- 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>
5.1 KiB
5.1 KiB