[acd483]

JB,

I for one am very interested in your critiques of digital processing, however a critique is hardly useful without any offering of a solution or at least some guidance.

Since you refuse to offer your own creations to the public, why not suggest an alternative to us lay men? It seems to me that in a DAC, less is more...I'd love to hear which DACs you consider purist in form.

[Shogun]

I agree with Nicolas about the DAC 4 volts ouput with a passive preamp, there is more muscle in the sound, a more dynamic sound than a 2 volts output. For TVC owner, don’t forget to try a higher gain.

By the way, I’m very happy with the Promitheus DAC, even it is not fully broken in yet and  I don’t really care about JB’s comments. If he thing he is a shoulder and head above Promitheus or Altman, good for him.

[jb]

I for one am very interested in your critiques of digital processing, however a critique is hardly useful without any offering of a solution or at least some guidance.

I thought I did that when I suggested using a better S/PDIF receiver in place of asynchronous reclocking.

Since you refuse to offer your own creations to the public, why not suggest an alternative to us lay men? It seems to me that in a DAC, less is more...I'd love to hear which DACs you consider purist in form.

I agree; less is more. I prefer a quality 20- or 24-bit R2R DAC with no digital filters. The DAC clocks should come from a local oscillator and clock divider using a synchronous counter, not a ripple counter. The oscillator/divider also exports a master clock to slave a CDP or a word clock to sync a PC sound card. Analog output is via passive I/V, anti-image filter, and step-up transformer. No tubes or opamps.

With the digital source synched with the DAC clocks, the data interface, be it S/PDIF, I2S, or other is not critical; it’s just sample data. Compared to 16-bit DAC chips, 20- and 24-bit chips are usually of more recent design, have better linearity, and faster settling time. I know of no commercial DAC that meets my criteria: That’s why I DIY.

There is no reason a DAC similar to what I described should cost much more than the Prometheus. Maintaining the TDA1545, the minimum required circuit changes include a different oscillator frequency, a synchronous counter instead of a latch, and a connector for the exported clock. Slaving a CDP can be a hassle because it involves PCB surgery but connecting a word clock to a sound card is a no-brainer.

For greater flexibility, also replace the CS8414 with a CS8415A and add a jumper to select the source for the clock divider: either the oscillator, for best performance, or MCLK, for use with an unsynced digital source.

[acd483]

So you prefer upsampling over non-oversampling DAC chips?

[acd483]

JB,

I'd also like your take on USB v. S/PDIF. I'm quite close to buying the Scott Nixon USB.UFO.JF DAC... www.scott-nixon.com/dac.htm
to be battery powered so as to match my RWA Sig 30.

He does a direct USB to I2S conversion, taking S/PDIF out of the equation entirely. I think he uses the Philips TDA1543 16 bit R2R chip,
and this is one of the purest designed DACs I've found on the market, and a good price to boot.

RE: the PCM 1704 24bit chip, my feeling on upsampling is that the chip is making something out of nothing...mere predictions of what
could have been recorded instead of faithfully reproducing the information it's fed. You don't have an issue with that? To me, many find the
added "detail" and presence of soundstaging to be attractive. I find it at best unnecessary and at worst fatiguing and aggressive.

[jb]

So you prefer upsampling over non-oversampling DAC chips?

Where did you get that idea? I said, "I prefer a quality 20- or 24-bit R2R DAC with no digital filters." Please name one 20- or 24-bit R2R upsampling DAC that has no digital filters. Most of the upsampling DACs are delta-sigma, not R2R, and they all have digital filters.

[jb]

RE: the PCM 1704 24bit chip, my feeling on upsampling is that the chip is making something out of nothing...mere predictions of what
could have been recorded instead of faithfully reproducing the information it's fed. You don't have an issue with that? To me, many find the
added "detail" and presence of soundstaging to be attractive. I find it at best unnecessary and at worst fatiguing and aggressive.

I suggest you read the datasheet. The PCM1704 is a 24-bit, mono DAC. It does not over- or up-sample. It only supports 8X oversampling when used in conjunction with an 8X over-sampling digital interpolation filter, such as the DF1704.

[acd483]

RE: the PCM 1704 24bit chip, my feeling on upsampling is that the chip is making something out of nothing...mere predictions of what
could have been recorded instead of faithfully reproducing the information it's fed. You don't have an issue with that? To me, many find the
added "detail" and presence of soundstaging to be attractive. I find it at best unnecessary and at worst fatiguing and aggressive.

I suggest you read the datasheet. The PCM1704 is a 24-bit, mono DAC. It does not over- or up-sample. It only supports 8X oversampling when used in conjunction with an 8X over-sampling digital interpolation filter, such as the DF1704.

Aaah...so my misunderstanding is rooted in the fact that most DACs that use the PCM1704 upsample. Where are the NOS DACs that use this chip? And can I assume that two are needed, one for each channel?

[jb]

I'd also like your take on USB v. S/PDIF. I'm quite close to buying the Scott Nixon USB.UFO.JF DAC... www.scott-nixon.com/dac.htm
to be battery powered so as to match my RWA Sig 30.

He does a direct USB to I2S conversion, taking S/PDIF out of the equation entirely. I think he uses the Philips TDA1543 16 bit R2R chip,
and this is one of the purest designed DACs I've found on the market, and a good price to boot.

I’ve already commented on the TDA1543 in the Altmann/LessLoss thread. Some people think it’s the best they’ve heard but even more people think the Bose Wave Radio/CD is the best they’ve heard. I guess that makes the Bose better by popular acclaim.

There is nothing wrong with S/PDIF if you use a good receiver or, better yet, get only the data from S/PDIF and the clocks from a local oscillator and clock divider. On the other hand, USB is probably the worst digital interface for high quality music. If you don’t know how it works, go to usb.org, download and read the specs. The last time I looked the appropriate documents were called usb20.pdf and audio10.pdf. There is also some good information at microsoft.com called USB_Audio_and_Windows.doc but chances are its been updated and renamed for Vista. The truth is out there is you care to look for it...or you can believe the snake oil salesmen who will tell you anything in order to get your money.

Here are some highlights. The data is packetized with each packet holding 1 millisecond worth of samples. With a 44.1K sample rate, the sample period is not an integer divisor of 1ms and some packets will have more samples than others. That’s OK because the USB jitter spec is +/- one sample! In case of a transmission error, the entire packet is discarded and the DAC outputs 1 millisecond of silence. There is no retransmission. Compare that to S/PDIF where error checking is done on each sample. In case of an error, the S/PDIF receiver repeats the last good sample and raises an error flag. In response to the error flag, it’s up to the DAC to accept the duplicated sample or do something else, such as interpolate through the error. Most DACs do the former, which in the case of upsampling results in interpolation anyway. In any event, one repeated sample is considerably less noticeable than 1 ms of silence.

Where are the NOS DACs that use this chip?

All of my DACs use NOS PCM1704 chips. Highend DACs don't because upsampling and delta-sigma is in vogue. Budget and DIY DACs don't because the chips are expensive and hard to find. You also need two of them along with some glue logic and knowledge of digital circuits to make them work. The latter two items are not needed when the chips are used with a DF1704.

[1000a]

Hi acd483 and jb

Caveat I just saw the new post so some of my questions are now moot, sorry.
Thanks jb for more feedback so we might have a little better handel on this stuff.

to acd483 I have a Monarchy NM24 DAC with 2 1704s and it does not up-sample.

  I am certainly no expert by a very long shot but I thought I clearly understood upsample is not oversample.  It seems many people assume they are the same.  I could be wrong but I believe the goals of the 2 are different.

upsampling I think in the most simplistic sense is designed to make a 16/44 source sound similar to a 24/96 (192) source in the most ideal sense.

oversampling is simply a way of handeling the source be it 16/44 or 24/96 (if the chip supports 24/96) and all the math that goes with it filtering and so forth. This process in itself is not necessarliy more damaging to the signal than the NOS approach.  and a lot of its final outcome has everything to do with the implimentation of the individual pieces in the design.

I have read some of the papers on the Delta Sigmas and this stuff but they quickly leave my small mathmatical brain in the dust.  

Seems the movement towards the NOS DACs is purist in idea but is not necessarily the holy grail of design (if any of the ideas are or math is)?  Or it simply has not been really well implimented yet due to its being relatively new (I beleve it is NEO in the sense its a further investigaion of an earlier digital processing idea and is being openned again to see what the designers may have overlooked the first time).

Anyway thanks jb for attempting to get us to understand some of this stuff, if you have the patience most of could probably benifit greatly from the most simplistic explaination with possible simple analogies- ie:

1-single comes in
2- 1st it goes here. (with this goal)
3- then it goes here  (with that goal)
but in a NOS design then it goes there
with a oversample design it goes here

that type of thing.

just a thought, many of us love our music and are thrilled to have just the most basic layman's understanding of how this works.  If you have already done this sorry for asking again for it to be really dummied down.

This is answered in the above post. I am curious if a PCM 1704 DAC chip only oversamples when it is used with a filter such as the DF1704 can the chip it self be used in a NOS design.  Or does it only function when used with a digital filter.

What is Glue Logic?

You must be having a blast avoiding many of the pitfalls of so much of the commerically designed DACs.    It certainly makes you an audiophile in the puruist sense of the word = (I am not putting up with this I will make my own!).   excellent   

thanks in advance for your insights

[acd483]

I'd also like your take on USB v. S/PDIF. I'm quite close to buying the Scott Nixon USB.UFO.JF DAC... www.scott-nixon.com/dac.htm
to be battery powered so as to match my RWA Sig 30.

He does a direct USB to I2S conversion, taking S/PDIF out of the equation entirely. I think he uses the Philips TDA1543 16 bit R2R chip,
and this is one of the purest designed DACs I've found on the market, and a good price to boot.

I’ve already commented on the TDA1543 in the Altmann/LessLoss thread. Some people think it’s the best they’ve heard but even more people think the Bose Wave Radio/CD is the best they’ve heard. I guess that makes the Bose better by popular acclaim.

There is nothing wrong with S/PDIF if you use a good receiver or, better yet, get only the data from S/PDIF and the clocks from a local oscillator and clock divider. On the other hand, USB is probably the worst digital interface for high quality music. If you don’t know how it works, go to usb.org, download and read the specs. The last time I looked the appropriate documents were called usb20.pdf and audio10.pdf. There is also some good information at microsoft.com called USB_Audio_and_Windows.doc but chances are its been updated and renamed for Vista. The truth is out there is you care to look for it...or you can believe the snake oil salesmen who will tell you anything in order to get your money.

Here are some highlights. The data is packetized with each packet holding 1 millisecond worth of samples. With a 44.1K sample rate, the sample period is not an integer divisor of 1ms and some packets will have more samples than others. That’s OK because the USB jitter spec is +/- one sample! In case of a transmission error, the entire packet is discarded and the DAC outputs 1 millisecond of silence. There is no retransmission. Compare that to S/PDIF where error checking is done on each sample. In case of an error, the S/PDIF receiver repeats the last good sample and raises an error flag. In response to the error flag, it’s up to the DAC to accept the duplicated sample or do something else, such as interpolate through the error. Most DACs do the former, which in the case of upsampling results in interpolation anyway. In any event, one repeated sample is considerably less noticeable than 1 ms of silence.

Where are the NOS DACs that use this chip?

All of my DACs use NOS PCM1704 chips. Highend DACs don't because upsampling and delta-sigma is in vogue. Budget and DIY DACs don't because the chips are expensive and hard to find. You also need two of them along with some glue logic and knowledge of digital circuits to make them work. The latter two items are not needed when the chips are used with a DF1704.

It's my understanding that transmission errors in USB transmission are extremely rare. I'm not sure how much attention you've paid to USB chips, but the latest do a much better job dealing with data transmission/clocking than the original ones.

[jb]

It's my understanding that transmission errors in USB are extremely rare.

No more rare than S/PDIF errors. Read the damn specs!

Earlier today I read a post on another audio forum from an audiophile who had a very high quality USB-based PC audio system but he was bothered by occasional random clicks, maybe one or two a day. He tweaked and changed everything in his system but clicks would not go away until he changed his interface from USB to S/PDIF.

[acd483]

Haha, I read your specs...from 9 years ago. Hmmm...wonder if USB technology has improved since then...

And anyway, I wasn't comparing S/PDIF to USB in terms of errors made. We're not talking relativity here. In proper USB integration, it's an error every MONTH or so, not many times a day. But let's be honest, old USB interfaces are junk. USB interfaces that convert to S/PDIF first, rather than directly to I2S are junk. PC computers are junk. You haven't given a clue as to his setup. The truth is, his system may be so flawed, that while S/PDIF removed the clicking...which is a result of the DAC and the computer not talking to each other correctly, it may not have improved the SOUND at all.

[jb]

Haha, I read your specs...from 9 years ago. Hmmm...wonder if USB technology has improved since then...

And anyway, I wasn't comparing S/PDIF to USB in terms of errors made. We're not talking relativity here. In proper USB integration, it's an error every MONTH or so, not many times a day. But let's be honest, old USB interfaces are junk. USB interfaces that convert to S/PDIF first, rather than directly to I2S are junk. PC computers are junk. You haven't given a clue as to his setup. The truth is, his system may be so flawed, that while S/PDIF removed the clicking...which is a result of the DAC and the computer not talking to each other correctly, it may not have improved the SOUND at all.

I'd also like your take on USB v. S/PDIF.

You asked for my take and I told you. If you were already convinced that USB, with its +/- one sample jitter and 1 ms dropout in case of error is superior, why did you ask?

When it comes to clock quality, there is little difference between S/PDIF and USB when each is implemented properly. Both interfaces use a VCO that is phase-locked to the incoming signal. S/PDIF locks on the biphase preambles, which are timed by the word clock of the digital source. USB locks on the 1ms ticks from the host and adjusts the bit/word clocks to output all the samples it received in the last packet over the next millisecond interval. Neither VCO-derived clock is as good as the output of a crystal oscillator and clock divider clocking the DAC chip directly.

[brynaus]

Hi guys, just received my Promitheus Dac (USB with all silver upgrades) last week and have been burning it in since (about 100 hours total).  It now sounds a little silkier than when it first arrived but quite frankly it was fantastic from the beginning (with just a little harshness to remind you it was a new component).  Wow !  What a sound ! 

This was my first experience with a NOS dac and I actually like it far more (from memory) than the Vacuum State Electronics modified Sony SACD player I used to own (and which has always received rave reviews).  The Promitheus is far more musical.  Though the modified SACD was more transparent and had a better soundstage (and was the only CD player/SACD player to use a low pass filter only, and to entirely bypass the DAC altogether!), it always seemed a little too sterile and cautious/civilised to me (and this was with Parallel Single-Ended 300b monoblocks into Lowthers).  It didn't get up and boogy or get down and dirty (dirty in a fun sense, not a "Christina Aguillera" sense ... though Christina dirty might actually be fun too) !

This DAC is the antithesis of all that - It sounds a little like a very, very relaxed Naim component (I still have a CD 3 and Nait amplifier I love very much) but with MUCH greater transparency and bass.  That means it is dynamic, emotion bearing, crisp when required, soft when appropriate and always amazingly "coherent".  In fact, whilst I have enjoyed the levels of detail, natural warmth (and natural harshness when the recording is harsh - this isn't a "cloying" syrypy component by any stretch of the imagination), phenomenal bass depth and agility, it is this intoxicating cohesiveness which I find it's most amazing attribute.  The music hangs together beautifully, makes musical sense, the bass isn't slow, the treble isn't fast and everything is "cut from the same cloth" in the same way that good analogue is.  Music is joyful and vibrant, but as I said it is relaxed - this gives an interesting juxtaposition of a component which is cohesive and has great PRAT, but which doesn't subjectively sound "fast" (like the Naims do ... I know it's a colouration, but oh what a glorious colouration).  It sounds laid-back, and composed (not in a boring way), but still has plenty of energy and verve.  The sound is magnificent, but the musicality is even better according to my tastes.  I now wish I had ordered the upgraded caps to see just how good this thing gets
Like Bob, I have not yet received my USB module (which Nic has taked out of the main DAC casing due to noise issues), so bear in mind I have it sub-optimally set up at the moment (borrowed a friends Trends USB convertor and using a Radioshack crappy interconnect as digital coax - bad I know !).  I am actually happy with this as it shows Nic's perfectionist streak and integrity ... I know I will get a USB convertor when the USB convertor is good enough to go with the DAC and not sully its sound.  Oh, by the way,  Nic is adding a pulse transformer - at his own expense - to the USB convertor so that he is happy with the sound !!   Now THAT'S integrity.  Will post further review with more breaking in when the USB convertor arrives, but at this stage I feel I have snared an incredible bargain - and that feels good.

[tanchiro58]

brynaus,

Great review. It is happened exactly like my Promi DAC but I use RCA/BNC. I also bypassed the Panasonic output capacitors with Vitamin Q PIO. Now the DAC is going to another higher level of sound. 

[brynaus]

Hi Tanchiro,

It was partly your reviews which led to me purchasing this DAC, so thank you so much for your pioneering efforts !  It looks like I am going to have to learn how to solder very soon 

[tanchiro58]

It looks like I am going to have to learn how to solder very soon 

Brynaus,

If you would like to start soldering I suggest to buy a good solder station and quality lead from Vampire. Here are the links:

http://www.circuitspecialists.com/prod.itml/icOid/7307

http://www.soniccraft.com/products/connections/solder/vampire.htm

If you need help just PM me. Good luck

[jb]

Seems the movement towards the NOS DACs is purist in idea but is not necessarily the holy grail of design (if any of the ideas are or math is)?  Or it simply has not been really well implimented yet due to its being relatively new (I beleve it is NEO in the sense its a further investigaion of an earlier digital processing idea and is being openned again to see what the designers may have overlooked the first time).

Anyway thanks jb for attempting to get us to understand some of this stuff, if you have the patience most of could probably benifit greatly from the most simplistic explaination with possible simple analogies:

There is confusion because there is no universally agreed to definition of the terms oversampling and upsampling. Here are selected highlights and a simplified overview of the evolution of CD playback.
 
The standards for CDs represent the absolute minimum for recording sound, not music. The samples are too small and the sample rate is too low. According to sampling theory, the low sample rate required a very steep, ‘brick wall,’ anti-image filter, which contributed to the bad sound of the first generation DACs.

At the time, microprocessors were too slow to be able to manipulate the sample data in real time and DAC designers did what they could to increase the clock rate of the DAC to reduce the severe requirements of the anti-image filter. This brand new idea was called 2X over-sampling. It involved doubling the clock rate and inserting a null sample in between each recorded sample. Today, that technique is sometimes called decimation. The problem is decimation halves the energy content of the signal and adds noise. I believe preserving the energy content is essential to accurately reconstruct music.

The next attempt, called 4X oversampling, involved using 4 DACs that operate on successive quarters of the sample period. The outputs of the DACs are summed. The sample rate is quadrupled and the effective sample size is increased by two bits. However, instead of a vertical step between recorded samples, as there would be with no oversampling, there is a diagonal line between the two points. Instead of a square wave, the DAC outputs more of a triangular wave. In terms of energy content, a triangular wave does not approximate the energy in a sine wave as well as square wave does. Today we call it linear analog interpolation.

When microprocessors were fast enough to manipulate the sample data in real-time we got 8X oversampling and the digital filter. The technique, more properly called digital interpolation, attempts to recreate the shape of the signal in between each pair of samples from the original recording. To do so, a large number of samples preceding and following the sample period being interpolated are examined to determine the shape of the signal through the period in question. The accuracy of the interpolation depends on the accuracy of the samples and a similar shape of the signal leading up to and following the interpolated period. The effective sample size is increased by three bits.

So far, only the added samples are the result of calculations, the original samples are preserved, and the sample rate is increased by an integer power of two. The next generation of digital interpolation used non-integer sample-rate multipliers and was called UPsampling to differentiate it from all previous incarnations of OVERsampling. The only thing that is different is the non-integer multiplier and the fact that now every sample is the result of a calculation. The output is essentially a digitally synthesized version of the original recording.

Asynchronous sample rate conversion is a variation of upsampling. Instead of a fixed sample rate multiplier applied to the input sample clock, the sample rate multiplier is determined by continuously computing the difference between the input sample clock and another reference or output sample clock.

In my opinion, the evolution of CD playback has gone from bad to worse. Although upsampling improves the quality of steady-state sine waves, it doesn’t improve the quality of recorded music. That conjecture is proven by the renaissance of NOS DACs. If the newest generation of CD playback technology were as good as the promoters say it is, there would be no desire for music lovers to want to revisit the past.

Unfortunately, in their zeal to reject everything that was wrong with prior attempts to reproduce music, NOS proponents have thrown the baby out with the bath water. Yes, the brick-wall filter is bad and so is the digital filter, but a proper reconstruction filter, also called an anti-image filter, is an absolute requirement for any digital to analog converter. While 8X upsampling and extending the 16-bit sample to 24-bits with what is essentially noise doesn’t enhance the music, 24-bit DACs are superior to 16-bit DACs whether or not you use all the bits. Also, there’s nothing wrong with upping the sample rate provided you preserve the energy content of every sample.

[NewBuyer]

What's the story with 128X (or 256X) oversampling - anything gained from this? (Thinking of the AK4393 etc DAC chips here...)