speakers for 24/96 audio

Byron,

I appreciate your questions. You are definitely curious enough to look into this and I commend you on your interest.

However, poor Kunchur seems a very confused individual.

His test simply shows how two pure tones can interfere with eachother in a way that becomes audible. However, his conclusions are completely bogus. The listener is NOT hearing temporal time-domain effects of microseconds. The listener is actually hearing changes in the combined resultant waveform which has been altered by offsetting one source to the other (combined - meaning both waves and including all room reflections).

As I explained, this will lead to TOTAL destructive interference of the primary direct signal as heard by the listener at an offset of 2.5 CM. This is like a signal that is TOTALLY out of phase. The direct sound will be inaudible and all the listener hears is all the sound around the room (reflected sounds). Since we detect the direction of sound from the relative timing of the wave front (or nerve bundle triggers) across each ear then we lose that ability when a signal is out of phase.

Poor Kunchur is conflating things in a bad way - this is bad science.

However, his remarks about speaker alignment and panels are partly valid. It is almost certain that large radiating surfaces can cause the kind of interference at certain frequencies like what he achieved in this experiment. This manifests itself in a speaker response that has many suckouts across the frequency spectrum. In fact the anechoic response of a large panel response will look like a comb with many total suckouts across the frequency range. The result is that some sounds and some frequencies will not be as tightly imaged as with a point source speaker. Since most sounds are made up from many harmonics this effect will not be complete but on the whole it will lead to a larger more diffuse soundstage with some sounds imaging precisely and others more diffuse than when compared to a point source speaker. There is an audio tool called a flanger that is used for electric guitar - it achieves a similar effect but even stronger.

Also Jitter is not audible in the sense you describe. It is audible when non-random jitter over a great many 1000'sa and 100,000's of samples combines in a way that introduces new frequencies. We hear those new frequencies that are created by the non-random modulation of the clock (random jitter is just white noise at very low inaudible levels).

We are totally UNABLE to hear jitter effects on a few samples.

07-05-11: Almarg
...we are not hearing the nanoseconds or picoseconds of timing error itself. What we are hearing are the spectral components corresponding to the FLUCTUATION in timing among different clock periods...

That's what I suspected, Al, but I wasn't sure.

And thanks for your explanation of jitter. I was aware that jitter resulted in frequency modulation, but I didn't know that it was a kind of intermodulation distortion. Your explanation is much appreciated.

Shadorne - You may be right that Kunchur's methodology is flawed. I've read a few other experiments on human temporal resolution with similar methodologies, but my memory of them is a little vague. In any case, I have a question about your observation that "Some sample rates are noted for being better than others for reducing audible jitter." I'd be interested to hear a technical explanation for why that is the case.

Finally, I have a general question about high resolution audio that anyone might be able to answer:

My understanding is that the principal advantage of larger bit depth is greater dynamic range. What is the principal advantage of higher sampling rates, if it is not better temporal resolution?

Bryon

What is the principal advantage of higher sampling rates, if it is not better temporal resolution?

None above redbook CD except it allows cheaper and better filtering which may improve very slightly the audible band. However, higher sample rates do allow you to go to one bit resolution (like SACD format which is a DSD stream but SACD has very high levels of out of band noise - so to be honest I am not sure I accept that it is even as good as 24 bit/96)

What is the principal advantage of higher sampling rates, if it is not better temporal resolution?

Yes, as Shadorne indicated the principal advantage is that it dramatically relaxes the rolloff requirements for anti-aliasing filters (in the recording process) and reconstruction filters (in the playback process). Or it makes it possible to avoid the use of techniques that have been used to relax those requirements, which have their own tradeoffs (e.g., oversampling + noise shaping).

It should be kept in mind that not only will 44.1kHz sampling be unable to capture signal frequencies at or above 22.05kHz, but the a/d converter used in the recording process must not be exposed to those frequency components. Otherwise "aliasing" will occur, resulting in those ultrasonic frequencies appearing in the digital data as audible frequencies.

Therefore an a/d converter that doesn't use oversampling or other special techniques must be preceded by a low pass filter that is flat to 20kHz, but has rolled off to the point of inaudibility in about 1/10th of an octave, at 22.05kHz. That is an EXTREMELY sharp rolloff, and, besides being expensive to manufacture, that kind of filter can have the sonic effects Kijanki described above in his post of 6/27, and the effect described in my second post of 6/30.

In contrast, 96kHz sampling would make it possible to allow more than a full octave for the same rolloff to occur (at 48kHz rather than 22.05kHz).

Similar considerations apply to the playback process, with respect to the "reconstruction filter," which refers to a low pass filter used to eliminate the stepped character of the d/a converter device's output.

Best regards,
-- Al

Sounds like the consensus is that the original CD redbook format engineers did a more than adequate job, at least in theory.

So does that mean that when we hear deficiencies in specific redbook CDs compared to other formats (say R2R or very good vinyl even) that it is because of poor execution somewhere in the implementation , either in the recording or playback process or equipment, or most likely even both?

I like to think so but I have not heard the near perfectly created CD on the near perfectly executed system in a viable test scenario compared to other high quality formats yet that would confirm this, so I am not so sure reality reflects the theory in practice quite yet?

Has anybody else heard something specifically that has them convinced?

On my system, I think the issue is a wash, but I have done some imperfect a/b comparisons on very high end dealer reference systems where it was not, especially in comparison to R2R and with better large scale orchestral recordings involving massed strings in particular.