Charlie- You should post your conclusions here when you get a chance. Who knows, might spur Outlaw a little...

(Get it? "Spur" ? Hee, hee.)

(I'm personally a little curious too, because I have a high quality stereo analog parametric eq that I'm going to use for room node taming for my mains. But I haven't gotten around to hooking it up yet, because it's going to take a lot of manual measurements to do it right...)


[This message has been edited by Kevin C Brown (edited October 05, 2002).]

Just a reminder, people. Nyquist said that the minimum sample rate for reproducing a singnal is 2x the frequency of that signal. Not that you'd be able to reproduce the correct shape of the signal at that sample rate. A 22 kHz tone sampled at 44.1 kHz basically turns into a triangle wave. There are benefits of sampling higher than 44.1 kHz, even if human ears can't hear all of the frequencies that become possible at higher sampling rates.

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Matthew J. Hill
matt@idsi.net

Actually that's not true. It seems like it should be, but its not. Also, a 20kHz tone, by definition, is a sine wave.

Nothing personal Matt.

EDIT:

OK - here are some ways to think of it that don't involve numbers and letters dancing together in some sort of unholy rite:

- The simple 'connect the dots' shape is the stored data, not the waveform. The information about the shape of the waveform is spread over many samples and is no longer stored in a single sample.

- The simple 'connect the dots' shape contains frequencies higher than the Nyquist frequency. Or we could say the D/A step produced these frequencies, however we want to look at it. In any case, they will be removed in the next step.

- One really cool 'high level' way of looking at it is this - the dots are not the waveform they are just data, and the only waveform that can connect the dots and not contain elements over the Nyquist frequency is the original waveform.

[This message has been edited by charlie (edited October 07, 2002).]

But what about waveforms which are not sine waves? Now I know that Fourier (I think it was Fourier?) said that they can be broken down into a sum of sine waves, but is it not still possible that a non-sine wave could be at a frequency that's within our hearing range, whereas overtones that may be present in it may not be? Couldn't our ears still tell the difference between that waveform and one that's at the same frequency but doesn't contain the overtones? If you approximate it with a sine wave I still believe that audible information will be lost.

I did not think that DACs would round off the data to produce a sine wave but now that you say it, it seems like the natural thing for them to do.

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Matthew J. Hill
matt@idsi.net

The short answer is no.

Here are some things to do and think about:

The case you opened with is easy to visualize, where the wave is exactly at 1/2 the sample rate. Try a more complex case. Use a piece of graph paper and draw a 15kHz sine wave on it. Now draw vertical lines to represent the sample points at 44.1kHz. Now, either transfer those dots to a second sheet or connect them with line segments or something to show the resulting sample data more clearly. The result is a ghastly looking mess, yet the wave can be recovered.

The human ear responds to tones up to roughly 20kHz, tones being sine waves. Any tones higher are (to most people) not audible.

Obviously there will be some errors due to quantization and some problems due to flaws in the actual implementation, but it seems like the Ultracurve has a good implementation since it measures very well and has a lot of very favorable listening tests behind it as well.

I'm looking forward to trying them out.

If Outlaw was interested in this space they could implement and build a 10in/10out magic box and imbue it with capabilities far beyond what the Ultracurve does just by virtue of having all those channels in a single box, although the digital interconnect looks very flexible on the Ultracurve too.

For instance Outlaw could implement not just graphic/parametric EQ, gain and delays, but also a flexable BM system, phase vs. frequency adjustment and so on. The potential of something like this is almost unlimited, and if they could implement a digital audio out from their next higher end prepro they could even avoid that D->A->D->A issue, letting the 'box' do D->A at the last instant.

Obviously something like that would be prefered, esp. if they went 24/96 or something.

Well I got my first UltraCurve tonight, but the matching microphone is backordered until the end of the month. Grrrr.

It's a nice looking unit, anyway. And maybe I can have the menu system mastered by then....

In reading this thread I am a little confused by the DAC sampling at 24Hz issue. As a HT newbie the whole digital to analog conversion process is quite confusing. While reading a review of the new Audigy 2 soundcard over at tomshardware.com I came across a description that made sense at the time, but doesn't jive with what I (think I) read here. Could one of our myriad of knowledable sound engineer types that hang out in this forum glance at this description of the DAC process and maybe comment if what is written is accurate? I keep hearing things about differences not being audible - and sometime's I am thinking Hz refers to higher pitch or lower pitch and other times to sampling frequency... ughhhh... anyway the link is here: http://www.tomshardware.com/video/02q4/021106/index.html

On a side... the soundcard seems quite sophisticated using a Cirrus Logic CS4382 (maybe a sister of the 950 Cirrus) for decoding.

Hertz is a shorthand for 'cycles per second' and can be applied to many recurrent things.

http://www.audiovideo101.com/dictionary/hz.asp

There is an important theorem that states a digital sampling system can store all the data needed to restore an arbitrary waveform if (1) the waveform has no frequency components above 'N' and (2) it is sampled at a rate of at least N x 2.

This is where 'sample rate' becomes important.

EDIT:

I should answer the question is guess:



This is no joke - in real life it works a lot better than the picture might indicate. Also, as noted in discussion above, there is a tremendous amount of variation in quality of implementation. And for purposes of simplification toms' has omitted a few steps. See the link above for a good accurate rundown on digital recording.

[This message has been edited by charlie (edited November 14, 2002).]

I think that the sampling rate has to be higher than twice the highest frequency of interest. Just 2X won't cut it, but 2.0000001X will. A subtle (to many) distinction, but important.

I am seriously considering the purchase of one (or two) of these Audigy 2 cards to use as an audio o'scope and spectrum analyser for my audio tinkering hobby. I have never purchased a creative brand sound card because of the quality (or lack of) of the record/playback result. I have prefered the Turtle Beach soundcards because they have given me much better results, but the new Audigy 2 looks really inviting.

Paul

Paul

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the 1derful1

[This message has been edited by Paul J. Stiles (edited November 14, 2002).]