I am really confused by this discussion so I went to my trusty Wikipedia to look up DAC. There is a good article but it disagrees with what is being put out here. The number of bits is an OUTPUT signal and the more bits the better resolution and less need for interpolation between pulses. This means that the higher numbers of bits the closer the DAC can come to a true analog output. The input to a DAC is how fast it can read the digital material (sampling Frequency) and still be accurate on its output. This is expressed in terms of frequency such as 192 kHz. Audio must be sampled at a rate at least twice the expected spectrum. This is why even the low end DACs have a sampling rate of at least 44.1 kHZ because the audio spectrum for humans goes to just a bit more than 20 kHz. Do I have a misuderstanding of the process somewhere?

The following definitions came straight out of the article:

Resolution: This is the number of possible output levels the DAC is designed to reproduce. This is usually stated as the number of bits it uses, which is the base two logarithm of the number of levels. For instance a 1 bit DAC is designed to reproduce 2 (21) levels while an 8 bit DAC is designed for 256 (28) levels. Resolution is related to the effective number of bits (ENOB) which is a measurement of the actual resolution attained by the DAC.

Maximum sampling frequency: This is a measurement of the maximum speed at which the DACs circuitry can operate and still produce the correct output. As stated in the Nyquist–Shannon sampling theorem, a signal must be sampled at over twice the frequency of the desired signal. For instance, to reproduce signals in all the audible spectrum, which includes frequencies of up to 20 kHz, it is necessary to use DACs that operate at over 40 kHz. The CD standard samples audio at 44.1 kHz, thus DACs of this frequency are often used. A common frequency in cheap computer sound cards is 48 kHz—many work at only this frequency, offering the use of other sample rates only through (often poor) internal resampling.

Monotonicity: This refers to the ability of a DAC's analog output to move only in the direction that the digital input moves (i.e., if the input increases, the output doesn't dip before asserting the correct output.) This characteristic is very important for DACs used as a low frequency signal source or as a digitally programmable trim element.
THD+N: This is a measurement of the distortion and noise introduced to the signal by the DAC. It is expressed as a percentage of the total power of unwanted harmonic distortion and noise that accompany the desired signal. This is a very important DAC characteristic for dynamic and small signal DAC applications.

Dynamic range: This is a measurement of the difference between the largest and smallest signals the DAC can reproduce expressed in decibels. This is usually related to DAC resolution and noise floor.

There are many downloadable sources that are encoded / recorded / mastered at 24/192 bit rate. http://www.computeraudiophile.com/state-of-audiophile-music-downloads has many of the sites where you can download high res music.

Good point.

The 990 has 24/192 DAC's, so I daresay the 997 will as well. I think that Jimna was saying 32-bit/192kHz sources didn't exist, though.

yes i was speaking of 32/192 sources not existing. i record all my audio in 24 bit so i have a ton of material to pick from, more than most im sure.

FYI, a ton is available on the Live Music Archive for free and legal download. http://www.archive.org/details/etree

support live music

The benefit of 32 bit DAC's with 24 bit source material should be minute, similar to dithering. Presumably the extra 8 bits will just be assigned all zeros and thus not contain any real data anyways. An extra bit or two can come in handy for the multiple serial calculations (which are likely with more sound processing, i.e. room EQ) to prevent build-up of error, and may give better tolerance to noise, but the difference certainly won't be as significant as the jump from 16 bits to 24 bits. Law of diminishing returns. I'd be impressed if someone could hear the difference between 23 bit vs 24 bit source material on an ultra-high-end system anyways.

Needless to say, 32 bit processing will give no benefit over 24 bit processing when using a 16 bit source.

As mentioned above, the implementation (and supporting circuitry) of the DAC can be far more relevant to the quality.

well then with no real benefit of adding bits, i see this as a point of fault in design. the best programming and processors out there can cause artifacts when changing bit rates (dithering), so i call BS on the process and wouldnt want it in any of my gear. processing in native bitrates is always preferred if possible, and if not it should only happen when a real benefit is releavent. why else has there been such a effort to make bit perfect transports and interfaces in digital audio? ...because otherwise it intruduces artifacts and impurities to source material which is audible on a quality system, thats why. this is exactly why analog still consistantly sounds better than digital sources. the vinyl guys are pointing and laughing at us.

i call marketing shannagans.

I should clarify that dither is typically used when needing to reduce bit-depth, and perhaps that is why Jimna doesn't like it. Personally I have nothing against dithering as done properly it can benefit a recording. Perhaps I shouldn't compare the two since since the processes are significantly different.

I would also much rather use a digital source when applying advanced processing to the signal, as analog has all sorts of phase & linearity problems. And all else being equal (ie, quality & cost), I would choose a 32-bit DAC over a 24-bit DAC because it would allow maintaining the full 24-bit source resolution throughout the processing. The question is how much more does a 32-bit DAC cost than a 24-bit one and are the analog output drivers just of good in terms of quality? Gonk has already covered this. The difference between Audyssey and Trinnov will be more significant than the difference between 24 bit vs 32 bit, just like the brand of DAC (e.g., implementation) will make more difference too.

I don't see it as a marketing ploy when Onkyo doesn't dwell on using 32-bit DAC's. It's possible that they choose them because the parts will have longer availability than the previous generation, granted I'm sure public relations didn't mind.

I would think that the 32 bit DAC would be able to better approach the ability to produce an analog output than a 24 bit DAC as it can resolve the output to a much finer degree. It doesn't mean it will sound better, only that it will cost more.
To me, digital music sounds as close to the original right now as it is ever likely to. There are so many manipulations between the actual generation of the sound of an instrument and the playing of music in your music area that no one can actually discern what equipment sounds closest. If it sounds good to you then it must be the best.

thats a fact, sage words.

XeononMan is essentially correct. There are a number of factors here.

The two processes going on in audio systems are: Analog to Digital conversion (ADCs or A/D) and then Digital to Analog conversion (DACs or D/A). Digital media, which is pretty much everything now, has already undergone the A/D process, which leave our processors to do the D/A conversion.

BITS
More Bits is usually better as the number of bits is directly proportional to Signal-to-Noise ratio (SNR) for the A/D conversion process. Each bit in the A/D adds approximately 6 dB to the achievable SNR of the sampled system (e.g. 24 bit A/D has a theoretical SNR around 144 dB) .

The rule of thumb for DACs is that you should have at least 2 more bits of accuracy in the D/A process than in your digital stream. Why? A well designed DAC typically has errors of +/- 2-4 LSBs (Least Significant Bits). By using a DAC with a couple more bits than your data you can feel pretty confortable that you DAC isn't adding noise.

Sampling Frequency
Higher sampling frequencies help too. As noted the highest frequency representable in the digital data is 1/2 the sampling frequency. The bad news is that any frequencies above 1/2 the sampling frequency fold back into the band of interest unless they are filtered out. This is called aliasing. With lower sampling frequencies you need a very sharp filter to minimize aliasing.

The process works in reverse on the D/A (DAC) side. The DAC creates an analog stepped waveform with frequencies up to half the reconstruction frequency and it needs to be filtered in order to recover the pure analog that we want to hear. Lower frequency DACs require steep filters.

Digital Processing
The third thing going on (I lied when I said two) is the digital processing which is such a huge topic that I'm not really going to go into it. I'll just say this: Digital processing can actually increase the sampling frequency and resolution (# of bit) of the original sampled data which can result in a higher fidelity signal than that recorded IF DONE PROPERLY. This doesn't create new data, you still have what was receorded, but it allows some more advanced processing to remove noise.

Bottom Line: It is possible to start with 24/44.1k bit data and process it up to 32/192k preserving the music and decreasing the noise.

Beware the Specs
The magic is still in the design and execution though. A bad design can make a 32 bit DAC perform worse than a well designed 24 bit DAC. Forget all the theory and let your ears decide.