Capture processing, the jitter buffer and decoding were desktop classes,
so any other platform would have had to copy them. They now live in core
and reach the platform only through two interfaces:
- AudioIo lists devices and opens capture/playback lines; the desktop's
AudioDevices implements it over PipeWire and Java Sound.
- OpusCodec creates encoders and decoders; the desktop binds libopus
through the FFM API as before.
DesktopVoiceInput becomes CaptureVoiceInput unchanged in behaviour.
DesktopVoiceOutput splits into VoiceStream, one speaker's jitter buffer
and decoder, paced by whoever pulls it, and StreamingVoiceOutput around
it. Speakers reach the device either on a line each (the desktop, so
each shows up in the PipeWire mixer) or mixed onto one shared line, as
mobile audio APIs want.
The settings dialog's device lists and microphone test now go through
the AudioBackend instead of desktop classes.
Co-Authored-By: Claude Opus 5.5 <noreply@anthropic.com>
Ports WebRTC's rnn_vad (as TS3 embeds it) to Java: LPC, pitch
estimation, spectral features and the RNN itself, feeding a
speech-probability detector that replaces the old SpeechDetector.
Also switches the volume-gate threshold from raw dBFS to
InputLevel's scale, matching TS3's own slider and range, with a
migration for settings saved under the old key.
