## What changed
- Add a `GstAudioSink` subclass backed by a bounded mono `F32LE` playback writer, with partial writes, cancellation on speaker epoch changes, and delay accounting for queued samples and pending device output.
- Recreate the CPAL output stream and discard buffered audio when speaker controls change the epoch, preventing stale audio from surviving suppression.
- Add GStreamer dependencies, adjust Windows Bazel native-tool selection, and allow the helper handshake time to load startup-linked libraries.
- Separate installed-client tests from protocol lifecycle tests and supply native startup libraries in test packages.
## Testing
Add unit coverage for partial writes, sample accounting, suppression cancellation, stale writers, device delay, invalid samples, and stalled consumption.
GitOrigin-RevId: c48c48d2fee00659cf6f8a4039a6745385cb77cd
## Why
`voice-host` previously drained captured audio locally without sending it to the peer. Connect capture to the outgoing media track while preserving mute boundaries and limiting stale audio.
## What changed
- Resample capture and render references to 48 kHz with Rubato, apply Sonora echo cancellation, noise suppression, and gain control, and encode capture as 20 ms mono Opus packets.
- Attach an Opus track to the WebRTC peer and advance RTP timestamps across mute gaps without introducing sequence-number gaps.
- Prioritize queued controls and shutdown between sends. Clear pending audio and processing history on microphone mute transitions, reject delayed pre-unmute buffers, and reset partial history after capture gaps.
- Size bounded capture and render queues for device rates, validate callback timing, enforce a 500 ms processing deadline, and time out stalled sends after 100 ms.
Receive/decode and TUI integration remain unimplemented. The bundled Opus encoder build requires CMake and a C compiler.
## Testing
Add synthetic-audio tests for Opus decoding at 44.1 and 48 kHz input rates, echo reduction, capture gaps, stale audio, queue capacity, and control priority. A loopback WebRTC test verifies RTP delivery, decoded signal, mute isolation, and timestamp and sequence continuity across mute gaps.
GitOrigin-RevId: 38b8438577cfadb8124f8aeb426dffbab4845964
## What changed
- Add helper protocol requests to create a WebRTC offer, apply an answer, and report readiness once the ordered `oai-events` data channel opens.
- Support UDP and TCP candidates while bounding signaling size, remote candidates, inbound TCP streams, and WebRTC buffers. Redact session descriptions and peer errors from diagnostics.
- Close unexpected remote data channels and tear down the peer during helper shutdown, including when the parent pipe is lost during startup.
## Testing
- Exercise offer/answer negotiation and clean shutdown over real UDP and TCP sockets.
- Cover signaling validation, candidate and connection limits, early packet loss, slow TCP connection setup, and blocked-startup parent loss.
GitOrigin-RevId: 5f5e60a3a4d1d8eaca2e4724e66284f54061b6af
## Why
The voice helper previously supported only handshake and shutdown, without
validating that its packaged native runtime could be initialized safely.
## What changed
- Add an `initializeRuntime` protocol exchange that loads GStreamer and the
required plugins from physical package paths without opening audio devices.
- Restrict plugin discovery, registry access, and native library search paths,
and keep loaded libraries alive until the helper exits.
- Give initialization a dedicated deadline and terminate the owned helper when
initialization is cancelled or fails.
- Allow binary-only Rust targets to disable the default Bazel library target.
## Testing
- Cover helper-only packages, cancellation, and environment filtering.
- Add an ignored integration test for initialization from a relocated prepared
runtime and rejection of duplicate initialization.
GitOrigin-RevId: 8fff68fe26e52cb2e0722dc98fb5f124eddb0d4d
## What changed
- Add `VoiceHost` to resolve the packaged voice helper, launch it with an
allowlisted environment, perform the protocol handshake, and enforce bounded
shutdown and process cleanup.
- Preserve native executable path encoding in the pipe process APIs.
- Add `third_party/voice/assemble_package.py` to create a fresh package copy
containing a target-compatible helper and a provenance manifest with file
hashes.
## Testing
- Cover installed helper lifecycle, build matching, missing and symlinked
helpers, non-UTF-8 package paths, environment filtering, package validation,
target pairing, and failure cleanup.
GitOrigin-RevId: f893074b36ae6bb9bcedc00fbe7af6bb72745f4c
## What changed
- Add `codex-realtime-webrtc` with a length-prefixed, 256-byte JSON control protocol for `hello`, `ready`, `close`, and `closed` messages.
- Add a hardened `codex-voice-host` binary that validates protocol and build compatibility, handles orderly shutdown and parent pipe loss, and exposes its build commit.
- Keep this initial foundation limited to process lifecycle management; it does not open audio devices, load native plugins, negotiate WebRTC, or enable voice in the TUI.
## Testing
- Exercise the helper as a subprocess, covering handshake, explicit close, parent pipe loss, malformed frames, incompatible builds, and invalid message order.
GitOrigin-RevId: 6d8d7dd7ba5c4886879882d79314d235f6ab7cf4
