eager-crowd-1678
07/14/2025, 6:36 AMtall-airline-16756
07/14/2025, 6:36 AMtall-airline-16756
07/14/2025, 6:39 AMbytes
format and the frame length matches the configured output audio frame structure .
• Audio Buffer Size: For local/continuous streaming, buffer underruns may cause capture errors. Increase queue_size_ms
(try 1000-2000ms) in your rtc.AudioSource
configuration to stabilize buffering, which helps prevent dropped or delayed audio frames .
• Memory Usage: High memory can come from unclosed asyncio channels and stale generation objects. Ensure all message and audio channels are cleaned up after each response/generation, and monitor process memory with psutil
to identify leaks .
• Performance Monitoring: Track detailed timing metrics (duration, tokens/sec, time to first token) via RealtimeModelMetrics
for bottleneck analysis, and assess if session resumptions or connection failures coincide with output errors .
• Compute and Network: If running everything locally, ensure your system has adequate CPU/memory. High compute utilization can cause both slow inference and audio interruptions.
Monitor your logs for issues when audio frames are generated or channels are closed, and verify audio frame type/size. Adjust buffer sizes and perform regular resource cleanup after each interaction. Let me know your findings or if you have additional log details for targeted diagnostics.
Sources: livekit-plugins/livekit-plugins-google/livekit/plugins/google/beta/realtime... | Speech & Audio Processing | livekit/livekit_composite | DeepWiki | AI Agents Framework | livekit/livekit_composite | DeepWiki | Voice Assistants | livekit/livekit_composite | DeepWiki