Any use of going beyond 44100?

[kuniklo]

Below the nyquist frequency (Sample rate / 2) there is no theoretical advantage to the apparently higher resolution of 96k. Any benefit lies in the lower order (potentially more linear) anti-alias filter that can be used with the higher rate.
Whether this can be heard is a moot point.

Some vsts sound very noticeably better at higher sampling rates. The difference can be very obvious.

[kuniklo]

Kuniklo: Could you elaborate on the logic behind that procedure?

It's pretty straightforward. Some synths either sound duller at 44k or have very audible aliasing that goes away at higher sampling rates. So, you render at 96k/24bit to get the best possible sound and fewest artifacts. The resulting file can then be downsampled and dithered to the highest possible 44k/16 bit quality. If you render at 44k there's really nothing you're going to be able to do to remove the resulting artifacts.

I think this is the way a lot of pro studios work. They record at the highest practical resolution and then downsample/dither to CD quality for release.

[freeztar]

It's pretty straightforward. Some synths either sound duller at 44k or have very audible aliasing that goes away at higher sampling rates. So, you render at 96k/24bit to get the best possible sound and fewest artifacts. The resulting file can then be downsampled and dithered to the highest possible 44k/16 bit quality. If you render at 44k there's really nothing you're going to be able to do to remove the resulting artifacts.

I think this is the way a lot of pro studios work. They record at the highest practical resolution and then downsample/dither to CD quality for release.

So how does this help with aliasing? I understand that you don't have aliasing with 96k, but when you downsample to 44k doesn't it result in aliasing still?

[freeztar]

Not too long ago, I read from someone here that higher sample rates=lower latency. Is this true, it doesn't seem right? (wish I had asked the person when I read it)

[fishbowl.tucson.az]

Despite claims either way, some synths are said to sound better rendered as 96khz. Whether there is any point in recording at 96, there is probably a strong basis for this, and it should be easy to do a blind test of the claim.

But the community as a whole seems satisfied in dismissing any benefits to either wide bit depth or high sampling rates, and this ends up raising the barrier for people who are trying to do things with sound other than music. In particular, it is obnoxiously difficult to find amplifiers that can work with sonar frequencies, and nearly impossible to locate speaker drivers that can accurately reproduce very high frequencies.

This makes life difficult for researchers in the biological and physical sciences. I hope you don't think all university researchers have an unlimited budget, or any budget. Oh well, gotta get on my plane.

[contrast]

Some vsts sound very noticeably better at higher sampling rates. The difference can be very obvious.

Which does not neccessarily say anything about whether the higher sampling rate sounds "better", just that the particular VST sounds better at that sampling rate.

I used to do a lot of reading on this subject, of posts and papers written by people far more knowledgable than I. What I basically got from all of it was that 44.1 is not ideal, 96 is probably a bit more than is useful but fine to work with, and 192 is very excessive and can be worse when doing AD/DA. The simplied and rephrased explanation of this being that you trade speed for accuracy, 192 is too fast, you lose accuracy, therefore losing quality. I've frequently seen "60-80" listed as being (probably) optimal for music use, with 88.2 and 96 being cited as the most practical rates to achieve the best quality.

I'm just regurgitating the recommendations that came out of the arguments, though, and won't argue the theory behind it all. I think that's best left to people with degrees and years of professional experience in this stuff.

edit- Freeztar: latency is generally measured in samples, 256 samples is a longer period of time at 44.1 (44100 samples per second) that 88.2 (88200, of course), so yes; assuming you can run at the same latency settings.

[kuniklo]

So how does this help with aliasing? I understand that you don't have aliasing with 96k, but when you downsample to 44k doesn't it result in aliasing still?

It depends on how you downsample. I'd imagine most good downsampling utilities will put some kind of steep filter around the nyquist frequency to eliminate frequencies that could alias down into the audible range.

[kuniklo]

Which does not neccessarily say anything about whether the higher sampling rate sounds "better", just that the particular VST sounds better at that sampling rate.

I used to do a lot of reading on this subject, of posts and papers written by people far more knowledgable than I. What I basically got from all of it was that 44.1 is not ideal, 96 is probably a bit more than is useful but fine to work with, and 192 is very excessive and can be worse when doing AD/DA.

Right. I think it's important to distinguish here between the behaviour of vsts at various sampling rates and theoretical considerations. The fact that some vsts sound much better at higher sampling rates doesn't mean that we hear anything above 22khz at all. I'd probably still record at a higher resolution, just because I like to have the extra headroom for manipulation, but I suspect that well mastered audio at 44k final resolution should be good enough for just about anything.

[freeztar]

So how does this help with aliasing? I understand that you don't have aliasing with 96k, but when you downsample to 44k doesn't it result in aliasing still?

It depends on how you downsample. I'd imagine most good downsampling utilities will put some kind of steep filter around the nyquist frequency to eliminate frequencies that could alias down into the audible range.

Aha, that makes sense now.
Thanks.

[freeztar]

edit- Freeztar: latency is generally measured in samples, 256 samples is a longer period of time at 44.1 (44100 samples per second) that 88.2 (88200, of course), so yes; assuming you can run at the same latency settings.

Ok, but I'm still uncertain about something here. Sure, you get more samples in a shorter amount of time, but it is still the same audio. It seems to me that the extra processor time of computatiing the additional samples would offset any benefits you would gain from extra samples.
In other words, the time domain of heard audio is the same (latency), but the sample content (processor usage) is higher.

[contrast]

The processor usage will be higher at higher sample rates, yes. That only means that you will be able to do less before running into CPU issues; crackling noises, etc, I'm sure you've asked a bit too much of your CPU at some point.

You may even then decide to increase the latency of your soundcard or plugins (where possible) to help free up the CPU a bit.

But if you can or choose to run at the same latency settings (measured in samples) the latency in terms of actual time will go down as the sampling rate goes up.

[freeztar]

But if you can or choose to run at the same latency settings (measured in samples) the latency in terms of actual time will go down as the sampling rate goes up.

Will you please elaborate on this? First of all, the latency settings I have access to are measured in milliseconds (ie Tracktion, My ASIO Aardvark card is at 4.3ms). Secondly, how (exactly) do more samples in the same amount of time=lower latency. I know I'm being thick, but this information could be very useful to my ignorance.

[contrast]

Well, what is confusing you here is that sometimes latency is measured in samples, and sometimes it is measured in milliseconds.

If the latency settings are measured in MS, then that's the latency you get. 4.2 MS is 4.2 MS. The software takes care of figuring out how many samples this works out to on its own.

In the case that latency settings are measured in samples, you only need to understand that it takes more time to play X number of samples at 44.1KHz than it does at 88.2KHz. Twice as long in fact, because half as many samples are played every second.

[xoxos]

at 44100, you've got 100 samples in a 441Hz wavecycle.. 2 octaves higher, still well within musical range, you've got 25 samples per wavecycle..

12.5 samples per 180 degrees..

go.. 2 octaves higher.. 3.125 samples per 180 duty cycle..

that's not especially accurate, tho i don't worry about it myself since i use audio semantically.

[contrast]

That's not a good way to think about it. The number of samples per cycle is irrelevant. In theory, at 44.1, any frequency below 22050 will be reproduced perfectly. A sine wave at 441Hz is not somehow better or higher-res than a sine wave at 882Hz, they will both come out as perfect sines.

In practice, how well they come out depends on how well your AD/DA works, but that's a separate issue.

This begs the question of what, exactly, is being lost; there are fewer samples per cycle, that means less information per cycle, so logically we are losing something. What is being lost is, and is only, the portion of the wave that contains frequencies above 22050.