7aed4d9c3478a299762bd7455a042716a2766d98
- 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>
Description
AI Radio Show - web-based podcast production with multiple TTS providers
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