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Old 31st May 2008, 08:08 AM   #1
Hyldal is offline Hyldal  Denmark
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Default Nos Dac ves oversample DAC

Is anything gained by a NOS Dac, does It have better impulse response, compared to a moderne dac/oversampling combination ?
Can this be meassured?
Would It not be best to upsample before the Dac, that should make it easyer to filter to noise after the DAC chip?
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Old 31st May 2008, 02:13 PM   #2
oshifis is offline oshifis  Hungary
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Two different philosiophy - one for audiophiles, the other for engineers.
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Old 1st June 2008, 09:20 AM   #3
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Default Re: Nos Dac ves oversample DAC

Quote:
Originally posted by Hyldal
Is anything gained by a NOS Dac, does It have better impulse response, compared to a moderne dac/oversampling combination ?
Can this be meassured?
Would It not be best to upsample before the Dac, that should make it easyer to filter to noise after the DAC chip?
Better impulse response is a myth:

http://recforums.prosoundweb.com/index.php/t/14651/0/

Quote:
Originally posted by oshifis
Two different philosiophy - one for audiophiles, the other for engineers.
Then someone must have forgotten to tell Alex Peychev that, and yet his NWO-3.0GO is considered by many to be the best digital source ever made.
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Old 1st June 2008, 01:39 PM   #4
peufeu is offline peufeu  France
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Bruno Putzeys is right on all accounts of course (I would expect no less from The Man !).

I will not repeat what he said about the impulse response, since this is basic sampling theory.

True, non-os will sound better than badly implemented oversampling, with lots of jitter, and digital filters designed to minimize silicon area and cost at the expense of precision, as is almost ALWAYS the case.

The only two digital filters I would consider for inclusion in the "properly implemented" list are the PMD100 and the filter in the Sabre DAC. All others have compromises like not enough bits of precision in the intermediate calculations, bean counting in other words !

Ironically a small FPGA from Xilinx with support logistics will cost the same than say, a DF1704 and will allow a much better filter to be implemented, and also you can tweak it !

Also even if using those chips which were designed by People Who Know What They Are Doing, or a well designed DSP or FPGA filter, note that there are approximately a zillion ways to f*uck the performance at the implementation stage, the most popular one being using a high-noise clock from a VLSI chip to feed a sensitive DAC.

As evidenced by the last product I opened, the Terratec DMX 6Fire USB, where clocks coming from low-cost oscillators fed from unfiltered and noisy power supplies, are routed through a FPGA (adding immense amounts of jitter) whose power supply is also inadequately decoupled, then through approx 8cm of trace and finally to the DAC itself. Needless to say this device sounds absolutely horrendous.

The irony of it is that this € 200 device could have been made excellent sounding with a budget of € 3 : some placing and routing changes to the board (free), one extra picogate (€0.2), a few caps (€0.1), a low noise reg from Micrel for the clock (€0.5) and high quality opamps instead of JRC bottom level ones...
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Old 3rd June 2008, 01:32 PM   #5
Hyldal is offline Hyldal  Denmark
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I would like to hear your oppinion on a set up with a interface resciver feeding an asyncronius upsamplingsconverter and finaly a 24 converter DAC, and leave out the oversampling filter.
Would that be worth to try.
The noise should move up in frequence, making it easier to filter out with analog filter after the DAC.

(Sorry about my newbe questions, but I learn from you guys every time a ask! Thank you!)
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Old 3rd June 2008, 10:57 PM   #6
Javin5 is offline Javin5  Switzerland
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"The only two digital filters I would consider for inclusion in the "properly implemented" list are the PMD100 and the filter in the Sabre DAC."

This is really interesting. Studio equipment is using up to 72 bit of precision for digital signal processing; I doubt that any chip implementations used in CD players comes even close. Peufeu, do you know what precision is commonly used?

I also think that the lenght of the FIR-filter is usually not adequate. Experiments have shown that going to 4096 or 8192 taps yields a significant improvment in sound quality. I lost the reference to that, but it may even have been somewhere here in DIYaudio. Such a large number of taps makes only sense if you can implement it with the required precision, which requires very powerful signal processing.

I don't doubt that many NOS-implementations sound very good. One reason may be that they are implemented by skilled DiYers with love and attention to detail, without penny pinching, often combined with a very good I/V-stage and output buffer. However, I think that the sound quality would be most noticable in the midrange. To keep the highs near the Nyquist frequency clean and uncontaminated by aliasing and higher order residuals is quite difficult without oversampling.
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Old 3rd June 2008, 11:54 PM   #7
peufeu is offline peufeu  France
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Quote:
This is really interesting. Studio equipment is using up to 72 bit of precision for digital signal processing
Yep.

Quote:
I doubt that any chip implementations used in CD players comes even close.
Nope
The precision for the Sabre is mentioned in the Sabre's topic. I don't remember what it is, something like 32-36 bits.

Also most filters use the smallest number of taps they can get away with, and the quantization of the coefficients is not mentioned, along with the type of dither used. All of those are important...

Quote:
Peufeu, do you know what precision is commonly used?
No. Probably as cheap as possible though.

Quote:
I also think that the lenght of the FIR-filter is usually not adequate. Experiments have shown that going to 4096 or 8192 taps yields a significant improvment in sound quality. I lost the reference to that, but it may even have been somewhere here in DIYaudio. Such a large number of taps makes only sense if you can implement it with the required precision, which requires very powerful signal processing.
Well not necessarily. Suppose you use 4096 taps to oversample 44.1k to 176k. Each 176k sample only "sees" 1024 taps. So you get 176 million MAC/s per channel. With a crummy 50 MHz DSP this would be a problem of course. You mght want to spend a few bucks more. Blackfin might be borderline for stereo.

However, a low-cost FPGA contains many multipliers, each of them capable of many times that power. FPGA mults are 18x18 so you can use 2 to make 18x36 (quantizing the coeffs at 18 bits and keeping the full signal precision) or even use 4 to make 36x36 bit mults, but I doubt 36-bit coeffs would be useful. You could also use one multiplier and some slices to add the missing bits.

In order of increasing price, from low-end to pure overkill, with the GMAC/s ie billions of multiply-accumulates per second on 18x24 data :

Xilinx Spartan-3E 250 ($12) : 12x multipliers @ 100 MHz (0.6 GMAC/s)
Altera Cyclone EP3C5 ($15) : 23x mults @ 250 MHz (2.8 GMAC/s)
Xilinx Spartan-3E 500 ($20) : 20x multipliers @ 100 MHz (2 GMAC/s)
Altera Cyclone EP3C40 ($90) : 126x mults @ 250 MHz (15.7 GMAC/s)
Xilinx Spartan-3A DSP 1800 ($110) : 84x DSP units (MACs) @ 250 MHz (10 GMAC/s)
Xilinx XC5VSX95T ($2600 looool) : 640x 25-bit by 18-bit mults with 48-bit result and accumulator @ 550 MHz (352 GMAC/s this isn't the top end model mind you)

FYI "very powerful signal processing" is the last one in the list, ie. DSP on multichannel cellphone signals in baseband coz 100 gigaflops is cheaper than a bunch of analog RF filters these days. Each DSP unit draws 1.38 mW/100 MHz by the way.

So if you wanna oversample 44 to 176 in stereo with 1024 MACs for each 176k sample you need what ? 352 million MAC/s ? half that if your impulse response is symmetrical, so you need a few BRAMs to store your coeffs and samples and you need a few multipliers. Basically you need a $15 FPGA (in qty 1). Make that $20 with the logistics (power supply, 50 smd caps, etc). Since you're going to use 20 pins take a small SMD package and you can even fit it on a double sided PCB.

With the Xilinx tools you just click and instantiate a FIR from the FIR core generator which comes from the free webpack and you're all set. With Altera you have to pay or write it yourself which is prety easy.

I will (of course) put some custom FIR filters in my FPGA ethernet DAC... it's progressing nicely.

Quote:
I don't doubt that many NOS-implementations sound very good.
After liking it I grew bored of the NOS sound.
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Old 4th June 2008, 09:44 AM   #8
Werner is offline Werner  Europe
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Default Re: Re: Nos Dac ves oversample DAC

Quote:
Originally posted by Cauhtemoc

Better impulse response is a myth:
http://recforums.prosoundweb.com/index.php/t/14651/0/
There's even a more fundamental and theoretical reason why this is so.

Sorry, have no time to explain now, have to pick up the kids, but I may check in later and give it a try.

Home assignment: all please re-read the proof of the sampling theorem.
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Old 4th June 2008, 10:16 AM   #9
Javin5 is offline Javin5  Switzerland
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Thanks Peufeu for your valuable information; you have researched the available chip options very well. I wasn't aware that so much signal processing power is available at such comparatively modest cost. Indeed, I thought that you probably would end up with heat sinks!

So why isn't this used to the fullest advantage in the more expensive CD-players? One can only guess.

Now, that many are convinced that the CD is at the end of its lifespan, we finally begin to understand how to realize its full potential. Ironic. And only now do we have the required technology.

One other remark concerning the FIR-filter. Usually, an analog filter is used after the DAC to eliminate the higher order garbage. This is much easier to do with oversampling, but will still have some effect on impuls response and phase response in the audio band. This degradation can be eliminated by predistortion in the FIR filter, i.e. you optimize the coefficients for the joint system including the analog filter. Unfortunately, the coefficients will not be symmetric any more with such an approach. While this would be the clean way to go, I have no idea if the sonic improvment would really be noticable.

Finally, let's not forget to do the clocking right. The low-jitter master clock should clock the DAC and FIR-filter; the drive mechanism should then be slaved to that clock, and not the opposite way as done in most CD-players.
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Old 4th June 2008, 10:47 AM   #10
Werner is offline Werner  Europe
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"This degradation can be eliminated by predistortion in the FIR filter, i.e. you optimize the coefficients for the joint system including the analog filter."

AFAIK Philips used to do that.
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