Comment by fl0wenol | Hacker News Reader

Comment by fl0wenol | Hacker News Reader

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1) Userspace sleeps until audio data is available because there's an eternity of clock cycles between each sample; do you want to be woken up after every sample if you're done doing work or a handful? You could also busy wait but that kills battery life.

2) In order to hand you off some samples you have to at least make one copy. It's convenient to be able to copy an entire _something_ without worrying about the sound card trying to DMA into it (and any hardware-specific details to make that possible).

The difference here is that with something like jackd (similar conceptually to CoreAudio or ASIO) there is just the hardware buffer in the kernel and the user buffer in jack which can be basically "shared" by all jack-enabled apps without additional copying on the user-space side. On the other hand you can't do sample-rate conversion and per-app volume control with something like that.

But if you're doing audio software you're not worrying if flash is too loud and Skype is too soft. It's a whole-system thing enabled by the software and all the buffers and latency are being managed there.

Why is (2) dependent on anything regarding the number of samples you get at once? Sure, suppose there is a maximum block size; why does anything regarding copying "an entire something" require you to have filled that entire block size with live data? Why can't you just copy however much is available in the buffer?

I don't understand why copies are even relevant: you can make several extra copies and nobody will ever notice. Audio data is trivial in modern systems. Let's say there are two channels coming in; 48000 * 2 * 2 bytes per second is an absolutely trivial amount of data to copy and has been for many years. Building some convoluted (and unreliable) system just to prevent one copy per application, when each application is going to be doing a lot of nontrivial processing on that data, strikes me as foolish. But don't listen to me, look at the fact that Linux audio is still famously unreliable. If the way it's done were a good idea, it would actually work and everyone would be happy with it.

Typically because the hardware delivers audio in blocks. I know mostly about how USB works, but I imagine it applies to other hardware types as well.

USB has several transfer types: interrupt, bunk and isochronous.

Interrupt is initiated by the device so it wouldn't help you read variable amounts.

Bulk is good for mass data transfers, but has guaranteed access to the bus, so you'd possible get audio dropouts when accessing another USB device.

Isochronous can reserve bus bandwidth and have latency guarantees, but must occur on a fixed schedule. Since they are on a fixed schedule, they always have the same amount of data, hence a fixed block size.

Since data is arriving at the OS in fixed blocks, the lowest latency way to handle the data is deal with the blocks when the arrive. If you wanted to read variable amounts of data, you'd need to add a buffer on top that could hold a variable amount of data which would add latency.

Copying variable amounts of data isn't slow, but dealing with dynamic memory allocation is. If you needed to allocate a different sized block every few milliseconds you'd be spending the majority of your time allocating memory rather than processing audio.

This system can work very well: OS X, iOS, Windows and Linux can all get very low latencies. The issues with Android has nothing to do with block sizes, but something else in it's architecture.

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