| Commit message (Collapse) | Author | Age | Files | Lines |
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Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1871>
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LDAC is an audio coding technology developed by Sony that enables the
transmission of High-Resolution (Hi-Res) audio contents over Bluetooth.
Currently Adaptive Bit Rate (ABR) as supported by libldac encoder is not
implemented.
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1621>
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Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1730>
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Wrapper on the iSAC reference encoder and decoder from webrtc,
see https://en.wikipedia.org/wiki/Internet_Speech_Audio_Codec
Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/-/merge_requests/1124>
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Status:
- scan QR code with low resolution
- Scan barcode with high resolution
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The SVT-HEVC (Scalable Video Technology[0] for HEVC) Encoder is an
open source video coding technology[1] that is highly optimized for
Intel Xeon Scalable processors and Intel Xeon D processors.
[0] https://01.org/svt
[1] https://github.com/OpenVisualCloud/SVT-HEVC
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The provider for now only detects and handles rtsp devices, but
more protocols should be easy to add.
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This patch introduces the bootstrap code from the AVTP plugin (plugin
definition and init) as well as the build system files. Upcoming patches
will introduce payloaders, source and sink elements provided by the AVTP
plugin. These elements can be utilized by a GStreamer pipeline to
implement TSN audio/video applications.
Regarding the plugin build system files, both autotools and meson files
are introduced. The AVTP plugin is landed in ext/ since it has an
external dependency on libavtp, an opensource AVTP packetization
library. For further information about libavtp check [1].
[1] https://github.com/AVnu/libavtp
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Fixes https://gitlab.freedesktop.org/gstreamer/gst-plugins-bad/issues/965
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Merged into the existing opengl plugin in -base.
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Not so useful now that all effort is focused on AV1 instead.
Was only ever enabled with --enable-experimental anyway.
Fixes #848
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Replaced by gme plugin. Fixes #467
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The wpe element is used to produce a video texture representing a web page
rendered off-screen by WPE. This element can be used to overlay HTML on top of
another video stream for instance.
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- Add meson build definitions
- Add necessary API decorators
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We now have options for all plugins, so we will just disable these in
the cerbero recipe instead. These require external deps, so they won't
affect gst-build either.
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The only plugins remaining are those that haven't been ported to Meson
yet, and msdk. Also, the tests are still automagic.
https://bugzilla.gnome.org/show_bug.cgi?id=795107
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The rest will be converted later, these are necessary for gst-build to
set options correctly.
https://bugzilla.gnome.org/show_bug.cgi?id=795107
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Add meson build system support to bs2b plugin.
https://bugzilla.gnome.org/show_bug.cgi?id=794346
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No upstream (website disappeared), no maintainer, and
pretty much a fringe format anyway.
https://bugzilla.gnome.org/show_bug.cgi?id=776215
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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
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https://bugzilla.gnome.org/show_bug.cgi?id=754094
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Note that I only support lilv 0.22+ for now, because it's simple and old
enough. If anyone needs to use the 0.16 support it will need to be
added.
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https://bugzilla.gnome.org/show_bug.cgi?id=784160
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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
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https://bugzilla.gnome.org/show_bug.cgi?id=784646
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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
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https://bugzilla.gnome.org/show_bug.cgi?id=774223
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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
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https://bugzilla.gnome.org/show_bug.cgi?id=774493
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