Protools 192k

[Park Seward]

Thanks for contributing, Nika. Your attempts at education will indeed help clarify digital processing to those less informed.

[Hardnox]

Nika, when you say "new system," does that imply that this new product will be replacing an entire Mix cubed system, or just the converter/software?

[ernesto]

96kHz and 192kHz really don't gain you much, except in the way of psychoacoustics and more accurately capturing transients. At 48kHz, if you sample a 23.99999 kHz square wave, you're going to get a sine wave back because you only have two points to sample on, and the reconstruction filter will round it off to a sine wave. With 96kHz, you'll get more of a flattened sine wave, and with 192kHz, you'll get a better approximation. While these tones aren't audible, the high frequencies fold over into the LF domain making kick drums snappier, snare drums more crisp, and preserves more of the ambiance of the listening environment. The problem comes when you downsample back to 44.1 for CD... you lose that fidelity. If you're planning on doing DVD or SACD production though, you might want to go that route.

[Nika]

<BLOCKQUOTE><font size="1" face="Verdana, Arial">quote:<HR>Originally posted by Hardnox:
Nika, when you say "new system," does that imply that this new product will be replacing an entire Mix cubed system, or just the converter/software?<HR></BLOCKQUOTE>

Obviously I'm not allowed to discuss certain things. I hope you'll understand.

Nika.

[Nika]

[QUOTE]Originally posted by ernesto:
96kHz and 192kHz really don't gain you much, except in the way of psychoacoustics and more accurately capturing transients.

This is not correct. higher sampling frequencies will not help capture transients that are below 20kHz.

At 48kHz, if you sample a 23.99999 kHz square wave, you're going to get a sine wave back because you only have two points to sample on, and the reconstruction filter will round it off to a sine wave.

First, we should clarify that the ear can't hear 23.99999 kHz. Secondly, the anti aliasing filter (not the reconstruction filter) will filter it into a sinewave, much the way that the human ear does.

The human ear can't discern between square waves and sine waves in it's upper octave. If it could it would show that the ear can hear the second harmonic of a fundamental that is in it's upper octave. Clearly this would indicate that the ear could hear higher than it's highest note. Since this paradox is indeed paradoxical and not correct, the human ear can't discern between waveforms (with a first harmonic of an octave or higher) in it's upper octave - just the same as the anti-aliasing filter in an A/D converter.

With 96kHz, you'll get more of a flattened sine wave, and with 192kHz, you'll get a better approximation.

True, but this is unnecessary since the ear can't hear the differences.

While these tones aren't audible, the high frequencies fold over into the LF domain making kick drums snappier, snare drums more crisp, and preserves more of the ambiance of the listening environment.

This is absolutely not correct. There is no possible way that the LF content in kick drums is benefitted by a "fold over" of square wave data in the upper register. Your statement is going to need a lot of explanation as "snare drums more crisp" and preserving "more of the abiance of the listening environment" wouldn't possibly be benefitted by any sort of "LF fold over", whatever that is.


The problem comes when you downsample back to 44.1 for CD... you lose that fidelity.

What fidelity? Please, do explain.

If you're planning on doing DVD or SACD production though, you might want to go that route.

Be my guest. With an understanding of the digital process I'm prepared to conserve tracks and DSP by staying at base rate as long as possible, but if you're developing for DVD and think that it will somehow benefit you, go right ahead.

If you're working on SACD then the entire conversation about PCM sampling rates is moot. I sure hope you wouldn't sample at 192kS/s or 96kS/s if you're planning on going to a 2.833MHz SACD DSD sampling rate!

Nika.

[Nika]

<BLOCKQUOTE><font size="1" face="Verdana, Arial">quote:<HR>Originally posted by emilano:
\He mentions a study by Japanese researches that demonstrated this was true at least up to 50 khz. After mentioning this research, Dave goes on to say that even though he takes this study as truth, he never let it stop him from making digital his medium of choice given all its other benefits and basically that it sounds good enough (great really) and that sometimes people go a little nuts trying to squeeze out the last bit of detail long after the returns start diminishing.<HR></BLOCKQUOTE>

The study you mentioned was spearheaded by a gentleman by the name of Oohashi and his counterparts.

The study specifically addresses that people were NOT able to hear any frequencies above 20kHz as people were unable to discern from two playback systems - one of which had an additional tweeter that rolled in aspects of a gamelan recording above 26kHz.. According to the study, not a single person was able to detect the presence of that tweeter.

The study goes on to discuss that, even though the differences were imperceptible to the human sensory system, brainwave patterns in the participants changed such that the playback system with the added high frequency data yielded more Alpha Electroencephelograms to be transmitted from the brain. Again, the participants could not detect this.

I have two copies of this study in my files. One was presented by Oohashi to the AES, and the other to the American Physiological Society. I had been discussing this study with George Massenburg in regards to a committee that I'm on with him in the AES that is developing some high frequency listening protocols. He had mentioned the study when he was in Europe recently at a gathering of some group of engineers and the buzz was that the Oohashi study had been brought into question as of late for reasons that I am unclear on. It had been completely discredited by this particular group of engineers.

In any event, I just wanted to clear up what the study was that you were referring to, and how it may factor in to this discussion.

Cheers!
Nika.

[Noiz2]

<BLOCKQUOTE><font size="1" face="Verdana, Arial">quote:<HR>Originally posted by Nika:


The anti aliasing filter in an A/D converter works much like an ear - brickwalling everything above a fixed point. A notable difference is that a good anti-aliasing is inherently higher than most human ears' frequency range.

So what data is it that you'd like to capture that is higher than the anti-aliasing filter in an A/D, which is higher than the human ear? And why would these subtle changes somehow make a difference to a band limited ear that is being fed everything from the signal that is within it's band?

Nika.<HR></BLOCKQUOTE>

The filtering that happens at the output stage of the D/A averages between the steps created by your samplerate. A higher samplerate has smaller steps more information and therefore less averaging going on. Any "real" information that falls between the samplerate "steps" is lost so by it's nature a higher sample rate will give a more acurate picture of your original source. I'm male and over the age of 12 so I probably won't hear it, but that doesn't mean it doesn't exist. Personaly this whole discusion is a bit pointless since the new hardware is already designed and isn't going to be changed because you or I say we don't care. You can do everthing at 22k if you want to, most people wont notice - whats the point? More information improoves quality, period. Thats why studio decks ran at 30ips. Whether its worth it is something for individuals to decide, but don't tell me it doesn't make a diference to me. I like to slow things down, way down, so it will make a diference to me.

[agradl]

, ...but I am indeed budgeting for the new system because of the other advantages it has in store, ...

Looking forward to see the new stuff, but since it's OMS I am not so excited!?

cheers, ag [img]images/icons/wink.gif[/img]

[daish]

Hello guys,

Have you seen this interesting topic about new Protools HD specifications ?


EQmag forums/Protools HD


Interesting...

[Greg Malcangi]

Hi Nika,

Thanks alot for your response.

<BLOCKQUOTE><font size="1" face="Verdana, Arial">quote:<HR>We should clarify that your test did not compare 48k to 96k and 192k, but rather that it compared the differences on a specific box, with a specific set of filters and a specific design of A/D converter circuit.<HR></BLOCKQUOTE>

Absolutely.

<BLOCKQUOTE><font size="1" face="Verdana, Arial">quote:<HR>BTW, which converters were they that you tried? I'm looking for converters right now, and I'll eagerly avoid those...<HR></BLOCKQUOTE>

The ADC's were all prototypes and for that reason it wouldn't be fair to the company to name and possibly shame them. BTW, the best commercially available converter I've heard is the Prism ADA8.

Thanks again,

Greg