I am trying to create an rtsp restream server using rust and intersink/intersrc. The idea is to have a base pipeline that connects to the source and potentially modifies the image and multiple client pipelines that are created, once new rtsp clients connect.
I did this in Rust:
use anyhow::Error;
use derive_more::derive::{Display, Error};
use gstreamer::prelude::*;
use gstreamer as gst;
use gstreamer_rtsp_server::prelude::*;
use futures::prelude::*;
#[derive(Debug, Display, Error)]
#[display("Could not get mount points")]
struct NoMountPoints;
#[tokio::main]
async fn main() -> Result<(), Error> {
gst::init()?;
let main_loop = glib::MainLoop::new(None, false);
let src_pipeline = gst::parse::launch("rtspsrc location=rtsp://localhost:8554/test latency=0 ! queue ! rtph264depay ! h264parse ! avdec_h264 ! videoconvert ! videoscale ! videorate ! video/x-raw,framerate=10/1,width=640,heigh=480 ! queue ! intersink producer-name=test").unwrap();
src_pipeline.set_state(gstreamer::State::Playing)?;
let mut stream = src_pipeline.bus().unwrap().stream();
while let Some(msg) = stream.next().await {
use gstreamer::MessageView;
match msg.view() {
MessageView::StateChanged(..) => {
if let Some(o) = msg.src() {
if o.name() == src_pipeline.name()
{
let k: gstreamer::State = msg.structure().unwrap().value("new-state").unwrap().get()?;
if k == gstreamer::State::Playing {
break;
}
}
}
}
_ => ()
}
}
let server = gstreamer_rtsp_server::RTSPServer::default();
let mounts = gstreamer_rtsp_server::RTSPMountPoints::default();
server.set_mount_points(Some(&mounts));
server.set_service("8555");
let mounts = server.mount_points().ok_or(NoMountPoints)?;
let factory = gstreamer_rtsp_server::RTSPMediaFactory::default();
factory.set_launch("intersrc producer-name=test ! queue ! videoconvert ! x264enc speed-preset=veryfast tune=zerolatency ! queue ! rtph264pay name=pay0 pt=96 config-interval=-1");
factory.set_suspend_mode (gstreamer_rtsp_server::RTSPSuspendMode::None);
mounts.add_factory("/test", factory);
let id = server.attach(None)?;
println!(
"Stream ready at rtsp://127.0.0.1:{}/test",
server.bound_port()
);
main_loop.run();
id.remove();
Ok(())
}
(this is a shortened down version to show the problem)
When trying to connect to this server using ffmpeg:
ffmpeg -i rtsp://localhost:8555/test -hide_banner -c:v copy -rtsp_transport tcp "%Y-%m-%d-%H-%M-%S_Test.mp4"
I get a ton of “Dropping old item buffer” Warnings from the underlying appsrc element of the intersrc.
The reason I am using rsinter in contrast to intervideosink is, because I require potential metadata from the source and secondly because I want to move the encoding to the base-pipeline as well.
I tried the following:
- Setting the underlying appsrc of intersrc to
is-live=trueanddo-timestamp=true - Increasing the buffer size
- Setting the pipelines to identical clocks
- Replacing the rtspsrc with
videotestsrc is-live=true
But (as one may have noticed), Im tapping in the dark.
Is there something I have missed?