Update libwebrtc to m144 and fix audio device lifecycle/processing#5958
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akleshchev
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Jun 26, 2026
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Looks like it included a bunch of changes from develop that shouldn't be here. |
- Update libwebrtc to version m144 (autobuild.xml). - Use WebRTC's software APM exclusively; disable built-in (hardware/OS) AEC/AGC/NS, including after each device (re)deploy. - Only run the output device once a peer connection's audio is established (and bring devices up with the user's selected device at that point), fixing the buzz heard before/without an active connection. - Keep capture warm across mute/unmute to avoid the AEC cold-start hiss; stop recording 30s after a sustained mute so the OS mic indicator clears. - Reliably (re)select and (re)start capture/playout after teleport or voice restart so audio sends/records again. - Don't suspend the voice channel when entering tuning mode (Vivox-era behavior that dropped the peer connection). Co-Authored-By: Claude Opus 4.8 (1M context) <noreply@anthropic.com>
Mute now zeroes captured gain and disables the sender tracks instead of stopping the capture device, so unmuting no longer cold-starts the AEC (no hiss) and Bluetooth devices no longer drop/restart as they switch between mono and stereo. Capture is gated on voice being enabled rather than on mute: it starts when voice is enabled and runs across calls and mute/unmute, and is released when voice is disabled (setVoiceEnabled). Playout stays gated on there being a connection to render. As a result the OS "mic in use" indicator is on for the length of the session and only clears when voice is disabled. Co-Authored-By: Claude Opus 4.8 (1M context) <noreply@anthropic.com>
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akleshchev
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Jul 7, 2026
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Description
Related Issues
improves #5953 - reduced the chance of hisses occurring
closes #5825 - removed some old vivox code that dropped calls when editing audio settings
Test Guidance
In addition to the reproduction steps for #5953 and #5825, this will need a thorough voice test run, as we updated libwebrtc.
This includes: