Hacker News Reader

I don't know much about audio encoding, but do the speakers not have to buffer the incoming packets? Large enough buffer size would introduce drift between speakers even if everything is fine network-wise.

View on HN

Just make sure they have a large enough buffer, and buffer enough, so that all speakers can play the same frames at the same time (or with the exact delay you want for each specific speaker).

You only care about delay between the speakers, not about what latency any speaker has relative to the source.

Ah, that works. I also didn't know NTP was precise enough for this. Cool.

A fairly typical and simple approach is to set an intentional, fixed delay, say 500ms, to absorb network latency / inconsistency. The sender sends a target playback timestamp ~500ms in the future with each block of audio. Then the actual delay at the playback side can expand or contract as necessary to take up network delay. The lower you make this delay, the more care you need to take on the network side to guarantee timely delivery.

NTP is accurate enough for this, but I think most of the modern protocols in the wild e.g. AES67, AirPlay2 are using PTP. It is both more accurate and in some ways simpler for this use case.

Reply on news.ycombinator.com