Bruno Putzeys Talks DACs

photo credit: John Atkinson Stereophile

Eureka! I remembered our conversation at RMAF! Kidding. Philip O'Hanlon was not only smart enough to record Bruno explaining aspects of his approach to DAC design to me, he was kind enough to allow me to share it here.

Bruno's Mola Mola Preamp/DAC

Here's Bruno Putzeys on his Mola Mola DAC:

So we're looking at a discrete DAC although these days I think that you can have very respectable DAC chips. They all still have shortcomings which I felt had to be resolved. For example, one of the major shortcomings you have with a classical DAC chip is that the noise floor is signal dependent. Whereas analog noise is just a steady hiss that's always there at the same level, these DAC chips typically have a noise floor where you have spikes popping up depending on what the signal is.

Also linearity at low and high levels is an important factor for me. I've found that if you want to make a converter where the output signal really sounds the same as the input converter, and having made an A to D converter I can obviously do that test, you really need staggeringly low distortion figures at all these signal levels. That's why for Mola Mola I've decided to base the design on a pulse width modulator (PWM), that I designed in 2004, that has enormously good linearity.

That PWM then drives a string of discrete 1-bit DACs which are also time delayed by a single clock cycle. The clock frequency of the DAC is 100MHz and there are 32 single-bit DACs arranged as a shift register. All of these outputs are then added together electrically and converted into voltage.

The modulators run on DSPs. There are three DSPs in the converter. The first DSP is responsible for upsampling any input signal to 32-bits at 3.125MHz. Then each DAC channel has its own DSP that converts that into PWM with a clock rate of 100MHz and a sample rate 3.125MHz. Also included in that first DSP is a brand new asynchronous sample rate converter or de-jittering algorithm which will pretty much remove any jitter that it might encounter.

We have done tests with different input cables and what have you on the digital side and nothing happened, it always sounded the same, and I think you've heard it yourself—it sounds spectacularly analog.

I mumbled, "Absolutely, yea, yea." Philip O'Hanlon added, "Now what's interesting is Bruno said to me yesterday have you listened to the DAC with a regular CD file. I don't know if we played any for you but before we go over anything else let me do that. Let me play a well recorded piece of Redbook and the interesting thing is that with Bruno's topology he is upsampling this and I would say to you it's the only time I've ever heard a digital file upsampled where it sounds outstanding. I mean everything else makes it sound different but it doesn't make it sound better. That would be my ten cents worth."

Back to Bruno:

Well yes, the upsampling is of course a kind of a standard part of most D to A converters but how you do that upsampling, that is how you treat the...how exactly you work out what the intermediate samples are, there are a number of shortcuts that practically anyone makes which are absolutely deadly to the sound. What you typically find is that the DAC is supposedly perfectly flat but if you zoom in microscopically then you find these tiny ripples, like a hundredth of a decibel, which repeat themselves all over the audio band and which corresponds to a small pre-echo and a small post-echo. Not just the ringing but also the echo. And that completely kills all body in the sound.

The reason why they do this is to save on computing power. Another classicical short cut is making a fairly sharp filter like the classical brick wall filter which is then centered around exactly half the sampling rate which means that you still get some aliasing which is not as evil as you would think but which is just kind of a mathematical gimmick again to reduce the number of cycles and the amount of processing power that you need. But it's not necessarily psychoacoustically optimum. Psychoacoustic optimum actually starts rolling off a bit earlier so I'm actually sort of getting the perfect specifications amiss when we're playing Redbook. It's not flat to 0dB at 20kHz it's already 0.1 of a decibel down at 20kHz which is tiny but it allows you to reduce the amount of ringing on the impulse response. Also as I said a tiny bit of aliasing is not a problem sonically it's not that dramatic so actually allow the transition band to be a bit wider again to make it a bit shorter.

In listening tests I just found that the classical transition band, which is literally 10% of the Nyquist bandwidth, is just too narrow. It's just enough to be audible. Make it a tiny bit wider and immediately the signature of the brick wall disappears.

Me again, "Fascinating. What you said before, thin, I think you described the sound as 'thin'", Bruno interjects, "As lacking in body", "Yea lacking in body, I find that to be the case." (Gosh I really did not have much of value to say, did I?)

"I mean there's, there's a lot of, um, the way I say it," this is me muttering again, "is there's lots of digital out there that sounds like OK digital, then better digital, but it hardly gets beyond sounding like really good digital." (sometimes I get thoughts stuck in my head and I'll use any excuse to get them out).

Bruno responds:

Yea, and ideally of course you want to use a similar design procedure on the recording side but of course we don't have any power over that. But even by just doing it correctly on the replay end, you're already halving the problem. And that really makes a significant difference even with quite garden variety Redbook recordings.
I think I'll bring my digital recorder to every hi-fi show.

COMMENTS
wikeeboy's picture

Thanks for the report Michael, i'm a big fan of Bruno's work.

The technical stuff makes my head spin. Wondering though if this is like the new breed of FPGA DAC's that can be upgraded/changed significantly by a mere firmware update a la PS Audio DS?

Michael Lavorgna's picture
...is unlike the PS Audio DirectStream in that it does not handle D to A conversion in a FPGA. As Bruno describes above, this conversion is processed by 32 single-bit DACs. However, the processing that is done in the DSPs in Bruno's DAC, upsampling, dejittering, etc., can certainly be tweaked, in theory.
DH's picture

I'm really interested in hearing his new Kii line of powered/DSP speakers. Kii claims analog in, digital conversion, analog out (speaker) is totally transparent.

Wavelength's picture

I am a little confused, isn't a PWM actually a 1 bit dac? So how can it feed a DAC? Something in the translation is a little off.

https://en.wikipedia.org/wiki/Pulse-width_modulation

Thanks,
Gordon

Michael Lavorgna's picture
There was no translation - I just typed out what was said.
Michael Lavorgna's picture
Wavelength's picture

Michael,

I would guess (and it's only a guess) that the DAC shown and mentioned is really passive details required for PWM to convert to analog current output which is then summed with the other PWM/DAC channels into a single analog voltage output.

At least that is what I get out of it.

Thanks,
Gordon

Michael Lavorgna's picture
I sent your question along to my contact for Bruno. I hope he has the time to respond.

Your guess is better than mine ;-)

Bruno Putzeys's picture

Hi all,

Pulse Width Modulation is not a piece of hardware but a type of signal. It's a square wave with a constant frequency but a variable mark/space ratio. That is, you can encode information on it by making it spend variable amounts of time in the high and low states. You can recover that information by averaging out the waveform over time, for instance through a low-pass filter. The exact PWM signal in the Mola-Mola DAC has a 3.125MHz pulse frequency with 32 possible pulse widths. Through correct noise shaping that gives you about the same information capacity as a 6 to 8 speed DSD signal, but with much less wideband noise going round.

Anyhow. That signal is effectively a 100 MHz 1-bit signal with long strings of ones and zeros in it. You could simply feed that into a 1-bit DAC and low-pass filter it to get the audio back but I chose to use thirty-two 1-bit DACs, each of which gets the signal with a 10ns delay from the previous one. Parallelling the 32 outputs then results in a running average that smooths away the square wave part of the PWM signal so you get something that's very easy for a low-pass filter to deal with.

PWM is a very close cousin of DSD which does not vary pulse width but instead varies the average number of pulses over time. For that reason DSD is technically called Pulse Density Modulation (DSD is a genericised trademark).

The common goal of PDM and PWM is to create a 1-bit signal of which the low-frequency part of the spectrum is the encoded signal and the high-frequency part is noise so that a low-pass filter is all you need to separate the two. You could say the audio is the payload and the supersonic noise is padding that is so precisely tailored to the signal that the whole package is precisely binary.

Both PDM and PCM are precursors for D/A (or A/D) conversion. Their only merit lies in the fact that the circuitry to turn them into analogue is, on first approximation, fairly trivial. It's mostly a fast switch and a very clean DC voltage source. For admittedly arcane but nevertheless very real reasons, PDM is at a disadvantage compared to PWM. The quality of a PDM D/A conversion is highly sensitive to whether that switch turns off exactly as fast as it turns on and the usual solution (shortening the pulses) increases jitter sensitivity. This is why actual PDM converters went mostly out of fashion by 2000, certainly in the IC world. One or two really good discrete PDM converters have been made since but most are plagued by the very same issues that chipmakers have been trying to avoid. So this is why I'm happy to doff the ideology hat and don the engineering cap instead, separating the delivery format (PCM or DSD) from the conversion format, picking the conversion format I'm most comfortable with (PWM) and digitally converting all incoming data into that one format.

Cheers,

Bruno

Michael Lavorgna's picture
And thank you for sharing your knowledge here. It's very much appreciated.

Cheers.

Venere 2's picture

It's pretty cool that two well known digital audio experts, take the time to question and explain this technical stuff right here on an open forum. A credit to this passion/hobby they both are, by sharing their expertise about the reproduction of music.

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