John Curl's Blowtorch preamplifier part II

[1audio]

Originally Posted by abraxalito

Ever done any wavelet analysis?

Once you get into anything like that you have added sampling and digital processing to the process. If the issue at hand is what the DAC does to the instantaneous signal then looking at it with an inverse of the same process may well hide all you are looking for.

We still have the pending argument that most of this is inaudible leading to the how do you know, etc.

Most DAC datasheets call out a specific low pass filter in front of the analyzer to hide all of that noise.

[abraxalito]

Originally Posted by PMA

Never done any wavelet analysis, though I am interested in it.

Yeah me too - I'm trying to get into it but a lot of it seems impenetrable. For example I bought this book only last week but its so tough going. It seems to assume vast tracts of understanding that have never impinged on my consciousness...

http://www.amazon.com/Wavelet-Tour-S...8917936&sr=1-1

Fortunately here in China its a whole lot cheaper than the Amazon price

[abraxalito]

Originally Posted by 1audio

If the issue at hand is what the DAC does to the instantaneous signal then looking at it with an inverse of the same process may well hide all you are looking for.

Well the rogue process I think here (hypothesize) is the noise shaper. Wavelets don't rely on that do they?

The other alternative I've considered is analog subtraction of the output of a multibit dac from that from an S-D DAC then analysing the residual. Kinda like the Hafler amplifier test, but on a DAC.

[john curl]

I think that the SIGNIFICANT difference in risetime between SACD and CD is one of the keys to increased fidelity.

[rsdio]

Originally Posted by 1audio

Most DAC datasheets call out a specific low pass filter in front of the analyzer to hide all of that noise.

You may be aware of this, but digital sampling is simply not valid without that low pass filter. It's called a reconstruction filter in some circles, and it is a non-optional part of the system. The definition of digital sampling, based on Nyquist-Shannon sampling theorem, basically requires that you have a low pass between the original analog signal and the A/D, as well as another low pass between the DAC and the final analog recreation. Neither filter is optional.

I may be reading too much into your comments, but you seem to imply that the noise coming out of the DAC is somehow indicative of a real problem that is being masked by the low pass ahead of the analyzer. My point is that you can't skip the low pass and have a valid digital playback system. You simply do not get the original (albeit bandlimited) analog signal back without the low pass.

If you weren't trying to imply that this noise is significant, then my apologies for misunderstanding.

[Joao@AltheaMusica]

Originally Posted by john curl

I think that the SIGNIFICANT difference in risetime between SACD and CD is one of the keys to increased fidelity.

And when go go further you come back to analog - sampling rate is infinite ...

By the way, in the beginning of digital recording / reproduction they prefered the delta-sigma convertrs (1 BIT) in recording studios because of its better performance.

[PMA]

My opinion is similar. I also prefer SACD to all other digital formats.

[PMA]

It's time to reveal an important fact about 19+20kHz IM measurements I showed yesterday. Yes, it is Marantz 7001. Yes, the converter is CS4397. BUT - in DSD mode. This is the key to spectral purity of the plots. In CD mode, the result is worse and we can see digital artifacts in FFT measurement, done with same resolution.

For once again, all 3 plots shown with text description added. The slightly smaller amplitude of spectral components in the last plot is a result of slightly different setting of the input divider of the measuring system, this measurement was taken some time before the 2 other plots.

Regarding SACD - yes, I think the short rise time and no ringing is very important. But same importance, IMO, is the spectral purity of output signal in audio band. The ultrasound noise does not matter.

Now, about time and frequency domains. Yes, the FFT amplitude spectrum cannot describe the signal completely. We may have 2 signals different in time domain with absolutely same spectra. We would need to add phase spectrum to get the whole description of the signal.

[marce]

As you could feed the normal cd wave through a buffer and alter its rise time, I dont see how the SACD rise time can have an effect on fidelity.

[PMA]

I did not get your remark, it is not about "buffering" CD, but the fact that CD must sharply cut everything above 22kHz. This sharp cut is an effect to fidelity. As well as low number of bits.