- Reduce capture chunk from 4096 to 640 samples (256ms → 40ms)
- Replace BufferSource scheduling with AudioWorklet playback ring buffer
- Add 80ms jitter buffer with linear interpolation upsampling
- Reduce host mic and live caller stream blocksizes from 4096/2048 to 1024
- Replace librosa.resample with numpy interpolation in send_audio_to_caller
Co-Authored-By: Claude Opus 4.6 <noreply@anthropic.com>
