4 ms·

Well... it's like all audio chipsets work nowadays. Only some DSPs will be able to efficiently handle single-frame data processing, but they have the help of de

by cnvogel 12y ago

Well... it's like all audio chipsets work nowadays. Only some DSPs will be able to efficiently handle single-frame data processing, but they have the help of dedicated address generators and lightweight interrupts synchronized to the digital interface. If you write "process however much audio data is ready", then you already imply that your CPU will not be up to speed to process 48000 interrupts/second reliably and you need some buffering. And if you have to assume that sometimes you'll miss 100 samples (which, then, you'll process en-block), this means that to work reliably, you'll have to start at least 100 samples early so that you don't miss the deadline of the DAC, because the DAC will, with intractably output one sample every 48000th of a second. This already implies some kind of periodic processing of blocks, doesn't it? (and yes, such a scheme will theoretically allow you to half the latency from something to 2period-size to 1period-size + the time for processing) Third, a lot of the algorithms for processing audio can be implemented much more efficiently if you have a known block size and don't have to calculate your filters or convolutions with constantly changing number of samples for every step. Also efficiency of processing will decrease (reloading the cache after each interrupt when switching from processing plugin to processing plugin), so the time spent on calculating per frame will go up if your period size gets smaller. At one point you'll need exactly one "period size" to calculate one period size worth of samples: That's the maximum your machine can handle, and at that point you'll have a latency of your "period size"*2, which is exactly the same as running with a fixed period-size ;-). And as you can choose the period-size rather freely (maybe completely arbitrary, maybe 2^n, depends on the chipset/hardware) there's no disadvantage left.

jblow 12y ago
As someone who has actually done a good amount of soft-real-time audio programming, I can tell that you probably haven't. Everything you are saying about CPU speeds is made-up nonsense. Look into how these things are done on systems where folks actually care about latency (for example, commercial audio hardware, game consoles, etc).
- cnvogel 12y ago
  Please tell me what claim I made about CPU speeds is unsubstantiated nonsense?
  - jblow 12y ago
    I could type up a thorough explanation, but it would take about an hour, and I have a lot to do. It is actually not a bad idea to do such a write-up, but I don't think the appropriate venue for it is an ephemeral post on Hacker News ... I'd rather blog it somewhere that's more suitable for long-term reference. But I'll drop a few hints. First of all, nobody is talking about running interrupts at 48kHz. That is complete nonsense. The central problem to solve is that you have two loops running and they need to be coordinated: the hardware is running in a loop generating samples, and the software is running in a (much more complicated) loop consuming samples. The question is how to coordinate the passing of data between these with minimal latency and maximum flexibility. If you force things to fill fixed-size buffers before letting the software see them (say, 480 samples or whatever), then it is easy to see problems with latency and variance: simply look at a software loop with some ideal fixed frame time T and look at what happens when T is not 100Hz. (Let's say it is a hard 60Hz, such as on a current game console). See what happens in terms of latency and variance when the hardware is passing you packets every 10ms and you are asking for them every 16.7ms. The key is to remove one of these fixed frequencies so that you don't have this problem. Since the one coming from the hardware is completely fictitious, that is the one to remove. Instead of pushing data to the software every 10ms, you let the software pull data at whatever rate it is ready to handle that data, thus giving you a system with only one coarse-grained component, which minimizes latency. You are not running interrupts at 48kHz or ten billion terahertz, you are running them exactly when the application needs them, which in this case is 16.7ms (but might be 8.3ms or 10ms or a variable frame rate). You don't have to recompute any of the filters in your front-end software based on changing amounts of data coming in from the driver. The very suggestion is nonsense; if you are doing that, it is a clear sign that your audio processing is terrible because there is a dependency between chunk size and output data. It should be obvious that your output should be a function of the input waveform only. To achieve this, you just save up old samples after you have played them, and run your filter over those plus the new samples. None of this has anything to do with what comes in from the driver when and how big. Edit: I should point out, by the way, that this extends to purely software-interface issues. Any audio issue where the paradigm is "give the API a callback and it will get called once in a while with samples" is terrible for multiple reasons, at least one of which is explained above. I talked to the SDL guys about this and to their credit they saw the problem immediately and SDL2 now has an application-pull way to get samples (I don't know how well it is supported on various platforms, or whether it is just a wrapper over the thread thing though, which would be Not Very Good.)
- jblow 12y ago
  I understand that people are downvoting this because it is just a negative comment, or something. But, I felt it was VERY important to call out information that is clearly false. Someone who doesn't know about audio programming might read the above post and think "hey that sounds plausible, I learned something today" when in fact they were deeply misled. Registering dissent is important and I tried not to be rude about it. I did go on to give a sketch of reasons in the thread below (but it is a complex issue with a lot of details; exact situations differ on every platform; etc, etc.)
  - cnvogel 12y ago
    I didn't downvote you and was genuinely interested in why you were considering my information to be incorrect. And I now realize it's because I've always worked with systems where processing is always strongly synced to the central frame/sample/... clock. Also I read your initial comment as "why don't we use 'process every single sample' to reduce latency at all costs" which is -as you wrote- clearly a bad idea. Sorry for misrepresenting that.
  - brazzledazzle 12y ago
    I thought it was a tad condescending. Instead of: >As someone who has actually done a good amount of soft-real-time audio programming, I can tell that you probably haven't. Everything you are saying about CPU speeds is made-up nonsense. You could have wrote: >I've actually done a good amount of soft-real-time audio programming and everything you're saying about CPU speeds doesn't make sense. I think it's good to call out what you see as misleading information but that may have been going a bit too far.
- darkmighty 12y ago
  Doesn't a lot of audio processing rely on FFTs, for which you need large block sizes? I agree that constantly varying fractional block sizes don't seem like a good idea.

by cnvogel 12y ago

jblow 12y ago

As someone who has actually done a good amount of soft-real-time audio programming, I can tell that you probably haven't. Everything you are saying about CPU speeds is made-up nonsense. Look into how these things are done on systems where folks actually care about latency (for example, commercial audio hardware, game consoles, etc).

cnvogel 12y ago
Please tell me what claim I made about CPU speeds is unsubstantiated nonsense?
- jblow 12y ago
  I could type up a thorough explanation, but it would take about an hour, and I have a lot to do. It is actually not a bad idea to do such a write-up, but I don't think the appropriate venue for it is an ephemeral post on Hacker News ... I'd rather blog it somewhere that's more suitable for long-term reference. But I'll drop a few hints. First of all, nobody is talking about running interrupts at 48kHz. That is complete nonsense. The central problem to solve is that you have two loops running and they need to be coordinated: the hardware is running in a loop generating samples, and the software is running in a (much more complicated) loop consuming samples. The question is how to coordinate the passing of data between these with minimal latency and maximum flexibility. If you force things to fill fixed-size buffers before letting the software see them (say, 480 samples or whatever), then it is easy to see problems with latency and variance: simply look at a software loop with some ideal fixed frame time T and look at what happens when T is not 100Hz. (Let's say it is a hard 60Hz, such as on a current game console). See what happens in terms of latency and variance when the hardware is passing you packets every 10ms and you are asking for them every 16.7ms. The key is to remove one of these fixed frequencies so that you don't have this problem. Since the one coming from the hardware is completely fictitious, that is the one to remove. Instead of pushing data to the software every 10ms, you let the software pull data at whatever rate it is ready to handle that data, thus giving you a system with only one coarse-grained component, which minimizes latency. You are not running interrupts at 48kHz or ten billion terahertz, you are running them exactly when the application needs them, which in this case is 16.7ms (but might be 8.3ms or 10ms or a variable frame rate). You don't have to recompute any of the filters in your front-end software based on changing amounts of data coming in from the driver. The very suggestion is nonsense; if you are doing that, it is a clear sign that your audio processing is terrible because there is a dependency between chunk size and output data. It should be obvious that your output should be a function of the input waveform only. To achieve this, you just save up old samples after you have played them, and run your filter over those plus the new samples. None of this has anything to do with what comes in from the driver when and how big. Edit: I should point out, by the way, that this extends to purely software-interface issues. Any audio issue where the paradigm is "give the API a callback and it will get called once in a while with samples" is terrible for multiple reasons, at least one of which is explained above. I talked to the SDL guys about this and to their credit they saw the problem immediately and SDL2 now has an application-pull way to get samples (I don't know how well it is supported on various platforms, or whether it is just a wrapper over the thread thing though, which would be Not Very Good.)
jblow 12y ago
I understand that people are downvoting this because it is just a negative comment, or something. But, I felt it was VERY important to call out information that is clearly false. Someone who doesn't know about audio programming might read the above post and think "hey that sounds plausible, I learned something today" when in fact they were deeply misled. Registering dissent is important and I tried not to be rude about it. I did go on to give a sketch of reasons in the thread below (but it is a complex issue with a lot of details; exact situations differ on every platform; etc, etc.)
- cnvogel 12y ago
  I didn't downvote you and was genuinely interested in why you were considering my information to be incorrect. And I now realize it's because I've always worked with systems where processing is always strongly synced to the central frame/sample/... clock. Also I read your initial comment as "why don't we use 'process every single sample' to reduce latency at all costs" which is -as you wrote- clearly a bad idea. Sorry for misrepresenting that.
- brazzledazzle 12y ago
  I thought it was a tad condescending. Instead of: >As someone who has actually done a good amount of soft-real-time audio programming, I can tell that you probably haven't. Everything you are saying about CPU speeds is made-up nonsense. You could have wrote: >I've actually done a good amount of soft-real-time audio programming and everything you're saying about CPU speeds doesn't make sense. I think it's good to call out what you see as misleading information but that may have been going a bit too far.
darkmighty 12y ago
Doesn't a lot of audio processing rely on FFTs, for which you need large block sizes? I agree that constantly varying fractional block sizes don't seem like a good idea.