Digi 001 discontinued?!

[Chaasm71]

The phenomena you guys are describing is called 'beating'. As an example, it happens when you tune a guitar and get the notes close, but not quite equal. It produces a low frequency oscillation that our ears hear as 'beats'. The beat frequency is equal to the difference between the two input frequencies. So, if you have two high inaudible frequencies that are close to eachother, they may well 'beat' in the audible range. However, even if you sample at 44.1kHz, we've already captured all audible info! You don't need to sample at a higher rate and capture the inaudible stuff. You will have already captured all audible 'beat' frequencies during the recording. So, yes, inaudible frequencies can interact and produce audible ones. However, when we record at 44.1, we already catch those audible 'beats', so we don't gain anything more by sampling faster. We've got it all.

As for the boards sounding different at 44.1 and 96k. If I was a manufacturer of equipment, I might tweak my super expensive AD DA to sound best at its highest sample rate, because if I didn't, I'd probably have a bunch of really PO'd customers asking me why the heck my super expensive AD DA sounds identical running at 44.1kHz and 96 or 192! Of course, I don't have any proof of these remarks, but it is certainly a possibility. Are you sure that the circuits are using all the same filters etc when you switch sampling rates? In fact, I'd guess that they don't, since the requirements on the filters are probably quite different depending on the sample rate.

Charlie.

[froyo]

By Chaasm71

As for the boards sounding different at 44.1 and 96k. If I was a manufacturer of equipment, I might tweak my super expensive AD DA to sound best at its highest sample rate, because if I didn't, I'd probably have a bunch of really PO'd customers asking me why the heck my super expensive AD DA sounds identical running at 44.1kHz and 96 or 192! Of course, I don't have any proof of these remarks, but it is certainly a possibility. Are you sure that the circuits are using all the same filters etc when you switch sampling rates? In fact, I'd guess that they don't, since the requirements on the filters are probably quite different depending on the sample rate.

<font size="2" face="Verdana, Arial">Hello. Chaasm71 this is exactly why when I was expounding my blind test scenario I said if higher sampling rates sound better, FOR WHATEVER REASON, use them. It really doesn't matter what the reason is now does it, as long as the final product sounds better. It wouldn't matter if it was voodoo, butter pecan in the circuits or solely a product of your imagination. As long as it sounds better. Conversely, if it doesn't sound better to you, it really wouldn't matter at all if the gear and the software where in fact better. Even if they had better filters, circuitry, converters, etc. If in fact they sound the same then you don't need them. It's as simple as that.

[Park Seward]

Originally posted by B-Grade:
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">Almost. Please realize that Morse code is just like... DIGITAL AUDIO with 1s and 0s!

Nyquist's Signal Sampling Theory

In the late 1920s, the only technology to preserve musical recordings was to copy sound waves in wax. Harry Nyquist, an AT&T scientist, thought there was a better way. He wrote a landmark paper (Nyquist, Harry, "Certain topics in Telegraph Transmission Theory," published in 1928) describing the criteria for what we know today as sampled data systems. Nyquist taught us that for periodic functions, if you sampled at a rate that was at least twice as fast as the signal of interest, then no information (data) would be lost upon reconstruction. And since Fourier had already shown that all alternating signals are made up of nothing more than a sum of harmonically related sine and cosine waves, then audio signals are periodic functions and can be sampled without lost of information following Nyquist's instructions. This became known as the Nyquist frequency, which is the highest frequency that may be accurately sampled, and is one-half of the sampling frequency.

Harry Nyquist thought of a way to take an analog signal (such as voice) and code it (just like with the Morse code) using ones (1) and zeros (0). For this, he invented something called a "CODEC" or coder-decoder. This thing that today is the size of a fingernail (a microchip) measures the input analog signal, codes the result of the measurement and sends this code down the telephone lines and trunks. It does so often enough so its peer at the other end of the line can reconstruct the voice signal almost as good as it was at the calling side. N. Erd calls the measuring of the signal "sampling." Good old Harry Nyquist also recommended that the number of samples per second for a good representation of the signal has to be twice as big as the number of Hertz of the fastest sine wave contained in the analog signal. Since the telephone only allows 4 kHz through the phone line, sampling for voice is done 8000 times per second.

Harry had it right.

[Park Seward]

We all may be laboring under some misconceptions:

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?

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?

3. Drinking Pepsi makes you feel younger.

Perhaps but the power of marketing should not be dismissed too quickly.

[guitates]

Why do many-(many) Heavy Hitter Pro's still use analog?

If YOU understood that, you would not have offered all the UNinformed info in this thread.

Analog sounds Wonderful. Right..Why?

1 of many reasons is HARMONIC OVERTONE Frequencies up to 200,000 HZ with the Analog domain...OK?

Why does Analog sound so Warm-Fat-Rich?

1 of many reasons is HARMONIC OVERTONES up to 200,000 HZ...OK?

For this discussion to center around the range of human hearing is Not having an understanding of what causes Analog to still be used by MANY intellegent/Famous Audio Producers.

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...

Please study the Harmonic Overtone series, and stop guessing...

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.

They do everything Analog, even if you bring them digital--(oh, of course some hard cases insist on LOUD Square Waves).

I am still in shock how good my CD sounds, with the good OLD .001 PCI...(OK..Joe Meek Pre's)

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..

OK..You say we cannot hear above 20k...but we definitly sense the overtones, and who cannot notice the differences in Analog and Digital?

Please do not be offended with my $.002... [img]images/icons/cool.gif[/img]

[guitates]

Clorox--Excellent reasearch...They say Analog to 45k?? Soooo...96k sampling would be Extremely desirable...

I think I want a .002...soon...

I will find the article that talks about the 200k potential... [img]images/icons/cool.gif[/img]

[Burton]

Originally posted by clorox:
<blockquote><font size="1" face="Verdana, Arial">quote:<hr /><font size="2" face="Verdana, Arial">The trumpet with a mute shows significant energy up to 100 kHz before dropping into the noise floor. It is a scientific fact that we can hear and measure "beats" formed by the interaction of these ultrasonics. These are facts, not opinions, and the results have been documented in labs using a spectrum analyzer. Not up to debate.

<font size="2" face="Verdana, Arial"><hr /></blockquote><font size="2" face="Verdana, Arial">But why would you think that the audible "beats" (which fall below 20khz) would not be recorded at 48khz sampling rate, as was stated before?

[clorox]

Originally posted by Burton:
<blockquote><font size="1" face="Verdana, Arial">quote:<hr /><font size="2" face="Verdana, Arial">Originally posted by clorox:
<blockquote><font size="1" face="Verdana, Arial">quote:<hr /><font size="2" face="Verdana, Arial">The trumpet with a mute shows significant energy up to 100 kHz before dropping into the noise floor. It is a scientific fact that we can hear and measure "beats" formed by the interaction of these ultrasonics. These are facts, not opinions, and the results have been documented in labs using a spectrum analyzer. Not up to debate.

<font size="2" face="Verdana, Arial"><hr /></blockquote><font size="2" face="Verdana, Arial">But why would you think that the audible "beats" (which fall below 20khz) would not be recorded at 48khz sampling rate, as was stated before? <hr /></blockquote><font size="2" face="Verdana, Arial">Bear with me, because everyone seems to be missing my point.

I do not "think that the audible beats (which fall below 20khz) would not be recorded at 48khz sampling rate." I'm merely saying that preserving the ultrasonic information allows the track to interact in a more natural way when combined with other tracks.

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.

I agree that recording in 44.1kHz will perfectly capture a track that is intended to be played by itself in mono. However, as soon as you want to mix it, mathematically process it, create stereo images of transients within it, etc., then the extra granularity beyond human hearing makes a difference. Again, how BIG a difference is what's up for debate; but there IS a difference.

Again, I reference these docs. I especially like the first link, which has three separate, scientifically-sound reasons for ultrasonic recording:

http://www.digitalprosound.com/Htm/S...ap2_Apogee.htm
http://homerecording.about.com/libra...y/aa060998.htm
http://www.tannoyna.com/professional...p_wideband.pdf

[muspro]

Hey Guys,

I'm still reading and learning from this thread. I agree with Chaasm that it is a very mature and informing thread even though the thread title is not accurate.

Clorox, I will concede that there are frequencies in the 50k, 100k and maybe even 200k range that exist and influence the audible range of 20-20k. We are of course recording those results at 44.1. You clarified your point by saying that if the digital system was able to record the 100k range (192 comes close), it would sound more natural when we mix those tracks together. I too might agree with that.

However, I don't think it would be the digital recording system that would be the weak link. Do microphones and preamps capture 200k? And if so at what type of dB loss? I would assume it would be non-existant information in the mix and would not effect the 20-20k range once mixed.

[guitates]

This info from John Vestman.com

Analog Q&A

"...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.

Q) John I want to know if its possible to get a top quality project from a PC recording software from start to finish. Some examples Cooledit Pro, Paris, and Cakewalk9. -Sean

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.

Half inch Analog masters are the preferred format of the major stars.

Digital recordings that I've heard so far have sounded thin on the bottom and vacant in the mids. Like little holes in the sound, or something.

Key: Analog is *infinite* sampling. There are no gaps in the sound. Analog "holds" the bottom end better, and adds a sweetness to the top end that I still haven't heard from any black "analog emulation" box. Analog tape compression gives more emotion to the sound.

Digital is all about calculations, and whenever the sound is altered, there are calculations going on inside the box. [img]images/icons/cool.gif[/img]