Watts Up...?
Mar 31, 2016 at 5:39 PM Post #76 of 4,673
I was wondering if Rob Watts would mind commenting on the advantages of having more elements on the Pulse Array DAC. Obviously, the Chord QBD76 and DAVE have more elements in the Pulse Array compared to Mojo/Hugo/2Qute. Is it ultimately about noise floor performance? Can you use fewer elements and use a higher performance noise shaper to compensate? Or vice versa, can you get away with more elements and use a lesser performance noise shaper? Is there a limit where more elements is no longer better?
 
Mar 31, 2016 at 6:05 PM Post #77 of 4,673
  I'm trying to find a post (not sure if it was Rob's or not) I recall reading, about a year ago, which was very insightful on the topic of legitimacy (or otherwise 
wink.gif
) of 24/192, but it is, thus far, eluding me.
 
I'm also mindful that it'd be bordering on 'Sound Science' forum territory, so perhaps it is apt that I've been unable to locate it!
 


There were a number of Pono bashing articles at the time - could it have been this one?
 
Apr 1, 2016 at 1:52 AM Post #79 of 4,673

Mistah Currawong,
 
Nice hearing from y'all.   
 
Convicts is it?, if I pick up a couple of felonies will I start talking like an Aussie?,  I occasionally travel to North of London where the Brits are impossible for intelligent people to understand but they have a much wider range of vocabulary than us 4 letter word folks, in the USA.  The Brits have 6 or 7 meanings for each and every word, whilst we in the States keep it much simpler with mostly one meaning ( unless you count our versatile use of the word crap, which can mean everything ).  
 
I feel pleased having you quoting me.
 
Tony in Michigan 
 
ps.  I liked that recent review you did, might've been on the little Schiit Amp.  I almost wrote you about it but held-off, for some reason.  
 
Apr 1, 2016 at 1:55 AM Post #80 of 4,673
Rob, both Davina and your digital amp project seem to me that they will have the potential to be landmark products as they could well result in a level of performance never before achieved. It would seem to me that Davina has the potential to have the broader impact as all systems would benefit, Chord or otherwise, while your digital amp will only benefit owners of the DAVE. Is that correct?

Of the two projects, which is the bigger priority? Which one will be the bigger challenge?

 
Generally I have two primary motivations - getting closer to the un-amplified sound I hear in concert halls - or better sound quality is one. The second is about improving my understanding and knowledge. Davina mostly in the short term is about better understanding - it could take some time before we have substantial numbers of new recordings available. On the understanding side I have a number of issues I want answers too:
 
1. How much aliasing is acceptable? Conventional ADC's often use half band filters, and these have aliasing that peaks at -6dB. The argument is that you can't hear anything from 20 to 20k, so it does not matter. The fact that the aliasing has completely destroyed the timing of transients has not occurred to them. In my experiments, previous decimation had -140 dB performance, and I know that is not good enough measurement and SQ wise. 
 
2. Will we get lifelike depth reproduction? I want to hear an organ at 100m away sounding like its 100m away in your living room. For sure, a problem is noise shaper resolution, and I have designed 350dB (Dave standard) analogue noise shapers for Davina. Is this the whole answer?
 
3. Can we decimate then interpolate with WTA completely transparently? This will give an absolute indicator of how good the WTA filter actually is.
 
4. How do we get 16 bit data but with no depth problems? Will triangular dither do it perfectly?
 
Technically, I have already solved the issues of noise floor modulation, which is a major problem with ADC's - and I can do this because the analogue is all discrete. Also, distortion is mostly about the reference DAC, and I know from Dave how to solve that problem too. But there remains a number of very interesting issues to understand.
 
On the digital power amp - well this will initially make the biggest SQ change, as it will happen to all my music, and I know already what it will do to transparency for example. Some of the interesting stuff from this project (the second order analogue noise shaper) is already been proven with Dave.
 
They are both high priorities, but I guess Davina will need more work but I will learn much more from. Certainly, if the studios take it on, it will have much more long term impact as everybody will benefit.
 
Rob
 
Apr 1, 2016 at 2:08 AM Post #81 of 4,673
  I was wondering if Rob Watts would mind commenting on the advantages of having more elements on the Pulse Array DAC. Obviously, the Chord QBD76 and DAVE have more elements in the Pulse Array compared to Mojo/Hugo/2Qute. Is it ultimately about noise floor performance? Can you use fewer elements and use a higher performance noise shaper to compensate? Or vice versa, can you get away with more elements and use a lesser performance noise shaper? Is there a limit where more elements is no longer better?

 
To answer all the questions it's yes,yes,yes,and finally yes.
 
For example Davina ADC will have 40 elements as that was the only way I could get the 350 dB noise shaper small signal resolution performance, and have Dave levels of distortion.
 
Its actually a complicated mix of different factors - generally, all things being equal, you get 3dB lower noise for doubling the number of elements, and a 6 dB improvement in signal resolution - assuming the noise shaper time constants do not change.
 
Rob
 
Apr 1, 2016 at 11:26 AM Post #82 of 4,673
Thanks. That's fascinating with the elements for the pulse array. Dear Mr. Watts, I know in the past, you've explained your perspective on which measured parameters are important in DACs and how you think there are fundamental limitations with chip-based multi-bit SDM DACs/ladder DACs or PWM/DSD DAC designs with some of their measured parameters. I think you've discussed these quite extensively on the other Chord Head-Fi forums. Now that you're designing a new digital amplifier, I was wondering if you have some insights so far in terms of what parameters you find important and what your thoughts are with respect to existing classes of amplifier designs, especially since there has been a lot of advances in Class D amplifier designs with increasing uses of feedback and higher frequency oscillations. Do you also see other fundamental limits to other amplification designs in Class A or Class AB? Do you have a preferences in terms of power supply designs, such as regulated power supply vs unregulated but oversized power supplies (torroidal or switching)? Do you thinking clipping is still a big issues for people who use inefficient speakers? Or do you think the solution should be a form of digital soft clipping? And is the main challenge in high-end amplifiers similar to with DACs where low-level linearity and noise floor modulation are still the predominant problem to transparency and accurate musical reproduction? I know you may not be able to openly comment as much considering that Chord still sells their own amplifier designs and I believe there are other digital amplifiers currently on the market that has some of your design IPs. But any insights would be helpful although I doubt I would upgrade my amplifier until I've seen your latest product come to market.
 
Apr 2, 2016 at 1:15 AM Post #83 of 4,673
The biggest problem that Class D has is switching - and its kind of a fundamental problem - its the fact that the very act of switching the OP degrades the timing of transients, in the same way that digital does with sampling. Unfortunately, even when you switch at 1.6 MHz, you can't escape from the small timing errors that get created. So the digital power amp is actually non switching for the OP, so this eliminates this issue. So this by far is the biggest issue - but I also have all the same problems that you have mentioned with conventional power amps and with DAC's.
 
Rob
 
Apr 2, 2016 at 7:36 AM Post #84 of 4,673
Thanks, I never even thought of the issue of timing of transients for amplifiers before. That makes a lot of sense. But then for conventional amplifier designs, people also talk a lot about feedback. So even in the scenario where feedback is "properly done", is it just a matter of trade-offs currently where if you have more feedback, you're going to get better low-level linearity but it may cause more noise floor modulation and worsen the timing of transients? Whereas if you have less feedback, you're going to have more distortion and worse low-level linearity but then you're going to get slightly better timing of transients?
 
Apr 3, 2016 at 2:31 AM Post #85 of 4,673
Thanks, I never even thought of the issue of timing of transients for amplifiers before. That makes a lot of sense. But then for conventional amplifier designs, people also talk a lot about feedback. So even in the scenario where feedback is "properly done", is it just a matter of trade-offs currently where if you have more feedback, you're going to get better low-level linearity but it may cause more noise floor modulation and worsen the timing of transients? Whereas if you have less feedback, you're going to have more distortion and worse low-level linearity but then you're going to get slightly better timing of transients?

 
OK, you are talking about a number of different problems together, so lets try to un-tease your Gordian knot, without recourse to swords!
 
In class D we do indeed have a problem of timing of transients, and it sounds just like the problems I have with digital in converting sampled data back to the original continuous analogue signal - you can't hear the starting and stopping of notes correctly, and the music does not flow properly. In the case of Class D, the signal is being sampled but the timing is signal related, and the OP LC filter reconstructs the timing of transients - but it can not do an effective enough job of doing that. Another massive problem with Class D is noise floor modulation.
 
With linear amps, of whatever type, here are some notes:
 
1. Noise floor modulation - this is feedback independent, as the first stage is normally responsible for this - so subsequent gain and feedback can't fix an input stage problem. To do this you need lots of decoupling, RF filtering, and careful grounding and other techniques.
 
2. Low level non linearity - this isn't solved by feedback either, as most of the problems comes from the direct signal path - the coupling components and the feedback components themselves. You can only solve this by reducing the number of metal/metal interfaces and reducing the oxides and impurities at the interface..
 
3. There is nothing wrong with feedback per se. What the problem is is poor high frequency performance - and that can be caused by poor open loop distortion, or not enough feedback for high frequencies - its way more complex than simple "no global feedback" or "feedback is bad" argument.
 
4. Timing in linear amplifiers - this is not the same as the digital timing problem, as its more of a continuous non linearity problem. It is where the delay changes with signal level - this can occur with cored inductors or with OP stages that have signal related propagation delay. This distortion is known as phase intermodulation distortion (PID), and it sounds different to the timing of transients problem - reducing PID makes it sound smoother and darker, with better instrument separation and focus.
 
The problems of 3 and 4 can be solved by using an analogue noise shaper approach and this is something I have used well before with Dave. It pretty much eliminates HF distortion, and solves the PID problem of output stages too.
 
I hope this clarifies a very complex subject,
 
Rob
 
Apr 3, 2016 at 11:15 AM Post #86 of 4,673
Thanks. I think you sliced through that one pretty well. Clearly, high-end power amplifiers, like DACs, are not just about a bunch of technical specifications and "standard" measurements. There are lots of design subtleties that can tremendously alter the performance. Best to be patient and wait for your new product design.
 
Apr 3, 2016 at 9:20 PM Post #87 of 4,673
  The biggest problem that Class D has is switching - and its kind of a fundamental problem - its the fact that the very act of switching the OP degrades the timing of transients, in the same way that digital does with sampling. Unfortunately, even when you switch at 1.6 MHz, you can't escape from the small timing errors that get created. So the digital power amp is actually non switching for the OP, so this eliminates this issue. So this by far is the biggest issue - but I also have all the same problems that you have mentioned with conventional power amps and with DAC's.
 
Rob


Is a micro SD player for Mojo still in the works?
 
Apr 3, 2016 at 9:47 PM Post #89 of 4,673
Is a micro SD player for Mojo still in the works?
It is still in progress but according to Rob it could be months away. And I really hope it has a full sized card slot instead of a micro one.


Just a reminder:

http://www.head-fi.org/t/800264/watts-up/60#post_12465820
 
Apr 3, 2016 at 11:17 PM Post #90 of 4,673

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