mirror of
https://github.com/openai/codex.git
synced 2026-10-27 18:48:37 +00:00
## What changed - Add feature-gated `/voice`, `/voice mute`, and `/voice stop` commands with local WebRTC audio and app-server signaling. - Show live transcripts, conversation status, and microphone and speaker levels. Preserve captions and undelivered answers across thread switches. - Speak final answers from voice handoffs while keeping delegated reasoning and commentary hidden and typed answers unspoken. - Retry eligible startup failures once and clean up voice sessions on thread switches and disconnects. - Stop voice and block late handoffs after a misalignment policy violation. Remove realtime event payloads and spoken text from receipt and debug logs. ## Testing Add coverage for voice command mapping, rejected starts, disconnect cleanup, delegated final-answer speech, late handoff rejection, and transcript and footer rendering. GitOrigin-RevId: 11def1f86b0e023b21d0d92072d6636ccad89815
20 lines
551 B
Rust
20 lines
551 B
Rust
mod client;
|
|
mod message_reader;
|
|
mod protocol;
|
|
mod session;
|
|
|
|
pub use client::ConnectionError;
|
|
pub use client::VoiceHost;
|
|
pub use protocol::AudioControls;
|
|
pub use protocol::AudioState;
|
|
pub use protocol::MAX_FRAME_BYTES;
|
|
pub use protocol::Message;
|
|
pub use protocol::RUNTIME_ENVIRONMENT;
|
|
pub use protocol::SessionDescription;
|
|
pub use protocol::decode_frame;
|
|
pub use protocol::encode_frame;
|
|
pub use protocol::read_message;
|
|
pub use session::RealtimeWebrtcSession;
|
|
pub use session::RealtimeWebrtcSessionHandle;
|
|
pub use session::StartedRealtimeWebrtcSession;
|