Timestamp synthetic frames in `real_encoder_produces_twenty_millisecond_opus_packets`
from a single start time using each block's sample offset and input rate. This
keeps the test audio contiguous even when the test thread is descheduled.
GitOrigin-RevId: e66440fcc3b3c965a7b1b22b61246c2a622a9cc9
## 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
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
## 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
goodman-oai