Supposedly, the biggest advantage to 24/96 or 24/192 for that matter is the 24-bit data word, which allows a much lower noise floor, far more headroom during recording, and finer-grained loudness level tracking. To be honest I've never a difference between CDs and SACDs on my system, and I'm not sure the difference is audible. I have no doubt that for recording the extra headroom is a big advantage, but for playback I'm not convinced.
The higher sampling rates enable the use of less intrusive digital filtering circuits (better time-domain accuracy), which could make playback more accurate in the 20-20KHz bandpass, and maybe I just haven't heard the right equipment yet. Dinosaur that I am, I still haven't heard anything, from any source type or signal, that sounds better than my Levinson No39 CD player. |
"Maybe sufficient for sinewaves but not for the music because it would call for brick wall filters that have very uneven group delays (non-linear phase if you prefer) and will cause wrong summing of harmonics. Such setup will be OK for single frequency reproduction but will be very unpleasant with music (dynamic signal)."
I have no idea what you're talking about. The wrong summing of harmonics, and it'll be very unpleasant? I don't know about that, Kijanki. You talk like a technically competent person, but then you make these outlandish claims. So if these summed harmonics will be so screwed up, why is it that those of us with very good high frequency hearing and high quality speakers can't hear anything very unpleasant? And if they do sound so unpleasant, why when I listen to higher res stuff through a Benchmark DAC it doesn't sound noticeably better?
I think you're exaggerating the issues, and wrapping the arguments in technical-sounding reasons that really don't alter the music audibly. |
Kijanki, your example of a 15KHz sine wave and "three points", implying poor reproduction, isn't the case, according to the well-proven Nyquist Theorem. 20KHz is reproduced as accurately as 1KHz with a 44KHz sampling rate. As for the phase shifts from steep digital filters, this is what over-sampling was invented to address, and eases the difficulty of designing a good anti-aliasing filter. For playback 16/44 is probably better than your audio system. (As I mentioned, for recording you might want the headroom of a longer word-length.)
Bob R is correct about some amps being a limiting factor in hearing the better s/n ratios of longer word-lengths. Most high-end amps are rated as having noise levels about 100db below full power. Since amps usually have about 25-27db of gain that means that their s/n ratio is only about -75db at 1 watt, or well inside the capabilities of 16/44. Krell amps, for example, are about the best, and they have s/n ratios in the mid-high -80db range at 1 watt. |
Kijanki - 20KHz reproduction with a 44KHz sampling rate is perfect for sine waves, not "coarse". A higher sampling rate doesn't improve accuracy within the frequency response of the lower rate, it just extends the frequency response. That doesn't mean I think digital recording and reproduction is perfect overall, it just means that in terms of capturing the frequency domain information at 20KHz, 44.1KHz sampling is completely sufficient to perfectly capture the sine waves. I think people confuse digital sampling with analog interpolation, and it isn't similar. |
First of all, I totally agree about compression running rampant these days, especially for drum kits. My wife is a drummer, so I know what drums really sound like, and only a few recordings give you a hint of their dynamic range. In fact, she and I both lament that a lot of modern recordings don't even use real drums any more, only those electronic travesties, for ease of recording.
To expand about about differences I've heard with hi-res on the DAC1, sometimes I think I hear a difference, in that some hi-res recordings seem to reveal something I've never heard before, but then I go back to a CD and hear similar things. Or I find an awesome recording on CD that sounds better than anything I've heard on hi-res. For example, oddly, I still haven't heard a piano recording superior to the ancient Telarc CD of Malcom Frager playing Chopin. That old Soundstream recording even forces an odd conversion to 16/44, and it still sounds great.
As for Kijanki's comment that the s/n ratio of most amps is specified at 1W, I say check again. All of the obvious ones I've checked reference full power, and most can't break -85db at 1W into 8ohms. JA's measurements in Stereophile are very interesting in this regard. (His measurements are the only reason I read the magazine.) |
Amazon says unavailable, Al. Too bad. |
I dunno, Kijanki, I randomly looked at two good power amps in JA's testing, a Moon and a Pass, and they were both had measured s/n ratios of about -84db at 1 watt, which is actually excellent performance. You keep forgetting that most power amps have about 30db of gain. Measuring s/n ratio at full power is sort of cheating for marketing's sake. A speaker with 95db/2.83v/1m efficiency, like my old Legacy Focus, will let you hear the hiss from such an amp rather readily. So I still contend that for listening the amps are the limiting factor, not well implemented 16/44.
As for your comments about Nyquist, it would seem your real thesis is that digital reproduction isn't very good, even with a DAC1. I still wonder, why does it sound so good if you're correct? I'm missing something. |
"I think that our hearing ability ends up slightly above 16-bit perhaps 18-20bits but I'm more concerned with sampling rate because low sampling rate in addition to phase shifts in steep low pass filters increases quantization noise (or size of square steps to make it simpler)."
It's statements like that, Kijanki, that make me wonder if you know what you're talking about. The width of the data word has nothing to do with our hearing ability. How many bits per word determines how many loudness levels there are. It's the sampling rate in KHz that determines how high the frequency response goes. You know that, right? |
Kijanki, are you implying that 24bit data words have a "finer grain" than 16bit data words? That each bit represents a smaller incremental signal level? |
"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."
Maybe, but it is very difficult to believe this is the case when listening to music or other complex sounds, like movie dialog or foley. I've always been leery of effects 70db or more below the music level, regardless of the component in question. |
Well, Bryon, that was a very interesting article. I'm not sure what to think after reading it... is this yet another investigation into a micro-problem that doesn't really affect music reproduction, or is it a significant factor? I certainly don't know. I can't even venture a guess.
Anyway, Kunchur admits to listening to cassettes. I haven't heard cassettes for many years, but 16/44 CDs must sound like a revelation by comparison. ;-) |
"Sounds like the consensus is that the original CD redbook format engineers did a more than adequate job, at least in theory."
Well, I would have said the red book CD is "just adequate". I think reducing the word length or the sampling rate might have audible effects, so "just adequate" comes to mind.
As for comparisons to "other media", are you talking about analog? For me at least, vinyl doesn't come close, no matter what you spend. Analog tape can sound very, very good, but except for our own master tapes, where would one get source material? In the past I've heard 1/2" 15 or 30ips tape with Dolby sound spectacular in a studio, but how many of us have access to such a thing?
Here's a better question: is there any significant amount of content coming out of the recording industry these days that requires a medium superior to a 16/44 CD? Perhaps some rare examples in the classical venue, but it seems like none in jazz, rock, pop, country, new age, or whatever. Or am I wrong? |
