Zvukovik & Nullpath
Hey, I’ve been tinkering with an adaptive quantization codec that keeps the midrange sharp while cutting bandwidth. I’m curious how you’d design a distributed system to stream that audio in real‑time without any noticeable latency.
To keep latency negligible, split the stream into micro‑chunks, say 10 ms frames. Run the codec on a lightweight worker that pulls a frame, quantizes, and pushes it to a zero‑MQ broker. On the receiver, a single thread dequeues, decodes, and feeds a low‑latency audio buffer. Use UDP for transport, add sequence numbers, and drop stale packets. Keep the network close to the edge; if you need redundancy, have a second broker mirror the stream but only activate it if packet loss exceeds 1 %. That way you get adaptive quality and instant delivery.
Sounds solid, but 10 ms frames is still a bit chunky for high‑fidelity audio; you’ll hear the blockiness if the codec hiccups. Consider 2–3 ms windows and a more robust jitter buffer. Also, UDP + zero‑MQ is fine, but the broker itself can become a bottleneck—maybe use a ring‑buffer shared memory if the nodes are on the same host. And dropping packets silently will break the perceptual quality; a lightweight forward error correction would be better than just killing the frame. The idea of a secondary broker is clever, but make sure the handover doesn’t introduce any sync drift.
Got the points. Move to 3 ms frames, keep a 30 ms jitter buffer, and swap to a shared‑memory ring if the nodes share a host. Use Reed‑Solomon FEC over each block to recover silent losses instead of dropping them. For the failover broker, lock the sequence space and stamp the stream with a monotonic timestamp; when the backup takes over, sync on that timestamp to keep drift to zero. That should keep blockiness invisible and the latency tight.