Any use of going beyond 44100?

[freeztar]

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.

Ok, Still, however you measure it, it should equal the same amount of audio time (bpm) in the end. Sure you can play the same amount of samples in 44k as you do in half the time at 88k. But what hosts/vsts work this way? Also, samples are not strictly time related. Latency is however (cpu load).

Visually:

1---2---3---4 (44k)
1-2-4-5-6-7-8 (88k)

How does the latency change here? (assuming that latency is a measure of the time difference between input and output)


What you are seemingly saying (visually):

1---2---3---4--- (44k)
1-2-3-4- (88k)

Which model is correct? I am so confused.

[The Chase]

Visually:

1---2---3---4 (44k)
1-2-4-5-6-7-8 (88k)

How does the latency change here? (assuming that latency is a measure of the time difference between input and output)


What you are seemingly saying (visually):

1---2---3---4--- (44k)
1-2-3-4- (88k)

Which model is correct? I am so confused.

well first off, you left out the 3 in your first visual aid, so i guess the second one would be correct by default

seriously though, your visuals are the same except you showed only the first bit of the 88.2's run in the second one? what are you trying to say here?

[contrast]

Your model is correct. If the latency is measured in samples, it will be half as much in terms of milliseconds at the higher sampling rate, which you can see in your second picture.

You are thinking of "latency in samples" as being some absolute amount of time. This is not true. You can have one sample per minute if you want, though of course that would be completely useless for audio. In that absurd case 128 samples of latency would be 128 minutes; try playing that in realtime. With a sampling rate of 1Hz (once per second) it would be 128 seconds. With a sampling rate of 44.1KHz (44100 times per second) it works out to 2.9 milliseconds or something like that. If you set the latency in samples, then the delay in terms of actual time will vary with the sampling rate.

In your specific setup, if you are setting the latency in ms to begin with, none of this really matters except for educational purposes. If you set it to 4 ms you will get 4 ms.

[AndrewSimon]

I don't know the technical details (and I don't want to know) but I know for a fact that if I choose a higher sampling rate I can achieve better latency with the same hardware.

[freeztar]

Your model is correct. If the latency is measured in samples, it will be half as much in terms of milliseconds at the higher sampling rate, which you can see in your second picture.

You are thinking of "latency in samples" as being some absolute amount of time. This is not true. You can have one sample per minute if you want, though of course that would be completely useless for audio. In that absurd case 128 samples of latency would be 128 minutes; try playing that in realtime. With a sampling rate of 1Hz (once per second) it would be 128 seconds. With a sampling rate of 44.1KHz (44100 times per second) it works out to 2.9 milliseconds or something like that. If you set the latency in samples, then the delay in terms of actual time will vary with the sampling rate.

In your specific setup, if you are setting the latency in ms to begin with, none of this really matters except for educational purposes. If you set it to 4 ms you will get 4 ms.

Fisrt of all, thank you for answering my banal questions.

I *think* I understand now.
The only confusion left for me now is why latency has always been expressed in terms of ms (in my limited experience), as opposed to samples/second. I guess it's just more universal, for all sample rates?

[Mighty_Hero]

I think that greater bit-depth [24-bit or 32-bit] is far more important than higher sample-rate but if your PC can handle both then why not work in the highest quality you can? That said, I always record/render at 44.1/32-bit.

Yep usually the same here.

[Evan]

Ok, here's what my limited knowledge has to say on this matter...

First of all, different converters perform differently at various sample rate. A specific ADC going up to 192kHz may sound better at say 48kHz than anything else. This is up to the hardware on what works best.

Ultra high sample rates are not meant so much to cover ultrasonic frequencies, but rather to make the higher audible frequencies sound better. So, you may not be able to hear more than 18-20kHz due to your biological limitations... but higher sampling rates may cause your upper frequency content, that you CAN hear, to sound better.

As I said, this is all hardware dependant, you may end up with a converter which sounds no different or worse at its higher sample rates.

Now, there is a real noticeable difference when it comes to processing. Simply put, digital arithmetics need as much precision as possible in order to end up with the best result. Simple as that. You wouldn't ever need 64 bit audio to satisfy your aural senses BUT... digital algorithms will benefit from such precision. Less errors and rounding in the calculations. A low precision calculation may come up with, say, 1.02 as a result. A higher precision calculation will be something like 1.0234456. Given that millions such calculations may be occuring at any given moment on a DAW, the accuracy of the numbers affects the perceived quality.

Digital synths & effects use the sample rate as a clock and a basis on which the sound is generated. Higher sample rates will make an algorithm perform more calculations in the same time frame, hence the increased cpu usage. Some algorithms, need to calculate much faster than the sample rate clock, in order to keep accuracy up. 44.1kHz may be great for your audio track, but it may not be enough for the processing algorithms to maintain high precision. This is where oversampling comes along. Oversampling makes the algorithm work at double or more the original clock (i.e. sample) rate. It works much faster and therefore has more precision in order to minimise audio artifacts.

Some synths/fx work at the project sample rate. These may benefit if you increase this sample rate and provide you with a cleaner sound (remember, greater precision). It gets more complex than that... Some algorithms (usually from older synths/fx) may not handle high sample rates well, generating a worse sound than the original one. Finally, many algorithms already employ oversampling, and automatically work at higher sample rates internally even if you are still at 44.1kHz with your ASIO driver. These will most likely sound the same if you move to say 96kHz, you will usually only get higher CPU usage and no benefit in sound.

Summing it up... high precision s/rates and bit-depths are mostly good because of the digital processing involved. Not so much because the ear MAY hear ultrasonics, or some part of your brain respond to them etc etc. It's just that arithmetics gets more room to work with... and that may or may not sound better or worse depending on what kind of algorithms you are working with.

Hope this helps... and I hope I have my facts straight.

[nuffink]

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.

Exactly. This is why the connect the dots argument (although seemingly obvious) doesn't hold water.

[kritikon]

Personally I've never heard much difference in the ultrahigh sample rates - definitely not enough to be worth flogging my PC to death and reducing the number of FX, dynamics etc that I can use in any project.

If you have a quick enough CPU and enough RAM etc, then it may be worth the extra workload to some. To me it's not. Unless you are making extremely dynamic music that is very sparse and very delicate....I doubt most people here could actually tell any difference on a dense pop/dance/rock mix whether it was rendered highly, mixed highly or any mix of the above.

Personally I use 24bit because I can hear quite a difference during the rendering and recording stage with reverb tails - having said that, I seriously doubt if I blind tested myself with a 24bit full mix that was reduced down to the normal 16 bit, and a mix that was all done in 16bit, that I could actually hear any difference. To me the point is I like the quality at higher bit depth on soloed reverb channels when I'm mixing etc, but I don't think I actually gain any end result in the long run - my mixes all end up at 44.1/16bit for CD. So the same applies with high sampling rates. I'm already flogging my PC by doing everything at higher bitdepth...so why torture it more with higher rates. You really do have to ask yourself if you seriously hear any improvement in quality in your final CD mix.

If you're recording everything to DVD, obviously it's a different issue...but let's face it - the vast majority of all of us end up at 44.1/16bit.

[stefancrs]

The way I see it, anything above 44.1KHz is "oversampling". Oversampling is useful if effects or generators you use produce aliasing noise. So I look upon high sample rates as a workaround for sucky oscillators etc (I never use it though, I scrap the synths that produce too much aliasing noise instead...)

[pitchdrifter]

All this theorisation crap is so pointless. If and maybe and not sure blah.
For the 5th time, try it OUT. Vanguard is a good example. The superwave synths are good examples.
If you only have half decent monitors or headphones you will hear the difference. Whether or not you prefer the aliasing version or the alias-free version is up to everyone's taste. But unless someone's deaf above like 12khz, the difference is clearly audible.

Markus

[DevonB]

You're playing connect the dots with 96,000 points or 44,100 points over a one second segment.

No, you're not playing connect the dots; yes, in SOME VSTi there's a huge difference in sound between different sample rates (in my experience Lounge Lizard, Powercore 01, Superwave synths etc.; NOT in Pro53, B4 and some others), but it hasn't anything to do with the dots. Sorry, I'm not the best to explain what's happening exactly.

Ok, you tell me I'm wrong, but won't tell me why? Try again.

Sampling rate is the amount of times the audio device captures the amplitude of the incoming analog signal over a one second interval. It uses all those captured points to re-create the analog waveform. If we take 44,100 points captured over a one second interval, please explain how it's not 'connect the dots' between each sample?

Also, keep in mind higher sampling rates and high bit depths gives more opportunity to more accurately recreate a sound. Key word here is "REcreate". Even though we can represent a specific freqeuncy, it's how much closer we can get it to the original sound with higher sampling rates.

How much you'll HEAR the difference is a different story and depends on the source material, as always.

Devon

[stefancrs]

You guys seem to be a bit confused.
There's quite a huge difference between recreating material (recording and then playing back samples) and creating material (digitally syntheeizing it).

If an oscillator tries to create waveforms without bandwidth-limiting them, like for instance vanguard and the superwave synts, the result will be different depending on sample rate, regardless of which sample rates you compare. When using bandwidth-limited oscillators (limited to for instance 20KHz) the result of that oscillator will be the same for _any_ sample rate above bandwidth * 2 (40KHz in this example).

This is due to "faulty" constructed oscillators (or effects for that matter) and has nothing to do with the correctness of recording / playing back sampled material. In theory, the result when recording and playing back material in the range of 0-20KHz, the result will always be the same (exactly the same) regardless of how high above 40KHz in sample rate you go.

[nuffink]

Ok, you tell me I'm wrong, but won't tell me why? Try again.

You are wrong and I did explain why. Contrast then gave another explaination why you are wrong. I'll try again if you like?

[akisd28]

You're playing connect the dots with 96,000 points or 44,100 points over a one second segment.

No, you're not playing connect the dots; yes, in SOME VSTi there's a huge difference in sound between different sample rates (in my experience Lounge Lizard, Powercore 01, Superwave synths etc.; NOT in Pro53, B4 and some others), but it hasn't anything to do with the dots. Sorry, I'm not the best to explain what's happening exactly.

Ok, you tell me I'm wrong, but won't tell me why? Try again.

Devon

Both nuffink and contrast explained that better than I could.