[WIP] Rewrite wish-server in Rust
This should solve our GC pauses :) Works locally. I still need to implement the SQLite-backed authentication and stuff though.
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1
wish-server-rs/.gitignore
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wish-server-rs/.gitignore
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/target
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wish-server-rs/Cargo.lock
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2839
wish-server-rs/Cargo.lock
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16
wish-server-rs/Cargo.toml
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wish-server-rs/Cargo.toml
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[package]
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name = "wish-server-rs"
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version = "0.1.0"
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edition = "2021"
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# See more keys and their definitions at https://doc.rust-lang.org/cargo/reference/manifest.html
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[dependencies]
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axum = { version = "0.6.11", features = ["macros"] }
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miette = { version = "5.5.0", features = ["fancy"] }
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once_cell = "1.17.1"
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tokio = { version = "1.26.0", features = ["full"] }
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tower-http = { version = "0.4.0", features = ["trace", "cors"] }
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tracing = "0.1.37"
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tracing-subscriber = "0.3.16"
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webrtc = "0.7.1"
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379
wish-server-rs/src/main.rs
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wish-server-rs/src/main.rs
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use std::{
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collections::{hash_map::Entry, HashMap},
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fmt::Display,
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net::SocketAddr,
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sync::{
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atomic::{AtomicU64, Ordering},
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Arc, Mutex,
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},
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};
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use axum::{
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extract::State,
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http::{header, HeaderMap, Method, StatusCode},
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response::{IntoResponse, Response},
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routing, Router,
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};
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use miette::{miette, Context, IntoDiagnostic, Result};
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use once_cell::sync::Lazy;
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use tower_http::{
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cors::{Any, CorsLayer},
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trace::TraceLayer,
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};
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use tracing::{error, instrument};
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use webrtc::{
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api::{
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interceptor_registry::register_default_interceptors, media_engine::MediaEngine,
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setting_engine::SettingEngine, APIBuilder, API as WebRTC,
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},
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ice::network_type::NetworkType,
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ice_transport::ice_server::RTCIceServer,
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interceptor::registry::Registry as InterceptorRegistry,
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peer_connection::{
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configuration::RTCConfiguration, peer_connection_state::RTCPeerConnectionState,
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sdp::session_description::RTCSessionDescription, RTCPeerConnection,
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},
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rtp_transceiver::rtp_codec::RTCRtpCodecCapability,
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track::track_local::{track_local_static_rtp::TrackLocalStaticRTP, TrackLocalWriter},
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Error,
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};
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fn create_rtc_config() -> RTCConfiguration {
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RTCConfiguration {
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ice_servers: vec![
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RTCIceServer {
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urls: vec!["stun:stun.cloudflare.com:3478".into()],
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..Default::default()
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},
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RTCIceServer {
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urls: vec!["stun:stun.l.google.com:19302".into()],
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..Default::default()
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},
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],
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..Default::default()
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}
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}
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fn setup_webrtc() -> Result<WebRTC> {
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let mut media_engine = MediaEngine::default();
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media_engine
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.register_default_codecs()
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.into_diagnostic()
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.wrap_err("Failed to register default media engine codecs.")?;
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let interceptor_registry = InterceptorRegistry::new();
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let interceptor_registry =
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register_default_interceptors(interceptor_registry, &mut media_engine)
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.into_diagnostic()
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.wrap_err("Failed to register default interceptors.")?;
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let mut setting_engine = SettingEngine::default();
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setup_ice(&mut setting_engine)?;
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let api = APIBuilder::new()
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.with_media_engine(media_engine)
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.with_interceptor_registry(interceptor_registry)
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.with_setting_engine(setting_engine)
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.build();
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Ok(api)
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}
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fn setup_ice(setting_engine: &mut SettingEngine) -> Result<()> {
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setting_engine.set_network_types(vec![
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NetworkType::Tcp4,
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NetworkType::Tcp6,
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NetworkType::Udp4,
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NetworkType::Udp6,
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]);
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// TODO: Set up UDP muxing?
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Ok(())
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}
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#[derive(Clone)]
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struct OngoingStream {
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video_track: Arc<TrackLocalStaticRTP>,
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audio_track: Arc<TrackLocalStaticRTP>,
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viewer_count: Arc<AtomicU64>,
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}
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static STREAMS: Lazy<Mutex<HashMap<String, OngoingStream>>> =
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Lazy::new(|| Mutex::new(HashMap::new()));
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async fn get_ongoing_stream(channel: &str) -> Result<OngoingStream> {
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let mut streams = STREAMS.lock().expect("Mutex was poisoned");
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let stream = match streams.entry(channel.into()) {
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Entry::Occupied(o) => o.into_mut(),
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Entry::Vacant(v) => {
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let video_track = Arc::new(TrackLocalStaticRTP::new(
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RTCRtpCodecCapability {
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mime_type: "video/h264".into(),
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..Default::default()
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},
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"video".into(),
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"pion".into(),
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));
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let audio_track = Arc::new(TrackLocalStaticRTP::new(
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RTCRtpCodecCapability {
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mime_type: "audio/opus".into(),
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..Default::default()
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},
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"audio".into(),
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"pion".into(),
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));
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v.insert(OngoingStream {
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video_track,
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audio_track,
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viewer_count: Arc::new(AtomicU64::new(0)),
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})
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}
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};
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Ok(stream.clone())
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}
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async fn setup_whip_connection(
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channel: &str,
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webrtc: Arc<WebRTC>,
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) -> Result<Arc<RTCPeerConnection>> {
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let rtc_config = create_rtc_config();
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let peer_connection = Arc::new(
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webrtc
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.new_peer_connection(rtc_config)
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.await
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.into_diagnostic()?,
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);
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let OngoingStream {
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video_track,
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audio_track,
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..
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} = get_ongoing_stream(channel).await?;
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peer_connection.on_track(Box::new(move |track, _recv, _tx| {
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let local_track = if track.codec().capability.mime_type.starts_with("audio/") {
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&audio_track
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} else {
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&video_track
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}
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.clone();
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tokio::spawn(async move {
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let mut rtp_buf = vec![0u8; 1500];
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while let Ok((rtp_read, _)) = track.read(&mut rtp_buf).await {
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let (Ok(_) | Err(Error::ErrClosedPipe)) = local_track.write(&rtp_buf[..rtp_read]).await else { break };
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}
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Result::<()>::Ok(())
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});
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Box::pin(async {})
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}));
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Ok(peer_connection)
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}
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fn log_http_error<E: Display>(status_code: StatusCode, error: E) -> Response {
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error!("{error}");
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(status_code, format!("{}", error)).into_response()
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}
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#[instrument(skip_all)]
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async fn handle_whip(
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method: Method,
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headers: HeaderMap,
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State(webrtc): State<Arc<WebRTC>>,
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offer: String,
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) -> Response {
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if method != Method::POST {
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return (StatusCode::METHOD_NOT_ALLOWED, "Please use POST!").into_response();
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}
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let Some(auth) = headers.get(header::AUTHORIZATION) else { return (StatusCode::UNAUTHORIZED, "Authorization was not set").into_response() };
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let auth = auth
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.to_str()
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.into_diagnostic()
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.wrap_err("Failed to decode auth header")
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.unwrap();
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let auth = auth.strip_prefix("Bearer ").unwrap_or(auth);
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let Some((channel, _key)) = auth.split_once(':') else { return (StatusCode::UNAUTHORIZED, "Invalid Authorization header").into_response() };
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// TODO: Validate the stream key
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let peer_connection = match setup_whip_connection(channel, webrtc)
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.await
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.wrap_err("Failed to initialize peer connection")
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{
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Ok(peer_connection) => peer_connection,
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Err(e) => {
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return log_http_error(StatusCode::INTERNAL_SERVER_ERROR, e);
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}
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};
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let Ok(description) = RTCSessionDescription::offer(offer) else { return log_http_error(StatusCode::BAD_REQUEST, "Malformed SDP offer") };
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if let Err(e) = peer_connection.set_remote_description(description).await {
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return log_http_error(StatusCode::INTERNAL_SERVER_ERROR, e);
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};
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let mut gather_complete = peer_connection.gathering_complete_promise().await;
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let answer = match peer_connection.create_answer(None).await {
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Ok(answer) => answer,
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Err(e) => return log_http_error(StatusCode::INTERNAL_SERVER_ERROR, e),
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};
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if let Err(e) = peer_connection.set_local_description(answer).await {
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return log_http_error(StatusCode::INTERNAL_SERVER_ERROR, e);
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}
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let _ = gather_complete.recv().await;
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match peer_connection.local_description().await {
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Some(desc) => (StatusCode::CREATED, desc.sdp).into_response(),
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None => log_http_error(
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StatusCode::INTERNAL_SERVER_ERROR,
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miette!("No local description exists!"),
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),
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}
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}
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async fn setup_whep_connection(
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channel: &str,
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webrtc: Arc<WebRTC>,
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) -> Result<Arc<RTCPeerConnection>> {
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let rtc_config = RTCConfiguration::default();
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let peer_connection = Arc::new(
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webrtc
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.new_peer_connection(rtc_config)
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.await
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.into_diagnostic()?,
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);
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let OngoingStream {
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video_track,
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audio_track,
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viewer_count,
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} = get_ongoing_stream(channel).await?;
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peer_connection.on_peer_connection_state_change(Box::new(move |s: RTCPeerConnectionState| {
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if s == RTCPeerConnectionState::Connected {
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viewer_count.fetch_add(1, Ordering::Relaxed);
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}
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if s == RTCPeerConnectionState::Disconnected {
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viewer_count.fetch_sub(1, Ordering::Relaxed);
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}
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Box::pin(async {})
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}));
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peer_connection
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.add_track(video_track)
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.await
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.into_diagnostic()
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.wrap_err("Failed to add video track")?;
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peer_connection
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.add_track(audio_track)
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.await
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.into_diagnostic()
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.wrap_err("Failed to add audio track")?;
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Ok(peer_connection)
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}
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#[instrument(skip_all)]
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async fn handle_whep(
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method: Method,
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headers: HeaderMap,
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State(webrtc): State<Arc<WebRTC>>,
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offer: String,
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) -> Response {
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if method != Method::POST {
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return (StatusCode::METHOD_NOT_ALLOWED, "Please use POST!").into_response();
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}
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let channel = match headers
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.get(header::AUTHORIZATION)
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.ok_or(miette!("Authorization header was not set"))
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.and_then(|h| {
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h.to_str()
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.into_diagnostic()
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.wrap_err("Authorization header was malformed")
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}) {
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Ok(a) => a,
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Err(e) => return log_http_error(StatusCode::BAD_REQUEST, e),
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};
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let channel = channel.strip_prefix("Bearer ").unwrap_or(channel);
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let peer_connection = match setup_whep_connection(channel, webrtc).await {
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Ok(p) => p,
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Err(e) => return log_http_error(StatusCode::INTERNAL_SERVER_ERROR, e),
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};
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let Ok(description) = RTCSessionDescription::offer(offer) else { return log_http_error(StatusCode::BAD_REQUEST, "Malformed SDP offer") };
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if let Err(e) = peer_connection.set_remote_description(description).await {
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return log_http_error(StatusCode::INTERNAL_SERVER_ERROR, e);
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};
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let mut gather_complete = peer_connection.gathering_complete_promise().await;
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let answer = match peer_connection.create_answer(None).await {
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Ok(answer) => answer,
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Err(e) => return log_http_error(StatusCode::INTERNAL_SERVER_ERROR, e),
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};
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if let Err(e) = peer_connection.set_local_description(answer).await {
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return log_http_error(StatusCode::INTERNAL_SERVER_ERROR, e);
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}
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let _ = gather_complete.recv().await;
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match peer_connection.local_description().await {
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Some(desc) => (StatusCode::CREATED, desc.sdp).into_response(),
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None => log_http_error(
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StatusCode::INTERNAL_SERVER_ERROR,
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miette!("No local description exists!"),
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),
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}
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}
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#[tokio::main]
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async fn main() -> Result<()> {
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let webrtc = Arc::new(setup_webrtc()?);
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tracing_subscriber::fmt::init();
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// TODO: CORS
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let app = Router::new()
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.route("/api/wish-server/whip", routing::any(handle_whip))
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.route("/api/wish-server/whep", routing::any(handle_whep))
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.layer(TraceLayer::new_for_http())
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.layer(
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CorsLayer::new()
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.allow_methods(Any)
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.allow_origin(Any)
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.allow_headers(Any),
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)
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.with_state(webrtc);
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let bind_addr: SocketAddr = "127.0.0.1:3001"
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.parse()
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.into_diagnostic()
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.wrap_err("Couldn't parse bind address")?;
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println!("Listening at http://{bind_addr}/ ...");
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axum::Server::bind(&bind_addr)
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.serve(app.into_make_service())
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.await
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.unwrap();
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Ok(())
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}
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