44.1kHz/16bit & 192kHz/24bit versus 2.822MHz/1bit! - Page 2

Think of it like that: You could easily increase the information content of DVD-Audio by a factor of 10 by increasing the sampling rate to 1,92 MHz. Would this mean any audible increase of accuracy? It's doubtable -- more likely it's a waste of data. And the latter applies in some way to DSD's ultra-high sampling rate: It's needed to achieve at least the mentioned decent amplitude resolution, but not for frequency resolution. On the other hand, a 4-bit data format with a sampling rate of 705.6 kHz (1/4 of DSD) would provide a higher amplitude resolution throughout the spectrum -- namely 9.5 bit at 16 kHz and 19 bit at 20 Hz. The ratio between usable resolution and the amount of data will become even better with even higher bit depth.


As logical as it sounds, I wouldn't rely on this. After all there are a lot of «experts» around who think that DSD is crap and at best usable for low-fi consumer level. That said, this doesn't correspond to my listening impression at all.

Why are you equating precision with S/N ratio? Whatever reduces noise to such fantastic depths, precision/accuracy for me expresses itself in the number of samples per second as well as the number (and accuracy) of possible amplitude values. It looks like this: 2.822 MHz enables 176.4 samples per wave cycle, that is 176 amplitude steps for a 1-bit system, at 16 kHz, which is the equivalent of said 7.46 bit. At 20 Hz, there's 141,120 samples = amplitude values per wave cycle, which is the equivalent of 17.1 bit.


I fear the expense/understanding ratio would be miserable in my case. Maybe you can explain it?

http://members.chello.nl/~m.heijlige...ory/hauser.txt

http://www.diyaudio.com/forums/showthread/t-50939.html

Hi Glassman,
Are all the new ADC chips 1-bit sigma delta design?
Is there multi-bit or non-sigma-delta ADC out there?

usually DACs are multibit sigma delta, but multi means 5bit for example.. 1bit DACs are history now.. ADCs, the cheaper ones still use 1bit modulators.. as an example cheap AKM ADCs or the new ADCs with DSD output capabilities from Burr Brown.. previous top of the line AKM ADCs have 2bit modulators I believe and the currently best ADC on the market has 5bit modulator..
I don't know of any non delta sigma ADC meant for audio use on the market todays..

this one probably is non-delta sigma, as you can see it's rather outdated and obsolete these days.. also notice the high price, you can find the best sigma delta ADCs much cheaper..

So this means almost all modern D/A and A/D use sigma-delta architectectue. That means PCM and DSD is basically the same thing, doesn't it.
With DSD, any processing has to go through PCM and convert back.
With PCM, the signal comes from low-bit sigma-delta stuff and has to be converted back for playback.
Only if things are mastered in the analog domain and the you can have pure DSD.
On the other han, pure PCM basically does not exist.
So the basic difference is where, when and how to interconvert low-bit delta-sigma signal between hi-bit PCM.
The so-called format war is just a mirage
While I have too few hi-rez PCM discs to discuss their merits, I am quite convinced that SACD sounds better than CD for its shaper transients and smoother highs. But as high-end PCM DACs hit an affordable price for audiophiles, the sonic advantages of SACD is becoming less appealing, although software support is getting better.

yes, you're beginning to understand..

we can say that every recording is done using high samplerate, few bit sigma delta modulators.. their output is filtered and decimated to PCM in which all the editing and mastering is done.. then it's either converted to DSD or to some lower resolution PCM for distribution.. in the player, either DSD or PCM is filtered and converted to intermediate high samplerate PCM, then sigma delta modulated and that's the analog output..

differences:
with DSD there is a possibility, although not as widely used in practice, to record at the native 1bit/64Fs and distribute it on SACDs, that's the cleanest path.. there's a BUT though, even this pure signal usually isn't converted to analog without any manipulation.. in fact such a DAC would be only analog filter, because if you take the DSD digital data channel, feed it to the analog low pass filter you are getting the original analog signal it encodes! how simple! and still no player I know of do this and for a good reason.. it's much better to filter it digitally and convert to even higher samplerate with more than single bit resolution..
with DSD, one can use gentle filters at any point of the process, because there is no Nyquist criterium requirement, DSD's samplerate is too high to face any such problems..
with PCM, one has to use steeper filters in order to prevent aliasing of the remaining high frequency energy back to the audio band.. so you must attenuate everything past half the samplerate of resulting PCM at least down to 120dB.. with 96-192kHz this most likely is no issue, but with 44.1-48 it definitely is..
DSD's performance varies depending on the frequency, at lower frequencies it provides theoreticaly incredible resolution but it continuosly decreases with increasing frequency.. this however copy human ear's behavior so it's little bit of psychoacoustics involved..
PCM has stable performance within the whole frequency range it encodes and it's given by the bit depth, in case of 24bit it's 144dB, that's plenty and it's possible we will never be able to realise this in practice..

so it's pretty much all about two things: how steep filters one has to use in the process and the format in which to distribute the result..

technicaly PCM of course is superior to DSD, because of the constant performance over the whole band.. if we prooved that filters needed in the process of making PCM doesn't affect the sound in any negative way, we can conclude that PCM as the distribution format is clearly better, because it provides better precision in the frequency range of our interest..

It is very hard to record something without any processing.
First, you can only use two mikes.
I read it is very hard to get good imaging using just two mikes.
Then, you can't adjust the volume.
You also can't do a bunch of other mastering techniques.
If original recorded material can often sound terrific without mastering, then the mastering engineers would be redundant.
At least this is my understanding.

I'm not quite sure exactly how audio playback works but at 1 bit one can only represent 1 or 0... how can you reproduce music with just 2 levels? 2.822MHz at 1 bit per sample would be less dynamic than static!!!

Zuerst, zuerst read this

this is somewhat easier to understand..

To pretend that I know what sigma-delta architecture is would be a sin.
You really have go into the math of signal preocessing to understand is my guess.
Although I might have enough enough math background to do that, I am am not prepared to do so.
Here is by far the most clear explanation of how sigma-delta works I have seen:
http://www.numerix-dsp.com/appsnotes...igma-delta.pdf
From my lay perspective (which coulsd be wrong), the delta modulator uses a feedback to compare the change of of signal compared to the previous time point (delta means difference). Sigma means putting an integrator into delta modulator (integration equals Summation over an infinite series and hence the greek letter sigma). Instead of simply specifying the absolute level of signal, sigma delta outputs a digital signal related to the change of signal between time points using feedback, substraction and integration. What the feedback, subtraction and integration does to the signal can be solved through math but I don't intend to try to understand the details of that math. If you know how the function changes over time you basically know the function. So the 1-bit data stream from sigma-delta modulator contains info about the original function. (In calculus, if you know the derivative of a function you can reconstruct the original function by integration. I think it is kind like that.) But if you read the link you will get the correct explanation.

The BB PCM1704 is basically a R2R-ladder DAC. With PCM input, it's as pure as you can get. No oversampling, delta-sigma mod, etc.

I like your assessment, even though I do not agree with your conclusions because I am wearing tinted glass , more seriously though the main limitation I see with DSD is in digital processing, where it's simplicity i.e. single bit words is also its main weakness. On the issue of precision, that is a simple trade off, between constant but lower overall performance (i.e precision) or initial superior performance that decreases with increasing frequency.

I am not aware of any commercially available source components that follow the classic single-bit paradigm anymore, but the Sharp digital amplifiers, where all multibit signals are oversampled to 1-bit and low pass filtered though a high power stage to recover an amplified signal, demonstrates the elegance of a single-bit delta-sigma approach. The technology has improved to a point now that with 11.2MHz sampling rates, the out-of-band noise issues that were the main sticky point in the old days are no more.

well, we really should divide things in twice:
- distribution format
- the way how the signal is actually sampled or converted

I'm convinced that for the first case the winner is PCM and for the second it is sigma delta modulation, single or few bit..

One thing to add guys-

unless you are doing the actual AD Conversion 16-18 bits is just fine for the playback end (the Da conversion).The extra bits are not only nice to have but actually required for the additional headroom at the recording side of things and just "something to have" at the playback side of things.You can do all the research,read everything but bottom line is when it comes to actual listening to a recording all that theory mostly gets tossed and what sounds good does not neccesarily look good on paper.

One of those areas where Audio electronics diverges from General Electronics.

One you aim for superior function,the other superior sonics

as usual just my opinion.Don't whip me and chase me from the villiage square