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  #51  
Old 06-18-2003, 07:26 PM
gavinchan gavinchan is offline
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Default Re: Digi 001 discontinued?!

Quote:
In summary, please show me a digram of an analog signal. Then the same signal digitized at 48k and 96k. Then show me the resulting analog wave out of the DA. If some how it is exaclty like the original, please explain how the DA derived the information that was lost during the digitizing process.

Thanks for the discussion.
<font size="2" face="Verdana, Arial">You may see a square wave in your computer, but as soon as it hits the voice coil in your speaker, guess what? It takes time to accelerate and decelerate, it is no longer a square wave! That is where the extra information comes from. Simple.

And unless you have speakers that can break the laws of physics and perfectly represent the square wave inside the computer, and thus move from point A to point B with no time taken, [img]images/icons/rolleyes.gif[/img] then I have to agree with Duardo.
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  #52  
Old 06-18-2003, 07:35 PM
muspro muspro is offline
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Default Re: Digi 001 discontinued?!

Great info Chaasm71! I am with you. I still don't understand (as in my example), how a DA can derive the missing information. It isn't represented at all by the digital information.

I am talking about EXACT reproduction of the original analog material. Are you saying this is what is being done? Exactly! I still have a hard time believing that.

If that were the case, it sounds more like a lossless digital codec not a DA conversion.

More info please!
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  #53  
Old 06-18-2003, 07:44 PM
Carl Z Carl Z is offline
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Default Re: Digi 001 discontinued?!

Hate to break it to ya but that's not a sine wave. If it were, it would show the same value for each sample in your example.

Furthermore, if you knew that the highest frequency that you allowed your system to convert was half the sample rate, then you could deduce that the same value at each sanple must mean it's a sine wave at half the sampling frequency.

And then, if it's true that you could do the math to do such a deduction (which you can) then it's true that no additional information is needed to better represent that waveform.

C
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  #54  
Old 06-18-2003, 07:56 PM
snurble snurble is offline
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Default Re: Digi 001 discontinued?!

I think a key distinction that Muspro is missing is that Duardo is emphasizing how a signal is "reproduced" as opposed to how it is represented. The reproduction of the audio signal is ultimately much more important than how we see it on the computer screen when we're doing that "sample accurate editing" that we're so fond of. Sorry if it sounds obvious, but those little analog air particles jostling back and forth hitting our eardrums should be our true focus. They need to accelerate and then decelerate, which is an inherrently analog process. There are no square waves in real life, just curves of varying degrees of tightness. If our D/A and our clock are cooperating, then they are telling the speaker(ultimately) where to be at all those predetermined moments to allow physics to put the curves between the dots.
I imagine that things get pretty funky as you approach the limit ( say 22049.99Hz at a 44.1kHz sampling rate ), but that's what the filters are for.
If you accept that 20kHz is the human limit ( see below) then any extra points that would allow higher freqs to be reproduced are unnecessary. They will just fall on the curve that is already defined by physics. This actually helps me envision why the clock source is so important. If it's jittery then the signal's curves will be squeezed and pulled around. I'm not sure where the practical limit to the bit depth/"vertical resolution" is, but that's probably measurable as how much distortion we hear. I'm sure we can only hear distortion down to a certain point, similar to how we can only hear freqs up to 20k ( or see light only in the visible spectrum, or taste chemicals down to so many ppm, etc.)

Now as far as hearing sound higher than 20k goes, maybe using the word "hear" is a misnomer. I think I've read about studies that have tested for physiological effects form high freq sounds ( sorry, can't quote anything) and I could see how that info could alter our "perception" of a piece of music. Maybe that's why nothing will ever quite compare to being near an acoustic instrument as it's played. I'm sure we get all those overtones into our brains, even if not via the auditory nerves. Actually, if anyone knows more about this, please post a link or the title of a book or something. I'd love to know more...
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  #55  
Old 06-18-2003, 07:57 PM
Duardo Duardo is offline
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Default Re: Digi 001 discontinued?!

Quote:
Duardo - I never said anything about HD/TDM systems PCI cards. They are out of the price point we are discussing here and therfore do not apply. HD/TDM cards are sonically superior & have a low level noise floor that nothing in the lower price line (Mbox, 001, 002) systems can touch.
<font size="2" face="Verdana, Arial">I know, but you did say that Firewire means "cleaner and quieter operation over PCI design". I was just pointing out that the HD system uses a PCI design...so Firewire isn't better by design. If you took an external converter and ran it into a 192 IO, a Digi001, a Digi 002 and an M Box you'd wind up with the same thing all around.

Quote:
No pissing match here either. I teach digital audio and video courses at a major University and if I am wrong, I need to know. All info I have ever read or studied is not supporting what you are saying.
<font size="2" face="Verdana, Arial">Understandable. A lot of the digital audio theory I learned in school was incorrect (from my classes in the music department...my Acoustics class in the physics department taught it correctly).

Quote:
If there is absolutly no difference what so ever in higher sample rate recording, why have it?
<font size="2" face="Verdana, Arial">I think Charlie nailed the main reason above...marketing. If you have the chance to the AES convention in New York visit the various manufacturers of high-end converters. Oftentimes you can talk directly to the engineers who designed the products. Many of them will tell you that their converters sound identical at 44.1 kHz and 96 khz and 192 khz. But the demand is there for it...just look at these forums. And go down to Best Buy. That same kid who thinks the MP3's he's downloaded from the web sound great, but if the DVD he's buying isn't at 96 khz, it must not sound as good as it can, right?

I didn't say that there is no difference whatsoever. Depending on the design of the converter, there may be very audible differences. But there doesn't have to be. There are other arguments for it as well, but most of them involve our ability to percieve stuff above 20 khz, which isn't what we're talking about here...

Quote:
I say it is not about recording higher frequencies, who cares about that. It is about more resolution throughout the audible spectrum.
<font size="2" face="Verdana, Arial">Well, that's where higher-resolution recording isn't necessary.

Quote:
Why are top engineers and producers around the world celebrating the sonic superiority of higher sample rates. It isn't marketing either. It is science.
<font size="2" face="Verdana, Arial">I haven't yet seen any science that shows that recording at higher sampling rates increases resolution in the audible spectrum. A lot of it is marketing. A lot of it may have more to do with the sound of a particular converter than the technology behind it in general. A lot of it may be people who buy a new 96 kHz converter, turn it on at 96 kHz and leave it there, not even realizing that it may sound just as good at 44.1 khz.

Quote:
Every basic digital audio text I have ever read, clearly shows the difference in resolution as sample the sample rate increses. Thus being more like the original analog signal. You have repeatedly said this is not the case. Can you please provide some diagrams that would contridict this fact and all digital audio books that I have read?
<font size="2" face="Verdana, Arial">I think if you go back and re-read those texts you may find otherwise. Some of them may be wrong...I don't know what you've read, but a lot of things I've read aren't all right. I don't know how I can provide a diagram here. Did the way I explained things earlier not make sense? Please explain to me how a sine wave can be "more accurately" represented than what I described in my examples in previous posts.

Quote:
quote: "I don't disagree with you on this entirely...yes, more "slices" represent the analog wave more accurately"
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THANK YOU. This has been my whole point.
<font size="2" face="Verdana, Arial">No, it hasn't...because the only reason that more slices represent the analog wave more accurately is because they're also representing frequencies higher than what we can hear. Which is why...

Quote:
quote: "but we don't hear that accuracy"
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That is a very subjective answer. Many people do.
<font size="2" face="Verdana, Arial">It's not a subjective answer. If they're hearing something different, they're not hearing anything that has to do with more "slices" in the audible range...it's most likely something that has to do with the design of the filters in the converter, which is affecting what we hear in the audible range. The argument could even be made that the differences people are hearing is because of the presence of higher-frequency content, but we're not talking about that here. We're talking about the information below the Nyquist frequency, which is no more accurate at higher rates.

Quote:
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quote: "No, I'm not. I'm talking about sampling that same analog sine wave at different sampling rates. As long as the sine wave is below the Nyquist frequency, it doesn't matter how many samples you take of it, it will be reproduced exactly. Perfectly.
-----
I completely disagree. You had just stated above "more slices represent the analog wave more accurately". You can't have it both ways.
<font size="2" face="Verdana, Arial">Sure I can. When I said "more slices represent the analog wave more accurately" I was referring to a "typical" audio signal that has frequency content well beyond what we can hear. In my second quote, I was talking about sampling a specific sine wave that is below the Nyquist frequency (of the lower sampling rate). They're both different situations, so I can have it both ways.

Quote:
Can you please show me a diagram where a stored digital signal is EXACTLY like the original analog signal.
<font size="2" face="Verdana, Arial">The stored digital signal will never be like the analog signal. The stored digital signal is not something we can listen to. It's useless until it's been converted back to an analog waveform. If you have access to a tone generator and an oscilloscope at your school, generate a 20 khz sine wave. Look at it on the oscilloscope. Sample it at 44.1 kHz. Play it back. You will hear a sine wave and see that sine wave on your oscilloscope just as it went in.

Quote:
To take it one step further, I see you qualify the above statment by using the word "reproduced" refering to the DA process. I would also like to know how a DA converter can take that digital signal that has been quantized (wich is representing the analog signal the best it can limited by it's sample rate) and recreate the original wave form EXACTLY. That is imposible. It would only be posible if we were talking about a lossless codec, which we are not.
<font size="2" face="Verdana, Arial">I qualify that statement because you keep referring to digital signals and square waves and the like, which isn't what we're hearing. What we're hearing is an analog waveform. And it's not impossible. I did a quick Google search and came up with a relatively detailed explanation from MIT which you can see at

http://216.239.57.100/google_univ?q=cache:ftvwwJJjsrcJ:www-math.mit.edu/~gs/papers/newsigproc.ps+nyq uist&hl=en&ie=UTF-8

Most of that is over my head. I'll quote one of the easier-to-swallow conclusions here:

"The conclusion of the Shannon Sampling Theorem is always amazing to me, that a band-limited analog signal (continuous time and low frequencies only) can be exactly recovered from a countable number of samples (discrete time). This fact is fundamental to communications and digital signal processing. Sinusoids can be recovered from samples that are taken faster than the Nyquist rate.

This is not just a statement of good approximation by linear interpolation. It is a case of perfect reconstruction through interpolation by "sinc functions." The interpolation formula (12) will be stated again (with proof). We are sampling the theorem twice in one book, as Shannon would have wished. [/quote]

Quote:
Again, I take exception to the word "exactly" and "just like what went in". The DA will simply convert the digital square waves into an analog wave. All of the detail greater than the sample rate is lost forever. This is very diferent than a lossless codec.
<font size="2" face="Verdana, Arial">Yes, all of the detail greater than the sample rate is lost for ever...and that detail is only the information above the Nyquist frequency (or, again, a little less than that because of the filters used). That detail is not detail we would hear.

Quote:
No, in my expample what is stored is 2 samples. The space bwtween the 3 lines represents time, 1/40,000 sec. The wave DOES move during this time. However a sample can only give it ONE value which can never represent it's movement.
<font size="2" face="Verdana, Arial">Actually, the samples would occur on the lines, so your example would require three samples. If it is a sine wave those three points would all be horizontally aligned. (Again, you can't accurately sample a pure sine wave right at the Nyquist frequency, but for point of illustration we'll assume that something more complex was going on before and will happen after the period we're looking at here). The samples would be represented as dots, not lines or blocks. The D/A converter's job is to interpolate and connect those dots. Since everything above the Nyquist frequency has been filtered out, there is only one way to connect those dots, and that's with a sine wave.

You are right, a sample only gives one value which can't represent the wave's movement. That sample represents an infinitely small point in time. They're 1/40,000th of a second apart, but they are not each 1/40,000th of a second long.

Quote:
In summary, please show me a digram of an analog signal. Then the same signal digitized at 48k and 96k. Then show me the resulting analog wave out of the DA. If some how it is exaclty like the original, please explain how the DA derived the information that was lost during the digitizing process.
Thanks for the discussion.
<font size="2" face="Verdana, Arial">No problem. I really hope I have explained myself better this time. Have I?

-Duardo
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  #56  
Old 06-18-2003, 07:58 PM
muspro muspro is offline
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Default Re: Digi 001 discontinued?!

Carl Z,

Your working backwards. My very crude hand drawn waveform represents one complete cycle at 20k and two samples.

The difficult task here is that the waveform does not start on the sample.

Please provide a diagram of what my example would look like digitally and what the resulting DA would look like.
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  #57  
Old 06-18-2003, 08:26 PM
Chaasm71 Chaasm71 is offline
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Default Re: Digi 001 discontinued?!

Muspro, lets be careful about applying the word EXACTLY! That's a pretty powerful word. If I told you the location of the center of a circle and its radius, could you draw me a perfect circle? Of course not! Your pencil will draw a line with a finite thickness (mathematically, the line of a circle is infinitely thin)...Your hand will wiggle slightly as you try to draw it...there will always be distortions in any real world system, right? However, in theory, you have all the info you need to draw a perfect circle. It's the same with a AD DA situation. In theory if you sample at rate higher than twice the Nyquist frequency, and then apply a filter (which in the real world is less than perfect) to remove all info above 20kHz, you will exacly reproduce (as well as your DA lets you! It can be flakey too!) the AUDIBLE part of the original signal can be exactly reproduced. How does a DA converter take these two points and make a sign wave? Some engineer designed a circuit that understands Fourier's theory! In fact, I don't personally know the specifics of how it makes the conversion, I just know the math behind it that makes it possible. If I give you two points, you know that there is only one line that connects them. You don't have any choice. If I give you two points and tell you that they lie on a sign wave of a given frequency and amplitude and also assure you that they are within half a period of the side wave to eachother, then you have no choice as to what that signwave looks like. It has been completely determined.

Charlie.
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  #58  
Old 06-18-2003, 08:26 PM
muspro muspro is offline
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Default Re: Digi 001 discontinued?!

Hey Duardo,

I understand what you are saying. So is everyone saying that all things being equal, there is absolutly no benifit for sample rates higher than 44.1/48? Considering only resolution, the same converter sounds EXACTLY the same at 44.1, 48, 96 and 192! Then why the hell do we have the high SR. I have had so many people tell me their HD3 system sounds better at high SR! Is it just because the converters perform better at higher SR?

I would love this if it were all true!

If I could just have someone clear up my diagram, I would be set free. This diagram is an example that I was given to show how the Nyquist frequency can not be reproduced very accurately and why increasing the sample rate will yeild a better represention of high frequency information.

Thanks guys - I have to get back to work somethime tonight!
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  #59  
Old 06-18-2003, 08:42 PM
Carl Z Carl Z is offline
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Default Re: Digi 001 discontinued?!

Sorry I messed you up. I was trying to say that it is uniform but alas I am sleepy.

C
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  #60  
Old 06-18-2003, 08:44 PM
Chaasm71 Chaasm71 is offline
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Default Re: Digi 001 discontinued?!

Yes, higher sample rates let you represent higher frequencies better. But at a certain sample rate (44.1 or 48), you are accurately representing the highest frequencies humans can hear(20kHz). See the sound proofing analogy given above (which made me laugh like crazy!). So, to answer your question, there is no theoretical reason why audio sampled at 44.1kHz should sound different (to humans) than audio sampled at 96kHz on the same converter (unless there is weird monkey business going on inside the converter that causes it to use different circuitry when the sample rate is changed).

Charlie.
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