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  #91  
Old 06-20-2003, 01:19 AM
Duardo Duardo is offline
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Default Re: Digi 001 discontinued?!

Quote:
Are all of these authors and engineers wrong?
<font size="2" face="Verdana, Arial">Not necessarily...because what we're talking about here is specific to the <20kHz information, and what they're talking about may be a little more broad. But sure, it's not all right.

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If they are, I need clear proof that higher SR have no apparent effect on the resolution of the digital audio by creating a more accurate representation of the original signal. And any noticable sound quality improvment is due to another factor.
<font size="2" face="Verdana, Arial">That MIT link I gave you does a good job, I think...otherwise a Google search under "Nyquist" will probably turn up some good information...but there's a lot of incorrect information out there. Maybe check the Audio Engineering Society's website as well.

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The 3D audio link has information about a listening test comparing 44.1, 48, 96 and 192. It is a good artical that I read a while back and is part of the basis for many of my statements.
<font size="2" face="Verdana, Arial">It's also from 1997, which is an eternity in "digital audio" years. Those findings are specific to that particular set of converters as well, and there's nothing there that indicates that what Lynn heard had anything to do with <20kHz audio being more accurately reproduced.

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Continuously varying sound is called an analog signal. Once the computer grabs the sound, it doesn’t have enough precision to store all the information about the sound in order to perfectly reproduce it. What the computer has stored is called a digital signal representation.
<font size="2" face="Verdana, Arial">This is correct, sort of...the computer does store a digital signal representation, and it doesn't have enough precision to store all the information about the sound...but it can be "perfectly" reproduced when run through a good D/A converter.

As always, you have to take "perfect" with a grain of salt...nothing's "perfect", not our ears, preamps, or anything...but it can be "perfectly" reproduced inasmuch as doubling the sampling rate wouldn't give us any added precision...just more points on that same curve that our D/A converter "redraws".

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Your sound card captures information about an analog sound signal by measuring its intensity at a given instant. This corresponds to one single point on the waveforms we’ve been looking at. In order to capture an entire waveform, the measurement process must be repeated at a high rate, usually thousands of times a second. Since the hardware has limited speed and memory capacity, there are only so many points it can capture. Any information between those points is lost forever.
<font size="2" face="Verdana, Arial">Yes, there are only so many points it can capture (in the case of CD's, 44,100 of them per second). And while it's true in a way that the information between those points is lost forever, if the information we're sampling is below the Nyquist frequency, all of that information is not lost forever, but can be recreated. Any information above the Nyquist frequency is lost forever, but we're not concered about that.

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To play back a sound, we just reverse the process and convert the digital samples back to an analog signal. Of course, the new signal will probably retain some of the staircase effect, so the reproduction won’t be perfect.
<font size="2" face="Verdana, Arial">Well, first off how scientific is somethign that says "probably"? And this isn't true. There is no "staircase" effect. You can see it if you look at a digital representation, but you can't hear it because the reconstruction filters in your D/A converter filter all of that stuff out. (And even if it were still there, it would all be out of our range of hearing anyhow, so it would be filtered out by our ears.)

You can sometimes hear what some describe as the "staircase" effect when listening to audio that's too close to the digital noise floor, but that doesn't relate to what we're talking about here.

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Number of samples or snapshots taken of a particular signal in a given amount of time (usually one second). Higher sampling rates result in more pieces of the true signal. When a sufficient number of pieces are generated, the pieces meld together to form a very close approximation of the original.
<font size="2" face="Verdana, Arial">No argument there. But again, I'll assume that this isn't a very scientific sourcesince it says the pieces "meld" together...but in any case, the digital samples themselves will always be an approximation. There would have to be an infinite number of samples for it to be exact...or it would have to be converted back to analog, which is what happens. The whole point of the Nyquist theorem is that once you have two samples per wave, your approximation is as close as it needs to be and any more information would be redundant.

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DVD technology holds data encoded at a 96 kHz sampling rate with a quantisation of 24 bits resulting a vastly superior audio reproduction format compared to standard CD.
<font size="2" face="Verdana, Arial">I don't argue that 24 bits can make an easily audible difference. 96 kHz, not necessarily...again, there's nothing here to back the statement up.

Quote:
The division of the sound into discrete samples means that there is some loss of information. The process fails to capture fluctuations in the signal which happen in the time interval between one sample and the next. Similarly it fails to capture fluctuations which are smaller than its measurement steps. So the resolution of the sampling process determines how much information about the sound is captured and how much is lost.
<font size="2" face="Verdana, Arial">Again...no argument with this one. 44.1kHz is sufficient to capture all of the sound up to 20kHz without losing any of it.

[quote]the cranesong hedd192's converters blow the 001's outta the water. i think perhaps the perceived diff between the 002 & 001 is just wishful thinking (it's certainly grossly exaggerated in the above statement). [/quopte]

I'd certainly aree about the HEDD. It's hard to say that the statement is "grossly exaggerated" though, since what constitutes beling "blown out of the water" is entirely subjective.

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You guys are talking about 2 different issues that are related to the same fact, and thats why this urination continues lol. If you guys would just add the two discussions together you will have the basis of digital technology (related to audio).
<font size="2" face="Verdana, Arial">What are those two issues? Digital representation of a sampled waveform vs the analog reconstruction of it?

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it points to another aspect of why higher sample rates may sound better: if the filters can be placed at higher frequencies, fewer artifacts reach the audible spectrum (?).
<font size="2" face="Verdana, Arial">That's absolutely true...but today's oversampling filters sample at an extremely high rate, having first filtered out everythign below whatever the Nyquist frequency may be with a gentle filter that doesn't touch what we can hear, then filter that digitized signal digitally with a very steep filter that wouldn't be possible in the analog domain, then decimate that huge sample down to a word at whatever sampling rate we're storing at. Pretty slick.

Gotta run now...

-Duardo
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  #92  
Old 06-20-2003, 09:23 AM
B-Grade B-Grade is offline
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Default Re: Digi 001 discontinued?!

Well, by guessing I mean anti-aliasing like oversampling and dithering. Dithering is a random "noise" put in to mask errors. Well, most dithering uses noise shaping today to hone in on typical problem frequencies in the conversion process.

Oversampling is technically an interpolation, which is a type of guesswork. Sure, an educated guess, but guesswork nonetheless.

My angle is more why do we need this correction if the sample rate is high enough not to be detected? Because we can hear the lower sample rate. If we couldn't, we wouldn't need all the fancy smoothing. Most DACs use the same chips. Its the analog componets used to mask the digital that make the great converters sound great. We souldn't need anti aliasing if the aliases weren't detectable.

On the 1mHz TC thing. A bit of marketing hype (pun intended). That is a 1 bit System, so divide 1 million by the 16 used bits and you get like 60 kHz. That system requires an extra couple of bits, so it effectively uses 18, which brings our math to around 48K. Many CD playes have used this technology for years. Fun stuff ain't it?

What's the subject again? 001...
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  #93  
Old 06-20-2003, 10:29 AM
clorox clorox is offline
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Default Re: Digi 001 discontinued?!

Once again, I must point out that Frequency and Sampling Rate are only related insomuch that you NEED a certain Sampling Rate to digitally record above a certain frequency. I think we're all clear on that.

But even if the only thing I was recording was the lowest note on a piano, higher sampling rates would still benefit the sound. This is because, unless I'm recording pure sin waves, each note is made up of infinitely complex waves that you can magnify time and time again and still find ever smaller and smaller waves and nuances. Increasing sampling rates allows us to more accurately portray these tiny subtleties.

Someone used a circle analogy, which is nice. How many square pixels (samples) do you need to build a circle? You could do it with 1 (horrible!), or 4 (better!) or 1 million (incredible!) or 1 billion or 1 billion billion or 1 billion billion billion.
The point is that each level of resolution brings you closer and closer to a perfect circle (which you will never, ever quite attain).

Whether or not we could tell the difference between a circle (or soundwave) made up of 1 million vs. 48000 squares is subjective, and may depend on the quality of our eyes and magnifying glass (or ears and speakers).

Bottom line: higher sampling rates more accurately depict true wave forms at any frequency.
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  #94  
Old 06-20-2003, 10:51 AM
clorox clorox is offline
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Default Re: Digi 001 discontinued?!

By the way, getting back to the ORIGINAL topic of this thread. . . .

What's the recording latency of the 002r?

Now, keep in mind I don't mean the Monitoring Latency, which is adjustable by setting HW Buffer size. I'm talking about the Recording Latency, which is a rock-solid, immutable number for each system.

I've posted a thread with a test to find this out here:
http://duc.digidesign.com/cgi-bin/ub...;f=24;t=022204

For Mac mbox, digi says it's 164 samples. For 001 , it's 51 samples. What about the 002 and 002r?

[img]images/icons/confused.gif[/img]
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  #95  
Old 06-20-2003, 02:58 PM
Chaasm71 Chaasm71 is offline
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Default Re: Digi 001 discontinued?!

Clorox, re-read my posts and Duardo's posts. We agree that sampling at higher and higher rates lets you SEE the 'wave form' with greater and greater detail. What you keep missing is that this greater detail corresponds to stuff human ears DON'T hear.

Imagine we have two samples recorded at 44.1kHz. I.e., they are separated by 1/44.1 thousandths of a second along the time axis. Any thing that causes the waveform to change appreciably (move up and down) between those two samples corresponds to something vibrating so fast that human ears don't hear it!

So, yes, if the job of a digital recording system was to make the most accurate graph of the audio waveform possible, then by all means, sample infinitely fast if you can! However, that isn't the job of a digital recording system. The job of a digital recording system is to make as accurate a graph of the part of the audio wave form that corresponds to what is audible to humans. That job is done by 44.1kHz, at least in theory. Now, when we start to talk about the imperfections of filters and what to do about it, that's another matter. The point is, the oversampling isn't done to get more resolution.

Charlie.
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  #96  
Old 06-20-2003, 03:07 PM
Chaasm71 Chaasm71 is offline
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Default Re: Digi 001 discontinued?!

Clorox, you misunderstood the circle analogy in a completely analogous way to how you misuderstand digital signal regeneration, which is kind of interesting. Let me try to explain my analogy a little better because it gets to the heart of the matter. Our DA doesn't reconstruct the sounds we hear by 'connecting the dots'. If it did, then yes, we'd need to do a better job than two points per sine wave. The tool used is called an inverse Fourier transform, and what it does is analogous to the following: go back to the circle. If I sample enough points on a circle to figure out where it's middle is, and what it's radius is, then I know enough to CALCULATE as many additional points as I need to draw the circle with arbitrary precision, right? That is what the DA converter does with the digital audio info we give it. The DA is clever enough that it doesn't need us to give it data that 'looks' like the original wave from. It only needs a couple of points in one period of a sine wave to figure out EXACTLY what that sine wave looks like. Is that more clear?

Charlie.
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  #97  
Old 06-20-2003, 03:16 PM
clorox clorox is offline
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Default Re: Digi 001 discontinued?!

I agree. You guys are right.

Theoretically, you want to sample as fast as you can, to get things as perfect as you can, BUT there is resolution beyond which humans can tell the difference.

A scientist wants the wave done perfectly for the goal of unerring precision. An artist wants the soundwave perfectly recorded because the lost information represents something "intangible."

The ENGINEER knows that you can cut it at 44.1, tell the other two that it's 96, and no one will be able to tell the difference!

One more question, though. Wouldn't you want to sample sound at a much higher rate, say 96kHz or so, NOT because someone claims they can hear the difference, BUT so you could deal with it at a higher degree of precision and minimize the rude effects of mathematical rounding/truncation? My work in CS/EE has taught me to keep things in the highest possible precision until the very last step. . . Isn't that the REAL reason?

Isn't that the reason most of us record in 24 bit and then master to 16? Don't you want the highest possible internal precision for all the number crunching before you lose the information forever at the very end?

Just a thought.

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  #98  
Old 06-20-2003, 03:33 PM
Duardo Duardo is offline
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Default Re: Digi 001 discontinued?!

Quote:
Well, by guessing I mean anti-aliasing like oversampling and dithering.
<font size="2" face="Verdana, Arial">Anti-aliasing, oversampling, and dithering are all different things...and none of them are "guessing"...the closest thing to "guessing" would probably be dithering, which as you mentioned is random noise added to mask errors...more specifically, quantization errors that don't relate to sampling rate, but to bit depth.

Anti-aliasing filters are filters that prevent aliasing. Aliasing is what happens when you sample a frequency above the Nyquist frequency...if you sample a 30 kHz sine wave at 44.1 kHz, for instance, the system will sample that wave at certain points along the waveform. When it "reconnects" those dots, there's only one way they can be put back together, which would be wrong because the sampled frequency wasn't within the Nyquist limit in the first place. So anti-aliasing filters remove everything above (and even a little bit below) the Nyquist frequency so the D/A converter doesn't have to guess...it knows.

Quote:
Oversampling is technically an interpolation, which is a type of guesswork. Sure, an educated guess, but guesswork nonetheless.
<font size="2" face="Verdana, Arial">Oversampling is where the audio is initially sampled at a much higher rate so that we can use an analog filter that doesn't affect the frequencies we can hear. Once the audio has passed through the analog anti-aliasing filter and been (over)sampled, it's filtered again digitally (where you can have an extremely steep filter with no phase shift) and then downsampled to 44.1. Works in reverse on D/A conversion. The D/A converter doesn't have to "guess" because it knows what the highest frequency can be and what at least two points per waveform are, which is all it has to know to eliminate any "guesswork".

Quote:
My angle is more why do we need this correction if the sample rate is high enough not to be detected?
<font size="2" face="Verdana, Arial">I assume you mean if the frequency is high enough not to be detected at a given sampling rate? If that's what you mean, the audio information will still be there, and will still be "detected"...but it will be misinterpereted as a lower-frequency signal...a signal below the Nyquist frequency. The anti-aliasing filters get rid of all of that.

Quote:
Because we can hear the lower sample rate. If we couldn't, we wouldn't need all the fancy smoothing.
<font size="2" face="Verdana, Arial">No, you can't "hear" a sample rate. What you're hearing is the inadequacies of the filters used to get rid of the inaudible frequencies. If the filters are good enough, they'll be totally inaudible...but they'll always be necessary.

Quote:
Most DACs use the same chips. Its the analog componets used to mask the digital that make the great converters sound great. We souldn't need anti aliasing if the aliases weren't detectable.
<font size="2" face="Verdana, Arial">No, we'll always need anti-aliasing filters. Even when sampling at 96 kHz, or 192 kHz, we need anti-aliasing filters, because there's always going to be something out there that's higher than the Nyquist frequency at any sampling rate. It's true that it's the analog components (including the anti-aliasing filters) that make the better converters sound they way they do.

Quote:
Once again, I must point out that Frequency and Sampling Rate are only related insomuch that you NEED a certain Sampling Rate to digitally record above a certain frequency. I think we're all clear on that.
<font size="2" face="Verdana, Arial">I thought we all were as well...but why do you say they're "only" related in that way? That's a pretty significant relationship, don't you think?

Quote:
This is because, unless I'm recording pure sin waves,
<font size="2" face="Verdana, Arial">Haven't we established that every sound we record can be broken down into individual sine waves?

Quote:
each note is made up of infinitely complex waves that you can magnify time and time again and still find ever smaller and smaller waves and nuances.
<font size="2" face="Verdana, Arial">Right, you can "find" them, but you can't hear them, and with good filters you can remove them without affecting what you can hear.

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Increasing sampling rates allows us to more accurately portray these tiny subtleties.
<font size="2" face="Verdana, Arial">Yes, it does, but we don't need to portray them if we can't hear them. These tiny subtleties you're talking about are high-frequency sounds we can't hear. You can't say that 44.1 kHz isn't enough to accurately reproduce material below 20kHz because it can't reproduce the sound above that frequency.

[quote]Someone used a circle analogy, which is nice. How many square pixels (samples) do you need to build a circle? [quote]

To make a proper analogy to our situation here...in the case of a circle, if you know where the center of that circle is, all you need is one poing (NOT a square pixel). Your D/A converter acts like a protractor, and based on the center point of that circle and one point, you can draw a perfect circle. It doesn't matter if you have one, two, ten, twenty, or a thousand points...they'll all be part of that same circle, which you can recreate perfectly just knowing the center and one point on the circle.

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You could do it with 1 (horrible!), or 4 (better!) or 1 million (incredible!) or 1 billion or 1 billion billion or 1 billion billion billion.
<font size="2" face="Verdana, Arial">That makes sense, but it's just not the way digital audio works. Samples aren't like square pixels. They're infinitely small points on a curve. The D/A converter, based on the sampling rate and the samples it's given, will recreate the audio waveform with no guesswork involved, as those points can only be connected with one possible curve.

Quote:
Whether or not we could tell the difference between a circle (or soundwave) made up of 1 million vs. 48000 squares is subjective, and may depend on the quality of our eyes and magnifying glass (or ears and speakers).
<font size="2" face="Verdana, Arial">Well, if that were the way that it worked, it wouldn't be subjective...one million would sound better. But a soundwave isn't made of "squares" or "steps". It's science, it's not subjective.

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Bottom line: higher sampling rates more accurately depict true wave forms at any frequency.
<font size="2" face="Verdana, Arial">I'm sorry, that is just plain wrong.

-Duardo
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  #99  
Old 06-20-2003, 03:36 PM
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Park Seward Park Seward is offline
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Default Re: Digi 001 discontinued?!

Duardo, Charlie and Nyquest are 100% correct.

I'd like to compliment them since very few have their proper understanding of digital audio and even fewer can explain it as well.

As far as hearing higher frequencies with the higher sampling rates, have you measured the difference on a 001 at 44.1k and 48k? There is NO difference.

I asked Digidesign the frequency response of the HD system at 96k and 192k. They thought both had 20k top end response. The reason it is limited to 20k? Because of the increase in noise if they let higher frequencies through.

Please test this statement since all the people I talked to were unsure of the frequency response. But it would be funny to discover that the response at 96 and 192 are identical.

So if you have a system that records at 96k, please send it some frequencies above 20k and let us know what you get back.
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  #100  
Old 06-20-2003, 03:57 PM
clorox clorox is offline
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Default Re: Digi 001 discontinued?!

Quote:
Originally posted by Duardo:

<blockquote><font size="1" face="Verdana, Arial">quote:<hr /><font size="2" face="Verdana, Arial">Bottom line: higher sampling rates more accurately depict true wave forms at any frequency.
<font size="2" face="Verdana, Arial">I'm sorry, that is just plain wrong.

-Duardo
<hr /></blockquote><font size="2" face="Verdana, Arial">On the contrary, in theory, it is as right as 1+1=2, as you point out in your other quote:

Quote:
<blockquote><font size="1" face="Verdana, Arial">quote:<hr /><font size="2" face="Verdana, Arial">Increasing sampling rates allows us to more accurately portray these tiny subtleties.
<font size="2" face="Verdana, Arial">Yes, it does, but we don't need to portray them if we can't hear them. These tiny subtleties you're talking about are high-frequency sounds we can't hear. You can't say that 44.1 kHz isn't enough to accurately reproduce material below 20kHz because it can't reproduce the sound above that frequency.
<hr /></blockquote><font size="2" face="Verdana, Arial">

I've already conceded that while the wave may be getting more and more "perfect" with more and more samples, no one would be able to hear any difference.

My question now is with the internals. The whole 96K thing can't be ALL marketing.

You guys are obviously very smart, so let me ask again: mathematically, wouldn't you want to deal with a very, very precise digital signal, at the highest sampling rate/bit depth, to do calculations on and then dither it down at the last possible instant? It's the same reason why the internal path of a lot of audio software is more precise than the inputs and outputs and, in a more real-world example, why the internals of accounting software deals with numbers to the ten-thousandth of a penny. Because constant truncation affects the end results.

Sorry if the statement seems ignorant. Just trying to "get my learn on." [img]images/icons/grin.gif[/img]
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