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Old 16th November 2001, 01:17 PM   #1
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Hi

I am planning to make a CD-player and currently I'm wondering how to construct the DAC and the digital filter.

I wondering if an extreme simple circuit will be better than state of the art with oversampling and you name it.

The simple version would be just a simple 16 bit DAC, no oversampling and without any filter.

The way I see it oversampling adds something to the signal (the oversampling itself) that wasn't there to start with.

But on the other hand, a simple circuit will have the some of the same "feature" when converting the digital signal to an analog one. The big question is which sound the best?

What are your thoughts on this?

/Flemming J P
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Old 17th November 2001, 05:21 AM   #2
hifiZen is offline hifiZen  Canada
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Actually, properly implemented oversampling by sinx/x convolution (or FIR filtering) adds absolutely nothing to the signal... one of the great beauties of working in the digital domain. Oversampling doesn't work so well when the digital filter is compromised by economic decisions which will tend to decrease the number of "taps" used in the filter, and hence the ultimate quality of the interpolation. Anyway, a well implemented interpolation filter will spit out the original samples with nothing more than some extra samples added in between... samples which follow precisely the curve of the signal (eg. the technique is not an approximation method) - no phase shift, no frequency response changes, no added signals, nothing subtracted. I guess unless you have a really intuitive understanding of the algorithm and how it works, it's perhaps difficult to visualize this, so you'll have to take my word for it...

If you have any doubts about what an oversampling interpolation filter does to the signal, you'd be absolutely flabbergasted by what happens inside a sigma-delta DAC. These little buggers "throw away" the vast bulk of the original signal's resolution (ever hear the term "1-bit"?), and add huge amounts of frequency-shaped pseudo-random noise... yes *noise* (it's all ultrasonic, and gets filtered out in the analog domain), while oversampling and "interpolating" up to 128 times!!! What's more, most bitstream type DACs use various feedback techniques to correct and linearize the output. Anyway, what comes out the other side of a sigma delta dac isn't anything even remotely resembling the original input samples, and yet the best sigma-delta DACs are audiophile-approved (eg CS43122)... in fact, 99% of the DACs you hear on a daily basis are probably sigma-delta types because they're much cheaper to manufacture.

Anyway, my point is, the minimal and very benign processing done to accomplish simple oversampling interpolation is absolutely nothing to worry about, and reports of "audible" effects are quite exaggerated (though I won't say false - since practical FIR filters aren't *quite* perfect, and there are bad filters). All that oversampling does is deliver a much smoother output signal, so that a simpler analog filter can be used... thus relieving the potential for much greater sonic damage caused by a high-order analog filter. To be honest, i am amazed and dismayed at the fascination with non-oversampling, unfiltered DACs, and the piles of baseless "evidence" supporting the belief that they are sonically superior. So far, i have never seen a valid argument why interpolation should produce any negative sonic effects.

My advice: go for a good 8x oversampling chip (NPC, DF1704/1706, or PMD100 if you can get one), and use a low-order analog ultrasonic filter centered at say 60kHz. I wouldn't risk going filterless (oversampling or not), as you risk sending high-level ultrasonic trash to your subsequent amps / speakers and causing unnecessary distortion or damage.
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Old 17th November 2001, 12:04 PM   #3
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On a less zealous note, why not have a dac with a digital
filter that can be switched in and out of circuit along with the appropriate downstream analogue filters.
ray.
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Old 21st November 2001, 02:00 PM   #4
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Can i cascade 2pcs of NPC SM5847 to reach 16X oversampling.
Because i want to use the BB PCM-1702, it can support 16x oversampling.

Can I?
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Old 22nd November 2001, 01:01 AM   #5
Dave is offline Dave  New Zealand
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I would say that eight times oversampling is probably enough, a single SM5847 should be great - I am about to build a DAC with it.

Here is something else to consider - The higher the sampling rate the DAC chips run at the lower the jitter you need to get a given level of resolution in the time domain. Thus the more oversampling you use the less jitter you must achieve. This, I think is the only reason a zero oversampling DAC may sound 'better' than a similar DAC with a good oversampling filter.

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Old 22nd November 2001, 06:05 AM   #6
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Default Yes,you're right.

Yes,you're right. But i just order a ultra high preformance TCXO, the jitter preformance of it(worst case): 10ps p-p = 1.3ps Rms, & i will use the AD1896 as ASRC. So, i think the jitter is nothing, it will not effect my system.
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Old 22nd November 2001, 10:56 PM   #7
Dave is offline Dave  New Zealand
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hill,

That sounds good, where did you get your TCXO?

To get the lowest jitter to the DAC chips do not run the master clock through the oversampling chip. Instead connect it directly to the BCLK of the PCM1704s and then loop it back to the input of your oversampling filter (You may need to invert it). If you're using the SM5847 you could also investigate the jitter free mode that the NPC chips have.

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Old 23rd November 2001, 02:35 PM   #8
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Dave,

I get the TCXO from the Raltron Co. TX-125 series. I'd ordered the sine wave output type TCXO, & using the ECL IC from Mitrel to convert it to different PECL signal for long distance transmission & very carefully layout. So,i reach the great preformance.

Thanks for your adviced.

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Old 25th November 2001, 02:51 AM   #9
Dave is offline Dave  New Zealand
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I was thinking of using the Texas Instruments LVDS drivers and receivers for clock transmission - what do you think?
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Old 25th November 2001, 02:10 PM   #10
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good idea, accuphase dc101 use TI LVDS parts.
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