allocation queries on purpose can use the
"drop-allocation" property for that instead.
audioconvert now has a
"mix-matrix" property, which obsoletes
audiomixmatrix element. There's also mix matrix support in
the audio conversion and channel mixing API.
"insert-vui" property to disable VUI (Video Usability
Information) parameter insertion into the stream, which allows creation
of streams that are compatible with certain legacy hardware decoders that
will refuse to decode in certain combinations of resolution and
VUI parameters; the max. allowed number of B-frames was also increased
from 4 to 16.
dvdlpcmdec: has gained support for Blu-Ray audio LPCM.
appsrc has gained support for buffer lists (see above) and also seen
some other performance improvements.
flvmux has been ported to the GstAggregator base class which means it
can work in defined-latency mode with live input sources and continue
streaming if one of the inputs stops producing data.
jpegenc has gained a
"snapshot" property just like
pngenc to make it
easier to output just a single encoded frame.
jpegdec will now handle interlaced MJPEG streams properly and also handles
frames without an End of Image marker better.
v4l2: There are now video encoders for VP8, VP9, MPEG4, and H263. The v4l2
video decoder handles dynamic resolution changes, and the video4linux device
provider now does much faster device probing. The plugin also no longer uses
the libv4l2 library by default, as it has prevented a lot of interesting use
DMABuf, usage of
TRY_FMT. As the libv4l2 library
is totally inactive and not really maintained, we decided to disable it. This
might affect a small number of cheap/old webcams with custom vendor formats
for which we do not provide conversion in GStreamer. It is possible to
re-enable support for libv4l2 at run-time however, by setting the environment
rtspsrc now has support for RTSP protocol version 2.0 as well as ONVIF audio
backchannels (see below for more details). It also sports a new
signal for "manually" checking a TLS certificate for validity. It now also
prints RTSP/SDP messages to the gstreamer debug log instead of stdout.
shout2send now uses non-blocking I/O and has a configurable network
splitmuxsink has gained a
"split-now" action signal and new
"use-robust-muxing" properties. If robust
muxing is enabled, it will check and set the muxer's reserved space
properties if present. This is primarily for use with mp4mux's robust
qtmux has a new prefill recording mode which sets up a moov header with
the correct sample positions beforehand, which then allows software like
Adobe Premiere and FinalCut Pro to import the files while they are still
being written to. This only works with constant framerate I-frame only
streams, and for now only support for ProRes video and raw audio is
implemented. Adding support for additional codecs is just a matter of
defining appropriate maximum frame sizes though.
qtmux also supports writing of svmi atoms with stereoscopic video
information now. Trak timescales can be configured on a per-stream basis
"trak-timescale" property on the sink pads. Various new formats
can be muxed: MPEG layer 1 and 2, AC3 and Opus, as well as PNG and VP9.
souphttpsrc now does connection sharing by default: it shares its
with other elements in the same pipeline via a
GstContext if possible
(session-wide settings are all the defaults). This allows for connection
reuse, cookie sharing, etc. Applications can also force a context to use.
In other news, HTTP headers received from the server are posted as element
messages on the bus now for easier diagnostics, and it's also possible now
to use other types of proxy servers such as SOCKS4 or SOCKS5 proxies, support
for which is implemented directly in gio. Before only HTTP proxies were
matroskamux will now refuse caps changes of input
streams at runtime. This isn't really supported with these containers
(or would have to be implemented differently with a considerable effort)
and doesn't produce valid and spec-compliant files that will play everywhere.
So if you can't guarantee that the input caps won't change, use a container
format that does support on the fly caps changes for a stream such as
MPEG-TS or use
splitmuxsink which can start a new file when the caps
change. What would happen before is that e.g.
would simply send new SPS/PPS inband even for AVC format, which would then
get muxed into the container as if nothing changed. Some decoders will
handle this just fine, but that's often more luck than by design. In any
case, it's not right, so we disallow it now.
matroskamux has Table of Content (TOC) support now (chapters etc.) and
matroskademux TOC support has been improved.
matroskademux has also
seen seeking improvements searching for the right cluster and position.
videocrop now uses
GstVideoCropMeta if downstream supports it, which
means cropping can be handled more efficiently without any copying.
compositor now has support for crossfade blending, which can be used
via the new
"crossfade-ratio" property on the sink pads.
avwait element has a new
"end-timecode" property and posts
"avwait-status" element messages now whenever
avwait starts or stops
passing through data (e.g. because target-timecode and end-timecode
respectively have been reached).
'alsamidisrc' element has been broken for many many years and has now been
repaired allowing live capture from your MIDI HW.
h265parse will try harder to make upstream output the same
caps as downstream requires or prefers, thus avoiding unnecessary conversion.
The parsers also expose chroma format and bit depth in the caps now.
dtls elements now longer rely on or require the application to run a
GLib main loop that iterates the default main context (GStreamer plugins
should never rely on the application running a GLib main loop).
openh264enc allows to change the encoding bitrate dynamically at runtime now
nvdec is a new plugin for hardware-accelerated video decoding using
the NVIDIA NVDEC API (which replaces the old VDPAU API which is no longer
supported by NVIDIA)
The NVIDIA NVENC hardware-accelerated video encoders now support dynamic
bitrate and preset reconfiguration and support the
I420 4:2:0 video format.
It's also possible to configure the gop size via the new
The MPEG-TS muxer and demuxer (
tsdemux) now have support for JPEG2000
jpeg2000parse support 2-component images now (gray with alpha),
jpeg2000parse has gained limited support for conversion between JPEG2000
stream-formats. (JP2, J2C, JPC) and also extracts more details such as
colorimetry, interlace-mode, field-order, multiview-mode and chroma siting.
decklink plugin for Blackmagic capture and playback cards have seen
decklinkvideosrc now put hardware reference
timestamp on buffers in form of
This can be useful to know on multi-channel cards which frames from
different channels were captured at the same time.
decklinkvideosink has gained support for Decklink hardware keying with
two new properties (
"keyer-level") to control the
built-in hardware keyer of Decklink cards.
decklinkaudiosink has been re-implemented around
GstAudioBaseSink base class, since the Decklink APIs don't fit
very well with the
GstAudioBaseSink APIs, which used to cause various
problems due to inaccuracies in the clock calculations. Problems were
audio drop-outs and A/V sync going wrong after pausing/seeking.
support for more than 16 devices, without any artificial limit
work continued on the
msdk plugin for Intel's Media SDK which enables
hardware-accelerated video encoding and decoding on Intel graphics hardware
on Windows or Linux. Added the video memory, buffer pool, and context/session
sharing support which helps to improve the performance and resource utilization.
Rendernode support is in place which helps to avoid the constraint of having
a running graphics server as DRM-Master. Encoders are exposing a number rate control
algorithms now. More encoder tuning options like trellis-quantiztion (h264),
slice size control (h264), B-pyramid prediction(h264), MB-level bitrate control,
frame partitioning and adaptive I/B frame insertion were added, and more pixel formats
and video codecs are supported now. The encoder now also handles
force-key-unit events and can insert frame-packing SEIs for side-by-side
and top-bottom stereoscopic 3D video.
dashdemux can now do adaptive trick play of certain types of DASH streams,
meaning it can do fast-forward/fast-rewind of normal (non-I frame only)
streams even at high speeds without saturating network bandwidth or exceeding
decoder capabilities. It will keep statistics and skip keyframes or fragments
as needed. See Sebastian's blog post DASH trick-mode playback in GStreamer
for more details. It also supports webvtt subtitle streams now and has seen
improvements when seeking in live streams.
kmssink has seen lots of fixes and improvements in this cycle, including:
Raspberry Pi (vc4) and Xilinx DRM driver support
"render-rectangle" property that can be used from the command line
as well as
Plugin and library moves
MPEG-1 audio (mp1, mp2, mp3) decoders and encoders moved to -good
Following the expiration of the last remaining mp3 patents in most jurisdictions,
and the termination
of the mp3 licensing program, as well as the decision by certain distros
to officially start shipping full mp3 decoding and encoding support, these
plugins should now no longer be problematic for most distributors and have
therefore been moved from -ugly and -bad to gst-plugins-good. Distributors
can still disable these plugins if desired.
In particular these are:
GstAggregator moved from -bad to core
GstAggregator has been moved from gst-plugins-bad to the base
library in GStreamer and is now stable API.
GstAggregator is a new base class for mixers and muxers that
have to handle multiple input pads and aggregate streams into one output
stream. It improves upon the existing GstCollectPads API in
that it is a proper base class which was also designed with live streaming
in mind. GstAggregator subclasses will operate in a mode with
defined latency if any of the inputs are live streams. This ensures that
the pipeline won't stall if any of the inputs stop producing data, and that
the configured maximum latency is never exceeded.
audiointerleave moved from -bad to -base
GstAudioAggregator is a new base class for raw audio mixers
and muxers and is based on GstAggregator (see above). It provides
defined-latency mixing of raw audio inputs and ensures that the pipeline won't
stall even if one of the input streams stops producing data.
As part of the move to stabilise the API there were some last-minute API
changes and clean-ups, but those should mostly affect internal elements.
It is used by the
audiomixer element, which is a replacement
for 'adder', which did not handle live inputs very well and did not align input
streams according to running time.
audiomixer should behave
much better in that respect and generally behave as one would expected in most
audiointerleave replaces the 'interleave'
element which did not handle live inputs or non-aligned inputs very robustly.
GstAudioAggregator and its subclases have gained support
for input format conversion, which does not include sample rate conversion
though as that would add additional latency. Furthermore,
GAP events are
now handled correctly.
We hope to move the video equivalents (
to -base in the next cycle, i.e. for 1.16.
GStreamer OpenGL integration library and plugin moved from -bad to -base
The GStreamer OpenGL integration library and
have moved from gst-plugins-bad to -base and are now part of the stable API
canon. Not all OpenGL elements have been moved; a few had to be left behind
in gst-plugins-bad in the new
because they depend on the
GstVideoAggregator base class which we were not
able to move in this cycle. We hope to reunite these elements with the rest of
their family for 1.16 though.
This is quite a milestone, thanks to everyone who worked to make this happen!
Qt QML and GTK plugins moved from -bad to -good
The Qt QML-based
qmlgl plugin has moved to -good and provides a
qmlglsink video sink element as well as a
qmlglsink renders video into a
qmlglsrc captures a window from a QML view and feeds it as video
into a pipeline for further processing. Both elements leverage GStreamer's
OpenGL integration. In addition to the move to -good the following features were
A proxy object is now used for thread-safe access to the QML widget which
prevents crashes in corner case scenarios: QML can destroy the video widget
at any time, so without this we might be left with a dangling pointer.
EGL is now supported with the X11 backend, which works e.g. on Freescale imx6
The GTK+ plugin has also moved from -bad to -good. It
gtkglsink which both render
video into a
Cairo for rendering the video, which will work
everywhere in all scenarios but involves an extra memory copy, whereas
gtkglsink fully leverages GStreamer's OpenGL integration, but
might not work properly in all scenarios, e.g. where the OpenGL driver does not
properly support multiple sharing contexts in different threads; on Linux
Nouveau is known to be broken in this respect, whilst NVIDIA's proprietary
drivers and most other drivers generally work fine, and the experience with
Intel's driver seems to be mixed; some proprietary embedded Linux drivers
don't work; macOS works.
GstPhysMemoryAllocator interface moved from -bad to -base
GstPhysMemoryAllocator is a marker interface for
allocators with physical address backed memory.
sunaudio plugin was removed, since it couldn't ever have been built
or used with GStreamer 1.0, but no one even noticed in all these years.
schroedinger-based Dirac encoder/decoder plugin has been removed,
as there is no longer any upstream or anyone else maintaining it. Seeing
that it's quite a fringe codec it seemed best to simply remove it.
- some MPEG video parser API in the API unstable codecutils library in
gst-plugins-bad was removed after having been deprecated for 5 years.
The video support library has gained support for a few new pixel
NV16_10LE32: 10-bit variant of
NV16, packed into 32bit words (plus 2 bits padding)
NV12_10LE32: 10-bit variant of
NV12, packed into 32bit words (plus 2 bits padding)
GRAY10_LE32: 10-bit grayscale, packed in 32bit words (plus 2 bits padding)
GstDiscoverer have seen stability improvements
in corner cases such as shutdown while still starting up or shutdown in error
cases (hat tip to the oss-fuzz project).
floating reference handling was inconsistent and has been cleaned up across
the board, including annotations. This solves various long-standing memory
leaks in language bindings, which e.g. often caused elements and pads to be
gobject-introspection annotation improvements for large parts of the
library API, including nullability of return types and function parameters,
correct types (e.g. strings vs. filenames), ownership transfer, array length
parameters, etc. This allows to use bigger parts of the GStreamer API to be
allows static bindings (e.g. C#, Rust, Vala) to autogenerate more API
bindings without manual intervention.
The GStreamer OpenGL integration library has moved to gst-plugins-base and
is now part of our stable API.
new Mesa3D GBM backend. On devices with working libdrm support, it is
possible to use Mesa3D's GBM library to set up an EGL context directly on top
of KMS. This makes it possible to use the GStreamer OpenGL elements without a
windowing system if a libdrm- and Mesa3D-supported GPU is present.
Prefer wayland display over X11: As most Wayland compositors support
XWayland, the X11 backend would get selected.
gldownload can export
dmabufs now, and
glupload will advertise dmabuf
as caps feature.
Tracing framework and debugging improvements
New memory ringbuffer based debug logger, useful for long-running
applications or to retrieve diagnostics when encountering an error. The
GStreamer debug logging system provides in-depth debug logging about what
is going on inside a pipeline. When enabled, debug logs are usually
written into a file, printed to the terminal, or handed off to a log
handler installed by the application. However, at higher debug levels the
volume of debug output quickly becomes unmanageable, which poses a problem
in disk-space or bandwidth restricted environments or with long-running
pipelines where a problem might only manifest itself after multiple days.
In those situations, developers are usually only interested in the most
recent debug log output. The new in-memory ringbuffer logger makes this
easy: just installed it with
and retrieve logs with
when needed. It is possible to limit the memory usage per thread and set a
timeout to determine how long messages are kept around. It was always possible
to implement this in the application with a custom log handler of course, this
just provides this functionality as part of GStreamer.
'fakevideosink is a null sink for video data that advertises
video-specific metas and behaves like a video sink. See above for more details.
gst_util_dump_buffer() prints the content of a buffer to stdout.
gst_state_change_get_name() print pad link
return values and state change transition values as strings.
The latency tracer has seen a few improvements: trace records now contain
timestamps which is useful to plot things over time, and downstream
synchronisation time is now excluded from the measured values.
Miniobject refcount tracing and logging was not entirley thread-safe, there
were duplicates or missing entries at times. This has now been made reliable.
netsim element, which can be used to simulate network jitter, packet
reordering and packet loss, received new features and improvements: it can now
also simulate network congestion using a token bucket algorithm. This can be
enabled via the
"max-kbps" property. Packet reordering can be disabled
now via the
"allow-reordering" property: Reordering of packets is not very
common in networks, and the delay functions will always introduce reordering
if delay > packet-spacing, so by setting
guarantee that the packets are in order, while at the same time introducing
delay/jitter to them. By using the new
"delay-distribution" property the
user can control how the delay applied to delayed packets is distributed: This
is either the uniform distribution (as before) or the normal distribution;
in addition there is also the gamma distribution which simulates the delay
on wifi networks better.
gst-inspect-1.0 now prints pad properties for elements that have pad
subclasses with special properties, such as
This only works for elements that use the newly-added
API or the
convenience function to tell GStreamer about the special pad subclass.
gst-launch-1.0 now generates a gstreamer pipeline diagram (.dot file)
whenever SIGHUP is sent to it on Linux/*nix systems.
gst-discoverer-1.0 can now analyse live streams such as
GStreamer RTSP server
Initial support for RTSP protocol version 2.0 was
added, which is to the best of our knowledge the first RTSP 2.0
ONVIF audio backchannel support. This is an extension specified
by ONVIF that allows RTSP clients (e.g. a control room operator) to send audio
back to the RTSP server (e.g. an IP camera). Theoretically this could have
been done also by using the
RECORD method of the RTSP protocol, but ONVIF
chose not to do that, so the backchannel is set up alongside the other
streams. Format negotiation needs to be done out of band, if needed. Use the
new ONVIF-specific subclasses GstRTSPOnvifServer and
GstRTSPOnvifMediaFactory to enable this functionality.
The internal server streaming pipeline is now dynamically reconfigured on
PLAY based on the transports needed. This means that the server no longer
adds the pipeline plumbing for all possible transports from the start, but
only if needed as needed. This improves performance and memory footprint.
rtspclientsink has gained an
"accept-certificate" signal for manually
checking a TLS certificate for validity.
Fix keep-alive/timeout issue for certain clients using TCP interleave as
transport who don't do keep-alive via some other method such as periodic
OPTION requests. We now put netaddress metas on the packets from the
TCP interleaved stream, so can map RTCP packets to the right stream in the
server and can handle them properly.
Language bindings improvements: in general there were quite a few
improvements in the gobject-introspection annotations, but we also
extended the permissions API which was not usable from bindings before.
Fix corner case issue where the wrong mount point was found when there were
multiple mount points with a common prefix.
Improve DMABuf's usage, both upstream and dowstream, and
memory:DMABuf caps feature is also negotiated when the
dmabuf-based buffer cannot be mapped onto user-space.
VA initialization was fixed when it is used in headless systems.
VA display sharing, through
GstContext, among the pipeline, has
been improved, adding the possibility to the application share its
VA display (external display) via
VA display cache was removed.
libva's log messages are now redirected into the GStreamer log handler.
Decoders improved their upstream re-negotiation by avoiding to
re-instantiate the internal decoder if stream caps are compatible
with the previous one.
When downstream doesn't support
GstVideoMeta and the decoded
frames don't have standard strides, they are copied onto system
H.264 decoder has a
low-latency property, for live streams which
doesn't conform the H.264 specification but still it is required to
push the frames to downstream as soon as possible.
As part of the Google Summer of Code 2017 the H.264 decoder drops
MVC and SVC frames when
base-only property is enabled.
Added support for libva-2.0 (VA-API 1.0).
H.264 and H.265 encoders handle Region-Of-Interest metas by adding a
delta-qp for every rectangle within the frame specified by those
Encoders for H.264 and H.265 set the media profile by the downstream
H.264 encoder inserts an AU delimiter for each encoded frame when
aud property is enabled (it is only available for certain
drivers and platforms).
H.264 encoder supports for P and B hierarchical prediction modes.
All encoders handles a
quality-level property, which is a number
from 1 to 8, where a lower number means higher quality, but slower
processing, and vice-versa.
VP8 and VP9 encoders support constant bit-rate mode (CBR).
VP8, VP9 and H.265 encoders support variable bit-rate mode (VBR).
GstGLUploadTextureMeta handling for EGL backends.
H.265 encoder can configure its number of reference frames via the
Add H.264 encoder
mbbrc property, which controls the macro-block
bitrate as auto, on or off.
Add H.264 encoder
temporal-levels property, to select the number
of temporal levels to be included.
Add to H.264 and H.265 encoders the properties
qp-ib, to handle the QP (quality parameter) difference between
the I and P frames, and the I and B frames, respectively.
vaapisink was demoted to marginal rank on Wayland because COGL
cannot display YUV surfaces.
More details in Víctor's blog post GStreamer VA-API 1.14: what’s new?.
GStreamer Editing Services and NLE
Handle crossfade in complex scenarios by using the new
Add API allowing to stop using proxies for clips in the timeline
Allow management of none square pixel aspect ratios by allowing application to deal
with them in the way they want
Misc fixes around the timeline editing API
Handle running scenarios on live pipelines (in the "content sense", not the GStreamer one)
Implement RTSP support with a basic server based on gst-rtsp-server, and add RTSP
1.0 and 2.0 integration tests
Implement a plugin that allows users to implement configurable tests. It currently
can check if a particular element is added a configurable number of time in the
pipeline. In the future that plugin should allow us to implement specific tests of
any kind in a descriptive way
verbosity configuration which behaves in a similare way as the
verbose flags allowing the informations to be outputed on any running pipeline when
Misc optimization in the launcher, making the tests run much faster.
GStreamer C# bindings
Port to the meson build system, autotools support has been
Use a new GlibSharp version, set as a meson
Update wrapped API to GStreamer 1.14
Removed the need for "glue" code
Provide a nuget
Misc API fixes
Build and Dependencies
the new WebRTC support in gst-plugins-bad depends on the GStreamer elements
that ship as part of libnice, and libnice version 1.1.14 is required. Also
gst-plugins-bad no longer depends on the libschroedinger Dirac codec library.
srtp plugin can now also be built against libsrtp2.
some plugins and libraries have moved between modules, see the Plugin and
library moves section above, and their respective dependencies have moved
with them of course, e.g. the GStreamer OpenGL integration support library
and plugin is now in gst-plugins-base, and mpg123, LAME and twoLAME based
audio decoder and encoder plugins are now in gst-plugins-good.
Unify static and dynamic plugin interface and remove plugin specific static
build option: Static and dynamic plugins now have the same interface. The
--enable-shared toggle is sufficient. This
allows building static and shared plugins from the same object files, instead
of having to build everything twice.
The default plugin entry point has changed. This will only affect plugins
that are recompiled against new GStreamer headers. Binary plugins using the
old entry point will continue to work. However, plugins that are recompiled
must have matching plugin names in
GST_PLUGIN_DEFINE and filenames, as
the plugin entry point for shared plugins is now deduced from the plugin
filename. This means you can no longer have a plugin called
in a file called
libfoobar.so or such, the plugin filename needs to match.
This might cause problems with some external third party plugin modules
when they get rebuilt against GStreamer 1.14.
Note to packagers and distributors
A number of libraries, APIs and plugins moved between modules and/or libraries
in different modules between version 1.12.x and 1.14.x, see the Plugin and
library moves section above. Some APIs have seen minor ABI changes in the
course of moving them into the stable APIs section.
This means that you should try to ensure that all major GStreamer modules are
synced to the same major version (1.12 or 1.13/1.14) and can only be upgraded
in lockstep, so that your users never end up with a mix of major versions on
their system at the same time, as this may cause breakages.
Also, plugins compiled against >= 1.14 headers will not load with
GStreamer <= 1.12 owing to a new plugin entry point (but plugin binaries
built against older GStreamer versions will continue to load with newer
versions of GStreamer of course).
There is also a small structure size related ABI breakage introduced in the
gst-plugins-bad codecparsers library between version 1.13.90 and 1.13.91. This
should "only" affect gstreamer-vaapi, so anyone who ships the release
candidates is advised to upgrade those two modules at the same time.
ahcsrc (Android camera source) does autofocus now
macOS and iOS
- no major changes in macOS and iOS support, only bugfixes
wasapi plugin was rewritten and should not
only be usable now, but in top shape and suitable for low-latency use cases.
The Windows Audio Session API (WASAPI) is Microsoft's most modern method for
talking with audio devices, and now that the
wasapi plugin is up to scratch
it is preferred over the
directsound plugin. The ranks of the
wasapisrc elements have been updated to reflect this. Further
support for more than 2 channels
"low-latency" property to enable low-latency operation
(which should always be safe to enable)
support for the AudioClient3 API which is only available on Windows 10:
wasapisink this will be used automatically if available; in
it will have to be enabled explicitly via the
as capturing audio with low latency and without glitches seems to require
setting the realtime priority of the entire pipeline to "critical", which
cannot be done from inside the element, but has to be done in the
set realtime thread priority to avoid glitches
allow opening devices in exclusive mode, which provides much lower latency
compared to shared mode where WASAPI's engine period is 10ms. This can
be activated via the
Also see Nirbheek's blog post Low Latency Audio on Windows with GStreamer.
There are now
GstDeviceProvider implementations for the
directsound plugins, so it's now possible to discover both audio sources
and audio sinks on Windows via the
debug log timestamps are now higher granularity owing to
now being used as fallback in
gst_utils_get_timestamp(). Before that, there
would sometimes be 10-20 lines of debug log output sporting the same timestamp.
Aaron Boxer, Adrián Pardini, Adrien SCH, Akinobu Mita, Alban Bedel,
Alessandro Decina, Alex Ashley, Alicia Boya García, Alistair Buxton,
Alvaro Margulis, Anders Jonsson, Andreas Frisch, Andrejs Vasiljevs,
Andrew Bott, Antoine Jacoutot, Antonio Ospite, Antoni Silvestre,
Anton Obzhirov, Anuj Jaiswal, Arjen Veenhuizen, Arnaud Bonatti, Arun Raghavan,
Ashish Kumar, Aurélien Zanelli, Ayaka, Branislav Katreniak, Branko Subasic,
Brion Vibber, Carlos Rafael Giani, Cassandra Rommel, Chris Bass,
Chris Paulson-Ellis, Christoph Reiter, Claudio Saavedra, Clemens Lang,
Cyril Lashkevich, Daniel van Vugt, Dave Craig, Dave Johnstone, David Evans,
David Schleef, Deepak Srivastava, Dimitrios Katsaros, Dmitry Zhadinets,
Dongil Park, Dustin Spicuzza, Eduard Sinelnikov, Edward Hervey, Enrico Jorns,
Eunhae Choi, Ezequiel Garcia, fengalin, Filippo Argiolas, Florent Thiéry,
Florian Zwoch, Francisco Velazquez, François Laignel, fvanzile,
George Kiagiadakis, Georg Lippitsch, Graham Leggett, Guillaume Desmottes,
Gurkirpal Singh, Gwang Yoon Hwang, Gwenole Beauchesne, Haakon Sporsheim,
Haihua Hu, Håvard Graff, Heekyoung Seo, Heinrich Fink, Holger Kaelberer,
Hoonhee Lee, Hosang Lee, Hyunjun Ko, Ian Jamison, James Stevenson,
Jan Alexander Steffens (heftig), Jan Schmidt, Jason Lin, Jens Georg,
Jeremy Hiatt, Jérôme Laheurte, Jimmy Ohn, Jochen Henneberg, John Ludwig,
John Nikolaides, Jonathan Karlsson, Josep Torra, Juan Navarro,
Juan Pablo Ugarte, Julien Isorce, Jun Xie, Jussi Kukkonen, Justin Kim,
Lasse Laursen, Lubosz Sarnecki, Luc Deschenaux, Luis de Bethencourt,
Marcin Lewandowski, Mario Alfredo Carrillo Arevalo, Mark Nauwelaerts,
Martin Kelly, Matej Knopp, Mathieu Duponchelle, Matteo Valdina,
Matt Fischer, Matthew Waters, Matthieu Bouron, Matthieu Crapet, Matt Staples,
Michael Catanzaro, Michael Olbrich, Michael Shigorin, Michael Tretter,
Michał Dębski, Michał Górny, Michele Dionisio, Miguel París, Mikhail Fludkov,
Munez, Nael Ouedraogo, Neos3452, Nicholas Panayis, Nick Kallen, Nicola Murino,
Nicolas Dechesne, Nicolas Dufresne, Nirbheek Chauhan, Ognyan Tonchev,
Ole André Vadla Ravnås, Oleksij Rempel, Olivier Crête, Omar Akkila,
Orestis Floros, Patricia Muscalu, Patrick Radizi, Paul Kim, Per-Erik Brodin,
Peter Seiderer, Philip Craig, Philippe Normand, Philippe Renon, Philipp Zabel,
Pierre Pouzol, Piotr Drąg, Ponnam Srinivas, Pratheesh Gangadhar, Raimo Järvi,
Ramprakash Jelari, Ravi Kiran K N, Reynaldo H. Verdejo Pinochet,
Rico Tzschichholz, Robert Rosengren, Roland Peffer, Руслан Ижбулатов,
Sam Hurst, Sam Thursfield, Sangkyu Park, Sanjay NM, Satya Prakash Gupta,
Scott D Phillips, Sean DuBois, Sebastian Cote, Sebastian Dröge,
Sebastian Rasmussen, Sejun Park, Sergey Borovkov, Seungha Yang, Shakin Chou,
Shinya Saito, Simon Himmelbauer, Sky Juan, Song Bing, Sreerenj Balachandran,
Stefan Kost, Stefan Popa, Stefan Sauer, Stian Selnes, Thiago Santos,
Thibault Saunier, Thijs Vermeir, Tim Allen, Tim-Philipp Müller, Ting-Wei Lan,
Tomas Rataj, Tom Bailey, Tonu Jaansoo, U. Artie Eoff, Umang Jain,
Ursula Maplehurst, VaL Doroshchuk, Vasilis Liaskovitis,
Víctor Manuel Jáquez Leal, vijay, Vincent Penquerc'h, Vineeth T M,
Vivia Nikolaidou, Wang Xin-yu (王昕宇), Wei Feng, Wim Taymans, Wonchul Lee,
Xabier Rodriguez Calvar, Xavier Claessens, XuGuangxin, Yasushi SHOJI,
Yi A Wang, Youness Alaoui,
... and many others who have contributed bug reports, translations, sent
suggestions or helped testing.
Bugs fixed in 1.14
More than 800 bugs have been fixed during
the development of 1.14.
This list does not include issues that have been cherry-picked into the
stable 1.12 branch and fixed there as well, all fixes that ended up in the
1.12 branch are also included in 1.14.
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 1.14 branch
After the 1.14.0 release there will be several 1.14.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 1.14.x bug-fix releases will be made from the git 1.14 branch,
which is a stable branch.
1.14.0 was released on 19 March 2018.
The first 1.14 bug-fix release (1.14.1) was released on 17 May 2018.
This release only contains bugfixes and it should be safe to update from 1.14.0.
Noteworthy bugfixes in 1.14.1
- GstPad: Fix race condition causing the same probe to be called multiple times
- Fix occasional deadlocks on windows when outputting debug logging
- Fix debug levels being applied in the wrong order
- GIR annotation fixes for bindings
- audiomixer, audioaggregator: fix some negotiation issues
- gst-play-1.0: fix leaving stdin in non-blocking mode after exit
- flvmux: wait for caps on all input pads before writing header even if source is live
- flvmux: don't wake up the muxer unless there is data, fixes busy looping if there's no input data
- flvmux: fix major leak of input buffers
- rtspsrc, rtsp-server: revert to RTSP RFC handling of sendonly/recvonly attributes
- rtpvrawpay: fix payloading with very large mtu sizes where everything fits into a single RTP packet
- v4l2: Fix hard-coded enabled v4l2 probe on Linux/ARM
- v4l2: Disable DMABuf for emulated formats when using libv4l2
- v4l2: Always set colorimetry in
- asfdemux: Set stream-format field for H264 streams and handle H.264 in bytestream format
- x265enc: Fix tagging of keyframes on output buffers
- ladspa: Fix critical during plugin load on Windows
- decklink: Fix COM initialisation on Windows
- h264parse: fix re-use across pipeline stop/restart
- mpegtsmux: fix force-keyframe event handling and PCR/PMT changes that would confuse some players with generated HLS streams
- adaptivedemux: Support period change in live playlist
- rfbsrc: Fix support for applevncserver and support NULL pool in decide_allocation
- jpegparse: Fix APP1 marker segment parsing
- h265parse: Make caps writable before modifying them, fixes criticals
- fakevideosink: request an extra buffer if enable-last-sample is enabled
- wasapisrc: Don't provide a clock based on WASAPI's clock
- wasapi: Only use audioclient3 when low-latency, as it might otherwise glitch with slow CPUs or VMs
- wasapi: Don't derive device period from latency time, should make it more robust against glitches
- audiolatency: Fix wave detection in buffers and avoid bogus pts values while starting
- msdk: fix plugin load on implementations with only HW support
- msdk: dec: set framerate to the driver only if provided, not in 0/1 case
- msdk: Don't set extended coding options for JPEG encode
- rtponviftimestamp: fix state change function init/reset causing races/crashes on shutdown
- decklink: fix initialization failure in windows binary
- ladspa: Fix critical warnings during plugin load on Windows and fix dependencies in meson build
- gl: fix cross-compilation error with viv-fb
- qmlglsink: make work with eglfs_kms
- rtspclientsink: Don't deadlock in preroll on early close
- rtspclientsink: Fix client ports for the RTCP backchannel
- rtsp-server: Fix session timeout when streaming data to client over TCP
- vaapiencode: h264: find best profile in those available, fixing negotiation errors
- vaapi: remove custom GstGL context handling, use GstGL instead. Fixes GL Context sharing with WebkitGtk on wayland
- gst-editing-services: various fixes
- gst-python: bump pygobject req to 3.8; fix GstPad.set_query_function(); dist autogen.sh and configure.ac in tarball
- g-i: pick up GstVideo-1.0.gir from local build directory in GstGL build
- g-i: update constant values for bindings
- avoid duplicate symbols in plugins across modules in static builds
- ... and many, many more!
Cerbero build tool and packaging changes in 1.14.1
Toolchain updates on iOS and Android necessitated a fairly large number of
changes in our cerbero build tool used to create our binary packages for the
various platforms we support:
- Add support for Ubuntu 18.04 in cerbero
- Fix generation of fat shared libraries on macOS
- gnutls: also rename assembly functions on macos/ios to fix link errors
- gnutls: fix assembly symbol names for windows x86
- openssl: fix linking on android/armv7
- openssl: fix linker issue with Android NDK's r16 binutils
- ffmpeg: disable asm for android x86 to fix issues when linking with apps
- x264: disable asm for android x86 to fix issues when linking with apps
- gnutls: rename private symbols for armv8, x86 to not conflict with openssl
- mpg123: disable assembly on android/x86 to fix linker problems with relocations
- Check built version while loading recipe and rebuild if needed
- Fix packaging of libgcc_s_sjlj which was missing in Windows packages
- Make not-found in library search fatal so we don't accidentally ship broken packages
- ship the proxy plugin which was new in 1.14
- Fix git commands accidentally pulling in locally built libraries and failing
Contributors to 1.14.1
Antonio Ospite, Aurélien Zanelli, Brendan Shanks, Carlos Rafael Giani,
Edward Hervey, Emilio Pozuelo Monfort, Enrique Ocaña González, Garima Gaur,
Georg Lippitsch, Guillaume Desmottes, Havard Graff, Hoonhee Lee, Hyunjun Ko,
James Stevenson, Jan Alexander Steffens (heftig), Jan Schmidt, Joakim Johansson,
Jun Xie, Kai Kang, Kirill Marinushkin, Mark Nauwelaerts, Matej Knopp,
Mathieu Duponchelle, Matthew Waters, Matthias Fend, Michael Olbrich,
Mikhail Fludkov, Nicolas Dufresne, Nirbheek Chauhan, Olivier Crête, Omar Akkila,
Patrik Nilsson, Philippe Normand, Pierre Labastie, Sebastian Dröge, Seungha Yang,
Sreerenj Balachandran, Stian Selnes, Takeshi Sato, Thibault Saunier,
Tim-Philipp Müller, U. Artie Eoff, Víctor Manuel Jáquez Leal, Vivia Nikolaidou,
Whoopie, Xabier Rodriguez Calvar, Xavier Claessens, Zeeshan Ali, and
List of bugs fixed in 1.14.1
For a full list of bugfixes see Bugzilla. Note that this is
not the full list of changes. For the full list of changes please refer to the
GIT logs or ChangeLogs of the particular modules.
The second 1.14 bug-fix release (1.14.2) is scheduled to be released around
This release only contains bugfixes and it should be safe to update from 1.14.x.
webrtcdsp element (which is unrelated to the newly-landed GStreamer
webrtc support) is currently not shipped as part of the Windows binary
packages due to a build system issue.
gst-libav module currently won't build against the newly-released
ffmpeg 4.0 (as in RPM Fusion for Fedora 28). Use the internal ffmpeg copy
instead, if you build using autotools.
Schedule for 1.16
Our next major feature release will be 1.16, and 1.15 will be the unstable
development version leading up to the stable 1.16 release. The development
of 1.15/1.16 will happen in the git master branch.
The plan for the 1.16 development cycle is yet to be confirmed, but it is
expected that feature freeze will be around August 2018 followed by several
1.15 pre-releases and the new 1.16 stable release in September.
1.16 will be backwards-compatible to the stable 1.14, 1.12, 1.10, 1.8, 1.6,
1.4, 1.2 and 1.0 release series.
These release notes have been prepared by Tim-Philipp Müller with
contributions from Sebastian Dröge, Sreerenj Balachandran, Thibault Saunier
and Víctor Manuel Jáquez Leal.
License: CC BY-SA 4.0