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examples: webrtc: rust: Set keyframe-max-dist=2000 and picture-id-mode=15-bit for VP8 and perfect-timestamps=true for audio
This makes it in sync with the C sendrecv and generally behaves better. Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3758>
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2 changed files with 3 additions and 10 deletions
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@ -157,9 +157,9 @@ impl App {
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// Create the GStreamer pipeline
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let pipeline = gst::parse_launch(
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&format!(
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"videotestsrc is-live=true ! vp8enc deadline=1 ! rtpvp8pay pt=96 ! tee name=video-tee ! \
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"videotestsrc is-live=true ! vp8enc deadline=1 keyframe-max-dist=2000 ! rtpvp8pay pt=96 picture-id-mode=15-bit ! tee name=video-tee ! \
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queue ! fakesink sync=true \
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audiotestsrc wave=ticks is-live=true ! opusenc ! rtpopuspay pt=97 ! application/x-rtp,encoding-name=OPUS ! tee name=audio-tee ! \
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audiotestsrc wave=ticks is-live=true ! opusenc perfect-timestamp=true ! rtpopuspay pt=97 ! application/x-rtp,encoding-name=OPUS ! tee name=audio-tee ! \
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queue ! fakesink sync=true \
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audiotestsrc wave=silence is-live=true ! audio-mixer. \
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audiomixer name=audio-mixer sink_0::mute=true ! audioconvert ! audioresample ! autoaudiosink \
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@ -192,13 +192,6 @@ impl App {
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// Channel for outgoing WebSocket messages from other threads
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let (send_ws_msg_tx, send_ws_msg_rx) = mpsc::unbounded::<WsMessage>();
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// Asynchronously set the pipeline to Playing
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pipeline.call_async(|pipeline| {
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pipeline
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.set_state(gst::State::Playing)
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.expect("Couldn't set pipeline to Playing");
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});
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let app = App(Arc::new(AppInner {
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pipeline,
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video_tee,
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@ -115,7 +115,7 @@ impl App {
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> {
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// Create the GStreamer pipeline
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let pipeline = gst::parse_launch(
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"videotestsrc pattern=ball is-live=true ! vp8enc deadline=1 ! rtpvp8pay name=vpay pt=96 ! webrtcbin. \
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"videotestsrc pattern=ball is-live=true ! vp8enc deadline=1 keyframe-max-dist=2000 ! rtpvp8pay name=vpay pt=96 picture-id-mode=15-bit ! webrtcbin. \
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audiotestsrc is-live=true ! opusenc perfect-timestamp=true ! rtpopuspay name=apay pt=97 ! application/x-rtp,encoding-name=OPUS ! webrtcbin. \
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webrtcbin name=webrtcbin"
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)?;
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