vomit prevention from stream finish with clients still connected

This commit is contained in:
2026-07-06 13:01:49 +01:00
parent 528dff7e8a
commit 7d61d7b425
3 changed files with 20 additions and 13 deletions
+3 -7
View File
@@ -46,8 +46,8 @@ pub struct AppState {
#[derive(Clone, Debug)]
pub enum StreamCodec {
H264,
H265,
AV1, // Unsupported
H265, // Very shity suport
AV1,
}
pub struct StreamSession {
@@ -61,11 +61,7 @@ pub struct StreamSession {
#[tokio::main]
async fn main() -> Result<(), Box<dyn Error>> {
let env_filter = EnvFilter::builder()
.with_default_directive(LevelFilter::DEBUG.into())
.from_env()
// .parse("")
.unwrap();
let env_filter = EnvFilter::builder().from_env().unwrap();
tracing_subscriber::fmt().with_env_filter(env_filter).init();
+16 -5
View File
@@ -11,6 +11,7 @@ use std::{
use tokio::{
net::UdpSocket,
sync::mpsc::{Receiver, Sender},
time::sleep,
};
use tracing::{debug, error, info, warn};
@@ -137,7 +138,8 @@ impl Webrtc {
});
if !video_has_pts {
warn!(
request_id, stream_id,
request_id,
stream_id,
"no video codec negotiated — browser likely doesn't support {:?}; rejecting offer",
stream_codec
);
@@ -222,7 +224,10 @@ impl Webrtc {
video_pt = Some(params.pt());
video_mid = Some(ma.mid);
} else {
warn!(stream_id, "video writer has no payload params — codec not negotiated");
warn!(
stream_id,
"video writer has no payload params — codec not negotiated"
);
}
} else {
warn!(stream_id, mid = ?ma.mid, "no writer for video mid");
@@ -301,13 +306,16 @@ impl Webrtc {
for _ in 0..8 {
match stream.try_recv() {
Ok(frame) => {
debug!(stream_id, is_keyframe = frame.is_keyframe, ts = frame.timestamp_ms, bytes = frame.data.len(), "WebRTC received video frame");
if !saw_keyframe {
if !frame.is_keyframe {
continue;
}
saw_keyframe = true;
info!(stream_id, ts = frame.timestamp_ms, "first keyframe — starting RTP send");
info!(
stream_id,
ts = frame.timestamp_ms,
"first keyframe — starting RTP send"
);
}
let now = Instant::now();
let rtp_time =
@@ -316,7 +324,9 @@ impl Webrtc {
(Some(pt), Some(writer)) => {
match writer.write(pt, now, rtp_time, frame.data.to_vec()) {
Ok(_) => wrote_any = true,
Err(e) => error!(stream_id, "video RTP write error: {:?}", e),
Err(e) => {
error!(stream_id, "video RTP write error: {:?}", e)
}
}
}
_ => warn!(
@@ -330,6 +340,7 @@ impl Webrtc {
Err(async_broadcast::TryRecvError::Empty) => break,
Err(async_broadcast::TryRecvError::Closed) => {
warn!("video channel closed, stream ended");
sleep(Duration::from_secs(1)).await;
break;
}
Err(async_broadcast::TryRecvError::Overflowed(_)) => {