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  #171  
Old 06-23-2003, 09:16 AM
marty martin marty martin is offline
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Default Re: Digi 001 discontinued?!

This is getting real nerdy.
Forget the maths, use your ears.
If you can hear the difference with 96k and you like it, then buy the 002.
If you can't hear the difference stick with your little old 001 (and your really loud drumming) [img]images/icons/tongue.gif[/img]
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  #172  
Old 06-23-2003, 10:33 AM
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Park Seward Park Seward is offline
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Default Re: Digi 001 discontinued?!

I agree with the "use your ears" point.

But in order to give up half the tracks and processing power, I would have to hear a siginificant or obivious improvement to use a higher sampling rate.

My point is I don't. And I believe there is little there in higher frequencies to make much difference.
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  #173  
Old 06-23-2003, 11:11 AM
marty martin marty martin is offline
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Default Re: Digi 001 discontinued?!

Problem solved then.
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  #174  
Old 06-23-2003, 06:46 PM
metaltim metaltim is offline
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Default Re: Digi 001 discontinued?!

So uhh... is the digi 001 discontinued??? [img]images/icons/grin.gif[/img]

I thought that's what I would find out when I came in here, instead I got to take physics 208 again...


tim
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  #175  
Old 06-23-2003, 07:58 PM
Chaasm71 Chaasm71 is offline
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Default Re: Digi 001 discontinued?!

Hey there guys. I don't know if the digi001 is discontinued, so...sorry metaltim! Hope the Phys 208 flashbacks stop soon!

For any who are interested, I did find a cool link which does a great job describing Discrete Fourier Transform theory as it applies to signal processing. It's from a prof at Stanford's Center for Computer Research in Music and Acoustics. His name is Julius Orion Smith III (!), and his page on this stuff is http://ccrma-www.stanford.edu/~jos/mdft/mdft.html

If you are into such things...enjoy...otherwise...don't! [img]images/icons/wink.gif[/img]

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  #176  
Old 06-23-2003, 08:26 PM
Duardo Duardo is offline
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Default Re: Digi 001 discontinued?!

Sorry for the long post, I was sick over the weekend...and look at all the fun I missed.

Quote:
They are both 24bit digital representatives of the same wave (a very brief one with two sharp peaks), but the first one gets four samples; the second one only gets two samples to represent the same wave.
<font size="2" face="Verdana, Arial">Sorry, I don't quite understand your graph, but unless I'm understanding things wrong, if you're doing a wave with two peaks and only using two samples, you're represending a frequency that's above the Nyquist frequency.

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We run both signals through the same (very) simple amplifier (x2) and process them at their native sample rate.
<font size="2" face="Verdana, Arial">What kind of amplifier are you talking about? If you're just amlifying the level, the frequency won't change at all...

Quote:
Well, when I say Nyquist bastardized I mean it for several things. His analog to digital theories were based on morse code, not digital PCM audio.
<font size="2" face="Verdana, Arial">His theorem (not theory) is based on converting an analog to a digital signal. Doesn't matter whether it's audio, video, or whatever...the technology available to implement that theorem may change (filters, etc) but the theorem itself still stands.

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Another wrinkle:
Read an interview with Rupert Neve. He says he makes his consoles that can deal with 140+ db and 40-50K frequencies. He says this all affects the audio, weather you can hear those frequecies or not, they do affect the frequencies we can hear.
<font size="2" face="Verdana, Arial">Right...mostly in the phase relationships at lower frequencies, apparently...but that doesn't mean that we need to be able to capture those higher frequencies! If they are there and audible then they can be caputured by a converter that only operates within the 20 kHz bandwidth. It's only important that the analog circuitry be able to operate at those higher frequencies.

Quote:
whilst I agrre that better filtering schemes seem to be the most likely explanation of why some people prefer high sample rates, I'm just pointing out that higher sample rates also give you, at least theoretically, one or two more octaves of HF.
<font size="2" face="Verdana, Arial">No argument there. I've never stated otherwise.

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Personally, even if they actually do in practice, my listening has convinced me that I don't need these extra ocatves. But that's just me, some considerable people, including good old Rupert, seem to be claiming that it is possible to perceive these vey high frequencies, and I don't believe there is yet any irrefutable proof one way or the other.
<font size="2" face="Verdana, Arial">I don't know if there will ever be irrefutable proof either...but there have been no scientific studies that I'm aware of that prove that we can hear or perceive frequencies above 20 kHz on any conscious level, and there are plenty of explanations why we can perceive when a piece of analog gear has an anomaly at 50 kHz within the bandwidth we can hear...

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The little tiny up and down motions created by the high frequencies make the lower frequencies go up or down (that is relative to the null point).
<font size="2" face="Verdana, Arial">Right...and those lower-frequency oscillations can be captured by a 44.1kHz converter.

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An even bigger change is using a sony RMX-D100 console at 44.1 vs. 96, again using the radar. (the sony don't do 192).
<font size="2" face="Verdana, Arial">The fact that there's an "even bigger" change using different gear implies that there's something more to it than the sample rate alone, right? Otherwise it'd likely be consistent from one piece of gear to the next.

Quote:
1. 96k sampling systems will sample up to around 20-40k bandwidth.
Who says? Has anyone seen a spec that says higher sampling systems ACTUALLY record higher audio frequencies? What is the top end?
<font size="2" face="Verdana, Arial">Actually, Lucid's spec sheet for the AD9624 says it goes up to 40 khz, +/-1 dB, althogh both Apogee and Digidesign only spec their converters (the AD16 and 192 respecitvely) up to 20 kHz (at least with the data I was able to pull up with a short search, although I know Digi has more information buried in there somewhere as mentioned earlier in the thread).

Quote:
2. Low pass filters are more gentle with 96k systems than 44.1 systems.
Maybe not. How does the filter differ between 44.1 and 96? Is it the same filter?
<font size="2" face="Verdana, Arial">Well, if you were trying to filter out everything above 20 kHz with a simple low-pass filter, then sure, that would be the case. However, since that's not the case, I'm not sure how much of an issue this really is.

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3. Drinking Pepsi makes you feel younger.
Perhaps but the power of marketing should not be dismissed too quickly.
<font size="2" face="Verdana, Arial">Exactly. And I like Dr. Pepper.

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Why do many-(many) Heavy Hitter Pro's still use analog?
<font size="2" face="Verdana, Arial">Lots of reasons.

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Analog sounds Wonderful. Right..Why?
<font size="2" face="Verdana, Arial">Even that's not necessarily "right". Like everything, it's very subjective.

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1 of many reasons is HARMONIC OVERTONE Frequencies up to 200,000 HZ with the Analog domain...OK?
<font size="2" face="Verdana, Arial">To which piece of analog gear are you referring? Preamps, mixers, etc...sure. Tape machines...don't think so. There's a reason that many so-called "Heavy Hitter Pro's" track to tape, dump to digital, and mix analog.

[/quote]Why does Analog sound so Warm-Fat-Rich? [/quote]

Mostly due to distortion. "Good" distortion, yes, but distortion. Especially when you say "warm-fat-rich", that's what your'e talking about.

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1 of many reasons is HARMONIC OVERTONES up to 200,000 HZ...OK?
<font size="2" face="Verdana, Arial">No, I don't think so.

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Please do not continue to insult yourselves, by saying you do NOT Understand why 192 sampling has arrived, or why 48 would sound better than 44.1...HARMONIC OVERTONES...
<font size="2" face="Verdana, Arial">Actually, you may have a point with 192...but not at 48. If you understand the harmonic overtone series, you'd understand that you're not gaining any useful harmonic overtones with the extra tenth of an octave you gain going up to 48 kHz, even on a theoretical level.

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Please study the Harmonic Overtone series, and stop guessing...
<font size="2" face="Verdana, Arial">I'm very familiar with it, what's your point?

Quote:
I just mastered my CD at the Mastering Lab in Hollywood, and they went straight to analog-NO Dither, from my final mixs of .001/44.1 S/PDIF out-straight to my HHB. I have Waves Platinum and used the LinMB and L2 with 24bit Ultra on the dual mono imports of the sessions.
<font size="2" face="Verdana, Arial">Right, from a 44.1 kHz source...which just goes to show that it's not the sampling rate itself!

Quote:
This Digital disucssion about the Nyquest theory has confused the simple explanation that, in the Digital Domain, the rate you sample at---192 or 44.1--allows you to obtain the Harmonic (Overtone) frequencies of half the sampled rate, or 44.1=22.05 KHZ--or 192=96khz...Thats all Nyquest refers to..Very basic/simple..
<font size="2" face="Verdana, Arial">Right, we established that quite a while ago...

Quote:
OK..You say we cannot hear above 20k...but we definitly sense the overtones, and who cannot notice the differences in Analog and Digital?
<font size="2" face="Verdana, Arial">Why do you say "we definitely sense the overtones"? If your CD sounds so much better after having gone through all that analog gear, wouldn't you agree that it's something else? Especially since there are no overtones above about 20 kHz?

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Please do not be offended with my $.002...
<font size="2" face="Verdana, Arial">I'm not, although perhaps I should be.

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And, in fact, lots of people hear considerably less that 20Hz-20kHz because as we get older that range decreases.
<font size="2" face="Verdana, Arial">Yes, including many prominent engineers and designers.

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Ae you saying that there are microphones, mixers, recorders and speakers that can capture and reproduce 200kHz?
<font size="2" face="Verdana, Arial">Lots of preamps can do more than that. Lots of them have to to be able to reproduce sounds under 20 kHz accurately. Tannoy's Super Tweeters™ will go up to 50 kHz, and there are lots of microphones that go up to 40 kHz and beyond (Earthworks, Sennheiser and DPA come to mind) as well but again, just because an analog piece of equipment can go up so high doesn't mean that the digital gear we use with it needs to store all of that. If you could open up your converter (and maybe some of you can) I'd bet that if you measured the analog section of your converter, including the initial anti-aliasing filter, you'd find that it performend similarly.

Quote:
Park--It is Not the equipment...It is the TAPE...
<font size="2" face="Verdana, Arial">Actually, the tape (frequency-wise) is one of the weaker parts of the analog chain. Most people seem to like tape for the color it adds. That's why so many people these days track to tape and then dump immediately to HD...to capture that color and then not lose it due to the problems inherent to tape (shedding etc).

Quote:
I am saying (not my opinion, but many Audio Student Profesionals have suggested that) your Frequencies rising up from any fundamental, are Restricted by your sample rate in the Digital Domain, and un-restricted (rising upwards to 200,000 in rare cases) by analog TAPE.
<font size="2" face="Verdana, Arial">How can tape "un-restrict" overtones to 200kHz if we don't even have the transducers to get those frequencies to and from tape in the first place?

Quote:
On the nyquist thingie...
frequencies of 8K may be reproduced with that scenario, but not accurately. Square waves sound different than sine waves.
<font size="2" face="Verdana, Arial">Right...a square wave is a sine wave with an infinite number of odd harmonics...but once you get to a certain point (20 kHz, give or take) a square wave sounds like a sine wave to us. And long before then we're only hearing one of the harmonics.

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However, by themselves, theoretically the human brain cannot discern volume increments less than 3 db. 3 db! I can attest to mixing bands and making adjustments of instruments in .5 db and hearing a difference.
<font size="2" face="Verdana, Arial">I think we can all attest to that...it's not true.

Quote:
They are capturing color much past the range of human sight. Why? So they can further manipulate the image. This is their bit depth. Very analagous to our bit depth.
<font size="2" face="Verdana, Arial">Actually, no, that's analagous to our sampling rate. Bit depth is analogous to their DPI resolution.

Quote:
I'm wondering why nobody's freaked on SACD yet?
<font size="2" face="Verdana, Arial">People have...there's a thread on one of the Music Player forums where it's going on right now, I think in David Frangioni's forum...

Quote:
Keep all the info until the last step indeed. That is one of the nice things about the DMX-R100 is the 56 Bit summing bus. Pile it all on, paint the whole picture, then reduce for size.
<font size="2" face="Verdana, Arial">Right...but that's why all digital mixers and systems process at a higher bit depth...but very few at a higher sampling rate.

Quote:
The problem with our current sampling rate of 44.1k is that the timing difference between samples is 20-28 microseconds. This is not nearly enough to represent how we hear things in the real world.
<font size="2" face="Verdana, Arial">This isn't quite right...because although the timing difference between samples is less than how we hear it in the real world, we don't have to capture every "point" along a wave, just two...and that's enough to recreate them with the proper phase and everything. Plus, when you've run everything through our less-than-perfect transducers on both ends (including speakers that may be spaced three feet apart here, ten apart there, etc)...

Quote:
By the way, speaking of misleading, there are some EXTREMELY misleading charts in textbooks and on the web which show the frequency range of many instruments. Remember that these charts show FUNDAMENTALS ONLY.
<font size="2" face="Verdana, Arial">Most of them (the ones I've seen, at least) do state specifically that they're showing fundamentals only.

Quote:
What if you want to record the trumpet with two microphones at the same time (common practice to get that "room" sound"), or mix it in to a full brass band? You're gonna be missing out on some information necessary to 100% duplicate the interactions of all the sounds because the ultrasonic interactions (and their forays into the audible spectrum) will be gone forever. That's a scientific fact. How BIG a difference this missing data makes is the only debatable area.
<font size="2" face="Verdana, Arial">Sure, I won't debate that...but I do wonder how much of a difference that would really make when the tracks were combined together, and especially in the case where the instruments weren't actually recorded together, if you'd actually want to hear those beat frequencies? That's the type of things trained musicians listen for as they play, and if the musicians weren't actually playing together might we be better off not hearing those frequencies?

Please realize that that last paragraph was a purely speculative one.

Quote:
"...quite frequently after I've recorded an album, I'll dump the whole thing to analog before mixing it. And...I almost always use analog tape for mastering." --EQ Magazine interview with Gus Dudgeon: producer / engineer: Elton John, David Bowie, etc.
<font size="2" face="Verdana, Arial">This again shows that it's more likely the character of analog equipment (including tape) than the sampling rate itself since people keep talking about dumping back and forth between digital and analog.

Quote:
Key: Analog is *infinite* sampling. There are no gaps in the sound.
<font size="2" face="Verdana, Arial">There are no "gaps" in the "sound" with digital either.

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Analog tape compression gives more emotion to the sound.
<font size="2" face="Verdana, Arial">In other words, distortion....

Quote:
There exists a group out there who wants to capture high sample rates because they need it for precision placement of sharp transients in 3-d space because we can hear 5-15 microsecond delays between our ears. But that's another topic.
<font size="2" face="Verdana, Arial">Sure, if you want to change things on that level, just move your speaker a little...

Quote:
If the "tones" (harmonics) are out of the range of human hearing, then how do we hear them???
<font size="2" face="Verdana, Arial">There are plenty of harmonics (on the guitar example mentioned) that are within the range of our hearing.

Quote:
As you can see, a trumpet has 2% of its energy in the inaudible range. Crash cymbals have 40%. Jangling keys have 68%. I don't know what qualifies as "significant"
<font size="2" face="Verdana, Arial">I haven't had the chance to read the paper yet, and may amend my response when I have, but off the top of my head I'd say that when we're talking about energy in the inaudible range, none of it may be significant.

Quote:
but you need to realize that as you move higher up in frequency, the less energy it takes to produce sound, so 2% is a lot bigger than it seems.
In other words, 2% at 20kHz is a lot more than 2% at 20Hz. I don't have the energy to explain the physics; PLEASE trust me or research it yourself. Just think about how many more Watts a bass amp needs to produce the same level of sound as a guitar amp, and you get the idea.
<font size="2" face="Verdana, Arial">Just because less power is necessary to produce it doesn't mean that it's more significant...if you look at the Fletcher-Munson curves for human hearing that should give you a much better idea of what's significant at certain frequencies and what's not.

Quote:
Forget the maths, use your ears.
If you can hear the difference with 96k and you like it, then buy the 002.
If you can't hear the difference stick with your little old 001 (and your really loud drumming)
<font size="2" face="Verdana, Arial">Aaah, almost back to topic...sort of...

I'd change it to read like this:

•If you can hear the difference between the 001 and the 002, then buy the 002 (assuming it sounds better to you and not worse, or course)
•If you can hear the difference between recording at 44.1 kHz and 96 kHz with the 002, then record at 96kHz (again assuming it sounds better).

-Duardo
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  #177  
Old 06-23-2003, 09:19 PM
Park Seward's Avatar
Park Seward Park Seward is offline
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Default Re: Digi 001 discontinued?!

Thanks to an excellent recap from Duardo, I wanted to re-list some of the points made on this thread. Let's see what we have learned.

"Bottom line: higher sampling rates more accurately depict true wave forms at any frequency."

Wrong. As explained by Charlie:
"The point is, the oversampling isn't done to get more resolution.
Our DA doesn't reconstruct the sounds we hear by 'connecting the dots'.
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"

Duardo says:
quote:
------------------------------------------------------------------------
This is because, unless I'm recording pure sin waves,
------------------------------------------------------------------------
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.
------------------------------------------------------------------------
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. 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:
------------------------------------------------------------------------
Bottom line: higher sampling rates more accurately depict true wave forms at any frequency.
------------------------------------------------------------------------
I'm sorry, that is just plain wrong.

quote:
------------------------------------------------------------------------
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.
------------------------------------------------------------------------
Right, but we need to separate what we're talking about here...within the range of human hearing, the wave is not getting any more perfect with more samples.

But the main thrust of this discussion has been whether or not the material below 20 kHz is represented more accurately at higher sampling rates because there are more samples per waveform, which is not at all subjective...it is not.

I have to say that I've been involved in several threads similar to this one over the years, and I'm a pleasantly surprised that this one has progressed as it has. Usually people just get pissed off and start calling each other idiots and the thread just dies.


quote:
------------------------------------------------------------------------
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?
------------------------------------------------------------------------
Yes and no. Pro Tools LE, for instance, processes internally at whatever sampling rate you're running at, but at a higher bit depth (32-bit floating point). As far as frequencies are concerned (sampling rate), when we're mixing we're mainly just adding them together...we're not going to create harmonic content above our highest frequency by adding one frequency to another. Remember, any complex wave can be broken down into individual sine waves at various frequencies, and likewise adding multiple sine waves together will create a complex wave...but it won't create frequencies that are higher than any of the individual sine waves. (As mentioned earlier, other processes may do more complex things, and many of them do upsample to process and then downsample again...but we don't need to start off or end up with more detail than we can perceive, although I suppose there may be an advantage to avoiding even one of the extra processes...but then again, that doesn't have anything to do with a better representation of the audible range, which is what this thread seems to be about now.)


quote:
------------------------------------------------------------------------
Cut to the chase boys. Use the slide rules to tell why there IS a difference, not why there shouldn't be. I've heard nyquist bastardized for years. I can hear the difference. Everyone I've AB tested with can hear it.
------------------------------------------------------------------------
I've already answered that: You have more room for the filter to cut off in A/D and D/A conversions, so you don't need such a steep cutoff slope, and can create filters that create less distortion.

5) The quality of the Sennheiser mic is not relevant to this discussion. Stay on topic.
Err...you brought it up!

"Exactly. No one's claiming anyone can hear 40kHz. "

(From Apogee) Do we need to capture “audio” signals at up to 96 kHz? Obviously not – such signals don ’t exist. However, some recent research suggests that the human brain can discern a difference in a sound's arrival time between the two ears of better than 15 microseconds – around the time between samples at 96 kHz sampling – and some people can even discern a 5µS difference! So while super-high sample rates are probably unnecessary for frequency response, they may be justified for stereo and surround imaging accuracy. However, it should be noted that many authorities dispute this conclusion.]

"The trumpet with a mute shows significant energy up to 100 kHz before dropping into the noise floor."

That is not an accurate quote from the Tannoy paper. It says the energy drops to the noise floor at 100k, not that is has "significant energy up to 100 kHz". Very different statements.

So in one quote we hear Apogee say no signals exist above 96k and Tannoy say they extend to 100k.

"Alright. Why don't you put a 10k low pass filter on all your vocals? Because it would sound horrible, that's why. Sibilance in human speech has been shown to have energy above 40kHz."

I designed the production system for a network TV series where we recorded all the voice audio at a 22k sampling rate (to get four channels on a DV tape). As you know from Nyquest, 10k was the highest frequency we could record. NO one noticed the voices sounding any different from a 48k sampling, not the network executives, not the post team, not the producers and not any listeners.

"My idea of "top" quality is 2" 16 or 24/48 track analog recorded with a Neve, API, D&R, SSL or vintage Trident console, mixed to analog tape."

Ask Studer how many A827 Gold Edition recorders they sold last year. From their own web site, "But there is no doubt, the future of sound recording will be digital or already is in many places."

Thanks to all for an interesting thread. And I would bet that the 001 will be discontinued soon, if not already.
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  #178  
Old 06-23-2003, 09:23 PM
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Park Seward Park Seward is offline
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Default Re: Digi 001 discontinued?!

Quote:
Originally posted by Chaasm71:


For any who are interested, I did find a cool link which does a great job describing Discrete Fourier Transform theory as it applies to signal processing.

Charlie.
<font size="2" face="Verdana, Arial">Wow, Charlie. A paper discussing Gram-Schmidt Orthogonalization, Triangle Difference Inequality and the FFT of a Zero-Padded Sinusoid.

I think I died and went to Fourier heaven!!!
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  #179  
Old 08-01-2003, 08:20 AM
froyo froyo is offline
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Default Re: Digi 001 discontinued?!

Hello. I realize this post quickly turned into 5 pages on sampling rates. However the title and initial topic dealt with the 001. If you go to the products page on the Digidesign site the 001 is nowhere to be found. Nowhere. Not a mention. Nada. No Mix systems either. They are still in the compatibility documents, but then again so is the AMIII card which is no longer produced.

As for sampling rates a friend of mine will help me do some blind tests for my own benefit, which I may post into a different thread.
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  #180  
Old 08-01-2003, 08:31 AM
saggsy saggsy is offline
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Default Re: Digi 001 discontinued?!

yes is does rate a mention. It has a link to
Legacy Items:
- PT mix24
- Digi 001
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