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0 pointsjancsika1y ago0 comments

> If your DSP callback executes in 1ms 99.99% of the time but sometimes takes 10ms, you’re hosed.

I tend to agree, but...

From my recollection of using Zoom-- it has this bizarre but workable recovery method for network interruptions. Either the server or the client keeps some amount of the last input audio in a buffer. Then if the server detects connection problems at time 't', it grabs the buffer from t - 1 seconds all the way until the server detects better connectivity. Then it starts a race condition, playing back that amount of the buffer to all clients at something like 1.5 speed. From what I remember, this algo typically wins the race and saves the client from having to repeat themselves.

That's not happening inside a DSP routine. But my point is that some clever engineer(s) at Zoom realized that missing deadlines in audio delivery does not necessarily mean "hosed." I'm also going to rankly speculate that every other video conferencing tool hard-coupled missing deadlines with "hosed," and that's why Zoom is the only one where I've ever experienced the benefit of that feature.

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ssfrr1y ago

The context for this article is writing pro audio software, where that kind of distortion would generally be as bad as a dropout, if not worse.

wizardforhire1y ago

Yeah, 5ms is the threshold for noticeability as far as latency in pro-audio. Its like frame-rate for pro-gamers. The problem is your target user is highly specialized out side the the norms by a large margin. What makes audio even more difficult is that sub ms issues can cause phase and frequency distortion that can become even more noticeable than latency alone.

PaulDavisThe1st1y ago

1. you do not need to be a highly specialized target user to detect latency between pressing a key on a MIDI keyboard and the corresponding sound being produced.

2. 3ms is typical in-air latency between a typical DAW user and their near-field monitors, so claims about sensitivity to times much lower than 5msec should be taken with some skepticism

3. In live contexts, many drum + bass pairings have more than 10ms of air latency between them, so ditto #2

4. On the other hand, no good reason to add to latency

5. For performance purposes, jitter is much worse than latency. Pipe organ players rapidly learn to deal with even whole seconds of latency, but almost nobody can deal with jitter (essentially, variable, unpredictable latency)

6. There are no sub-ms issues that will cause phase and frequency distortion. Those come from DSP errors, not handling of latency, which is just about always a constant, fixed feature of the data signal path. You may be thinking of stuff like comb filtering, but this is not related to the latency in the signal path in a correct setup.

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