speakers for 24/96 audio


is it correct to assume that 24/96 audio would be indistinguishable from cd quality when listened to with speakers with a 20khz 3db and rapid hi frequency roll-off?

Or more precisely, that the only benefit comes from the shift from 16 to 24 bit, not the increased sample rate, as they higher freq content is filtered out anyhow?

related to this, which advice would you have for sub $5k speakerset with good higher freq capabilities for 24/96 audio?

thanks!
mizuno
"Kijanki, are you implying that 24 bit data words have a "finer grain" than 16 bit data words? That each bit represents a smaller incremental signal level? "

"That's the basic reason to us more bits in each sample in digital signal processing of any kind, isn't it?"

That isn't my understanding of how it works, Mapman. Each bit of word length corresponds to 6.02db of dynamic range, or, technically, s/n ratio. So, with 16 bit words you get about 96db of dynamic range, and with 24 bit words 144db. And, of course, 65,535 voltage levels with 16 bits, and 16,777,216 with 24bit words. So 24 bit is "finer grain"?

Yes and no. Yes, there are 16M levels, but there's so much more voltage range to cover. If you reference the maximum level to, say, the 2v max line level used in consumer audio, and that's 24 1s in a row, then all zeros will be 144db below 2v, which would take cryogenic circuits or whatever to achieve. That means you're wasting the bits below the resolution of modern amplification systems, which is probably something like 100db below 2v of power amplifier output, including all amplification stages (I'm being very generous), which means you're wasting 44db of dynamic range, or about 7 bits of word length. So that means you're probably using about 17 out of 24 bits in a real system. And, of course, I'm probably generous by several db of s/n ratio in a real system, so 16 bits isn't far off of what is the resolution limit in a home system, and "finer grain", meaning a better s/n ratio, won't be audible by most mortals.

What 24 bit words are good for is eliminating digital clipping in the recording studio. There's no such thing as a 144db peak in music. :-)

So unless I'm completely misguided (always possible) 24 bit audio isn't really "finer grain" in the 96db of dynamic range that 16 bit audio can encode. 24 bit just goes higher or lower, or a bit of both.
Irv, keep in mind that it is generally accepted that signal can be perceived at levels that are significantly below the level of random broadband noise that may accompany the signal. 15db or more below, iirc. So amplifier noise floor is not really a "floor" below which everything is insignificant.

Also, quantization noise is significantly correlated with the signal, at low signal levels, and is therefore perceived as distortion rather than noise. Dithering will minimize that effect, but it has its limitations and my understanding is that it is often not properly applied.

That said, I think we are all in agreement that the main usefulness of 24 bits is in the creation of the recording.

Regards,
-- Al
That said, I think we are all in agreement that the main usefulness of 24 bits is in the creation of the recording.

Agreed.

However, I would add that high resolution recordings are targeted at audiophiles - so this new high resolution media is useful in that you tend to get a better quality recording that has NOT been heavily compressed for mass consumption. So they ARE useful to audiophiles but not so much from the "improved resolution" but mostly because audio that gets formatted this way tends to be a better quality master rather than a master intended for restaurant, pub, iPod & car FM radio play.
This has been a very interesting thread, and I've learned a lot. I have a question that bears on the value of high resolution audio formats, particularly the value of sampling rates higher than 44.1. Here is the question:

Is the preference for high resolution audio formats (24/96, 24/192, etc.) partly attributable to the fact that those formats have better temporal resolution?

I don't know the answer to this question, but it's been on my mind since reading a number of papers with passages like this:

It has also been noted that listeners prefer higher sampling rates (e.g., 96 kHz) than the 44.1 kHz of the digital compact disk, even though the 22 kHz Nyquist frequency of the latter already exceeds the nominal single-tone high-frequency hearing limit fmax∼18 kHz. These qualitative and anecdotal observations point to the possibility that human hearing may be sensitive to temporal errors, τ, that are shorter than the reciprocal of the limiting angular frequency [2πfmax]−1 ≈ 9 μs, thus necessitating bandwidths in audio equipment that are much higher than fmax in order to preserve fidelity.

That quote is from a paper by Milind Kunchur, a researcher on auditory temporal resolution. More can be read in this article from HIFI Critic. Kunchur's research is somewhat controversial, but I have found a number of other peer reviewed papers that seem to confirm that the limits of human temporal resolution is quite low, on the order of MICROseconds.

If that is true, then part of the advantage of high resolution audio formats might be the fact that they have superior temporal resolution, thereby providing more information about very short alterations in the music, i.e., transients. Or so the argument goes.

Anyone have an opinion about this?

Bryon
Byron,

There is no solid evidence for this - so it is indeed controversial. If a mere few microseconds were important then speaker and listener position would be dependent down to a millimeter or less than a tenth of an inch. It is generally accepted that 1 msec is the point at which time differences become audible (roughly 1 foot). Our ears are roughly 6 to 8 inches apart. Since temporal differences are detected by the difference in arrival at each ear - this all suggests that our "resolution" is close to that length which is about 0.5 msec in time ( at the speed of sound in air).

What these findings may be related to is "jitter" - it has been shown mathematically that non random time errors can produce audible "sidebands" around musical signals and that jitter of 1 microsecond can be quite audible due to our ability to hear these non-musical sounds or tones or sidebands. If you increase the sample rate then you will change the way jitter affects the sound - a significantly higher sample rate would likely reduce the deleterious effects of jitter. Some sample rates are noted for being better than others for reducing audible jitter. Benchmark found that 110 Khz worked better than other rates with the DAC chip they use.