[wish-server-rs] Refactor :)
This commit is contained in:
parent
5df33492f4
commit
e15483198a
7 changed files with 409 additions and 338 deletions
9
wish-server-rs/.editorconfig
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9
wish-server-rs/.editorconfig
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@ -0,0 +1,9 @@
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root = true
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[*]
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indent_style = space
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indent_size = 4
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end_of_line = lf
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charset = utf-8
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trim_trailing_whitespace = false
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insert_final_newline = true
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@ -1,355 +1,33 @@
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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 miette::{Context, IntoDiagnostic, Result};
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use std::{net::SocketAddr, sync::Arc};
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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 axum::{routing, Router};
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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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mod streams;
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mod util;
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mod wish;
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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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use crate::wish::{setup_webrtc, whep::handle_whep, whip::handle_whip};
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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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pub struct AppState {
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pub webrtc: Arc<webrtc::api::API>,
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pub db: Arc<&'static str>,
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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 webrtc = Arc::new(setup_webrtc()?);
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let app_state = AppState {
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webrtc,
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db: Arc::new("ooo weee i'm the data base!!!"), // TODO: sqlx
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};
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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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@ -361,7 +39,7 @@ async fn main() -> Result<()> {
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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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.with_state(app_state);
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let bind_addr: SocketAddr = "127.0.0.1:3001"
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.parse()
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|
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57
wish-server-rs/src/streams.rs
Normal file
57
wish-server-rs/src/streams.rs
Normal file
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use std::{
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collections::{hash_map::Entry, HashMap},
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sync::{atomic::AtomicU64, Arc, Mutex},
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};
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use miette::Result;
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use once_cell::sync::Lazy;
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use webrtc::{
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rtp_transceiver::rtp_codec::RTCRtpCodecCapability,
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track::track_local::track_local_static_rtp::TrackLocalStaticRTP,
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};
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#[derive(Clone)]
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pub struct OngoingStream {
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pub video_track: Arc<TrackLocalStaticRTP>,
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pub audio_track: Arc<TrackLocalStaticRTP>,
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pub 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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pub 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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|
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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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}
|
11
wish-server-rs/src/util.rs
Normal file
11
wish-server-rs/src/util.rs
Normal file
|
@ -0,0 +1,11 @@
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use axum::{
|
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http::StatusCode,
|
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response::{IntoResponse, Response},
|
||||
};
|
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use std::fmt::Display;
|
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use tracing::error;
|
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|
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pub 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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}
|
68
wish-server-rs/src/wish/mod.rs
Normal file
68
wish-server-rs/src/wish/mod.rs
Normal file
|
@ -0,0 +1,68 @@
|
|||
use miette::{Context, IntoDiagnostic, Result};
|
||||
|
||||
use webrtc::{
|
||||
api::{
|
||||
interceptor_registry::register_default_interceptors, media_engine::MediaEngine,
|
||||
setting_engine::SettingEngine, APIBuilder, API as WebRTC,
|
||||
},
|
||||
ice::network_type::NetworkType,
|
||||
ice_transport::ice_server::RTCIceServer,
|
||||
interceptor::registry::Registry as InterceptorRegistry,
|
||||
peer_connection::configuration::RTCConfiguration,
|
||||
};
|
||||
|
||||
pub mod whep;
|
||||
pub mod whip;
|
||||
|
||||
fn setup_ice(setting_engine: &mut SettingEngine) -> Result<()> {
|
||||
setting_engine.set_network_types(vec![
|
||||
NetworkType::Tcp4,
|
||||
NetworkType::Tcp6,
|
||||
NetworkType::Udp4,
|
||||
NetworkType::Udp6,
|
||||
]);
|
||||
|
||||
// TODO: Set up UDP muxing?
|
||||
Ok(())
|
||||
}
|
||||
|
||||
pub fn setup_webrtc() -> Result<WebRTC> {
|
||||
let mut media_engine = MediaEngine::default();
|
||||
media_engine
|
||||
.register_default_codecs()
|
||||
.into_diagnostic()
|
||||
.wrap_err("Failed to register default media engine codecs.")?;
|
||||
|
||||
let interceptor_registry = InterceptorRegistry::new();
|
||||
let interceptor_registry =
|
||||
register_default_interceptors(interceptor_registry, &mut media_engine)
|
||||
.into_diagnostic()
|
||||
.wrap_err("Failed to register default interceptors.")?;
|
||||
|
||||
let mut setting_engine = SettingEngine::default();
|
||||
setup_ice(&mut setting_engine)?;
|
||||
|
||||
let api = APIBuilder::new()
|
||||
.with_media_engine(media_engine)
|
||||
.with_interceptor_registry(interceptor_registry)
|
||||
.with_setting_engine(setting_engine)
|
||||
.build();
|
||||
|
||||
Ok(api)
|
||||
}
|
||||
|
||||
pub fn create_rtc_config() -> RTCConfiguration {
|
||||
RTCConfiguration {
|
||||
ice_servers: vec![
|
||||
RTCIceServer {
|
||||
urls: vec!["stun:stun.cloudflare.com:3478".into()],
|
||||
..Default::default()
|
||||
},
|
||||
RTCIceServer {
|
||||
urls: vec!["stun:stun.l.google.com:19302".into()],
|
||||
..Default::default()
|
||||
},
|
||||
],
|
||||
..Default::default()
|
||||
}
|
||||
}
|
124
wish-server-rs/src/wish/whep.rs
Normal file
124
wish-server-rs/src/wish/whep.rs
Normal file
|
@ -0,0 +1,124 @@
|
|||
use std::sync::{atomic::Ordering, Arc};
|
||||
|
||||
use axum::{
|
||||
extract::State,
|
||||
http::{header, HeaderMap, Method, StatusCode},
|
||||
response::{IntoResponse, Response},
|
||||
};
|
||||
use miette::{miette, Context, IntoDiagnostic, Result};
|
||||
|
||||
use crate::{
|
||||
streams::{get_ongoing_stream, OngoingStream},
|
||||
util::log_http_error,
|
||||
AppState,
|
||||
};
|
||||
|
||||
use tracing::instrument;
|
||||
use webrtc::{
|
||||
api::API as WebRTC,
|
||||
peer_connection::{
|
||||
configuration::RTCConfiguration, peer_connection_state::RTCPeerConnectionState,
|
||||
sdp::session_description::RTCSessionDescription, RTCPeerConnection,
|
||||
},
|
||||
};
|
||||
|
||||
async fn setup_whep_connection(
|
||||
channel: &str,
|
||||
webrtc: Arc<WebRTC>,
|
||||
) -> Result<Arc<RTCPeerConnection>> {
|
||||
let rtc_config = RTCConfiguration::default();
|
||||
let peer_connection = Arc::new(
|
||||
webrtc
|
||||
.new_peer_connection(rtc_config)
|
||||
.await
|
||||
.into_diagnostic()?,
|
||||
);
|
||||
|
||||
let OngoingStream {
|
||||
video_track,
|
||||
audio_track,
|
||||
viewer_count,
|
||||
} = get_ongoing_stream(channel).await?;
|
||||
|
||||
peer_connection.on_peer_connection_state_change(Box::new(move |s: RTCPeerConnectionState| {
|
||||
if s == RTCPeerConnectionState::Connected {
|
||||
viewer_count.fetch_add(1, Ordering::Relaxed);
|
||||
}
|
||||
|
||||
if s == RTCPeerConnectionState::Disconnected {
|
||||
viewer_count.fetch_sub(1, Ordering::Relaxed);
|
||||
}
|
||||
|
||||
Box::pin(async {})
|
||||
}));
|
||||
|
||||
peer_connection
|
||||
.add_track(video_track)
|
||||
.await
|
||||
.into_diagnostic()
|
||||
.wrap_err("Failed to add video track")?;
|
||||
|
||||
peer_connection
|
||||
.add_track(audio_track)
|
||||
.await
|
||||
.into_diagnostic()
|
||||
.wrap_err("Failed to add audio track")?;
|
||||
|
||||
Ok(peer_connection)
|
||||
}
|
||||
|
||||
#[instrument(skip_all)]
|
||||
pub async fn handle_whep(
|
||||
method: Method,
|
||||
headers: HeaderMap,
|
||||
State(app): State<AppState>,
|
||||
offer: String,
|
||||
) -> Response {
|
||||
if method != Method::POST {
|
||||
return (StatusCode::METHOD_NOT_ALLOWED, "Please use POST!").into_response();
|
||||
}
|
||||
|
||||
let channel = match headers
|
||||
.get(header::AUTHORIZATION)
|
||||
.ok_or(miette!("Authorization header was not set"))
|
||||
.and_then(|h| {
|
||||
h.to_str()
|
||||
.into_diagnostic()
|
||||
.wrap_err("Authorization header was malformed")
|
||||
}) {
|
||||
Ok(a) => a,
|
||||
Err(e) => return log_http_error(StatusCode::BAD_REQUEST, e),
|
||||
};
|
||||
let channel = channel.strip_prefix("Bearer ").unwrap_or(channel);
|
||||
|
||||
let peer_connection = match setup_whep_connection(channel, app.webrtc).await {
|
||||
Ok(p) => p,
|
||||
Err(e) => return log_http_error(StatusCode::INTERNAL_SERVER_ERROR, e),
|
||||
};
|
||||
|
||||
let Ok(description) = RTCSessionDescription::offer(offer) else { return log_http_error(StatusCode::BAD_REQUEST, "Malformed SDP offer") };
|
||||
if let Err(e) = peer_connection.set_remote_description(description).await {
|
||||
return log_http_error(StatusCode::INTERNAL_SERVER_ERROR, e);
|
||||
};
|
||||
|
||||
let mut gather_complete = peer_connection.gathering_complete_promise().await;
|
||||
|
||||
let answer = match peer_connection.create_answer(None).await {
|
||||
Ok(answer) => answer,
|
||||
Err(e) => return log_http_error(StatusCode::INTERNAL_SERVER_ERROR, e),
|
||||
};
|
||||
|
||||
if let Err(e) = peer_connection.set_local_description(answer).await {
|
||||
return log_http_error(StatusCode::INTERNAL_SERVER_ERROR, e);
|
||||
}
|
||||
|
||||
let _ = gather_complete.recv().await;
|
||||
|
||||
match peer_connection.local_description().await {
|
||||
Some(desc) => (StatusCode::CREATED, desc.sdp).into_response(),
|
||||
None => log_http_error(
|
||||
StatusCode::INTERNAL_SERVER_ERROR,
|
||||
miette!("No local description exists!"),
|
||||
),
|
||||
}
|
||||
}
|
124
wish-server-rs/src/wish/whip.rs
Normal file
124
wish-server-rs/src/wish/whip.rs
Normal file
|
@ -0,0 +1,124 @@
|
|||
use std::sync::Arc;
|
||||
|
||||
use axum::{
|
||||
extract::State,
|
||||
http::{header, HeaderMap, Method, StatusCode},
|
||||
response::{IntoResponse, Response},
|
||||
};
|
||||
use miette::{miette, Context, IntoDiagnostic, Result};
|
||||
use tracing::instrument;
|
||||
|
||||
use crate::{
|
||||
streams::{get_ongoing_stream, OngoingStream},
|
||||
util::log_http_error,
|
||||
AppState,
|
||||
};
|
||||
|
||||
use webrtc::{
|
||||
api::API as WebRTC,
|
||||
peer_connection::{sdp::session_description::RTCSessionDescription, RTCPeerConnection},
|
||||
track::track_local::TrackLocalWriter,
|
||||
};
|
||||
|
||||
use super::create_rtc_config;
|
||||
|
||||
async fn setup_whip_connection(
|
||||
channel: &str,
|
||||
webrtc: Arc<WebRTC>,
|
||||
) -> Result<Arc<RTCPeerConnection>> {
|
||||
let rtc_config = create_rtc_config();
|
||||
|
||||
let peer_connection = Arc::new(
|
||||
webrtc
|
||||
.new_peer_connection(rtc_config)
|
||||
.await
|
||||
.into_diagnostic()?,
|
||||
);
|
||||
|
||||
let OngoingStream {
|
||||
video_track,
|
||||
audio_track,
|
||||
..
|
||||
} = get_ongoing_stream(channel).await?;
|
||||
|
||||
peer_connection.on_track(Box::new(move |track, _recv, _tx| {
|
||||
let local_track = if track.codec().capability.mime_type.starts_with("audio/") {
|
||||
&audio_track
|
||||
} else {
|
||||
&video_track
|
||||
}
|
||||
.clone();
|
||||
|
||||
tokio::spawn(async move {
|
||||
let mut rtp_buf = vec![0u8; 1500];
|
||||
while let Ok((rtp_read, _)) = track.read(&mut rtp_buf).await {
|
||||
let (Ok(_) | Err(webrtc::Error::ErrClosedPipe)) = local_track.write(&rtp_buf[..rtp_read]).await else { break };
|
||||
}
|
||||
Result::<()>::Ok(())
|
||||
});
|
||||
|
||||
Box::pin(async {})
|
||||
}));
|
||||
|
||||
Ok(peer_connection)
|
||||
}
|
||||
|
||||
#[instrument(skip_all)]
|
||||
#[axum::debug_handler]
|
||||
pub async fn handle_whip(
|
||||
method: Method,
|
||||
headers: HeaderMap,
|
||||
State(app): State<AppState>,
|
||||
offer: String,
|
||||
) -> Response {
|
||||
if method != Method::POST {
|
||||
return (StatusCode::METHOD_NOT_ALLOWED, "Please use POST!").into_response();
|
||||
}
|
||||
|
||||
let Some(auth) = headers.get(header::AUTHORIZATION) else { return (StatusCode::UNAUTHORIZED, "Authorization was not set").into_response() };
|
||||
let auth = auth
|
||||
.to_str()
|
||||
.into_diagnostic()
|
||||
.wrap_err("Failed to decode auth header")
|
||||
.unwrap();
|
||||
let auth = auth.strip_prefix("Bearer ").unwrap_or(auth);
|
||||
|
||||
let Some((channel, _key)) = auth.split_once(':') else { return (StatusCode::UNAUTHORIZED, "Invalid Authorization header").into_response() };
|
||||
// TODO: Validate the stream key
|
||||
|
||||
let peer_connection = match setup_whip_connection(channel, app.webrtc)
|
||||
.await
|
||||
.wrap_err("Failed to initialize peer connection")
|
||||
{
|
||||
Ok(peer_connection) => peer_connection,
|
||||
Err(e) => {
|
||||
return log_http_error(StatusCode::INTERNAL_SERVER_ERROR, e);
|
||||
}
|
||||
};
|
||||
|
||||
let Ok(description) = RTCSessionDescription::offer(offer) else { return log_http_error(StatusCode::BAD_REQUEST, "Malformed SDP offer") };
|
||||
if let Err(e) = peer_connection.set_remote_description(description).await {
|
||||
return log_http_error(StatusCode::INTERNAL_SERVER_ERROR, e);
|
||||
};
|
||||
|
||||
let mut gather_complete = peer_connection.gathering_complete_promise().await;
|
||||
|
||||
let answer = match peer_connection.create_answer(None).await {
|
||||
Ok(answer) => answer,
|
||||
Err(e) => return log_http_error(StatusCode::INTERNAL_SERVER_ERROR, e),
|
||||
};
|
||||
|
||||
if let Err(e) = peer_connection.set_local_description(answer).await {
|
||||
return log_http_error(StatusCode::INTERNAL_SERVER_ERROR, e);
|
||||
}
|
||||
|
||||
let _ = gather_complete.recv().await;
|
||||
|
||||
match peer_connection.local_description().await {
|
||||
Some(desc) => (StatusCode::CREATED, desc.sdp).into_response(),
|
||||
None => log_http_error(
|
||||
StatusCode::INTERNAL_SERVER_ERROR,
|
||||
miette!("No local description exists!"),
|
||||
),
|
||||
}
|
||||
}
|
Loading…
Reference in a new issue