## Why
Opening a Bluetooth microphone can change the speaker format, leaving the previously cached sample rate unsupported when the speaker stream restarts.
## What changed
Requery and validate the speaker configuration after stopping the old stream. Update playback and render resampling to the current sample rate, and revalidate callback timing. Discard stale echo references while preserving capture history and audio processing state.
## Testing
Extend the speaker reset test to cover 48, 24, and 44.1 kHz output rates, checking that render output matches a fresh converter while capture history and pending samples are preserved.
GitOrigin-RevId: 3a4fba577288a83a7509fc78be86171ac85a3c0a
## Why
Linux voice needs system ALSA plugins and enough buffering to accommodate PipeWire graph cycles without losing capture samples. Voice startup failures also need actionable diagnostics without exposing native error details.
## What changed
- Build and bundle GNU voice helpers and runtimes with primary Linux musl release archives, and sign the archives. Keep Python wheels free of these libraries to preserve `manylinux_2_17` compatibility.
- Discover ALSA plugins in fixed system directories and increase Linux capture and playback buffering to support larger PipeWire graph cycles.
- Report voice failures by stage, preserve negotiation timeout classification, and discard native error sources. Suppress the misleading `requested` closure message after failure cleanup.
- Add explicit Windows MSVC, pkgconf, and CMake toolchain configuration and preserve host architecture in native build environments.
## Testing
Add coverage for Linux release assembly, ALSA plugin discovery, PipeWire capture and playback, classified startup failures, failure cleanup rendering, and Windows build environment handling.
GitOrigin-RevId: d805eace96a669ce3a4489f12e2db6f68f9f7f53
## Why
Audio queue saturation and processing delays could terminate voice sessions. Muted sessions also need outgoing audio to keep the peer alive.
## What changed
- Drop stale or excess incoming, capture, render-reference, and playback audio so fresh audio can resume. Reset affected processing state while preserving queue bounds and mute generations.
- Send paced, generated Opus silence while muted, without using device or processing buffers.
- Add fixed helper exit stages and session failure categories to default TUI logs without logging raw native errors, SDP, or device error text. Preserve observed helper exit status for shutdown.
## Testing
Extend tests to cover decoded mute silence and RTP sequence continuity, recovery after stale audio and queue saturation, and shutdown after observing helper output closure. Verify the helper's exit stage when the parent pipe closes during blocked startup.
GitOrigin-RevId: 17d580a72acd4423fb0b4f5107d9b541b964d069
Set the GStreamer audio sink's `slave-method` to `none` instead of
`resample`, and disable sink synchronization with `set_sync(false)`.
GitOrigin-RevId: a74f3e6d70e0d817c75bd963b31f16c69c2732ab
## 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
## Why
The voice host drained incoming RTP packets without playing them. Route received audio through decoding and speaker output while preserving speaker suppression boundaries.
## What changed
- Add a GStreamer pipeline with RTP jitter buffering, Opus decoding, conversion, and resampling into the native speaker sink.
- Rebuild playout when the speaker is re-enabled, reject packets received before the new pipeline started, and discard queued audio and echo references on speaker transitions while preserving capture history.
- Limit each receive pass to 64 packets so incoming media cannot indefinitely delay audio controls.
- Prevent retired sink failures from failing a new speaker generation, and retain the last coherent playback delay estimate during callback contention.
## Testing
Add tests for real Opus playback and stale packet rejection, combined microphone mute and speaker suppression, retired sink failures, delay contention, and echo-reference reset without losing capture history.
GitOrigin-RevId: c91c94a2ba2347ac98af4e7fd1dc331ae766cc45
## 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
## What changed
- Intercept incoming Opus RTP before the upstream track queue, preserving arrival timestamps without retaining a second queued copy.
- Limit outstanding media to 64 packets and 2 MiB, with a 64 KiB per-packet limit. Keep packet and byte permits until the received data is dropped.
- Reject unexpected payload types, changes in SSRC, oversized packets, and exhausted budgets. Report an error when a packet is more than one second old.
- Start with incoming audio suppressed and connect suppression to `speaker_suppressed`. Discard packets from earlier suppression epochs.
- Drain received packets in the host loop; decoding and playback are not implemented by this change.
## Testing
Add unit coverage for RTP bytes and arrival timestamps, budget ownership, suppression epochs, and invalid or oversized packets. Extend transport tests to verify incoming Opus delivery over UDP and TCP, including simulated packet loss and delayed TCP connection setup.
GitOrigin-RevId: 60237efafdf2acc1c9c4c5d2728e978b8131d9e6
## What changed
- Add `RealtimeWebrtcSession` and cloneable handles for startup, answer negotiation, audio controls, level meters, and error reporting.
- Open devices after answer negotiation and apply the latest controls before enabling audio. Preserve subsequent control transitions in order and close the session if the command queue fills.
- Cancel pending work on explicit close, external cancellation, or final handle drop, with a shared runtime keeping helper reaping alive.
- Add audio-state polling to consume microphone and speaker peaks and detect helper, device, or connection failures.
- Check packaged helper/runtime availability and initialize build information in the standalone TUI for the helper handshake.
## Testing
Add unit and helper-process integration tests covering startup control ordering, queue overflow, cancellation, helper reaping and loss, package availability, SDP debug redaction, and peak accumulation, clamping, and consumption.
GitOrigin-RevId: 458cd80353697f7411cee785f380d1d53aef8416
## 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 `openDevices` and `setAudioControls` to the helper protocol. Open the default microphone and speaker after runtime initialization and transport negotiation, initially muted and suppressed.
- Use CPAL on macOS, GNU Linux, and MSVC Windows, with native audio dependencies confined to `codex-voice-host`. Add ALSA development dependencies to Linux CI.
- Pack callback samples into preallocated, bounded queues and service devices every 5 ms. Reject incompatible callback configurations, invalidate queued audio on control transitions, and reject capture backlog after unmute using device timestamps.
- Handle recoverable stream underruns and overruns without terminating the helper; terminate on device failure, active audio queue overflow, or stale queued audio.
Device opening only establishes local device readiness. Capture and rendered audio references are drained locally; encoding, decoding, and peer audio are not connected by this change.
## Testing
Add unit tests for mute boundaries, stale playback invalidation, sample bounds, partial-block packing and timestamps, callback size and timing limits, startup silence, queue overflow, and recoverable stream errors.
GitOrigin-RevId: 108aea3d411ef6c12908fe5fb67aaf9cddf4475f
## 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 `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
