How is more than 16 bits better for CD?

Transnova is correct. I have the Consonance CD Linear and it is a 16 bit, non
oversampling/upsampling (unless you punch in the 88.2 option) with no op-amps and most importantly, no digital filter to mess up the sound.
I know this will sound lame but it seems to me that when you increase the bits, its like tracing over and over on an original drawing.
Sure, it will be 'like' the original drawing, but now you have a bolder line that creates its own version of the original, while at the same time masking/destroying the subtle clues and character of the original. Hence, a 'bolder' bass, 'sparkling' highs, and a somewhat bland middle as it 'guesses' where the lines should be drawn.
I was told that when you go up to (what was it?) 20 or 24 bits, it akin to having a reference point every nine inches from LA to NY.
Sounds like overkill to me when all one had to do was remove the digial filter and properly implement 16 bits as intended.
Don't want to start a firestorm here but its my take and I love the sound.

My understanding is that the extra bits are to increase the accuracy of the most significant bit since a 16-bit converter cannot, in practice, do a 100% linear conversion.

I believe that the Consonance "Linear" CD Players use a 16 bit/non oversampling DAC

Do any of the chip manufacturers even currently make 16bit chips for audio use?

Shadorne wrote:

" So the extra bits of a 24 bit DAC playing 16 bit CD data only help improve performance when PROCESSING the signal, such as upsampling and EQ adjustments. It does not improve the dynamic range of the original 16 bit CD data. A higher accuracy (24 bit DAC versus 16 bit DAC) will reduce artifacts from "rounding/truncation" in the processing. (Upsampling 44.1 Khz data being a processing step)."

I never thought of the truncating process that is surely ocurring during writing on a CD down to 16/44.1Khz from original 24/96 khz recording. It seems the increased word length and upsampling is trying to 'simulate' the Original signal that was truncated. Very interesting!!

Am I interpreting correctly?

Jason is right on. Thanks for clarifying it.

Arthur

but, to answer the original question, if it's just a 24 bit DAC, and it does not upsample the data, then the last 8 bits will just be filled in with zeroes, as Aball mentioned in the first reply.

Exactly. Thanks to Jason for clarifying completely.

Dither is used when decreasing bit depth and not really directly related to the original question (but important in making a 16 bit CD from higher bit studio recordings using Sony SBM or other dither techniques)

So the extra bits of a 24 bit DAC playing 16 bit CD data only help improve performance when PROCESSING the signal, such as upsampling and EQ adjustments. It does not improve the dynamic range of the original 16 bit CD data. A higher accuracy (24 bit DAC versus 16 bit DAC) will reduce artifacts from "rounding/truncation" in the processing. (Upsampling 44.1 Khz data being a processing step).

This is not to be confused with the inherent benefits of upsampling itself, which allow for better output filtering of the out of band noise with less artifacts on the in band signal. Another processing step that is often done both digitally and with an analog filter.

See Nika Aldrich, "Digital Audio Explained" fot the Audio Engineer for more details.

interesting thread here.
while sampling rate and bit depth are techincally "unrelated", you would want to have a great bit depth when changing sampling rates. you might want to upsample the 44.1 kHz data on CD's to 96 kHz to get the required filters further away from the audible range. an upsampling DAC might do this. now, whether upsampling via a DAC in real-time, or via a software application on your computer, it's just a set of mathematical calculations. the more bits you have, the less audible the "round off" errors will be. think of it as using 24 decimal places instead of 16 decimal places in all the calculations. in fact, when doing signal processing on a computer (including resampling), many software applications use 32 bit or even 64 bits when doing the mathematical calculations, and then drop down to 24 bit at the very end. so that's why you might want to have a higher bit depth when resampling data.

now, to commment on this thought:

Word length is increased by adding noise to the signal (aka, dithering)

actually, dither is usually used when decreasing the bit depth. dither is a very small amount of noise. the idea is to randomize the last bit of data that will be kept. if you have a 24 bit sampling depth, and you just truncate the data to 16 bit (chop off the 8 least significant bits), the result might sound a bit harsh. by adding dither, you essential add a small amount of white noise to the last bit, which the human ear finds more pleasing than digital errors produced by truncation. of course, then you get into a whole other topic of noise-shaping the dither. you can get the results as masking the errors of truncation even if you add noise (dither) that is mostly out of the range of hearing. that's what the Apogee uv22hr or the Sony SBM process is all about (to name just 2 of the many dithering schemes).

but, to answer the original question, if it's just a 24 bit DAC, and it does not upsample the data, then the last 8 bits will just be filled in with zeroes, as Aball mentioned in the first reply.

As interesting as the first response to your post is, sampling frequency and number of bits are unrelated. Adding bits doesn't have anything to do with Nyquist and filtering approaches.

Word length is increased by adding noise to the signal (aka, dithering). Interpolatuion (guessing) would only apply if you increased the sampling rate such that you had to 'fill in' the missing samples. In the case of more bits and a higher sampling rate, you would combine both interpolation and dithering.

It is technically called zero padding because the extra bits are in fact zeros. But its benefit is indirect. The real advantage is that it allows the specifications of the anti-aliasing filter to be looser by boosting the sample rate so the super sharp roll-offs needed to avoid the Nyquist limit (22.05kHz), and yet have enough bandwidth (20kHz) to reproduce all the music, aren't necessary. The filter's crossover can thus be higher in frequency and can also be more easily filtered out by the low-pass filter. This makes for a more linear outcome. My CD player is a 24bit Delta-Sigma DAC player and it is the best I have tried so the concept can work well despite the fact it seems improbable at first.

Arthur