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Bruce,
It's a great question and one I have spent a lot of time wondering about. I don't claim to have the complete answer but here's a couple of thoughts.
Firstly, in an open-systems world where there are so many (PC, O/S, sound card, media player - including arbitrary plug-ins, source file etc.) independent variables in the equation it is extremely difficult to assert for a generic case that a given file is being replayed bit perfectly. I saw a post on another forum that put it very nicely "with PCs there are very few absolutes". It is even harder to prove in practice and requires a fair degree of expertise and specialist equipment, not to mention a lot of time! But as usual, wherever there is an absence of fact mumbo jumbo races in to fill the vacuum.
Secondly, so few people appear to care! Witness the fact that afaik there isn't a single sound card on the market that was designed to simply produce a S/PDIF stream with the least amount of interface jitter possible. As you pointed out in your original post, compared to gamers and home theater users the PC world clearly does not consider audiophiles a significant target market.
It may not be the Rosetta stone but there are the beginnings of what looks like a really promising digital audio Wiki on the Benchmark Media website. It includes a set-up guide for computer audio and deals with issues like bit transparency, word length, even some surprising revelations about kmixer! It might go some way towards answering your questions, I found it very useful.
Rather than regurgitate the usual superstitions and half-truths these folks have taken nothing for granted and actually tested things themselves. For instance, they have mesaured the bit transparency of a number of media players using an Audio Precision digital signal generator and analyzer. Hallelujah!
Talk about a breath of fresh air. The site is "under construction" so some of the articles are only stubs, but given its solid roots in sound engineering practice and empirical measurment I think this is a really welcome injection of fact into this murky debate. Way to go Benchmark!
For my own part, I bought a 24/96 capable card in order to:
Bruce,
Just saw this piece in the Benchmark Media E-Update from May (Okay, I'm a little behind on my reading).
Bruce,
This is a great question. It is also a great example as to why it is initially quite hard to understand how to use a PC as a music server.
I have not seen a good, easy to understand article on how to make sure you are getting a bit-for-bit accurate data stream from a PC's digital output, whether it is S/PDIF, TosLink, USB, Firewire, whatever. All the manufacturers appear to want us to assume that their equipment is perfect. It is digital after all and digital is perfect, right?
JA's quick response is very good. You want to avoid Window's basic driver and to make sure that all "enhancement features", such as equalizers, etc. are off - as well as to make sure not to use any form of digital volume control.
Benchmark Media's guide is helpful and may be enough to answer your question on 16-bit transparency.
As to why we care about 24-bit: we are having fun trying to get the best possible sound, of course! Seriously, it matters to me as I record at 24/88.2 (or 176.4) and I want to be able to accurately monitor the sound without truncation, as well as put together and play back DVD-A's that play back everything I was able to capture in the recording. Thus, I care whether I can get all 24 bits out of my computer into a DAC and out of my DVD player as well. Now if I was only as good of a recording engineer as my equipment...
Trivia: I record at multiples of 44.1k as getting the best sound for a CD is most important to me. Those who are most concerned with DVD audio record in multiples of 48k as this is the standard audio sample rate for DVD's. I don't know the history behind the difference.
I think that a good, solid article on the basics of using a computer as transport would be great including affordable, accurate sound cards, suggested players, etc.
Sounds like rubbish to me. Again, as you noted, how can anything be "better" than the straight output of what is going in just by adding some zeros? I can believe that interpolating (extrapolating?) a larger word from preceding and subsequent words may actually improve the sound but I am not sure given my contradictory experience with it.
My Audio Alchemy DTI Pro32, which I have between the computer and my Accuphase CD player (used only for its DAC), can be set to interpolate to 24 length. The Accuphase can receive it without truncation. So I tried it of course and I did not like what it did to the sound. It definitely was doing something but I did not like it so I use it in its "straight though" mode.
But I also have experience that tells me it can improve the sound. A few years ago, I had the same AA unit but between an old CAL CD player modded to put out digital, and a Sonic Frontiers DAC (forget model but it was their top-end, Stereophile Class A unit). The SF unit could receive a 24 length and the AA in 24 bit mode did improve the sound. Go figure. From these two experiences, and given that the Accuphase is such a superior sounding machine, I can only conclude that having what poops out the back should be exactly as was fed in. In other words, maybe the bit manipulation was masking or compensating for the flaws of the SF unit. I dunno but I defer to my experience with the superior equipment. Of course I could be wrong as there are more than a few variables here and I do want to believe that these inference engines can and should be able to calculate better data but this is not my experience with my own, better equipment.
Now the AA unit does make a big improvement in sound in my system because of its jitter reduction function as that Audigy card apparently puts out quite a bit of jitter. I was shocked actually as the improvement was more than the correction from the old modded CAL unit (at least as I remember). Anyway, jitter is evil and must be addressed in these computer-as-a-transport systems. 24 bit? I just dunno.
Bruce,
I don't doubt your observations but I think your rationale is a bit off. You appear to be mixing up word length (bit depth) and sampling frequency which are two different things.
Each sample encodes the voltage of the signal at the instant the sample was taken so the word length defines the accuracy with which the voltage is measured. A longer word is like finer graduations on a ruler. If a distance is only measured to the nearest mile (c.f. 16-bit) there is no way on earth to subsequently figure out exactly how many inches (c.f. 24-bit) it was.
Neither interpolation nor extrapolation will help here; there is no way that increasing the word length can improve the resolution of the sample post hoc. Note, if the sample is being manipulated, e.g. digital gain or DSP is being applied, the extra 8 bits in a 24-bit word act like "decimal places" which can more accurately capture the result of a division applied to a 16-bit word. The resolution of the sample is not increased but as the manipulaton is more accurately applied it is decreased as little as possible. If the playback chain is completely bit-transparent however, all you end up with is a payload of 8 zeros at the LSB!
I would still be very interested to hear Benchmark's explanation for asserting that unmanipulated 16-bit programme sounds better played back at 24-bit resolution. I'm not quite ready to call it rubbish, but if true it would certainly debunk all this theory!
Successive samples capture the frequency information, i.e. how the voltage varies with time, and additional samples can indeed be interpolated from their neighbours. Converting to a higher sampling frequency ("up-sampling") cannot add any high frequency information (other than distortion) to the signal above the Nyquist frequency, for example a 16/44.1 ADC will have "brick walled" the original signal somewhere below 22.05kHz. It can however help to ameliorate some of distortion artifacts caused by the real-world imperfections of the DAC filter such as aliasing and amplitude ripple.
Or, summa summarum, increasing bit depth cannot really help you in a truly bit-transparent replay chain whereas up-sampling potentially can.
The problem with digital audio is just that there are so many moving parts, many of which are inter-dependent, that it is often really hard to tie effect back to cause with any certainty, you can just never eliminate enough of the variables! In other words your observations could be easily explained by completely different phenomena.
Hello! My apologies for the delay in response
Thank you, Elias.
Welcome to our forum, Elias.
John Atkinson
Editor, Stereophile
Thanks for that clarification Elias, makes perfect sense now.
To any non-gEEks out there curious to understand the concept of dither I would warmly recommend the Wikipedia piece on dither in digital audio. I have struggled to explain this concept without getting bogged down in the math myself on a number of occasions. IMHO this disarmingly clear and elegant explanation sweeps the board.
Unfortunately not all Wikipedia articles reach the same heights, as in the case of this choice piece of drivel from the entry for Bit Depth:
Frequency information encoded in one discrete sample? Hmmm...
Caveat lector!
I've edited the Wikipedia entry on bit depth: Bit Depth.
If you get a chance, take a look at it and see if anything is still unclear.
Thanks Elias, your rewrite is perfectly clear. I noticed one minor error which I fixed.
Oh yeah! We do have 0's in our decimal system these days, don't we? I remember when we had to use 'aught' before we had 0's.
Thanks John! I love 'talking shop', so the pleasure is mine.