Commit Graph

11 Commits

Author SHA1 Message Date
Siavash Sameni
12b6f30f9b feat: room-based calls + AudioWorklet for capture and playback
Rooms:
- URL-based: open /myroom to join a room
- Two clients in same room get bridged through relay
- Input field for room name, also supports URL path and hash
- Each room creates independent relay connections

AudioWorklet (replaces deprecated ScriptProcessorNode):
- capture-processor.js: accumulates mic samples, sends 960-sample frames
- playback-processor.js: pull-based output with 200ms buffer cap
- Falls back to ScriptProcessor if AudioWorklet unavailable
- Eliminates drift: worklet runs on audio thread, not main thread

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-03-27 20:16:06 +04:00
Siavash Sameni
ce6aacb25f fix: bridge pairing + auto-reconnect + test stability
Bridge mode rewrite:
- First client echoes while waiting, checks every 100ms if paired
- Second client triggers bridge immediately, first exits echo loop
- After bridge ends, slot is cleared for the next pair
- No more two tasks competing for the same transport recv

Web client auto-reconnect:
- On WebSocket close/error, automatically reconnects after 1s
- Keeps retrying as long as the user hasn't clicked Disconnect

Test fix:
- Install rustls crypto provider in transport config tests
  (fixes race condition when running full workspace tests)

Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
2026-03-27 19:49:27 +04:00
Siavash Sameni
38ae62b542 fix: raise drift cap to 1s — stops constant resetting on jittery links
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>
2026-03-27 19:41:45 +04:00
Siavash Sameni
709ad1ba7d fix: revert to scheduled playback with 200ms drift cap
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>
2026-03-27 19:31:26 +04:00
Siavash Sameni
1c91c4a1b5 fix: sample-accurate playback buffer eliminates robotic audio
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>
2026-03-27 19:29:52 +04:00
Siavash Sameni
4de72e2d98 fix: pull-based audio playback eliminates drift + rustls crypto provider
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>
2026-03-27 19:26:59 +04:00
Siavash Sameni
61d6fb173d feat: HTTPS support for web bridge (--tls flag)
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>
2026-03-27 18:56:00 +04:00
Siavash Sameni
66f720f1ee fix: cap web playback latency at 300ms — prevents drift accumulation
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>
2026-03-27 18:44:33 +04:00
Siavash Sameni
9ad21182a8 fix: web audio playback quality — gapless scheduling + sample rate debug
- 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>
2026-03-27 18:29:00 +04:00
Siavash Sameni
a7afe4ff21 fix: web audio capture buffer size + relay warning
- 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>
2026-03-27 18:27:08 +04:00
Siavash Sameni
3f128936c4 feat: web bridge — browser-based voice calls via WebSocket
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>
2026-03-27 18:23:39 +04:00