• WANTED: Happy members who like to discuss audio and other topics related to our interest. Desire to learn and share knowledge of science required as is 20 years of participation in forums (not all true). There are daily reviews of audio hardware and expert members to help answer your questions. Click here to have your audio equipment measured for free!

More Rob Wattage

Purité Audio

Master Contributor
Industry Insider
Barrowmaster
Forum Donor
Joined
Feb 29, 2016
Messages
5,390
Likes
5,135
Location
London
Rob Watts of Chord ( DACs) on power supplies, is there any real information here?

This thread got a little interesting last week with talk about the PSU within Dave. Now that the dust has settled, I thought it would be a good idea to discuss the thinking behind the power supply with Dave.

When it was mentioned that Dave's PSU was just a simple medical SMPS I had a little chuckle to myself.

Because the medical SMPS is not Dave's PSU - at least not the PSU that actually matters. You may as well talk about which power source you are using from the mains - nuclear, fossil or wind/solar? OK that's an extreme exaggeration, but power is simply a serial chain, and the medical SMPS is not the critical component within Dave's power structure. It's actually an extremely elaborate and complex interdependent system. And this is the reason why I found the post amusing. So that you can understand, lets talk about what you need from a PSU within a DAC:

1. RF noise. This by far is the most important thing in a PSU and analog electronics; it's something I talk about a great deal. The problem is that analogue audio components are non-linear, and very non-linear at RF (and by that I actually mean 20 kHz to several GHz). When a random RF noise gets into active audio components, it distorts with the wanted audio signal and creates inter-modulation distortion - and some of the inter-modulation distortion has an audio component - audible random noise. What happens with this is when there is no audio signal, you get no inter-modulation distortion, hence no extra noise. When the audio signal increases, the inter-modulation products increases, and noise goes up - so noise levels becomes linked to the signal level and you get noise floor modulation. Now this issue is easy to measure, and taking steps to remove RF noise lowers noise floor modulation. Additionally, you can improve the analogue electronics open loop RF linearity too.

The problem with noise floor modulation is the ear/brain is extremely sensitive to it, and certainly can detect levels of noise floor modulation that is below the ability to measure. My own tentative conclusions (or rule of thumb) are that one can hear levels of noise floor modulation down to -200dB - currently we can measure noise floor modulation at -180 dB, and Dave has zero measured noise floor modulation. In terms of SQ, if noise floor modulation is say around -120 dB (typical class D) you get considerable hardness and glare; at -140 dB its grain in the treble; below -160 dB then things sound much smoother with better instrument separation and focus. This continues until about -200dB (and perhaps even lower - reducing RF noise is not something that has an acceptable limit).

But to get this level of performance required me to do many things; and part of that was the PSU system. I call it a system because it is lots of parts working together, with sources of noise within the DAC contaminating other parts. One source is the mains power, so this is dealt with a filter that starts at a few 100 Hz to several GHz - each PSU line (+15v,-15v,+5v) from the SMPS is individually filtered with a complex multistage filter. That covers RF noise initially, but every analogue part is individually regulated and RF filtered again. Moreover, each digital module is individually RF filtered too, as each digital part of a DAC is a huge RF noise generator.

2. PSU impedance. The impedance of the power supply is crucial too as it can create distortion which is audible. The actual mechanism for distortion from an amplifier is due to the fact that an amp draws signal related current from the PSU; this current then creates a voltage drop in the power rails (an error) that is signal dependent; this error then is fed back to the amplifiers output by the amp power supply rejection ration (PSRR); the error then creates distortion in the output. For Class A it is second harmonic, for class B it is very serious HF harmonics extending to infinite harmonics. Now its very easy to design the amp to avoid this problem; simply use low enough output impedance PSU and an amp that has a large enough PSRR. It is easy to calculate your requirements; much more difficult to design a stage that meets those requirements. So with Dave I wanted no measurable effect from this; this meant a PSU impedance in the OP stage of 3 milli ohms, which would make this effect un-measurable at -180dB. And that's exactly what I achieve; loading the output stage shows no measurable PSU induced distortion at all.

3. Reference supply. So far I have talked about the requirements for the analogue section; and this is an interaction between amp topology (sensitivity to RF and PSRR) against the actual levels within the PSU. So you could use a poor PSU with an amp that had amazing RF and PSRR or vice versa. My approach with Dave is to use both strategies - get the best innate isolation with the best filtering I could do. But there is a PSU where whatever happens on the PSU will be directly on the output and this is the reference PSU. Now nobody talks about the reference for two reasons - it is buried inside a DAC chip, and because people are not aware of how crucial this component is. The reference supplies the voltage that is used to create the analogue voltage (or current) output on the DAC, so clearly if that voltage changes it will have an immediate and 100% impact on the output. Normally, with a silicon DAC the reference presents a problem; noise on the reference will appear on the output, so making the SNR and dynamic range (DR) depend on the reference. Its actually extremely difficult to have a 120dB reference voltage on chip; so to overcome this they use a differential structure (balanced + and -). This means the reference noise, when its reproducing small signals is cancelled as it becomes common mode. So you get good DR figures from a poor reference. But when the signal becomes larger, the cancellation stops and you then see the reference noise and you get noise floor modulation - which is very measurable and very audible. Its one reason why chip based DAC's have large amounts of noise floor modulation.

Now with all my DAC's I have the freedom to design my own discrete reference, so can use very low noise references - and I do not have to worry about substrate noise upsetting the reference, something that is impossible to eliminate on chip DAC's. So the reference is crucial for noise floor modulation, but its also crucial for distortion too. Any signal related currents on the reference will create an error voltage, which will modulate the output causing distortion. In the case of a differential structure it causes amplitude modulation - a problem that won't be seen with simple measurements of distortion. With my SE designs I get to see it directly, which means its easy to measure it and solve it - and because you are also eliminating amplitude modulation (which is very audible) it gives me a short cut to better sound quality.

With Dave I had to go to immense efforts to solve these problems - and its one reason why I get such good measured performance. But it took many years to design and get right. With Dave the reference OP is less than 0.1 milli ohms at the feedback point (that's a damping factor of about 100,000) and at the PCB power plane each flip flop sees only 0.5 milli ohms. Now the actual distortion you get with reference impedance does depend upon the flip flop currents, and this is a system design issue of balancing resistor noise with distortion - resistor noise being the dominant noise source for Dave. To eliminate measurable noise floor modulation I had to design the noise of the reference to be less than 0.3 uV 20 to 20k. Considering it is a 5v voltage, that is a DR of 144 dB. What other power supply or amplifier has 0.1 milli ohms and -144 dB noise?

So you can see my amusement about the PSU discussion. Would using a better SMPS give better SQ? Maybe, but the evidence suggests no. When I used this power supply, I initially used the DAC 64 unit. But I could see switching noise at -130 dB at 40 kHz - going to the better medical PSU eliminated this problem - but when I replaced the unit I could not hear a change (I did not do an AB test, just plugged it in and listened henceforth and I was not struck by my system sounding better). So far the listening evidence is that is good enough.

I always get amused by audiophiles listening with their eyes, so a big PSU with lots of decoupling caps must be better than a small PSU. But the actual reality is that PSU interaction and sound quality is a hugely complex interdependent system problem, and eliminating these problems are not visible, nor are they easy. Also, the techniques I use that I have discussed in this post is used from Mojo up to Dave; its just a question of how much budget I have available to solve these issues. Even Mojo has no measurable noise floor modulation at -170 dB; and it uses the same principles of a discrete reference circuit - albeit not as sophisticated and costly as Dave's.

Rob

Keith
 

amirm

Founder/Admin
Staff Member
CFO (Chief Fun Officer)
Joined
Feb 13, 2016
Messages
39,111
Likes
174,562
Location
Seattle Area
It is the usual Rob using technical words beyond the knowledge of audiophiles and not providing any proof of audibility or even measurements of the same. Gigahertz RF is a problem? Really?

Anyway, it is true that it is his problem to build a performant DAC however he does it. That said, it is bad PR statement for him to build his own custom DAC but when it comes to power supply, he buys a commodity switching one. I suspect he doesn't know how to design that power supply so he bought one off-the-shelf and post filtered.
 

solderdude

Grand Contributor
Joined
Jul 21, 2018
Messages
12,207
Likes
27,313
Location
The Neitherlands
I kind of agree with Rob that local decoupling, internal power supply rails, ground plane and references are much more important than the used power brick.
A medical grade UPS isn't any better than a normal powerbrick. It just has some specific requirements such as low leakage currents (good for audio) and higher voltage limits for power supply differences.

Designing a decent power supply is not a problem I think. He just picked a decent of the shelf one as he knows it isn't that important.
This has been shown to be the case by some of your power supply and 'PS cleaner' measurements.

This DAC measures and performs well. Might be over-engineered here and there increasing the price. There are other manufacturers that also reach those levels. Even the MoJo measures and sounds well so they know how to engineer this stuff.

The whole -200dB noisefloor modulation stuff I don't buy. He can't back out of this any more though. Didn't he even mention -300dB in the past ?
 
Last edited:

JJB70

Major Contributor
Forum Donor
Joined
Aug 17, 2018
Messages
2,902
Likes
6,053
Location
Singapore
Watts is doing what most audio manufacturers do, telling a story that appeals to his target audience and playing to what he sees as his strengths. I agree with Solderdude, much of what he has said this time is reasonable apart from silly noise floor stuff. If we stripped out all the solutions looking for problems from audio (distinct from outright fraudulent snake oil) we would have a much smaller range of equipment.
 

elberoth

Member
Joined
May 6, 2018
Messages
81
Likes
162
Location
Warsaw, Poland
I suspect he doesn't know how to design that power supply so he bought one off-the-shelf and post filtered.

Chord has been a pioneer in SMPS design in audio components - they have been using them in their amps from the late 80s / early 90s.

I'm sure they have enough engineering expertise to design a superb SMPS ;)

The main reason most of the DACs nowadays use an outboard SMPS is cost. Not the cost of real estate inside the component or the SMPS itself, but the cost assiociated with conformity tests. AFAIR there are 7 separate tests a (mains AC fed) component has to pass if you want to sell it worldwide (US have their own, China has their own etc).

An easy way around this to outsource SMPS to a manufacturer that makes hundreds of thousends of them, and can split the considerable cost of certificition across thousends of units.

You just buy a SMPS with all the certificates and voila - no further tests necessary! (low voltage DC fed ccomponents do not require those tests).
 
OP
Purité Audio

Purité Audio

Master Contributor
Industry Insider
Barrowmaster
Forum Donor
Joined
Feb 29, 2016
Messages
5,390
Likes
5,135
Location
London
Even more Rob Wattage, sorry to keep posting these, but his pronouncements never quite ring true, the problem is I am not clever enough to know whether they are not!
On digital versus analogue attenuation
Quote,
So when you have an app that has a volume control, and no bit perfect setting, then set it to full volume on the app on the assumption that this will keep the data closer to the original file. The volume control function on Mojo is much more sophisticated than the PC as I employ noise shaping and I do the function at a very high internal sample rate. Hopefully using the volume set to max on the app will mean the volume coefficient is 1.0000000... so it will return the original data."

"If I were to use rounding then what you say is correct, there would be small signal non-linearity.

Mojo categorically does not use anything as crude as rounding to convert bit depths as the volume function is running at 16FS (705.6 kHz or 768 kHz) - I use extensive noise shaping to change bit depths. Mojo's noise shaping from beginning to end through all the intermediate paths (that is digital input to the 4e pulse array outputs) ensures 200 dB performance in band - that's better than 32 bit performance. The benefit of this is small signal non-linearity is much better, and this is essential for depth perception - the tiniest error in small signal amplitude, no matter how small, is audible in terms of truncation of perceived depth of sound-stage.

Your assertion that analogue does not have these problems is incorrect. Any metal to metal interface contains oxides and other impurities - and copper oxide is diodic, and so attenuates small signals and creates small signal distortion. Moreover, carbon track volume controls are also non linear, as carbon composition has significant voltage dependency of resistance - another source of non-linearity. Analogue electronics additionally suffer from RF noise pick-up, which when added to an active stage will then create more noise floor modulation due to audio signal and random RF noise inter-modulation.

Mojo, unlike all other non Chord DAC's, has no measurable noise floor modulation, and zero distortion of small signals, with no measurable fundamental signal non-linearity. This is not something that other DAC's can do, nor is it something that an analogue volume control can do too. And the benefit of all this is refinement and transparency - key ingredients for musicality."

Keith
 

pozz

Слава Україні
Forum Donor
Editor
Joined
May 21, 2019
Messages
4,036
Likes
6,672

bennetng

Major Contributor
Joined
Nov 15, 2017
Messages
1,562
Likes
1,570
I don't own any Chord products so my knowledge are limited to Amir's measurements. Amir did not test Chord's volume control so I don't have any info to comment about. However I can write something about what I observed in Amir's measurement.

So let's use the test signal attached in this post and this foobar2000 resampler (SoX based) as example.

I used foobar's file converter and the resampler plugin with these settings and output to 32-bit float, one file is 99% and the other one is 90%.
filter.png


Then inspected them in DeltaWave:
intersample.png


While both files have about 155dB of stoppand attenuation in digital domain (200-45 as shown on the graph), the 99% file has a higher peak so the price for a steeper filter is higher intersample peaks (of course it depends on audio files), that means more digital attenuation is required to prevent clipping and thus reduces dynamic range at analog output.

Here are digital filters of the measured Chord products at analog output:
index.php

index.php


Now a typical DAC:
index.php


So even typical DACs these days have deeper attenuation (~100dB) than Chord, just not as steep. Also you can see that even Chord doesn't achieve full attenuation at fs/2.

DNR... I suppose the voltage digital level is controlled by AP? Interesting to see the "clipping" comment as well:
index.php


and there is "Chord hump" as well, makes me wonder how they can be "linear" up to 200 or 300dB.
index.php

index.php


Dave seems to have a steeper and deeper filter, but foobar's SoX plugin is free and its 155dB stopband attenuation is not going to bottleneck the analog output. Also, pay attention to JA's clipping comments.
https://www.stereophile.com/content/chord-electronics-dave-da-processor-measurements
617Davefig02.jpg
 
Last edited:

mansr

Major Contributor
Joined
Oct 5, 2018
Messages
4,601
Likes
10,330
Location
Hampshire
Even more Rob Wattage, sorry to keep posting these, but his pronouncements never quite ring true, the problem is I am not clever enough to know whether they are not!
On digital versus analogue attenuation
Quote,
So when you have an app that has a volume control, and no bit perfect setting, then set it to full volume on the app on the assumption that this will keep the data closer to the original file. The volume control function on Mojo is much more sophisticated than the PC as I employ noise shaping and I do the function at a very high internal sample rate. Hopefully using the volume set to max on the app will mean the volume coefficient is 1.0000000... so it will return the original data."

"If I were to use rounding then what you say is correct, there would be small signal non-linearity.

Mojo categorically does not use anything as crude as rounding to convert bit depths as the volume function is running at 16FS (705.6 kHz or 768 kHz) - I use extensive noise shaping to change bit depths. Mojo's noise shaping from beginning to end through all the intermediate paths (that is digital input to the 4e pulse array outputs) ensures 200 dB performance in band - that's better than 32 bit performance. The benefit of this is small signal non-linearity is much better, and this is essential for depth perception - the tiniest error in small signal amplitude, no matter how small, is audible in terms of truncation of perceived depth of sound-stage.
Any attenuation, whether digital or analogue by necessity causes an overall loss of dynamic range. What he's saying here is that instead of simply multiplying each sample value by the attenuation factor and rounding the result to the nearest representable value, he performs this multiplication after upsampling to a higher rate and uses shaped dither to maintain a high dynamic range in the audio band, naturally at the expense of increased noise at higher frequencies. There is nothing magical or mysterious about this, and it can easily be done in software on a PC. Of course, most DACs don't accept 768 kHz sample rate input (though some do). The question is, does any of this matter? That's very unlikely. Every DAC in the known universe has a noise limit equivalent to, at best, 22 bits or maybe a little more. As long as whatever noise is added by rounding/dither in the attenuation falls below this level, the details make no difference to the final analogue output. As usual, Rob Watts is making mountains out of single-atom mole hills.
 
Last edited:

Tks

Major Contributor
Joined
Apr 1, 2019
Messages
3,158
Likes
5,200
I'll never forget him talking about how scientifically he accepts it doesn't make sense to hear distortion artifacts -300dB, but alas, he's able to do it in his own personal hearing tests.

Still waiting on the day he demonstrates this in a scientific setting.. He (like many other audiophile buffoons) seems to think because we can't have something measured, we're somehow averse to being presented with a phenomena (explained or not regardless as long as the phenomena is demonstrated to be occurring). These sorts of infantile handwaving strawman mischaracterizations in order to excuse yourself from scrutiny are baffling to see from someone constantly reinventing the wheel to make products the rest of the industry does at far less cost.
 

RayDunzl

Grand Contributor
Central Scrutinizer
Joined
Mar 9, 2016
Messages
12,050
Likes
13,820
Location
Riverview FL

scott wurcer

Major Contributor
Audio Luminary
Technical Expert
Joined
Apr 24, 2019
Messages
1,503
Likes
2,803
I don't follow this much, but has Mr. Wattage ever proposed or revealed a hardware solution that produces these numbers? I don't know of any standard DSP chips that maintain full 64bit FP precision (~-300dB) input to output. This can be done with a CPU but I don't know if it can be done in real time at the highest sample rates.
 
Last edited:

March Audio

Master Contributor
Manufacturer
Joined
Mar 1, 2016
Messages
6,382
Likes
9,198
Location
Albany Western Australia
Even more Rob Wattage, sorry to keep posting these, but his pronouncements never quite ring true, the problem is I am not clever enough to know whether they are not!
On digital versus analogue attenuation
Quote,

Mojo categorically does not use anything as crude as rounding to convert bit depths as the volume function is running at 16FS (705.6 kHz or 768 kHz) - I use extensive noise shaping to change bit depths. Mojo's noise shaping from beginning to end through all the intermediate paths (that is digital input to the 4e pulse array outputs) ensures 200 dB performance in band - that's better than 32 bit performance. The benefit of this is small signal non-linearity is much better, and this is essential for depth perception - the tiniest error in small signal amplitude, no matter how small, is audible in terms of truncation of perceived depth of sound-stage.


Keith

All this is demonstrably untrue. Thats fantastic that the mathmatical process creates a theoretical 200dB performance. However the simple fact is that the electronics that turn this into analogue voltages dont come anywhere near to this level of performance. As mansr has already pointed out the best we can achieve is in the region of 22 to 23 bits maybe 138dB. Thats already beyond what we can hear.

In any case the Mojo has been tested here and its dynamic range is only 111dB!

Simply put, its just not physically possible to hear the things he claims makes a difference and the Mojo comes nowhere near being able to realise those numbers/performance.
 
Last edited:

March Audio

Master Contributor
Manufacturer
Joined
Mar 1, 2016
Messages
6,382
Likes
9,198
Location
Albany Western Australia
We're kind of beating the horse here aren't we. This is just someone with something to sell making shit up.
Well yes, of course its marketing, but its equally somewhat dishonest. It is taking advantage of the technically illiterate. This snake oil part of the HiFi scene that I think most find objectionable.
 
  • Like
Reactions: Tks

Thomas savage

Grand Contributor
The Watchman
Forum Donor
Joined
Feb 24, 2016
Messages
10,206
Likes
15,940
Location
uk, taunton
Or in other terms, standing at ground zero during a nuclear explosion, and being able to hear a dried leaf drift to the ground 1000 miles away at the same instant. Laughable I'd say. (my distance estimate was a guess and probably conservative).
Maybe Rob has some kind of inter-dimensional hearing , so distance is not relevant or not as it seems at least.
 

Blumlein 88

Grand Contributor
Forum Donor
Joined
Feb 23, 2016
Messages
15,876
Likes
26,636
Okay let me try another example. Krakatoa in 1883. Recorded by weather instruments as 172 db at a distance of 100 miles from the erupting exploding volcano. Now at 194 db you cannot get any louder in air.

So it is like being at Krakatoa in 1883 and hearing a dried leaf drift to the ground on the opposite side of the world, say in England. Maybe Ray can convert in his shout-o-meter and gives us exact numbers.

Does this mean Rob has geologic hearing?
 

scott wurcer

Major Contributor
Audio Luminary
Technical Expert
Joined
Apr 24, 2019
Messages
1,503
Likes
2,803
Technically the shock wave from the super heated gas gets higher than that, but you have to have your head right in the caldera.

At 10:02 a.m. on August 27, Krakatoa erupted with a sound that is, to date, considered the loudest sound ever clocking in at 310 decibels. For reference, the sound from the atomic bombs dropped on Hiroshima and Nagasaki were 248 decibels.
 

maxxevv

Major Contributor
Joined
Apr 12, 2018
Messages
1,845
Likes
1,935
The 194dB theoretical limit is sound at sea level without clipping of waveform. It can go louder once you allow compression and clipping of waveform.
 
Top Bottom