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README.md
gst-meet: Integrate Jitsi Meet conferences with GStreamer pipelines
Note: gst-meet is in an alpha state and is under active development. The command-line options and the lib-gst-meet API are subject to change, and some important features (TURN, RTX, TCC, simulcast) are not yet fully functional.
gst-meet provides a library and tool for integrating Jitsi Meet conferences with GStreamer pipelines. You can pipe audio and video into a conference as a participant, and pipe out other participants' audio and video streams.
Thanks to GStreamer's flexibility and wide range of plugins, this enables many new possibilities.
Installation
You will need the dependencies glib
, gstreamer
and libnice
, as well as any GStreamer plugins you want to use in your pipelines, and a Rust toolchain (rustup is the easiest way to install one).
Then: cargo install gst-meet
To integrate gst-meet into your own application, look at lib-gst-meet
.
Pipeline Structure
You can pass two different pipeline fragments to gst-meet.
--send-pipeline
is for sending audio and video. If it contains an element named audio
, this audio will be streamed to the conference. The audio codec must be 48kHz Opus. If it contains an element named video
, this video will be streamed to the conference. The video codec must match the codec passed to --video-codec
, which is VP8 by default.
--recv-pipeline-participant-template
is for receiving audio and video from other participants. This pipeline will be created once for each other participant in the conference. If it contains an element named audio
, the participant's audio (48kHz Opus) will be sent to that element. If it contains an element named video
, the participant's video (encoded with the codec selected by --video-codec
) will be sent to that element. The strings {jid}
, {jid_user}
, {participant_id}
and {nick}
are replaced in the template with the participant's full JID, user part, MUC JID resource part (a.k.a. participant/occupant ID) and nickname respectively.
Examples
A few examples of gst-meet
usage are below. The GStreamer reference provides full details on available pipeline elements.
gst-meet --help
lists full usage information.
Stream an Opus audio file to the conference. This is very efficient; the Opus data in the file is streamed directly without transcoding:
gst-meet --web-socket-url=wss://your.jitsi.domain/xmpp-websocket \
--xmpp-domain=your.jitsi.domain \
--room-name=roomname \
--send-pipeline="filesrc location=sample.opus ! queue ! oggdemux name=audio"
Stream a FLAC audio file to the conference, transcoding it to Opus:
gst-meet --web-socket-url=wss://your.jitsi.domain/xmpp-websocket \
--xmpp-domain=your.jitsi.domain \
--room-name=roomname \
--send-pipeline="filesrc location=shake-it-off.flac ! queue ! flacdec ! audioconvert ! audioresample ! opusenc name=audio"
Stream a .webm file containing VP8 video and Vorbis audio to the conference. This pipeline passes the VP8 stream through efficiently without transcoding, and transcodes the audio from Vorbis to Opus:
gst-meet --web-socket-url=wss://your.jitsi.domain/xmpp-websocket \
--xmpp-domain=your.jitsi.domain \
--room-name=roomname \
--send-pipeline="filesrc location=big-buck-bunny_trailer.webm ! queue ! matroskademux name=demuxer
demuxer.video_0 ! queue name=video
demuxer.audio_0 ! queue ! vorbisdec ! audioconvert ! audioresample ! opusenc name=audio"
Stream the default macOS video & audio inputs to the conference, encoding as VP8 and Opus, display incoming video streams and play back incoming audio (a very basic, but completely native, Jitsi Meet conference!):
gst-meet --web-socket-url=wss://your.jitsi.domain/xmpp-websocket \
--xmpp-domain=your.jitsi.domain \
--room-name=roomname \
--send-pipeline="avfvideosrc ! queue ! videoconvert ! vp8enc buffer-size=1000 deadline=1 name=video
osxaudiosrc ! queue ! audioconvert ! audioresample ! opusenc name=audio" \
--recv-pipeline-participant-template="opusdec name=audio ! autoaudiosink
vp8dec name=video ! videoconvert ! autovideosink"
Record a .webm file for each other participant, containing VP8 video and Opus audio, without needing to do any transcoding:
gst-meet --web-socket-url=wss://your.jitsi.domain/xmpp-websocket \
--xmpp-domain=your.jitsi.domain \
--room-name=roomname \
--recv-pipeline-participant-template="webmmux name=muxer ! filesink location={participant_id}.webm
opusparse name=audio ! muxer.audio_0
capsfilter caps=video/x-vp8 name=video ! muxer.video_0"
Play a YouTube video in the conference. By requesting Opus audio and VP9 video from YouTube, and setting the Jitsi Meet video codec to VP9, no transcoding is necessary:
YOUTUBE_URL="https://www.youtube.com/watch?v=vjV_2Ri2rfE"
gst-meet --web-socket-url=wss://your.jitsi.domain/xmpp-websocket \
--xmpp-domain=your.jitsi.domain \
--room-name=roomname \
--video-codec=vp9 \
--send-pipeline="curlhttpsrc location=\"$(youtube-dl -g $YOUTUBE_URL -f 'bestaudio[acodec=opus]')\" ! queue ! matroskademux name=audiodemux
curlhttpsrc location=\"$(youtube-dl -g $YOUTUBE_URL -f 'bestvideo[vcodec=vp9]')\" ! queue ! matroskademux name=videodemux
audiodemux.audio_0 ! queue ! clocksync name=audio
videodemux.video_0 ! queue ! clocksync name=video"
Debugging
It can sometimes be tricky to get GStreamer pipeline syntax and structure correct. To help with this, you can try setting the GST_DEBUG
environment variable (for example, 3
is modestly verbose, while 6
produces copious per-packet output). You can also set GST_DEBUG_DUMP_DOT_DIR
to the relative path to a directory (which must already exist). .dot
files containing the pipeline graph will be saved to this directory, and can be converted to .png
with the dot
tool from GraphViz; for example dot filename.dot -Tpng > filename.png
.
License
gst-meet
, lib-gst-meet
, nice
and nice-sys
are licensed under either of
- Apache License, Version 2.0, (LICENSE-APACHE or http://www.apache.org/licenses/LICENSE-2.0)
- MIT license (LICENSE-MIT or http://opensource.org/licenses/MIT)
at your option.
The dependency xmpp-parsers
is licensed under the Mozilla Public License, Version 2.0, https://www.mozilla.org/en-US/MPL/2.0/
The dependency gstreamer
is licensed under the GNU Lesser General Public License, Version 2.1, https://www.gnu.org/licenses/old-licenses/lgpl-2.1.en.html.
Contribution
Any kinds of contributions are welcome as a pull request.
Unless you explicitly state otherwise, any contribution intentionally submitted for inclusion in these crates by you, as defined in the Apache-2.0 license, shall be dual licensed as above, without any additional terms or conditions.
Acknowledgements
gst-meet
development is sponsored by AVStack. We provide globally-distributed, scalable, managed Jitsi Meet backends.