Remove the unused realtime WebRTC crate (#33639)

## What changed

- Remove `codex-realtime-webrtc` from the Cargo workspace and Bazel build,
  along with its native WebRTC dependency graph.
- Use `blake3`'s pure Rust implementation on
  `x86_64-pc-windows-gnullvm`, where the Bazel lint toolchain cannot
  reliably materialize the crate's native x86 assembly archives.

GitOrigin-RevId: c382d888cd7ada7a17a7a0dcbdecbe00744f5055

codex-rs/realtime-webrtc/BUILD.bazel

@@ -1,7 +0,0 @@
-load("//:defs.bzl", "MACOS_WEBRTC_RUSTC_LINK_FLAGS", "codex_rust_crate")
-
-codex_rust_crate(
-    name = "realtime-webrtc",
-    crate_name = "codex_realtime_webrtc",
-    rustc_flags_extra = MACOS_WEBRTC_RUSTC_LINK_FLAGS,
-)

codex-rs/realtime-webrtc/Cargo.toml

@@ -1,21 +0,0 @@
-[package]
-name = "codex-realtime-webrtc"
-version.workspace = true
-edition.workspace = true
-license.workspace = true
-
-[lib]
-name = "codex_realtime_webrtc"
-path = "src/lib.rs"
-test = false
-doctest = false
-
-[dependencies]
-thiserror = { workspace = true }
-tokio = { workspace = true, features = ["rt-multi-thread"] }
-
-[target.'cfg(target_os = "macos")'.dependencies]
-libwebrtc = { version = "0.3.26", git = "https://github.com/juberti-oai/rust-sdks.git", rev = "e2d1d1d230c6fc9df171ccb181423f957bb3c1f0" }
-
-[lints]
-workspace = true

codex-rs/realtime-webrtc/src/lib.rs

@@ -1,90 +0,0 @@
-#[cfg(target_os = "macos")]
-mod native;
-
-use std::fmt;
-use std::sync::Arc;
-use std::sync::atomic::AtomicU16;
-use std::sync::mpsc;
-
-#[derive(Debug, thiserror::Error)]
-pub enum RealtimeWebrtcError {
-    #[error("{0}")]
-    Message(String),
-    #[error("realtime WebRTC is not supported on this platform")]
-    UnsupportedPlatform,
-}
-
-#[derive(Debug, Clone, PartialEq, Eq)]
-pub enum RealtimeWebrtcEvent {
-    Connected,
-    LocalAudioLevel(u16),
-    Closed,
-    Failed(String),
-}
-
-pub type Result<T> = std::result::Result<T, RealtimeWebrtcError>;
-
-pub struct StartedRealtimeWebrtcSession {
-    pub offer_sdp: String,
-    pub handle: RealtimeWebrtcSessionHandle,
-    pub events: mpsc::Receiver<RealtimeWebrtcEvent>,
-}
-
-pub struct RealtimeWebrtcSessionHandle {
-    #[cfg(target_os = "macos")]
-    inner: native::SessionHandle,
-    local_audio_peak: Arc<AtomicU16>,
-}
-
-impl fmt::Debug for RealtimeWebrtcSessionHandle {
-    fn fmt(&self, f: &mut fmt::Formatter<'_>) -> fmt::Result {
-        f.debug_struct("RealtimeWebrtcSessionHandle")
-            .finish_non_exhaustive()
-    }
-}
-
-impl RealtimeWebrtcSessionHandle {
-    pub fn apply_answer_sdp(&self, answer_sdp: String) -> Result<()> {
-        #[cfg(target_os = "macos")]
-        {
-            self.inner.apply_answer_sdp(answer_sdp)
-        }
-        #[cfg(not(target_os = "macos"))]
-        {
-            let _ = answer_sdp;
-            Err(RealtimeWebrtcError::UnsupportedPlatform)
-        }
-    }
-
-    pub fn close(&self) {
-        #[cfg(target_os = "macos")]
-        self.inner.close();
-    }
-
-    pub fn local_audio_peak(&self) -> Arc<AtomicU16> {
-        self.local_audio_peak.clone()
-    }
-}
-
-pub struct RealtimeWebrtcSession;
-
-impl RealtimeWebrtcSession {
-    pub fn start() -> Result<StartedRealtimeWebrtcSession> {
-        #[cfg(target_os = "macos")]
-        {
-            let started = native::start()?;
-            Ok(StartedRealtimeWebrtcSession {
-                offer_sdp: started.offer_sdp,
-                handle: RealtimeWebrtcSessionHandle {
-                    inner: started.handle,
-                    local_audio_peak: Arc::new(AtomicU16::new(0)),
-                },
-                events: started.events,
-            })
-        }
-        #[cfg(not(target_os = "macos"))]
-        {
-            Err(RealtimeWebrtcError::UnsupportedPlatform)
-        }
-    }
-}

codex-rs/realtime-webrtc/src/native.rs

@@ -1,227 +0,0 @@
-use std::fmt::Display;
-use std::sync::mpsc;
-use std::thread;
-
-use libwebrtc::MediaType;
-use libwebrtc::peer_connection::OfferOptions;
-use libwebrtc::peer_connection::PeerConnection;
-use libwebrtc::peer_connection_factory::PeerConnectionFactory;
-use libwebrtc::peer_connection_factory::RtcConfiguration;
-use libwebrtc::peer_connection_factory::native::PeerConnectionFactoryExt;
-use libwebrtc::rtp_transceiver::RtpTransceiverDirection;
-use libwebrtc::rtp_transceiver::RtpTransceiverInit;
-use libwebrtc::session_description::SdpType;
-use libwebrtc::session_description::SessionDescription;
-use libwebrtc::stats::RtcStats;
-
-use crate::RealtimeWebrtcError;
-use crate::RealtimeWebrtcEvent;
-use crate::Result;
-
-enum Command {
-    ApplyAnswer {
-        answer_sdp: String,
-        reply: mpsc::Sender<Result<()>>,
-    },
-    Close,
-}
-
-pub(crate) struct StartedSession {
-    pub(crate) offer_sdp: String,
-    pub(crate) handle: SessionHandle,
-    pub(crate) events: mpsc::Receiver<RealtimeWebrtcEvent>,
-}
-
-pub(crate) struct SessionHandle {
-    command_tx: mpsc::Sender<Command>,
-}
-
-impl SessionHandle {
-    pub(crate) fn apply_answer_sdp(&self, answer_sdp: String) -> Result<()> {
-        let (reply, reply_rx) = mpsc::channel();
-        self.command_tx
-            .send(Command::ApplyAnswer { answer_sdp, reply })
-            .map_err(|_| RealtimeWebrtcError::Message("realtime WebRTC worker stopped".into()))?;
-        reply_rx
-            .recv()
-            .map_err(|_| RealtimeWebrtcError::Message("realtime WebRTC worker stopped".into()))?
-    }
-
-    pub(crate) fn close(&self) {
-        let _ = self.command_tx.send(Command::Close);
-    }
-}
-
-pub(crate) fn start() -> Result<StartedSession> {
-    let (command_tx, command_rx) = mpsc::channel();
-    let (events_tx, events_rx) = mpsc::channel();
-    let (offer_tx, offer_rx) = mpsc::channel();
-
-    thread::Builder::new()
-        .name("codex-realtime-webrtc".to_string())
-        .spawn(move || worker_main(command_rx, events_tx, offer_tx))
-        .map_err(|err| {
-            RealtimeWebrtcError::Message(format!("failed to spawn realtime WebRTC worker: {err}"))
-        })?;
-
-    let offer_sdp = offer_rx
-        .recv()
-        .map_err(|_| RealtimeWebrtcError::Message("realtime WebRTC worker stopped".into()))??;
-
-    Ok(StartedSession {
-        offer_sdp,
-        handle: SessionHandle { command_tx },
-        events: events_rx,
-    })
-}
-
-fn worker_main(
-    command_rx: mpsc::Receiver<Command>,
-    events_tx: mpsc::Sender<RealtimeWebrtcEvent>,
-    offer_tx: mpsc::Sender<Result<String>>,
-) {
-    let runtime = match tokio::runtime::Builder::new_multi_thread()
-        .enable_all()
-        .thread_name("codex-realtime-webrtc-tokio")
-        .build()
-    {
-        Ok(runtime) => runtime,
-        Err(err) => {
-            let message = format!("failed to start realtime WebRTC runtime: {err}");
-            let _ = offer_tx.send(Err(RealtimeWebrtcError::Message(message.clone())));
-            let _ = events_tx.send(RealtimeWebrtcEvent::Failed(message));
-            return;
-        }
-    };
-
-    let peer_connection = match runtime.block_on(create_peer_connection_and_offer()) {
-        Ok((peer_connection, offer_sdp)) => {
-            let _ = offer_tx.send(Ok(offer_sdp));
-            peer_connection
-        }
-        Err(err) => {
-            let message = err.to_string();
-            let _ = offer_tx.send(Err(err));
-            let _ = events_tx.send(RealtimeWebrtcEvent::Failed(message));
-            return;
-        }
-    };
-
-    for command in command_rx {
-        match command {
-            Command::ApplyAnswer { answer_sdp, reply } => {
-                let result = runtime.block_on(apply_answer(&peer_connection, answer_sdp));
-                if result.is_ok() {
-                    let _ = events_tx.send(RealtimeWebrtcEvent::Connected);
-                    start_local_audio_level_task(
-                        &runtime,
-                        peer_connection.clone(),
-                        events_tx.clone(),
-                    );
-                }
-                let _ = reply.send(result);
-            }
-            Command::Close => {
-                peer_connection.close();
-                let _ = events_tx.send(RealtimeWebrtcEvent::Closed);
-                return;
-            }
-        }
-    }
-
-    peer_connection.close();
-    let _ = events_tx.send(RealtimeWebrtcEvent::Closed);
-}
-
-async fn create_peer_connection_and_offer() -> Result<(PeerConnection, String)> {
-    let factory = PeerConnectionFactory::with_platform_adm();
-    let peer_connection = factory
-        .create_peer_connection(RtcConfiguration::default())
-        .map_err(|err| message_error("failed to create WebRTC peer connection", err))?;
-
-    let audio_transceiver = peer_connection
-        .add_transceiver_for_media(
-            MediaType::Audio,
-            RtpTransceiverInit {
-                direction: RtpTransceiverDirection::SendRecv,
-                stream_ids: vec!["realtime".to_string()],
-                send_encodings: Vec::new(),
-            },
-        )
-        .map_err(|err| message_error("failed to add audio transceiver", err))?;
-    let local_audio_source = factory.create_audio_source();
-    let local_audio_track = factory.create_audio_track("realtime-mic", local_audio_source);
-    audio_transceiver
-        .sender()
-        .set_track(Some(local_audio_track.into()))
-        .map_err(|err| message_error("failed to attach WebRTC audio track", err))?;
-
-    let offer = peer_connection
-        .create_offer(OfferOptions {
-            ice_restart: false,
-            offer_to_receive_audio: true,
-            offer_to_receive_video: false,
-        })
-        .await
-        .map_err(|err| message_error("failed to create WebRTC offer", err))?;
-
-    peer_connection
-        .set_local_description(offer.clone())
-        .await
-        .map_err(|err| message_error("failed to set local WebRTC description", err))?;
-
-    Ok((peer_connection, offer.to_string()))
-}
-
-async fn apply_answer(peer_connection: &PeerConnection, answer_sdp: String) -> Result<()> {
-    let answer = SessionDescription::parse(&answer_sdp, SdpType::Answer)
-        .map_err(|err| message_error("failed to parse WebRTC answer SDP", err))?;
-    peer_connection
-        .set_remote_description(answer)
-        .await
-        .map_err(|err| message_error("failed to set remote WebRTC description", err))?;
-    Ok(())
-}
-
-fn message_error(prefix: &str, err: impl Display) -> RealtimeWebrtcError {
-    RealtimeWebrtcError::Message(format!("{prefix}: {err}"))
-}
-
-fn start_local_audio_level_task(
-    runtime: &tokio::runtime::Runtime,
-    peer_connection: PeerConnection,
-    events_tx: mpsc::Sender<RealtimeWebrtcEvent>,
-) {
-    runtime.spawn(async move {
-        let mut interval = tokio::time::interval(std::time::Duration::from_millis(200));
-
-        loop {
-            interval.tick().await;
-            if matches!(
-                peer_connection.connection_state(),
-                libwebrtc::peer_connection::PeerConnectionState::Closed
-                    | libwebrtc::peer_connection::PeerConnectionState::Failed
-            ) {
-                return;
-            }
-
-            if let Some(peak) = local_audio_level(&peer_connection).await {
-                let _ = events_tx.send(RealtimeWebrtcEvent::LocalAudioLevel(peak));
-            }
-        }
-    });
-}
-
-async fn local_audio_level(peer_connection: &PeerConnection) -> Option<u16> {
-    let stats = peer_connection.get_stats().await.ok()?;
-    stats.into_iter().find_map(|stat| match stat {
-        RtcStats::MediaSource(stats) if stats.source.kind == "audio" => {
-            Some(audio_level_to_peak(stats.audio.audio_level))
-        }
-        _ => None,
-    })
-}
-
-fn audio_level_to_peak(audio_level: f64) -> u16 {
-    (audio_level.clamp(0.0, 1.0) * i16::MAX as f64).round() as u16
-}