luke 7aed4d9c34 Fix live caller audio latency and choppiness
- 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>
2026-02-05 16:32:27 -07:00
S
Description
AI Radio Show - web-based podcast production with multiple TTS providers
MIT 18 MiB
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