stuff and pain
This commit is contained in:
Nickiel12
parent
7b41a78fc5
commit
39f26c19c6
6 changed files with 373 additions and 216 deletions
172
Cargo.lock
generated
172
Cargo.lock
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@ -124,7 +124,7 @@ checksum = "965c2d33e53cb6b267e148a4cb0760bc01f4904c1cd4bb4002a085bb016d1490"
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@ -147,7 +147,7 @@ checksum = "7b18050c2cd6fe86c3a76584ef5e0baf286d038cda203eb6223df2cc413565f7"
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@ -170,7 +170,7 @@ checksum = "6e0c28dcc82d7c8ead5cb13beb15405b57b8546e93215673ff8ca0349a028107"
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[[package]]
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@ -295,9 +295,12 @@ dependencies = [
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@ -313,9 +316,9 @@ dependencies = [
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@ -327,12 +330,6 @@ version = "1.0.0"
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@ -409,9 +406,9 @@ dependencies = [
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@ -475,16 +472,6 @@ dependencies = [
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@ -508,7 +495,7 @@ checksum = "f46882e17999c6cc590af592290432be3bce0428cb0d5f8b6715e4dc7b383eb3"
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@ -584,7 +571,7 @@ checksum = "97369cbbc041bc366949bc74d34658d6cda5621039731c6310521892a3a20ae0"
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@ -751,7 +738,7 @@ checksum = "87750cf4b7a4c0625b1529e4c543c2182106e4dedc60a2a6455e00d212c489ac"
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@ -824,9 +811,9 @@ checksum = "40ecd4077b5ae9fd2e9e169b102c6c330d0605168eb0e8bf79952b256dbefffd"
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@ -837,9 +824,9 @@ dependencies = [
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[[package]]
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@ -854,27 +841,26 @@ dependencies = [
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"libc",
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[[package]]
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@ -1149,9 +1135,9 @@ checksum = "bbd2bcb4c963f2ddae06a2efc7e9f3591312473c50c6685e1f298068316e66fe"
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[[package]]
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"pin-utils",
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[[package]]
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"pest_meta",
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|
||||
"backtrace",
|
||||
"bytes",
|
||||
|
|
@ -2210,6 +2199,7 @@ dependencies = [
|
|||
"mio",
|
||||
"parking_lot",
|
||||
"pin-project-lite",
|
||||
"signal-hook-registry",
|
||||
"socket2",
|
||||
"tokio-macros",
|
||||
"windows-sys",
|
||||
|
|
@ -2223,7 +2213,7 @@ checksum = "693d596312e88961bc67d7f1f97af8a70227d9f90c31bba5806eec004978d752"
|
|||
dependencies = [
|
||||
"proc-macro2",
|
||||
"quote",
|
||||
"syn 2.0.72",
|
||||
"syn 2.0.75",
|
||||
]
|
||||
|
||||
[[package]]
|
||||
|
|
@ -2315,7 +2305,7 @@ checksum = "34704c8d6ebcbc939824180af020566b01a7c01f80641264eba0999f6c2b6be7"
|
|||
dependencies = [
|
||||
"proc-macro2",
|
||||
"quote",
|
||||
"syn 2.0.72",
|
||||
"syn 2.0.75",
|
||||
]
|
||||
|
||||
[[package]]
|
||||
|
|
@ -2433,9 +2423,9 @@ checksum = "d4c87d22b6e3f4a18d4d40ef354e97c90fcb14dd91d7dc0aa9d8a1172ebf7202"
|
|||
|
||||
[[package]]
|
||||
name = "unicode-xid"
|
||||
version = "0.2.4"
|
||||
version = "0.2.5"
|
||||
source = "registry+https://github.com/rust-lang/crates.io-index"
|
||||
checksum = "f962df74c8c05a667b5ee8bcf162993134c104e96440b663c8daa176dc772d8c"
|
||||
checksum = "229730647fbc343e3a80e463c1db7f78f3855d3f3739bee0dda773c9a037c90a"
|
||||
|
||||
[[package]]
|
||||
name = "universal-hash"
|
||||
|
|
@ -2492,7 +2482,6 @@ dependencies = [
|
|||
"anyhow",
|
||||
"async-channel",
|
||||
"config",
|
||||
"ctrlc",
|
||||
"gstreamer",
|
||||
"gstreamer-app",
|
||||
"log",
|
||||
|
|
@ -2511,7 +2500,18 @@ dependencies = [
|
|||
[[package]]
|
||||
name = "vcs-common"
|
||||
version = "0.1.0"
|
||||
source = "git+https://git.nickiel.net/VCC/vcs-common.git?branch=main#898d5181e08ed7cbfef745bf9036eae71a08be52"
|
||||
source = "git+https://git.nickiel.net/VCC/vcs-common.git?branch=main#1e37189866720e8d99b22492a78f47d220742831"
|
||||
dependencies = [
|
||||
"async-channel",
|
||||
"bincode",
|
||||
"futures-util",
|
||||
"serde",
|
||||
"serde_json",
|
||||
"tokio",
|
||||
"tokio-tungstenite",
|
||||
"tracing",
|
||||
"webrtc",
|
||||
]
|
||||
|
||||
[[package]]
|
||||
name = "version-compare"
|
||||
|
|
@ -2741,7 +2741,7 @@ dependencies = [
|
|||
"lazy_static",
|
||||
"libc",
|
||||
"log",
|
||||
"nix 0.26.4",
|
||||
"nix",
|
||||
"portable-atomic",
|
||||
"rand",
|
||||
"thiserror",
|
||||
|
|
@ -2928,7 +2928,7 @@ checksum = "fa4f8080344d4671fb4e831a13ad1e68092748387dfc4f55e356242fae12ce3e"
|
|||
dependencies = [
|
||||
"proc-macro2",
|
||||
"quote",
|
||||
"syn 2.0.72",
|
||||
"syn 2.0.75",
|
||||
]
|
||||
|
||||
[[package]]
|
||||
|
|
@ -2948,5 +2948,5 @@ checksum = "ce36e65b0d2999d2aafac989fb249189a141aee1f53c612c1f37d72631959f69"
|
|||
dependencies = [
|
||||
"proc-macro2",
|
||||
"quote",
|
||||
"syn 2.0.72",
|
||||
"syn 2.0.75",
|
||||
]
|
||||
|
|
|
|||
|
|
@ -8,7 +8,6 @@ edition = "2021"
|
|||
[dependencies]
|
||||
anyhow = "1.0.86"
|
||||
config = "0.14.0"
|
||||
ctrlc = "3.4.4"
|
||||
|
||||
gstreamer = { version = "0.23.0", features = ["v1_22"] }
|
||||
gstreamer-app = { version = "0.23.0", features = ["v1_22"] }
|
||||
|
|
@ -16,7 +15,7 @@ log = "0.4.22"
|
|||
serde = { version = "1.0.204", features = ["derive"] }
|
||||
serde_json = "1.0.122"
|
||||
snafu = "0.8.4"
|
||||
tokio = "1.39.2"
|
||||
tokio = { version = "1.39.2", features = ["signal"] }
|
||||
tokio-tungstenite = "0.23.1"
|
||||
toml = "0.8.19"
|
||||
tracing = "0.1.40"
|
||||
|
|
|
|||
45
src/gst.rs
Normal file
45
src/gst.rs
Normal file
|
|
@ -0,0 +1,45 @@
|
|||
use gstreamer::{
|
||||
self as gst,
|
||||
prelude::{Cast, GstBinExtManual},
|
||||
ElementFactory,
|
||||
};
|
||||
use gstreamer_app as gst_app;
|
||||
|
||||
pub struct Pipeline {
|
||||
pub pipeline: gst::Pipeline,
|
||||
|
||||
pub sink: gst_app::AppSink,
|
||||
}
|
||||
|
||||
pub fn new_pipeline() -> Pipeline {
|
||||
let pipeline = gst::Pipeline::builder()
|
||||
.name("camera_to_rtp_pipeine")
|
||||
.build();
|
||||
|
||||
let source = ElementFactory::make("v4l2src").build().unwrap();
|
||||
|
||||
let video_convert = ElementFactory::make("videoconvert").build().unwrap();
|
||||
|
||||
let vp8enc = ElementFactory::make("vp8enc").build().unwrap();
|
||||
|
||||
let rtp = ElementFactory::make("rtpvp8pay").build().unwrap();
|
||||
|
||||
let app_sink = gst_app::AppSink::builder().build();
|
||||
|
||||
pipeline
|
||||
.add_many([
|
||||
&source,
|
||||
&video_convert,
|
||||
&vp8enc,
|
||||
&rtp,
|
||||
app_sink.upcast_ref(),
|
||||
])
|
||||
.expect("Could not add all the stuff to the pipeline");
|
||||
|
||||
gst::Element::link_many(&[&source, &video_convert, &vp8enc, &rtp]).unwrap();
|
||||
|
||||
Pipeline {
|
||||
pipeline,
|
||||
sink: app_sink,
|
||||
}
|
||||
}
|
||||
206
src/main.rs
206
src/main.rs
|
|
@ -1,12 +1,29 @@
|
|||
use std::sync::{atomic::AtomicBool, Arc};
|
||||
use std::sync::{atomic::{AtomicBool, Ordering}, Arc};
|
||||
|
||||
use anyhow::Error;
|
||||
use gstreamer::{prelude::*, Element, ElementFactory, Pipeline, State};
|
||||
use gstreamer::{prelude::ElementExt, State};
|
||||
use gstreamer::Buffer;
|
||||
use gstreamer_app as gst_app;
|
||||
use tokio::sync::Notify;
|
||||
use tracing::{error, info};
|
||||
use webrtc::{
|
||||
api::media_engine::{MIME_TYPE_VP8, MIME_TYPE_VP9},
|
||||
ice_transport::ice_connection_state::RTCIceConnectionState,
|
||||
peer_connection::{peer_connection_state::RTCPeerConnectionState, sdp::session_description::RTCSessionDescription},
|
||||
rtp_transceiver::rtp_codec::RTCRtpCodecCapability,
|
||||
track::track_local::{
|
||||
track_local_static_rtp::TrackLocalStaticRTP, TrackLocal, TrackLocalWriter,
|
||||
},
|
||||
};
|
||||
|
||||
use vcs_common::{AppReceiver, AppSender, ApplicationMessage};
|
||||
|
||||
mod config;
|
||||
mod webrtc;
|
||||
mod gst;
|
||||
mod web_rtc;
|
||||
|
||||
fn main() -> Result<(), Error> {
|
||||
#[tokio::main]
|
||||
async fn main() -> Result<(), Error> {
|
||||
// TRACING SETUP
|
||||
let sub = tracing_subscriber::FmtSubscriber::new();
|
||||
if let Err(e) = tracing::subscriber::set_global_default(sub) {
|
||||
|
|
@ -18,24 +35,185 @@ fn main() -> Result<(), Error> {
|
|||
// EXIT HANDLER
|
||||
let to_quit = Arc::new(AtomicBool::new(false));
|
||||
let to_quit_2 = to_quit.clone();
|
||||
|
||||
ctrlc::set_handler(move || {
|
||||
to_quit_2.store(true, std::sync::atomic::Ordering::SeqCst);
|
||||
})?;
|
||||
let to_quit_3 = to_quit.clone();
|
||||
let to_quit_4 = to_quit.clone();
|
||||
let to_quit_5 = to_quit.clone();
|
||||
|
||||
// GSTREAMER SETUP
|
||||
gstreamer::init()?;
|
||||
|
||||
loop {
|
||||
if to_quit.load(std::sync::atomic::Ordering::SeqCst) {
|
||||
println!("Recieved Ctrl+C, stopping");
|
||||
let pipeline = gst::new_pipeline();
|
||||
|
||||
break;
|
||||
let (to_stream, mut stream) = tokio::sync::mpsc::channel::<Buffer>(10);
|
||||
|
||||
let (send_oneshot, recv_oneshot) = tokio::sync::oneshot::channel::<
|
||||
Result<(AppSender, AppReceiver), tokio_tungstenite::tungstenite::Error>,
|
||||
>();
|
||||
|
||||
// connect to remote server
|
||||
tokio::spawn(vcs_common::connect_to_server(format!("ws://{}:{}", config.destination_ip, config.destination_port), send_oneshot));
|
||||
|
||||
let app_sender: AppSender;
|
||||
let app_receiver: AppReceiver;
|
||||
match recv_oneshot.await {
|
||||
Err(e) => {
|
||||
panic!("Could not get connection status from oneshot!! {e}");
|
||||
}
|
||||
Ok(Err(e)) => {
|
||||
panic!("Could not connect to vcs websocket! {e}");
|
||||
}
|
||||
Ok(Ok((sender, recvr))) => {
|
||||
app_sender = sender;
|
||||
app_receiver = recvr;
|
||||
}
|
||||
std::thread::sleep(std::time::Duration::from_millis(200));
|
||||
}
|
||||
|
||||
println!("Success!");
|
||||
|
||||
|
||||
let peer_connection = web_rtc::setup_webrtc().await;
|
||||
|
||||
|
||||
|
||||
while let Ok(msg) = app_receiver.recv().await {
|
||||
match msg {
|
||||
ApplicationMessage::WebRTCPacket(pkt) => {
|
||||
peer_connection.set_remote_description(pkt).await.expect("The remote description caused an error! {e}");
|
||||
break;
|
||||
}
|
||||
}
|
||||
}
|
||||
|
||||
let answer = peer_connection.create_answer(None).await.expect("Couldn't create an answer from the peer connection");
|
||||
|
||||
let mut gather_complete = peer_connection.gathering_complete_promise().await;
|
||||
|
||||
peer_connection.set_local_description(answer).await.expect("Could not set local description of peer connection");
|
||||
|
||||
let _ = gather_complete.recv().await;
|
||||
|
||||
if let Some(local_desc) = peer_connection.local_description().await {
|
||||
app_sender.send(ApplicationMessage::WebRTCPacket(local_desc)).await.expect("Could not send message to the socket message channel");
|
||||
}
|
||||
|
||||
|
||||
// Set the handler for ICE connection state
|
||||
// This will notify you when the peer has connected/disconnected
|
||||
peer_connection.on_ice_connection_state_change(Box::new(
|
||||
move |connection_state: RTCIceConnectionState| {
|
||||
println!("Connection State has changed {connection_state}");
|
||||
if connection_state == RTCIceConnectionState::Failed {
|
||||
to_quit_3.store(true, Ordering::SeqCst);
|
||||
}
|
||||
Box::pin(async {})
|
||||
},
|
||||
));
|
||||
|
||||
// Set the handler for Peer connection state
|
||||
// This will notify you when the peer has connected/disconnected
|
||||
peer_connection.on_peer_connection_state_change(Box::new(move |s: RTCPeerConnectionState| {
|
||||
println!("Peer Connection State has changed: {s}");
|
||||
|
||||
if s == RTCPeerConnectionState::Failed {
|
||||
// Wait until PeerConnection has had no network activity for 30 seconds or another failure. It may be reconnected using an ICE Restart.
|
||||
// Use webrtc.PeerConnectionStateDisconnected if you are interested in detecting faster timeout.
|
||||
// Note that the PeerConnection may come back from PeerConnectionStateDisconnected.
|
||||
println!("Peer Connection has gone to failed exiting: Done forwarding");
|
||||
to_quit_4.store(true, Ordering::SeqCst)
|
||||
}
|
||||
|
||||
Box::pin(async {})
|
||||
}));
|
||||
|
||||
let notify_tx = Arc::new(Notify::new());
|
||||
let notify_video = notify_tx.clone();
|
||||
|
||||
let video_track = Arc::new(TrackLocalStaticRTP::new(
|
||||
RTCRtpCodecCapability {
|
||||
mime_type: MIME_TYPE_VP8.to_owned(),
|
||||
..Default::default()
|
||||
},
|
||||
"video_test".to_owned(),
|
||||
"webrtc_test".to_owned(),
|
||||
));
|
||||
|
||||
let rtp_sender = peer_connection
|
||||
.add_track(Arc::clone(&video_track) as Arc<dyn TrackLocal + Send + Sync>)
|
||||
.await
|
||||
.unwrap();
|
||||
|
||||
tokio::spawn(async move {
|
||||
// handle incoming packets that have been processed by the
|
||||
// interceptors
|
||||
let mut rtcp_buf = vec![0u8; 1500];
|
||||
while let Ok((_, _)) = rtp_sender.read(&mut rtcp_buf).await {}
|
||||
webrtc::util::Result::<()>::Ok(())
|
||||
});
|
||||
|
||||
tokio::spawn(async move {
|
||||
notify_video.notified().await;
|
||||
|
||||
info!("Starting video stream!");
|
||||
|
||||
// It is important to use a time.Ticker instead of time.Sleep because
|
||||
// * avoids accumulating skew, just calling time.Sleep didn't compensate for the time spent parsing the data
|
||||
// * works around latency issues with Sleep
|
||||
// Send our video file frame at a time. Pace our sending so we send it at the same speed it should be played back as.
|
||||
// This isn't required since the video is timestamped, but we will such much higher loss if we send all at once.
|
||||
|
||||
while let Some(map) = stream.recv().await {
|
||||
if to_quit_2.load(std::sync::atomic::Ordering::SeqCst) {
|
||||
stream.close();
|
||||
break;
|
||||
}
|
||||
if let Ok(buf) = map.map_readable() {
|
||||
if let Err(err) = video_track.write(&buf).await {
|
||||
if webrtc::Error::ErrClosedPipe == err {
|
||||
println!("The peerConnection has been closed.");
|
||||
} else {
|
||||
println!("video_track write err: {}", err);
|
||||
}
|
||||
}
|
||||
};
|
||||
}
|
||||
});
|
||||
|
||||
pipeline.sink.set_callbacks(
|
||||
gst_app::AppSinkCallbacks::builder()
|
||||
.new_sample(move |app_sink| {
|
||||
let sample = app_sink
|
||||
.pull_sample()
|
||||
.map_err(|_| gstreamer::FlowError::Eos)?;
|
||||
let buffer = sample.buffer_owned().ok_or_else(|| {
|
||||
gstreamer::element_error!(
|
||||
app_sink,
|
||||
gstreamer::ResourceError::Failed,
|
||||
("Failed to get buffer from appsink")
|
||||
);
|
||||
gstreamer::FlowError::Error
|
||||
})?;
|
||||
|
||||
if let Err(e) = to_stream.blocking_send(buffer) {
|
||||
error!("Error sending to stream of buffers: {e}");
|
||||
to_quit_5.store(true, Ordering::SeqCst);
|
||||
}
|
||||
|
||||
Ok(gstreamer::FlowSuccess::Ok)
|
||||
})
|
||||
.build(),
|
||||
);
|
||||
|
||||
if let Err(e) = pipeline.pipeline.set_state(State::Playing) {
|
||||
panic!("Could not start pipeline! {e}");
|
||||
}
|
||||
|
||||
tokio::spawn(async move {
|
||||
match tokio::signal::ctrl_c().await {
|
||||
Ok(()) => to_quit.store(true, std::sync::atomic::Ordering::SeqCst),
|
||||
Err(e) => {
|
||||
error!("Could not watch ctrl_c signal! {e}");
|
||||
}
|
||||
}
|
||||
});
|
||||
|
||||
Ok(())
|
||||
}
|
||||
|
|
|
|||
49
src/web_rtc.rs
Normal file
49
src/web_rtc.rs
Normal file
|
|
@ -0,0 +1,49 @@
|
|||
use std::sync::Arc;
|
||||
|
||||
use tracing::info;
|
||||
use webrtc::{
|
||||
api::{
|
||||
interceptor_registry::register_default_interceptors, media_engine::MediaEngine, APIBuilder,
|
||||
},
|
||||
ice_transport::ice_server::RTCIceServer,
|
||||
interceptor::registry::Registry,
|
||||
peer_connection::{configuration::RTCConfiguration, RTCPeerConnection},
|
||||
};
|
||||
|
||||
pub async fn setup_webrtc() -> Arc<RTCPeerConnection> {
|
||||
// WebRTC stuff
|
||||
let mut m = MediaEngine::default();
|
||||
|
||||
if let Err(e) = m.register_default_codecs() {
|
||||
// replace this with VP8 specific?
|
||||
panic!("Could not register default codecs for webrtc! {e}");
|
||||
}
|
||||
|
||||
info!("Default codecs registered");
|
||||
|
||||
// Create a InterceptorRegistry. This is the user configurable RTP/RTCP Pipeline.
|
||||
// This provides NACKs, RTCP Reports and other features. If you use `webrtc.NewPeerConnection`
|
||||
// this is enabled by default. If you are manually managing You MUST create a InterceptorRegistry
|
||||
// for each PeerConnection.
|
||||
let mut registry = Registry::new(); // may be able to prune for
|
||||
|
||||
registry = register_default_interceptors(registry, &mut m).unwrap();
|
||||
|
||||
let api = APIBuilder::new()
|
||||
.with_media_engine(m)
|
||||
.with_interceptor_registry(registry)
|
||||
.build();
|
||||
|
||||
let config = RTCConfiguration {
|
||||
ice_servers: vec![RTCIceServer {
|
||||
urls: vec!["stun:stun.l.google.com:19302".to_owned()], // you can set username and
|
||||
// password info here too
|
||||
..Default::default()
|
||||
}],
|
||||
..Default::default()
|
||||
};
|
||||
|
||||
// TODO: remove this unwrap
|
||||
// this is the offering for the remove device
|
||||
return Arc::new(api.new_peer_connection(config).await.unwrap());
|
||||
}
|
||||
114
src/webrtc.rs
114
src/webrtc.rs
|
|
@ -1,114 +0,0 @@
|
|||
use std::{sync::Arc, time::Duration};
|
||||
use std::io::Write;
|
||||
|
||||
use gstreamer::Buffer;
|
||||
|
||||
use tokio::sync::Notify;
|
||||
use tracing::{error, info};
|
||||
use webrtc::{
|
||||
api::{
|
||||
interceptor_registry::register_default_interceptors,
|
||||
media_engine::{MediaEngine, MIME_TYPE_VP9},
|
||||
APIBuilder,
|
||||
},
|
||||
ice_transport::ice_server::RTCIceServer,
|
||||
interceptor::registry::Registry,
|
||||
peer_connection::configuration::RTCConfiguration,
|
||||
rtp_transceiver::rtp_codec::RTCRtpCodecCapability,
|
||||
track::track_local::{track_local_static_rtp::TrackLocalStaticRTP, TrackLocal, TrackLocalWriter},
|
||||
};
|
||||
|
||||
pub async fn setup_webrtc() {
|
||||
let mut m = MediaEngine::default();
|
||||
|
||||
if let Err(e) = m.register_default_codecs() {
|
||||
// replace this with VP8 specific?
|
||||
panic!("Could not register default codecs for webrtc!");
|
||||
}
|
||||
|
||||
info!("Default codecs registered");
|
||||
|
||||
// Create a InterceptorRegistry. This is the user configurable RTP/RTCP Pipeline.
|
||||
// This provides NACKs, RTCP Reports and other features. If you use `webrtc.NewPeerConnection`
|
||||
// this is enabled by default. If you are manually managing You MUST create a InterceptorRegistry
|
||||
// for each PeerConnection.
|
||||
let mut registry = Registry::new(); // may be able to prune for
|
||||
|
||||
registry = register_default_interceptors(registry, &mut m).unwrap();
|
||||
|
||||
let api = APIBuilder::new()
|
||||
.with_media_engine(m)
|
||||
.with_interceptor_registry(registry)
|
||||
.build();
|
||||
|
||||
let config = RTCConfiguration {
|
||||
ice_servers: vec![RTCIceServer {
|
||||
urls: vec!["stun:stun.l.google.com:19302".to_owned()], // you can set username and
|
||||
// password info here too
|
||||
..Default::default()
|
||||
}],
|
||||
..Default::default()
|
||||
};
|
||||
|
||||
// TODO: remove this unwrap
|
||||
// this is the offering for the remove device
|
||||
let peer_connection = Arc::new(api.new_peer_connection(config).await.unwrap());
|
||||
|
||||
let notify_tx = Arc::new(Notify::new());
|
||||
let notify_video = notify_tx.clone();
|
||||
let notify_audio = notify_tx.clone();
|
||||
|
||||
let (done_tx, mut done_rx) = tokio::sync::mpsc::channel::<()>(1);
|
||||
let video_done_tx = done_tx.clone();
|
||||
let audio_done_tx = done_tx.clone();
|
||||
|
||||
let video_track = Arc::new(TrackLocalStaticRTP::new(
|
||||
RTCRtpCodecCapability {
|
||||
mime_type: MIME_TYPE_VP9.to_owned(),
|
||||
..Default::default()
|
||||
},
|
||||
"video_test".to_owned(),
|
||||
"webrtc_test".to_owned(),
|
||||
));
|
||||
|
||||
let rtp_sender = peer_connection
|
||||
.add_track(Arc::clone(&video_track) as Arc<dyn TrackLocal + Send + Sync>)
|
||||
.await
|
||||
.unwrap();
|
||||
|
||||
tokio::spawn(async move {
|
||||
// handle incoming packets that have been processed by the
|
||||
// interceptors
|
||||
let mut rtcp_buf = vec![0u8; 1500];
|
||||
while let Ok((_, _)) = rtp_sender.read(&mut rtcp_buf).await {}
|
||||
webrtc::util::Result::<()>::Ok(())
|
||||
});
|
||||
|
||||
let (to_stream, mut stream) = tokio::sync::mpsc::channel::<Buffer>(10);
|
||||
|
||||
tokio::spawn(async move {
|
||||
notify_video.notified().await;
|
||||
|
||||
info!("Starting video stream!");
|
||||
|
||||
// It is important to use a time.Ticker instead of time.Sleep because
|
||||
// * avoids accumulating skew, just calling time.Sleep didn't compensate for the time spent parsing the data
|
||||
// * works around latency issues with Sleep
|
||||
// Send our video file frame at a time. Pace our sending so we send it at the same speed it should be played back as.
|
||||
// This isn't required since the video is timestamped, but we will such much higher loss if we send all at once.
|
||||
|
||||
while let Some(map) = stream.recv().await {
|
||||
if let Ok(buf) = map.map_readable() {
|
||||
if let Err(err) = video_track.write(&buf).await {
|
||||
if webrtc::Error::ErrClosedPipe == err {
|
||||
println!("The peerConnection has been closed.");
|
||||
} else {
|
||||
println!("video_track write err: {}", err);
|
||||
}
|
||||
}
|
||||
};
|
||||
}
|
||||
});
|
||||
|
||||
// TODO: Initialize the gstreamer app, and start sending frames to the to_stream mpsc
|
||||
}
|
||||
Loading…
Reference in a new issue