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authorRoxie Linden <roxie@lindenlab.com>2026-06-26 00:07:52 -0700
committerRoxie Linden <roxie@lindenlab.com>2026-07-07 10:14:47 -0700
commit95120bc676fc68793da2d4c06542cf7cb3272d00 (patch)
tree04a992e23898438622589335339dd305218f09aa /indra/newview/llscripteditor.cpp
parent1c8144b1708420b30c54472a5139d451034e89de (diff)
Update libwebrtc to m144 and fix audio device lifecycle/processing
- 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>
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