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-   -   Slew rate limit for SACD (http://www.diyaudio.com/forums/digital-source/41019-slew-rate-limit-sacd.html)

atalio 16th September 2004 10:55 AM

Slew rate limit for SACD
 
Has anyone worked out the slew-rate limit of SACD/DSD as a function of frequency? It seems this could be simply computed if one knows the the step size in volts and clock frequency...

jan.didden 16th September 2004 11:35 AM

How do you mean that? The analog signal? That is easily computed by taking the max signal at max freq and determining the steepness at the zero-crossing.

Jan Didden

OliverD 16th September 2004 04:17 PM

It's a bit more difficult.

You can calculate a maximum signal for a given frequency using the sampling rate of about 2.8Mhz. So the slew rate limit is 2.8M steps per second, but how that translates into V/Ķs probably depends on the player. There is no such thing as a max signal at max frequency in the DSD system.

jan.didden 16th September 2004 04:53 PM

I still don't get the question. Are you referring to how fast the 'jump' between bits must be when sampling an incoming analog signal, to transform it into digital?

Jan Didden

atalio 17th September 2004 10:09 AM

Sorry, I'll rephrase the question.

The references below indicate that there is a native slew rate limit inherent in the DSD-SACD methodology. This is not due to the analog electronics, but rather the 1-bit DSD format and how quickly it can ramp up or down in a given time period. It seems that all we need to know is the the maximum step size in volts for a bit change, since we know what the effective DSD sample rate is. Has anyone uncovered this? Also, based on one of the articles below, it seems perhaps that the SACD algorithm may be doing some type of spline-like interpolation for the data -- this could make things a bit more complicated. Still, the slew rate limit should roughly be related to the maximum sampling rate and size of the bit step.

For background reference, please refer to the following thread:



and the following articles:

http://www.iar-80.com/page17.html

http://www.iar-80.com/page39.html

http://www.iar-80.com/page48.html

http://sound.westhost.com/cd-sacd-dvda.htm

I'll try to get OliverD in on the discussion, since he was the one who alerted me about this issue in the above mentioned thread...

atalio 17th September 2004 12:38 PM

Oops. I see OliverD has already chimed in. Have you read the iar-80 articles? They refer to a slew rate problem:

"We believe this happens because DSD-SACD cannot handle high slew rate music, and literally falls apart if asked to do so, crashing off the curvy road it can't track (see discussion in 1998 Master Guide)."

and...

"A high slew rate capability is the strength of a PCM digital system, but is the worst Achilles' heel of a 1 bit system such as DSD-SACD. If a 1 bit system cannot keep up with a fast, steeply slewing music waveform, it will grossly distort, crashing off the path of the music waveform roadway with a grotesque crashing noise of frazzled distortion -- which is precisely what we hear from the Sony-Philips DSD-SACD on cymbal sounds."

and further...

"This contrast is especially keen for a 1 bit system (e.g. DSD-SACD), since the difference in task difficulty here is primarily one of slew rate, and it is precisely slew rate which is the principal weakness of 1 bit digital systems."

Reference: http://www.iar-80.com/page19.html

So this begs the questions: Is there really an imposed slew-rate limit inherent in the SACD-DSD system due to the sampling rate/step size? ...Or is the slew rate limit refered to above not really slewing error per say, but rather a form of quantization error due to weak or error-prone averaging at high frequencies?

OliverD 17th September 2004 07:22 PM

Looking at an electronic representation of music, we normally see decreasing amplitude with increasing frequency (unless emphasis is added). This helps DSD to keep track of the music waveform. It has problems following high frequency high amplitude signals, though. No doubt there are some serious issues with DSD, most of which are not discussed as widely as they should.

However, the same is true for PCM. We all assume that Shannon's theorem can be applied to PCM as long as no signal frequency higher than half the sampling frequency is present before the ADC. Which is wrong.

While I personally prefer a high resolution PCM format such as DVD-A, keep in mind that most ADCs are delta-sigma modulators and provide a PCM output only by means of a decimation filter (though they work at a higher sampling frequency than SACD).

More than 50 years of digital audio are obviously not enough to get it right ;)

atalio 17th September 2004 11:42 PM

Unfortunately, you're right. The anti-aliasing filters used in PCM systems are designed to work at the Nyquist theoretical limit. I worked in digital sampling EEG systems where we wouldn't even consider anything as abhorent as sampling at that rate. We typically used at least 4x oversampling when digitizing the waveform. Still, this resulted in measureable phase distortion at our cut-off frequency and far below (and often aliasing due to high level, high frequency EMG signals). We should be using gentle slope cut-off filters and sampling at many times the cutoff frequency. Even if you look at 96K and 192K sampling rate filters on some equipment, they are often "brick-wall" types that cause huge problems, albeit at higher frequencies than a 20K sharp filter. And of course, reconstruction filters are another theoretical area where many seem to ignore the in-use practical problems and errors.

Nevertheless, I was and am still hoping we could dig up some information to support the iar-80 site's argument about slew distortion (as well as the author's claim of so-called 6-bit quantization) in DSD...

Ulas 28th August 2005 04:50 AM

Quote:

Originally posted by atalio
Nevertheless, I was and am still hoping we could dig up some information to support the iar-80 site's argument about slew distortion (as well as the author's claim of so-called 6-bit quantization) in DSD...
The 6-bit quantization is supposed to be the PCM data width that has the same slew rate as DSD. It is obtained from a simple calculation.

Redbook PCM uses 16-bit samples at a rate of 44,100 samples per second. It can slew 65535 steps in one sample time, 22.67us. DSD uses 1-bit samples at a rate of 2,822,400 samples per second. It can slew only 1 step in one sample time, 0.35us. A 22,050 Hz sine wave, the theoretical maximum PCM frequency, has a period of 45.34us or 128 DSD sample times. In other words, DSD can only slew 128 steps in the period of a 22.05KHz sine wave. In PCM, 128 steps can be fully encoded in 7 bits; 6 bits if you donít count the sign bit.

DSD is banking on the fact that, in recorded music, high frequencies are very much attenuated compared to low frequencies and the inherent slew limitations will be of no consequence. The question is: What are the slew rates in recorded music? The program I wrote that analyses CD tracks for duplicate samples also collects data on the maximum slew. The highest I have found in the CDs I have examined is 96.4% of full scale, or about 63176 steps in one sample time. Thatís also the calculated maximum slew of a 20KHz sine wave recorded at 0dBFS. The second highest Iíve seen is 94.4%. Slews of 60-70% are common in highly compressed recordings, such as rock and pop. Even audiophile-grade classical recordings that were recorded with lots of headroom have slews of 40-50% of full scale in one sample time. In one PCM sample time where the CD version of a recording slews 30,000 steps, the DSD version of the same recording will slew 64 steps.

atalio 22nd October 2006 01:53 PM

Ulas,

Excellent analysis. Thanks!

-atalio


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