This is a sanitized engineering note about server-side audio noise reduction for WebRTC calls. Source article: https://www.lodan.me/posts/server-side-webrtc-noise-reduction-pion-ffmpeg-rnn/ What the prototype tests The goal is not to replace WebRTC's built-in audio processing. The narrower test is: receive a WebRTC Opus track with Pion read RTP packets in OnTrack decode Opus payloads to PCM pipe raw PCM into FFmpeg apply the arnndn RNN noise reduction filter validate the output as a file before considering real-time forwarding Why this boundary matters RTP, Opus, PCM, and FFmpeg raw audio input are different boundaries. If the PCM format is wrong, FFmpeg may still produce a file, but the result should not be trusted. For example, if the Go side writes int16 PCM, the FFmpeg input format should be reviewed as s16le , not casually treated as s32le .…