Russell Ballestrini
1c52e09b37
Fox 2026-06-05: "the remote phones are transcribing more accurate than
the host version. figure out why and make host version better."
Two causes diagnosed:
1. Aliasing. The whisper-capture worklet decimates 48 kHz → 16 kHz by
taking every 3rd sample with NO anti-alias filter. Listener path
feeds Opus-decoded peer audio which is already band-limited
(~12 kHz max). Host path feeds RAW mic — all 48 kHz of it —
which means anything above 8 kHz folds back into the audible band
as garbage when decimated. Whisper sees noisier input on host.
2. Amplitude / dynamic range. Music-mode mic has NO AGC/NS/EC
(raw broadcast for music). Quiet passages are too low for
Whisper to confidently transcribe; loud passages can clip. The
listener path's audio has been through Opus encode/decode which
normalizes implicitly.
Both addressed via Web Audio nodes inserted between the
MediaStreamSource and the whisper-capture worklet:
Self-capture chain:
micStream → DynamicsCompressorNode → BiquadFilter (lowpass 7 kHz)
→ AudioWorkletNode (whisper-capture)
DynamicsCompressor: voice-friendly settings (threshold -30 dB,
knee 30, ratio 4:1, attack 3 ms, release 250 ms). Flattens
amplitude so Whisper sees a normalized waveform regardless of how
the user has the mic gain set or whether music mode is on.
BiquadFilter lowpass at 7 kHz (below Nyquist of 16 kHz = 8 kHz):
removes content that would alias when the worklet decimates.
Remote captures: just the lowpass (no compressor — Opus has already
normalized the peer audio adequately). Mostly a no-op on voice-mode
peers (Opus already cuts <12 kHz) but helps music-mode peers where
the encode preserves more high-frequency content.
stopCaptureForUuid / stopSelfCapture cleanly disconnect the new
nodes so re-toggle doesn't leak audio graph references.
Expected: host self-transcribe accuracy now comparable to (or better
than) the phone listener transcribes. Telemetry `emt=` should
increase per tick on host once the user is talking.
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