SDP's are generated and consumed according to the W3C PeerConnection API
available from https://www.w3.org/TR/webrtc/
The SDP is either created initially from the connected
sink pads/attached transceivers as in the case of generating an offer or
intersected with the connected sink pads/attached transceivers as in
the case for creating an answer. In both cases, the rtp payloaded streams
sent by the peer are exposed as separate src pads.
The implementation supports trickle ICE, RTCP muxing, reduced size RTCP.
With contributions from:
Nirbheek Chauhan <nirbheek@centricular.com>
Mathieu Duponchelle <mathieu@centricular.com>
Edward Hervey <edward@centricular.com>
https://bugzilla.gnome.org/show_bug.cgi?id=792523
SRT[0] is an open source transport technology[1] that optimizes
streaming performance across unpredictable networks.
Although SRT is based on UDP, it works like connection-oriented
protocol. However, it doesn't mean that the SRT server or client
is necessarily to link to a receiver or a sender so, here, the
pairs of source and sink elements are introduced.
- srtserversink: SRT server to feed SRT stream
- srtclientsrc: SRT client to get SRT stream from srtserversink
- srtclientsink: SRT client to send SRT stream
- srtserversrc: SRT server to listen from srtclientsink
[0] https://github.com/Haivision/srt
[1] http://www.srtalliance.org/https://bugzilla.gnome.org/show_bug.cgi?id=785730
If they were not ported after 4+ years it seems unlikely that anybody is
ever going to need them again. They're still in the GIT history if
needed.
https://bugzilla.gnome.org/show_bug.cgi?id=774530
This was used by MSN messenger in prehistoric times, it's safe
to say no one needs or wants this any more these days. For
decoding old recordings there's still a decoder in ffmpeg.
https://bugzilla.gnome.org/show_bug.cgi?id=597616
It only offers one metric for now, "dssim", available if
https://github.com/pornel/dssim was installed on the system
at the time the plugin was compiled.
The spearman correlation for dssim against the TID2008 dataset
is 0.81, against 0.70 for the old ssim implementation, and
it runs 15 times faster.
https://bugzilla.gnome.org/show_bug.cgi?id=751324
This DSP library can be used to enhance voice signal for real time
communication call. In implements multiple filters like noise reduction,
high pass filter, echo cancellation, automatic gain control, etc.
The webrtcdsp element can be used along, or with the help of the
webrtcechoprobe if echo cancellation is enabled. The echo probe should
be placed as close as possible to the audio sink, while the DSP is
generally place close to the audio capture. For local testing, one can
use an echo loop pipeline like the following:
autoaudiosrc ! webrtcdsp ! webrtcechoprobe ! autoaudiosink
This pipeline should produce a single echo rather then repeated echo.
Those elements works if they are placed in the same top level pipeline.
https://bugzilla.gnome.org/show_bug.cgi?id=767800
Very much in the same spirit as the Gtk GL sink
Two things are provided
1. A QQuickItem subclass that renders out RGBA filled GstGLMemory
buffers that is instantiated from qml.
2. A sink element that will push buffers into (1)
To use
1. Declare the GstGLVideoItem in qml with an appropriate
objectName property set.
2. Get the aforementioned GstGLVideoItem from qml using something like
QQmlApplicationEngine engine;
engine.load(QUrl(QStringLiteral("qrc:/main.qml")));
QObject *rootObject = engine.rootObjects().first();
QQuickItem *videoItem = rootObject->findChild<QQuickItem *> ("videoItem");
3. Set the videoItem on the sink
https://bugzilla.gnome.org/show_bug.cgi?id=752185
gmyth seems to be unmaintained upstream, and no one has asked
for this to be ported for a very long time, so let's just
remove it. Neither debian nor Fedora seem to ship libgmyth
any longer, and in any case it's most likely deprecated by
the UPnP support in MythTV.
It's not developed any more and replaced by the
libschroedinger-based elements in gst-plugins-good.
(The libschroedinger 1.0.9 release notes state "This
is an exciting release: most of the encoding tools in
dirac-research have been ported over to Schrödinger, so
now schro has the same or better compression efficiency
as dirac-research.")
TRM IDs are MusicBrainz' old audio fingerprinting system from
Relatable, they were phased out in favour of MusicIPs PUIDs.
https://wiki.musicbrainz.org/History:TRM
