mirror of
https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs.git
synced 2024-10-23 10:54:08 +00:00
87c6719e1d
When congestion control is used for a session with multiple encoders, the default implementation simply divides the overall bitrate equally between encoders. This is not always desirable, and this patch exposes a new signal that users can register to, with two arguments: * The overall bitrate to allocate * A structure with an encoder.stream_name -> bitrate mapping Handlers should return a similar structure with a custom mapping. An example is also provided. Part-of: <https://gitlab.freedesktop.org/gstreamer/gst-plugins-rs/-/merge_requests/1792>
85 lines
2.7 KiB
Rust
85 lines
2.7 KiB
Rust
// The goal of this example is to demonstrate how to affect bitrate allocation
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// when congestion control is happening in a session with multiple encoders
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use anyhow::Error;
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use gst::glib;
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use gst::prelude::*;
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fn main() -> Result<(), Error> {
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gst::init()?;
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// Create a very simple webrtc producer, offering a single video stream
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let pipeline = gst::Pipeline::builder().build();
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let videotestsrc = gst::ElementFactory::make("videotestsrc").build()?;
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let queue = gst::ElementFactory::make("queue").build()?;
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let webrtcsink = gst::ElementFactory::make("webrtcsink").build()?;
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webrtcsink.connect_closure(
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"define-encoder-bitrates",
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false,
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glib::closure!(|_webrtcsink: &gst::Element,
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_consumer_id: &str,
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overall: i32,
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_in_structure: gst::Structure| {
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let out_s = gst::Structure::builder("webrtcsink/encoder-bitrates")
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.field(
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"video_0",
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overall.mul_div_round(75, 100).expect("should be scalable"),
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)
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.field(
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"video_1",
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overall.mul_div_round(25, 100).expect("should be scalable"),
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)
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.build();
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Some(out_s)
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}),
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);
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webrtcsink.set_property("run-signalling-server", true);
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webrtcsink.set_property("run-web-server", true);
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pipeline.add_many([&videotestsrc, &queue, &webrtcsink])?;
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gst::Element::link_many([&videotestsrc, &queue, &webrtcsink])?;
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let videotestsrc = gst::ElementFactory::make("videotestsrc").build()?;
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let queue = gst::ElementFactory::make("queue").build()?;
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pipeline.add_many([&videotestsrc, &queue])?;
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gst::Element::link_many([&videotestsrc, &queue, &webrtcsink])?;
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// Now we simply run the pipeline to completion
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pipeline.set_state(gst::State::Playing)?;
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let bus = pipeline.bus().expect("Pipeline should have a bus");
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for msg in bus.iter_timed(gst::ClockTime::NONE) {
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use gst::MessageView;
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match msg.view() {
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MessageView::Eos(..) => {
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println!("EOS");
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break;
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}
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MessageView::Error(err) => {
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pipeline.set_state(gst::State::Null)?;
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eprintln!(
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"Got error from {}: {} ({})",
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msg.src()
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.map(|s| String::from(s.path_string()))
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.unwrap_or_else(|| "None".into()),
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err.error(),
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err.debug().unwrap_or_else(|| "".into()),
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);
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break;
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}
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_ => (),
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}
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}
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pipeline.set_state(gst::State::Null)?;
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Ok(())
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}
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