Being a complete dweeb at all of this a couple of questions.
Keep them coming
Have all your questions dried up now Lloyd?
I am following you here and was wondering while you are talking about 16bit DAC chips how are the higher sampling rate CDP's and other devices reaching 24bit.
I could write at length on this subject because there's a fair degree of sleight of hand going on with the manufacturers of the chips on this point. For me a 24bit DAC doesn't just accept 24bits of data, rather it resolves to 24bits, meaning that each output code can be individually recognised. No DAC chip that I've come across is very close to resolving 24bits because they're too noisy. Yet you do even get some manufacturers claiming '32 bits' for their chips. I call BS
24bits would mean a noise floor out of the DAC around -144dB to get the resolution of all the codes. The ESS Sabre comes closest I think around -133dB but this is just off the top of my head, I've not read the datasheet in a long time.
So to answer your question very briefly, no DAC reaches 24bits, but they do nowadays do a fair bit better than 16. They mostly do this with S-D modulators which shift the quantization noise of their low-bit DAC up to higher frequencies with noise shaping. So the high resolution does totally depend on filtering the output sufficiently to get a decent noise level in the audio band. If you measured the output in the full bandwidth you'd just see the resolution of the DAC part of the chip itself which nowadays is typically 6 bits.
Is this a series of 16bit DAC's working in parallel or in series or are there in fact 24bit chips that you can use for your design?
Given that the vast majority of audio products no longer use multibit DACs there's no advantage in paralleling DACs nowadays. That's because the noise between them isn't random, it would be correlated so no reduction when they're paralleled. That hasn't stopped at least one manufacturer I've heard of doing it though (can't recall who it was though, unmemorable over-priced product from what I vaguely remember)
Going by your conversation about the Nyquist relationship and the 1/2 value that seems to be what is driving the selection of your analogue filters lower 17khz selection. If you were using a higher frequency such as 96khx instead of the 48Khz that I think you are referring to wouldn't that solve the upper frequency cutoff situation?
I'm not sure here if you're talking about changing the sample rate we record at, or whether you're suggestion upsampling the 44k1 data we get off the CD? I'm strictly designing this DAC for the existing music out there, so the first is ruled out. The second brings some of its own issues in relation to SQ. Those are in my estimation good enough reason to stick with 44k1 for now until I exhaust all the potential design space...
I think that you have been basing all of your design based on the original Redbook 16/48Khz standard correct? Why lock yourself into that standard? Why not a higher bit rate and clock speed?
I recall a while ago reading that Willie Sutton was once asked why he robbed banks and the answer came 'Because that's where the money is'. I have a similar answer to his to give you - 'Because that's where the music is'.
