Sample Rate

[Tone Deft]

^didn't know all that before. EE?

since you 'know your shit' you might like to geek out on these pages.
http://www.cirrus.com/en/applications/a ... APP25.html <-- Cirrus Logic
http://focus.ti.com/en/download/aap/sel ... o/tool.htm <--- TI
http://www.analog.com/en/audiovideo-pro ... index.html <-- ADI, page not as hobbyist friendly.

they pretty much lays out how you can make your own sound card and most audio devices. analog in, SPDIF in, USB/Firewire, digital out, analog out, headphone out. FWIW the key to reading spec sheet tables is to just read the right side where the units are. you want stuff measured in dB, not V, A, W, ns etc.

check the DACs and ADCs to see how their distortions are rated, the nonlinearities of the devices, etc. they're not perfect but they're really good.

did you test your hearing range? it only take a minute.

[OneZeroMusic]

Yea I'm an electrical engineer, although I didn't enjoy the actual coursework very much I'm happy that it gave me a good understanding when it comes to the processing of analog and digital signals. I've seen similar function block diagrams in the past and have to say I'm not interested in creating my own sound card, I want to make music and I won't let myself get too side tracked with other things (right now I'm focusing on sound design, if it comes to the point where I need to pay insanely close attention to activity in DACs and ADCs I'll work on it then). I also did the tone generator, my hearing peaked at approximately 10.8 kHz.

[3phase]

.. and on 192 k recordings you might see some stuff above 50 k you really dont like to feet your mix engine with..

ermmm... wrong. Actually, it's the lower sampling rates that cause high-freq (higher than audible) noise to distort your audio in the audible range, it's called aliasing. In theory, a 192kHz sample rate would protect the audible range from a number of aliasing bins' worth of noise that lower 44kHz sample rates would not protect against. Of course, good hardware filtering right before the ADCs will have a great deal of impact on the recorded quality as well.
-hamish

you dont live next to a railway..the above 50 khz content i would catch with 192 k is really nothing you want to build up in a mix. so you need extra highcuts..

i see the 88/96 k as the best compromise and as said before would be nice if internal oversampling would be an option because the biggest benefots frm the higher samplerate really come form the internal processing within live .. but ok.the audio files sound a little better too.. just the difference to 192 is really little on the plain audio file..

a highend converter does a better job on 44 k than a cheap firface one on 96 k..

so it also relates to your converter ..

for multitrack recordings projekts get way to big with 192 k... at leats aslong we are not in 64 bit domain...

[mr.ergonomics]

I have only time for an short answer atm so apologies if its a bit fast written..

brief: the sampling theorem says that a bandlimited analog signal that has been sampled can be perfectly reconstructed if the sampling rate is more than 2x higher than the highest frequency in the original analog signal. this is proven by math. every audio signal is just an addition from many sinewaves. a bit simplified.. but the sampling points are "used" to see which sine waves you need to reconstruct the signal. you don't connect just the points from the sampling point you see in the editor.

the so called "more detailed" picture from audio is wrong. when you sample a signal with 192 khz and 44 khz the content below 22 khz (in reality due to not perfect filters a bit lower) is exact the same!

the difference with 192 khz is that you can get higher frequencies (you don't need). more sampling frequency just means higher possible content.


again: a sampled 10 khz sine wave with 44 khz is exactly the same as sampled with 192 khz. when you have a signal with a sine at 10 khz and a sine at 25 khz there is a difference, with 44 khz sampling rate you don't get that 25 khz sine. but we can't hear a 25 khz sine anyway...

it can happen that a da converter sounds better at higher sampling rate, but that is only due to bad and not sharp enough filters in the converter. but even if you consider that you dac has a bad filter... using 96 khz should be absolute enough to prevent this.

Actually it can't be "perfectly" reconstructed, and it isn't proven by math. I know, I took courses in filter design during university and have done the math by hand, I know my shit. However, you are correct about reconstruction by Sine waves, although I believe there are more ways than this to reconstruct signals. And the more detailed picture is correct, like I said I've done the math by hand and it has gotten to the point where I don't care if your dumb stubborn ass believes it or not.

interesting that you have to insult. you don't give a singe answer why it's wrong what I said. I'm always interested in learning and I'm really thankful when one give me a hint that my knowledge is wrong. but I don't want to waste my time her, I have no interests to convince you and I don't like your discussion style. and it doesn't matter if your a engineer or not, all what I've learned, some at uni too, indicates that your digital audio model is wrong. you should read lavrys paper on sampling theory (just google if you are interested).

[OneZeroMusic]

The reason I didn't explain in my previous response is because I have already done so 3 times, which is why I have such a snippy response. I'll check out that paper when I finish installing Komplete 7.

[TobiasHahn]

Well,

• in theory, every signal which does not contain frequencies above half of the sampling frequency can be perfectly reconstructed from its samples (Shannon-Whittaker sampling theorem).
• In practice, you cannot reconstruct a sampled signal perfectly. How well depends on the quality of your equipment much more, than on the sample rate (given the conditions of the sampling theorem are satisfied).

In Live, though, the sampling rate does matter for certain audio effects and instruments. Some effects like Overdrive may sound differently when used at different sample rates, and you may very well be able to hear the difference. The reason is that these devices can generate very high frequencies internally which will lead to different amounts of aliasing artifacts when used at different sample rates. It is important to note that this does not mean that such a device sounds "better" or "worse", but rather just different at different sample rates.

This is the main reason why we recommend not to change the sample rate after you have started working on a project.

Best wishes,
Tobias

[mr.ergonomics]

Thanks for your answer tobias.

* In practice, you cannot reconstruct a sampled signal perfectly. How well depends on the quality of your equipment much more, than on the sample rate (given the conditions of the sampling theorem are satisfied).

You don't necessarily need to explain it detailed, but can you give a hit or name the effect why? bazzwords are ok.

And your answer is very "compressed" so it's hard to understand if we mean the same with "perfect" reconstructed (it's clear that the lpf and the analog parts in the converter have an impact and can't be perfect). With perfect reconstructed I mean only the digital part.

From my understanding the recorded data with a specific converter from a 8 khz sine wave - sampled with 48khz and 96 khz - should be the same (given that the alias-filter in the DAC don't affect the 8 khz sine wave at all). The signal is not "more detailed" with 96khz than with 48 khz. (...of course you have more sampling points with 96khz, but they don't make the signal "more detailed" or "more precise" when you try to reconstruct the signal. the sampling points you get with 48 khz are enough).

edit: added a sentence to be better comprehensive.

[agent314]

Playing with the test tone generator, I was able to hear all the way up to 19.1khz.

I'm actually kind of flabbergasted, as I've spent way too many hours in front of loudspeakers at shows.

[simpli.cissimus]

32bit/88.2Hz for CD !!!
32bit/96Hz for DVD !!!

I can hear that it sounds better and the more effects I use
the more benefits it has !

The effects just give better results and reverbs are like cream...

Recording everything at higher bit and sample rate is recommended too.

The doubled bit and sample rate are cool too, when you later converting
your audio down to 16bit/44.1Hz or 48Hz !
It's simple math and not complicated down converting because of the fact
it just needs to be converted down to half.

[OneZeroMusic]

Well,

• in theory, every signal which does not contain frequencies above half of the sampling frequency can be perfectly reconstructed from its samples (Shannon-Whittaker sampling theorem).
• In practice, you cannot reconstruct a sampled signal perfectly. How well depends on the quality of your equipment much more, than on the sample rate (given the conditions of the sampling theorem are satisfied).

In Live, though, the sampling rate does matter for certain audio effects and instruments. Some effects like Overdrive may sound differently when used at different sample rates, and you may very well be able to hear the difference. The reason is that these devices can generate very high frequencies internally which will lead to different amounts of aliasing artifacts when used at different sample rates. It is important to note that this does not mean that such a device sounds "better" or "worse", but rather just different at different sample rates.

This is the main reason why we recommend not to change the sample rate after you have started working on a project.

Best wishes,
Tobias

The reconstruction of the signal would be based upon the algorithm used to calculate the interpolated points outside of the sampled points, would it not? And while we may be able to accurately calculate the interpolated points there still exists a possibility of small variation. Which would lead to the conclusion that the greater the amount of sampled points there are, the lower the amount of interpolated points that would be required and reduce the risk of error in the interpolation calculation?

[Hermanus]

When a externa soundcard is blocked at 192khz this often means no drivers has been installed.

Once the driver is installed, you have access to all sample rates available.
One good point to know for windows users is that you cant efficiently change the sample rate from within windows audio settings for most of the cards... If you don't have a specific console app running background in your tasks, maybe it's a driver issue and As tone deft gently told you, you'd surely have to check your audio driver [updates?].

E-mu are poor in updates I know.
Try ASIO then. my E-mu 0404 has it.

I had bitchin for longtime about my wrong buy... once I turned up ASIO, my poorish usb audio card turned to marvel in terms of latency and sample rate.
go in live preferences, choose asio and then in the sub menu, you should see your E-mu tracker card. VOILÀ!


My main concern is about making music and practising is the way to get better...
Doing a lot of gigs is the way to increase as well.

Ok about sound quality but at this point, really?
Are you not getting far away from the essential

Oh You'd be surprised too when someone will ask you for a wav file for vynil press at 44.1khz 16bits

Just my tiny two cents here, I don't wanna debate or argue.
Everyone do like he wants after all

[docprosper]

Some background on this:
http://en.wikipedia.org/wiki/Oversampling

I mentioned the aliasing bit earlier; although there isn't audible audio above 20khz there is always noise. The higher your sample rate, the less of this noise rolls down into the audible region. This won't reduce noise on previously sampled audio but would on audio you are currently sampling into Live.

The resolution section also applies, by oversampling you get a higher effective bit depth as you are is essence averaging a number of samples that are at the set bit depth.

Hamish

[TobiasHahn]

And your answer is very "compressed" so it's hard to understand if we mean the same with "perfect" reconstructed (it's clear that the lpf and the analog parts in the converter have an impact and can't be perfect). With perfect reconstructed I mean only the digital part.

I guess we are talking about the same thing: What you call the "digital part" is what I call the theoretical part.

The perfect reconstruction of the signal in the theoretical (or digital) world works via sinc-interpolation. Translated into analog gear this means you need so-called ideal low-pass filters to reconstruct the signal. Since you can only approximate an ideal filter there is no way you can perfectly reconstruct the original signal. How well depends on the DAC of your audio interface and the low pass filter it uses.

You are right that, in the theoretical domain, an 8 kHz sine sampled at 48 kHz contains exactly the same amount of information as when sampled at 96 kHz. Putting it differently, you can theoretically reconstruct the 8 kHz sine sampled at 96 kHz perfectly from the 8 kHz sine sampled at 48 kHz. (In practice, you still need an ideal low-pass filter to make sure you do not get artifacts above 24 kHz when upsampling from 48 kHz to 96 kHz.)

Best,
Tobias

[Tarekith]

The doubled bit and sample rate are cool too, when you later converting
your audio down to 16bit/44.1Hz or 48Hz !
It's simple math and not complicated down converting because of the fact
it just needs to be converted down to half.

False, that's not at all how downsampling works.

[simpli.cissimus]

The doubled bit and sample rate are cool too, when you later converting
your audio down to 16bit/44.1Hz or 48Hz !
It's simple math and not complicated down converting because of the fact
it just needs to be converted down to half.

False, that's not at all how downsampling works.

...maybe not all, but in my practical tests this gave me the best results.