150ms cap was too tight for Iran relay (high jitter), causing constant
audio drops. Raised to 1s — packet bursts are absorbed smoothly,
drift reset only fires on real accumulation.
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
Pull-based ScriptProcessor approach broke audio completely.
Back to createBufferSource scheduling which worked, but with
tighter 200ms max drift (was 300ms). Snaps back when exceeded.
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
Previous version output 960 samples into 1024-sample callback frames,
causing 64 samples of silence per frame (choppy/robotic sound).
Now accumulates float samples in a continuous buffer, output callback
pulls exactly 1024 at a time regardless of input frame size.
Buffer capped at 200ms to prevent drift.
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
Web playback rewritten from push-scheduling to pull-based ring buffer:
- ScriptProcessorNode pulls frames from buffer every ~21ms
- Buffer capped at 10 frames (~200ms) — drops oldest on overflow
- Latency permanently bounded, no drift over time
Also: install ring crypto provider for rustls TLS on Linux,
build on debian-12 to match mequ glibc.
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
Generates a self-signed certificate at startup for HTTPS.
Required for mic access on Android/remote browsers (getUserMedia
needs a secure context).
Usage: wzp-web --port 9090 --relay 127.0.0.1:4433 --tls
Browser: accept the self-signed cert warning, then mic works.
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
When playback buffer drifts beyond 300ms ahead of real-time, reset
to 40ms. This prevents the unbounded latency growth over long sessions.
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
- Schedule each playback buffer to start exactly where the last ended
(was causing gaps/overlaps with fixed 60ms offset)
- Log AudioContext sample rate to console for debugging
- Reset playback timeline when falling behind
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
- Use power-of-2 buffer (1024) for ScriptProcessorNode
- Accumulate samples and send exact 960-sample frames
- Remove unused watch import from relay
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
New wzp-web crate serves a web page with:
- Browser mic capture via Web Audio API (48kHz mono)
- WebSocket transport for raw PCM audio
- Server-side Opus encode/decode + FEC through wzp relay
- Real-time audio playback in browser
- Level meter and connection stats
Usage:
wzp-relay --listen 0.0.0.0:4433 # start relay
wzp-web --port 8080 --relay 127.0.0.1:4433 # start web bridge
Open http://localhost:8080 in browser
Two browsers connected to the same relay get bridged for a call.
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
