[AndrewC88]

To add to what @Currawong says, I don't see a need to stick to any standard between MScaler and DAC - it is not like other manufacturers will want to support that link. There could even be a commercial advantage in having proprietary interface there and in fact, the dual BNC link may already be non-standard. Having said all that however, the most optimal solution might be to one day integrate the MScaler with the DAC with an internal optical coupler?

The Dual-BNC link is in fact standards based, on the Dual-AES specification extension to AES, with AES-3id (75 Ohm BNC). Chord's Dual-AES implementation interworks perfectly with dCS

 

[Tims Fong]

I was replying to a post in the Dave thread, and my reply started to get pretty big. So I have turned it into a blog post about how pulse array works....

Quote:
"Originally Posted by JaZZ

Thanks, Jawed, for explaining!

I have always asked myself what exactly this «Pulse Array» could be. Now it turns out to be based on Pulse Width Modulation (PWM), which I happen to be somewhat familiar with."


Pulse array is much more than PWM, as a simple PWM system has a considerable number of problems.

Let's backtrack a bit to DSD so we can see why you would want to move to a PWM based system. Imagine a DSD 64, so you only have 64 unitary value pulses to encode a 22.05 kHz sine wave, and you want to encode a -140 dB sine wave to reasonable amplitude accuracy.

The answer is - forget it. There simply are not enough pulses to accurately encode high frequency information in spite of what DSD fan boys may tell you. The best you can get with DSD 64 is 120 dB THD and noise 20 to 20k. So if we want more resolution, we need to encode more data (n bits not 1 bit) or run at a much faster rate.

The second problem that DSD has is down to switching activity. Now you may say that the benefit of DSD is that it is a single element, so therefore no distortion (unlike R2R where it is impossible to have acceptable distortion performance due to resistor mismatch and switching activity). But examine this DSD sequence:

0101010101010101

Or this sequence

1100110011001100

They both are reproducing analogue 0, both legal sequences that a noise shaper may adopt. But if you run with those values for ever, you will measure very different DC values with a real DAC - even one with a perfect clock. The reason for this is switching activity - the first sequence is switching at double the rate. Now a real output has N and P ch FET's in the output buffer, and these devices have different propagation delays, different OP impedances, and physically different layout with different inductance - the result of this is that a real device will always have a mismatch or error when reproducing a pulse - ringing or glitch energy, differing rise times against fall times, and this creates a switching error that then will non-linearly demodulate the noise shaper sequence. The result - distortion, noise, and idle pattern noise - and this is all before we start talking about jitter and RF intermodulation effects. So although DSD is claimed to have one element and so no distortion - in practice that is not true, as it has two states and those states are different.

So how do we solve the resolution problem and the switching activity problem but retain the single element benefit?

Use PWM. Let's take our DSD 64 and use a 2048 FS clock. Run the noise shaper at 64 FS, and use the extra clock cycles to encode more resolution - in this case we have 0 to 32 possible states, so instead of DSD's 1 bit we now have 6 bits (to encode 32 we need 6 bits), so much more resolution. Let's also ban 0 and 32 as a legal state, now we have a system where it is always switching at exactly the same frequency so now the error introduced by switching becomes constant and signal independent - which just means a DC error, no distortion or noise!

Audio nirvana, we have solved the switching error and increased the noise shaper resolution by 32 times so now we will get 150 dB performance and no distortion from switching!

Err - no. We still have a jitter problem - the output swings from fully + to fully -, and any master clock jitter will create noise - now it's not as bad as DSD, as the jitter induced noise is not so signal dependent (with DSD we get maximum switching activity reproducing zero, none at 100% modulation) - but it is still an issue.

And PWM as a modulation process creates HF distortion. Now you can reduce that by running the noise shaper at 2048FS and feed the switching behaviour back into the feedback loop, so the feedback of the noise shaper will solve it. But we have another problem - timing. Any reader of my posts will know that the timing of transients is a very important concept for me - hence the WTA filter etc. But with PWM you innately have a timing of transients' issue, as the amplitude of the signal affects the timing of the output - simply a higher amplitude signal will be earlier than a low amplitude signal.

Oh dear. So now we have timing problems too. But we had timing problems with DSD anyway - a small amplitude transient has a bigger delay than a large amplitude transient. This is because a small value transient will not get thru the OP quantizer, so needs the noise shaper integrators to build up in value to trigger the OP. You can see this problem in simulation, and it's of the order of uS - in timing terms massive. It's this error that gives DSD its unnatural soft sound - because transients have amplitude errors, the brain can't process the transient information properly, and hence you can't perceive (hear) the transients. And when you can't hear a transient then things sound soft - it's the audible equivalence of an image going out of focus.

So PWM helps but has other issues. Where do we go from here?

Let's take a step back, and examine the scale of the problem. In order to solve the resolution problem, what do we need for perfect transparency? And what level do we need for perfect timing?

Ask that question, how much resolution do we actually need will depend on doing listening tests to find out - theory won't help you as we have no idea how the brain process's ear data to determine depth for example. You can only do it by designing noise shapers and actually listening to them. In the Dave project I did exactly that and was shocked to the core by what I discovered - there is no limit to how good the noise shaper needs to be.

That's actually quite a profound statement. I can substantiate this by saying that a 200 dB (THD and noise 20 kHz BW) has much better depth perception than a 220dB one. Furthermore, listening to a noise shaper with better than 350 dB has audibly deeper depth than a 330 dB one (albeit it's getting to be a much smaller change). These are staggeringly small numbers - better than 350 dB has a trillion times more resolving power than DSD 64 - no wonder DSD has such poor depth perception - the noise shaping is simply not good enough at resolving small signals. Indeed, the required performance is still 30 billion times better than the PWM example I talked about earlier. Now it's possible that the requirement for better than 350 dB performance on the noise shaper was simply a proxy for something else going on in the analogue section - but that when I use this performance for digital noise shapers that are purely digital (truncation noise shaping) I still hear the benefits in terms of depth reproduction. So the evidence based on my listening is that the brain is sensitive to small signal amplitude error of any size - even if almost zero and we need > 350 dB noise shaping

What can we say about timing errors?

If you would have asked me this a few years ago, I would have said uS accuracy was needed. Now I make no such assumption - there is perhaps no limit to how good the timing of transients need to be. So how can I substantiate that bold statement? Unlike noise shapers, it's rather difficult to put a number to timing accuracy. I guess I ought to state what I mean by transient timing accuracy. I do not mean - unlike the rest of the audio business - ringing performance; this is absolutely not what I am thinking about when I talk about the time domain or timing accuracy. Ringing uses an illegal signal from sampling theory POV as it is not bandwidth limited, so you would not actually get a perfect impulse from a perfect legal bandwidth limited ADC. So why worry about a signal you will never get? So it is actually pointless talking about it. What I mean is the accuracy of the timing of transients. Imagine a bandwidth limited analogue signal that is being sampled in the ADC - it is fully negative, goes positive and at some time crosses through zero. Let us say it is sampled at 44.1 kHz, so every 22,676 nS it's sampled. Let us imagine that the signal is sampled, and then crosses through zero at exactly 20,155 nS after sampling. Of course, when it gets sampled again at +22,676 nS it will now be a positive value. The question is, when the DAC reconstructs the sampled data - converting sampled data back to a continuous analogue signal - when will the signal cross thru zero? Theory is completely clear and undeniable - if we use an infinite oversampling FIR filter with a sinc response at 22,676 nS and a perfect DAC we will reconstruct the time it crosses thru zero absolutely perfectly at 20,155 nS. But with a finite non sinc function reconstruction filter, it will not cross thru at exactly 20,155 - maybe at 19,000 nS or 21,000 nS. And it is these differences in the timing of transients, are what I am talking about. Now in the past I would have said that getting it right to a uS was perhaps OK (timing errors can be as big as 100uS in conventional filters) - now I know that instead of worrying about uS we need to worry about getting it correct to nS's.

What is the evidence for that view? In designing Dave, I wanted to discover what I had done in the Hugo design (it was a happy accident) to give me the timing performance that I so enjoyed with it. By this I mean the ability to hear the stopping and starting of notes. After trying different things, I chased down this quality to the interpolation filters after the WTA filter. Now with Hugo, I used a 16FS WTA filter, followed by a linear interpolator and a two stage IIR filter filtering up to 2048 FS. Changing this to a 256 FS WTA filter followed by my usual 3 stage filtering gave a massive change in sound quality - at this point Dave was sounding impossibly rich and smooth and almost soft sounding. By changing it to 256FS WTA gave a substantial change in character - it was still smooth, but very fast and you could hear the starting and stopping much more easily. It went in character from soft and smooth to fast and sharp - when the occasion demanded.

Now replacing the WTA from 16FS (data every 1,417 nS) to 256 FS (data every 89 nS) is technically very small in the sense that transient accuracy using a WTA against an IIR filter at this speed is not a vast change in the time domain - it is a very subtle difference, but was nonetheless extremely audible. What it tells me is that very small - impossibly small - timing errors are very significant for the brain's ability to process the ear data.

So we have timing uncertainty due to the filter reconstruction, and timing errors due to the noise shaper. To eliminate filter errors we need a filter that oversamples to the max possible, with as close as possible to a sinc response - hence why 256 FS WTA filters, and further digital filtering to 2048FS. But we also need a DAC that has no amplitude errors either - by this I mean small amplitude transients has exactly the same delay as large amplitude transients - otherwise the brain can't perceive the starting and stopping of notes and things sound soft. Now we can't use DSD - the 1 bit noise shaper errors amplitude timing errors are greater than a uS - we can't use PWM, as this also creates amplitude related timing errors - we can't use R2R or ladder DAC's as they can't go faster than 16FS so will have interpolation filter timing errors.

So you can see now why I wanted to solve these problems and others well over 20 years ago by inventing pulse array. So how does pulse array solve these issues? The problem with PWM is the switching activity of one element. If we have 16 PWM states, and then use 16 different elements with different timing, then we can arrange the system so that a rising edge is always balanced by a falling edge for a constant OP, so that when they are added together there is no net switching activity. In short, with 16 elements and 16 PWM time states we can completely balance switching activity - this in a nutshell is how pulse array works.

The primary benefit of the system are that when the noise shaper asks for a value 8 say, then the output will immediately give a value 8 constantly with no variation. With all rising edges being matched by a falling edge, for a constant noise shaper value, the clock jitter has no effect whatsoever, so the system is innately jitter insensitive.

The key benefits are:

1. Element mismatch. The value of the resistor makes no difference to distortion at all as each element carries the wanted audio signal equally. Measurement and simulation has shown that resistor tolerance only creates a fixed unvarying noise. With Dave (20 pulse array elements) this noise is small compared to the amplifier and Johnson noise of the resistors.

2. Clock independent. The output from the array has a rising edge exactly balanced with a falling edge for all possible outputs actually used. Thus when it is reproducing a constant noise shaper value, common master clock jitter has no effect. The only possibility for jitter to play a part is when the noise shaper changes its output - and here the error is 1/16 of the DSD or PWM error. Simulations of adding jitter to the elements revealed negligible output of a fixed noise - negligible as it was very small compared to the resistor and amplifier noise.

3. Signal independent switching. The switching errors (rise time fall time and glitches) are constant so are signal independent - thus no noise floor modulation or distortion.

4. No transient timing problems. Since the analogue output is identical to the noise shaper output, the problems of PWM creating timing errors are eliminated. Since the noise shaper can run at 104MHz, I have failed to see any problems of transient timing errors; the output crosses thru zero with small transients and large transients with no consistent error (unlike DSD).

5. With the very fast clock and noise shaping running at 104 MHz, I can make noise shapers with better than 350 dB THD and noise performance - which I have found to be essential for realistic depth perception.

But talk is easy, measurements are the proof of the pudding. Take a look at this simulation. Now this is the actual analogue output from pulse array but with of course ideal analogue performance, but it proves that pulse array is innately capable of resolving a -301 dB signal perfectly, which is now my standard test for depth capability:



Here we have Dave's 20 elements reproducing a 6 kHz -301 dB sine wave. It is accurate to within 0.001 dB, and has no distortion whatsoever.

What is also very unusual is the high frequency performance - at 100 kHz the noise floor from the noise shaper is at -200dB. This is crucial, as it means that Dave needs very little analogue filtering, and it means the output amplifier does not have to deal with large amounts of out of band or RF noise. This is very important for two reasons - I do not need to have a complex analogue filter on the output, and the lack of RF noise innately gives much lower noise floor modulation.

Indeed, because the filtering needs are simple, I actually only need one amplifier, combining the roles of I to V, filtering, output buffering and headphone drive in one single amp and having very simple analogue electronics maximizes transparency.

Noise floor modulation is subjectively very important too as it adds hardness and grain to the sound when measurable; even when reducing it below measurable levels you can still hear it; as reducing it further makes things sound smoother and warmer. Pulse array is the only DAC technology that enables no measurable noise floor modulation - all other forms of DAC's I have seen have major noise floor modulation problems. To prove this, here is an actual measurement using an APX555:



The plot in blue shows 2.5v RMS 1 kHz, and red shows the AP with no signal. What you can see from this is the complete absence of any noise floor modulation at all - noise is just fixed and unvarying.

Of course with pulse array there are other things going on under the hood which are too complex to explain, in particular how I designed a noise shaper that can perfectly resolve a -301 dB signal. And of course, some will say that we simply do not need that level of performance. Indeed, before I actually listened to 350 dB capable noise shapers, I too would have agreed with that view!

Rob

I don't know any professional skills, but I have feelings. Listen to DSD music as if I was outside a sealed recording studio, listening to music with headphones on. And listen to the PCM music, it seems to be on the scene, music through the air into my ears. That's how I feel.

 

[JaZZ]

...Dave will be more transparent than the first DX amp via headphones.

[from the Hugo M Scaler thread)

Could you elaborate? Of course I don't expect the DX amp to improve the sound via (normal) headphones compared to the DAVE's headphone output, but I expected it to be about equal, since the DX amps are rumored to sound absolutely neutral and transparent – like a wire with gain. Or does your statement just refer to its headphone output, not the speaker terminals?

 

[Rob Watts]

It's just the headphone OP, and I was referring to the first DX amp...

 

[ecwl]

Of course I don't expect the DX amp to improve the sound via (normal) headphones compared to the DAVE's headphone output, but I expected it to be about equal, since the DX amps are rumored to sound absolutely neutral and transparent – like a wire with gain. Or does your statement just refer to its headphone output, not the speaker terminals?

I wonder how much Rob Watts is supposed to talk about DX amps since it's not a product in production yet and John Franks is always watching, haha...
But my gut feeling about DX amps, based on Hugo TT 2 and current design descriptions are as follows. First, DAVE has 20 element pulse array whereas Hugo TT2 uses 10 elements. It is not clear if DX amps can even have 20 elements so presumably the SNR is going to be better with DAVE if you're going into headphones. Moreover, I would presume with TT2's 8W? DX amp output, the SNR is for 8W so if you're only using 1W, if you run it through the TT2's 8W DX amp, you're automatically losing 8/1 = 18dB in SNR. But of course, if your headphones actually need the extra juice to go above 1W, you're getting what you want out of the DX amp.
For me, there are already a few thought experiments on the matter. For example, let's say your speakers only need 30W to drive it to the maximum volume you'll listen to, even if a DX amp exists for 200W, you'd probably actually want to buy the version that runs at 50W to get better SNR, because of the nature of the amp design (unless the 200W one has significantly better SNR than the 50W version which would be very unlikely). On the other hand, let's say you can get a 10-element 30W DX amp or you can pair your DAVE with a class A/AB uber 30W amp, my strong suspicion is that the transparency of the 10-element DX amp would be so much superior that you'll end up with far better SNR and transparency through the DX amp, rather than using DAVE + uber 30W analog amp.
As I write this, I realize all these issues are present with our current analog pre-amp/amp systems. It's just that we don't think that hard about them because manufacturers like to brag about how transparent their analog systems are. But I think if DX amps are going to push performance to the next level, these are factors that would start coming into play. For example, my DAVE when connected to my 500W amp is usually running between -50dB to -18dB. So clearly, I should have gotten a lower powered analog amp for even better transparency and SNR.

 

[JaZZ]

It's just the headphone OP, and I was referring to the first DX amp...

So it would seem logical to use the speaker terminal for driving insensitive headphones instead of the headphone output, no?

I recall that only the first (weakest) DX amp will have a headphone output at all. Does this still hold true?

I wonder how much Rob Watts is supposed to talk about DX amps since it's not a product in production yet and John Franks is always watching, haha...
But my gut feeling about DX amps, based on Hugo TT 2 and current design descriptions are as follows. First, DAVE has 20 element pulse array whereas Hugo TT2 uses 10 elements. It is not clear if DX amps can even have 20 elements so presumably the SNR is going to be better with DAVE if you're going into headphones. Moreover, I would presume with TT2's 8W? DX amp output, the SNR is for 8W so if you're only using 1W, if you run it through the TT2's 8W DX amp, you're automatically losing 8/1 = 18dB in SNR. But of course, if your headphones actually need the extra juice to go above 1W, you're getting what you want out of the DX amp.
For me, there are already a few thought experiments on the matter. For example, let's say your speakers only need 30W to drive it to the maximum volume you'll listen to, even if a DX amp exists for 200W, you'd probably actually want to buy the version that runs at 50W to get better SNR, because of the nature of the amp design (unless the 200W one has significantly better SNR than the 50W version which would be very unlikely). On the other hand, let's say you can get a 10-element 30W DX amp or you can pair your DAVE with a class A/AB uber 30W amp, my strong suspicion is that the transparency of the 10-element DX amp would be so much superior that you'll end up with far better SNR and transparency through the DX amp, rather than using DAVE + uber 30W analog amp.
As I write this, I realize all these issues are present with our current analog pre-amp/amp systems. It's just that we don't think that hard about them because manufacturers like to brag about how transparent their analog systems are. But I think if DX amps are going to push performance to the next level, these are factors that would start coming into play. For example, my DAVE when connected to my 500W amp is usually running between -50dB to -18dB. So clearly, I should have gotten a lower powered analog amp for even better transparency and SNR.

I'm always prepared to see Rob refuse to answer one of my questions, and I appreciate his patience and openness, which I don't take for granted.

I'm not sure if the noise floor is really such a big problem, from what I get it's rather about noise floor modulation, so the bigger DX amps may in fact not suffer from the mentioned issue.

 

[Triode User]

For example, my DAVE when connected to my 500W amp is usually running between -50dB to -18dB. So clearly, I should have gotten a lower powered analog amp for even better transparency and SNR.

Into my Pass Labs 60w monos I have my Dave (with Blu2 so that is 3dB down on just Dave) at about -15dB for cleaning the ear drums, -20dB for good rock music and quite a bit quieter if my wife is at home.

 

[514077]

I'm sorry @Rob Watts, but I'm a little confused. My admittedly layman's understanding about DSD64 was that it was sampled at 64X44.1K = 2.28...MHz. So your statement "m. Imagine a DSD 64, so you only have 64 unitary value pulses to encode," iseems to imply that DSD is extremely low in the information needed. I'm a bit lost. When you have time, could you clarify, so I can follow you a bit better? Thanks.

 

[Rob Watts]

Yes there are only 64 one bit pulses to encode the 44.1 kHz equivalent PCM sample; and this is the problem in that the high frequency resolution is innately not good enough.

 

[JaZZ]

Yes there are only 64 one bit pulses to encode the 44.1 kHz equivalent PCM sample; and this is the problem in that the high frequency resolution is innately not good enough.

How true!

That means, for a half-cycle of a 16 kHz wave there are only 71 samples available, resulting in 35.5 amplitude values in the best case of a pure triangle wave. For a sine wave things look even worse, not to speak of a frequency mix.

 

[musickid]

Hi Rob,

Slightly unrelated topic. I picked up some thin granite slabs from argos which look identical to the audiophile ones. Does granite possess the right qualities to act as a base for expensive items like H2/Mscaler? Sometimes it feels slightly cool to the touch. Thanks Mk.

 

[Triode User]

Hi Rob,

Slightly unrelated topic. I picked up some thin granite slabs from argos which look identical to the audiophile ones. Does granite possess the right qualities to act as a base for expensive items like H2/Mscaler? Sometimes it feels slightly cool to the touch. Thanks Mk.

It depends. Have you put the slabs in the fridge recently?

Seriously though, it probably feels cool to the touch because the stone is quite good at conducting heat and will be conducting heat away from your hand when you touch it. Hence why it feels cool.

What indeed are the right bases for H2/MScaler/Blu2 etc etc? I don't know that there is one. I just put my gear on the table. Any table where ever I happen to be.

If you have a slab of something though, maybe suspend it from one corner and tap it. If it rings or resonates then maybe it isn't such a good thing to use. If it has a dull thud then its probably OK. That's a rough and ready answer but probably not too far off the mark.

 

[musickid]

Good advice thanks.

 

[Triode User]

Good advice thanks.

Maybe, maybe not. I am obsessive about some aspects but try not to overthink others.

 

[Tims Fong]

Hi，Every Body，
Do you have any Chord Mojo computer Control Panel plugins available? Because I want to see the volume of chord mojo when playing.Thanks。