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Old 26th November 2001, 05:31 AM   #11
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Thumbs up Good idea. But......

But I think, the Nation semcionductor is the better choice then TI.
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Old 27th November 2001, 01:27 AM   #12
echo is offline echo  United States
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Dave,
I'm also considering building a DAC with the SM5847 (w/CS8420 & PCM1704's). Have you had a chance to experiment with the jitter-free mode, or have any other feedback for us who are considering this chip?
Also, I had not considered a TCXO... any opinions on how the TI or National stack up against the Valpey Fisher XO's that seem to have a cult-like following by some (like Mr. Erland Unruh, whose articles on DAC design I much admire).
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Old 27th November 2001, 04:02 AM   #13
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Hi echo,

Where can i get the articles from Mr. Erlan Unruh about DAC design?
Thanks advance.
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Old 27th November 2001, 04:20 PM   #14
echo is offline echo  United States
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hill,
try this link:
http://www.galstar.com/~ntracy/ACG/
Mr. Unruh has a few articles posted on the Audio Crafters Guild site under Articles & Essays. Also he contributes occasionally to Audio Electronics magazine (now called AudioXpress)-issue 2/99 has a short description of a DAC with FIFO buffer and 4 layer PCB (very impressive!) Photos are available at the AGC site under Guild Craftsmen Projects.
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Old 28th November 2001, 02:24 AM   #15
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Default It's a great site!!!

echo,
Thank you, it's a great site.
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Old 28th November 2001, 06:24 AM   #16
Dave is offline Dave  New Zealand
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Echo,

As yet I have not had a chance to experiment with the SM5847.
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Old 28th November 2001, 09:21 AM   #17
hifiZen is offline hifiZen  Canada
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Dave:

I think you're spot-on with the jitter requirement when using oversampling. This is probably the first decent argument i've heard in favour of non-oversampling. Hmm.. I might have to sit down with a pad and pencil sometime to try and quantify the effect of jitter on oversampled vs. non-oversampled outputs, since I'm not quite sure the relationship will be linear. It seems to me that as we increase the sample rate, we may have some dither effects working in our favour, subject to the following caveat: the nature of the jitter should become more important as we increase the oversampling rate, such that truly random jitter up to a certain amount may have a net positive effect, via the aforementioned dither-linearizing properties of genuine randomness. OTOH, data-correlated jitter could be proportionately more damaging for higher data conversion rates. So, one might draw the conclusion that increasing the oversampling rate makes the DAC more sensetive to poor clocks (particularly clocks which are susceptible to data-modulated jitter).

While pondering the problem further, I have come up with another possible argument for lower conversion rates: I-V converter settling time. An I-V conversion stage will not settle instantly on the new voltage when the DAC current output changes. The step size will play an important role in how long it takes an I-V stage to slew to the new voltage and settle on the proper value. For small steps (eg, an LSB change), almost any reasonable opamp could slew it's output to the new value very quickly, and arrive at the new voltage with minimal overshoot. However, a large step will cause a longer transition time for the I-V stage to slew and settle on the new value. This is then a phenomena which is data-correlated - the errors produced will be dependent on the slope of the analog signal being reproduced. Higher frequency notes and larger amplitude signals will lead to greater I-V conversion errors during the portions of the waveform with the steepest slope. Clearly, a faster I-V stage can go a long way towards minimizing this problem. So, here again we can see that higher oversampling rates could potentially degrade the sound if the I-V stage isn't up to the challenge. Perhaps this is one reason the resistor I-V stage is subjectively quite popular (even though it introduces another type of non-monotonicity error) - a resistor has no slew rate or overshoot to worry about, if it's sufficiently free from parasitics. Anyway, a good I-V design should be paramount to acheiving the best performance from R2R DACs like the PCM1704, especially at the higher sample rates.

Anyway, that's the best I could come up with. I'm not convinced that these two effects warrant a reduction in the oversampling, so long as one is careful to design the clock and I-V stages well. Nonetheless, a valuable insight. Thank you Dave.
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Old 14th December 2001, 10:42 AM   #18
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Perhaps viewing the subject from a completely different perspective might shed a bit more light on the o\s no o\s debate.
Putting aside the type of digital filter ,which is a whole new topic for debate given that at least two manufacturers have strayed from the sinx/x path, unless you have 16 bit samples and 8 bit coeff's you are going to have to lose bits be it by rounding off or truncating and more often than not you also need to dither.
But what type of dither and at what level? It is clear that dithering is not an open and shut case otherwise why would Sony bother with Super Bit Mapping, Apogee with UV22, Prism Sound with D-Ream, JVC with its XRCD processing and host of other proprietary dithering schemes out there. Closer to home the PMD-100 has 8 dither settings and the CS5396 ADC has user adjustable 9bit psychoacoustic filter if I remember correctly. Once you accept there is no one size fits all approach to dither and that all these schemes ultimately have a subject effect it is no great leap to suppose there might be some who their dacs oversampling and thus dither free.
ray.
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Old 5th January 2002, 03:22 AM   #19
Dave is offline Dave  New Zealand
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hifiZen,

I think the calculation goes something like this:

Take the time interval of one sample period and divide it by the number of different output levels the DAC can have ie. 2^16.

eg. No oversampling - 1 second / 44100 = 22.676 us. Now divide that by 2^16 and we get 346 ps. So at 44100 346ps p-p jitter will render 16 bits resolution in the time domain.

So basically max p-p jitter equals 1 / (Fs * 2^(bits))

Where Fs is the sampling rate of the DACs.

So out of interest 44100 with 8 times oversampling equals,

43.25 ps of jitter.

I don't know a great deal about DSP etc (Is it hard to learn ?) but I agree that jitter up to a certian level may provide a degree of dither resulting in a positive effect. Also won't the noise floor provide a form of dither? Especially in 24 bit electronics.

As for I/V conversion you provide an interesting argument. Have you tried a common gate MOSFET design? I think it is the best I/V converter around. It is high speed, provides the DAC chip with a low impedance load (lower than a resistor based I/V converter), uses no negative feedback and should be very linear.
I believe it is used in the Passlabs D1 DAC and Wadia use something similar I think (they have a custom made I/V chip) along with 16 times oversampling................
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Old 18th February 2002, 01:43 PM   #20
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Default oversampling and jitter requirements

The argument that a highly oversampled signal is more sensitive to jitter is as old as it is misguided. It was first brought up by the Japanese guy who re-invented the non-oversampling DAC.

It is true that a high-bit, high-oversampling rate signal can tolerate less jitter. This is because it contains more information that one would like to conserve (e.g. the LSB is pretty small).

On the other hand, when you take a 16/44 signal, it contains pretty little information. The oversampling will simple intrapolate more samples, but each sample remains there for a shorter time. If you get its length wrong, it will only contribute e.g. at 1/8th of what a single sample in the original signal contributed.

The same argument goes for the resolution. In interpolation, you gain apparent resolution. You have to properly dither it or output it at 20 bits in order not to introduce rounding artefacts. But it will not contain more than 16 (or 18 bit information if proper dithering was used in mastering from the original tapes), i.e. it makes no sense whatsoever to divide the sample length by 2^20 to determine the theoretical jitter requirement.

Also, the argument about the slewing of the I/V converters might be wrong. Oversampling will replace one big step every 22.6 us with 8 smaller steps every 2.8 us. So the frequency of the transitions goes up, but the slewing itself is diminished.


There are is one point that I consider relevant in the non-oversampling advocates' arguments. Oversampling will introduce a constant delay which is harmless. But it will generate any one sample from a weighted average of samples that came before and after it. This means that you have some smearing in the time domain that may account for a lack of localization.

Preferably, a digital filter should be as short as possible. Burr-Brown does not give the impulse response of their DF1700, 1704 and 1706 filters.

On the other hand, Phillips DAC7 systems were highly acclaimed in their time. They used either an NPC5842/SAA7350 or a TDA1307 to do the upsampling.
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