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Why aren't "power DACs" more prevalent?

NTomokawa

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By "power DACs" I mean PWM-based amplification throughout the entire signal chain. That is, the complete absence of a traditional DAC and true "digital" amplification.

Is it because such setups are not discernibly (measurably or otherwise) superior to more traditional DAC + analog amplification setups?

Thanks!
 

garbulky

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I've heard one of the power DAC amps. It was all right. Hugely expensive. Nothing stood out to me.
 

wiggum

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By "power DACs" I mean PWM-based amplification throughout the entire signal chain. That is, the complete absence of a traditional DAC and true "digital" amplification.

Is it because such setups are not discernibly (measurably or otherwise) superior to more traditional DAC + analog amplification setups?

Thanks!
It exists. All of Sony's DAPs and Bluetooth speakers use the S-Master amplifier which is exactly what you described.
 
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NTomokawa

NTomokawa

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It exists. All of Sony's DAPs and Bluetooth speakers use the S-Master amplifier which is exactly what you described.
Looks at my Walkman

So that's what S-Master is? The more you know.

Too bad none of the S-Master stuff seem to measure too well.
 

wiggum

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Looks at my Walkman

So that's what S-Master is? The more you know.

Too bad none of the S-Master stuff seem to measure too well.
There are very few objective reviews of Sony's gear. I don't know why. I saw some reviews of Sony DAPs on a Korean website. The IMD was much higher than traditional DACs.
 

Killingbeans

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One reason is that it's very hard (impossible??) to implement negative feedback through a power DAC. The time it takes to process the signal is too great.

That's one nut I'd really like to crack. Not that I have the technical skill or financial flexibility at the moment.. but one can dream.

Maybe even take it one step further and assimilate an SMPS as well. Designing an ampliverter DAC with THD+N in Hypex Ncore territory would be ever so sweet.

But like you said, it might be in the realm of perpetual motion and other pipe dreams.
 

DonH56

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Probably not enough demand for it. There are, or have been, a few power amplifiers with digital inputs.

PWM or class D amplification is not "digital", BTW. A SMPS is not a "digital" power supply, either. It's simply another way of modulating the analog output that achieves high efficiency. D happened to be the next letter after A, B, and C when they came up with class D amplifiers -- back in the 1930's, I think.
 

Killingbeans

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Browsing diyAudio looking for something completely different I stumbled upon a link to this white paper page at: Hypex.

It has a paper on power D/A conversion from 2002.

Notice the quote:

"We've come a long way since believing it made sense to go straight from digital to the power
stage. Have an inside look at the best power DAC ever built and how complicated it got
before the penny dropped."


I guess it's doable, but way to complicated to be economically feasible?
 

HammerSandwich

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While you're on that page, read the "All Amplifiers Are Analogue..." paper, which explains the feedback problem. TLDR: you need to digitize the output signal for comparison with the input.
 

Blumlein 88

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Browsing diyAudio looking for something completely different I stumbled upon a link to this white paper page at: Hypex.

It has a paper on power D/A conversion from 2002.

Notice the quote:

"We've come a long way since believing it made sense to go straight from digital to the power
stage. Have an inside look at the best power DAC ever built and how complicated it got
before the penny dropped."


I guess it's doable, but way to complicated to be economically feasible?

So his conclusion at the end of his AES paper:

The feasibility of direct power conversion of 1-bit deltasigma audio signals has been demonstrated. The initial drawbacks associated with 1-bit power conversion have been successfully overcome and actually turned into an advantage for the new topology when compared to traditional class D power stages.

As I've mentioned I have a couple of the Tact power amps. Digital conversion in the power output stage and they only accept digital inputs. They have some failings in interacting with speakers in the treble. With an appropriate load or some frequency contouring they work very well. As far as I know it is an updated version of this the Lygndorf amps still use. The Tacts had some built in crossover functionality if you wanted to use them in two or three way systems. They seemed to have been overlooked or ahead of their time or something. Mostly poorly marketed.
 

Theo

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PWM or class D amplification is not "digital"
I agree that the letter D was not chosen because of "D"igital. As the speakers are analog by definition, there must be some DA somewhere, any way. So when the final stage of a class D amplifier is driven by a microprocessor computing the Pulse Width from a digital input, is it a FDA? Or is it something different? How does feedback work with class D?
 

DonH56

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I am no expert in class D amplifier design. My career has been (mostly) low-level signals and, except for some DIY stuff many years ago, my class D work has been at RF including delta-sigma D/A converters with high-power (for GHz RF) output stages. I'll take a stab at it but more learned engineers may correct my discussion.

Class D amps usually have an analog input modulated to produce a PWM output. Below is the basic concept (see Wikipedia reference). The input signal is applied to a comparator (C) that switches between high and low depending on if the input is above or below the triangle wave. If it was just a DC (static) 0-V input instead of a triangle wave, you'd just get a square wave at the output at the signal frequency. If the DC (bottom) input was moved up and down, you'd get rectangular waves, still at the same frequency, with with different pulse width. The triangle generator means the pulse width varies with (modulates) the signal and thus you get a PWM output signal. The switching controller ensures the top and bottom output devices do not turn on at the same time -- that would short the +/- power rails with all the problems that implies. That also means there is a small "dead zone" when neither device is on during the switching phase. How much that matters depends upon how fast you switch, how large the dead zone (hysteresis), etc. Usually not a big deal in practice for audio amps now that switching frequencies have gotten well above the audio band.

1549733075399.png

https://en.wikipedia.org/wiki/Class-D_amplifier

This is what happens if we replace the triangle generator with a DC voltage above, at, and below the center point of the input signal (assumed 0 V):
1549729853174.png

The frequency is the same but the pulse width changes (assuming a fixed input frequency).

Feedback can be (and is) applied at several places. You can treat the blocks from the input to the speaker as an analog power amp (it is, technically) and then stick an input buffer (op-amp) in front and take the feedback from the output just like any other amplifier:

1549732886812.png

Making this stable means keeping the delay through the amp low enough and switching frequency high enough that the amp circuitry and output low-pass filter does not add too much phase shift. That was one of the biggest problems with early designs; the switching frequency was fairly low, so the output filter had to roll off just above the audio band, and that caused a lot of phase shift in the audio band itself. Too much phase shift and the feedback is in phase with the input so now it adds instead of subtracts and you have built an oscillator. A very powerful, speaker-eating oscillator. Switching frequencies have gotten much higher so this circuit can work well and is, I suspect, still the main compensation network for most of today's amplifiers.

An interesting approach is to take the feedback from point 2 back to point 1. That does a comparison of the output pulse before the filter to the digital input signal. It will not compensate for nonlinearity (distortion) in the input buffer, comparator, or output filter, but solves the phase lag problem of the output filter. I have only seen that used on low-frequency motor controllers and such where great linearity is not required (and sometimes the motor itself is the output filter).

Adding feedforward compensation is more prevalent today, at least in the very few schematics I have seen (remember this is not my day job). In this scheme some of the input signal is "fed forward" to the output to help bypass the switching stages and output filter. The output buffer is often a low-power class A or AB amplifier that handles not only error correction but also provides most of the output for low-level signals. You have to align the phase of the feed-forward circuit to the main signal path, of course, to add the right amount of feedforward compensation at the right time. Again, I do not know, but suspect many modern class D audio amplifiers are using this sort of approach. Some call it a hybrid design due to the class A/AB output driver.

1549732845342.png


Other schemes put switching output stages in parallel so the effective switching rate is much higher, use more complex generators to create PWM/PFM and other output signals, and so forth. Too complicated for one post.

HTH - Don
 

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Killingbeans

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I wote for Bruno Putzeys as president... of everything :D
 
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wiggum

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I am no expert in class D amplifier design. My career has been (mostly) low-level signals and, except for some DIY stuff many years ago, my class D work has been at RF including delta-sigma D/A converters with high-power (for GHz RF) output stages. I'll take a stab at it but more learned engineers may correct my discussion.

Class D amps usually have an analog input modulated to produce a PWM output. Below is the basic concept (see Wikipedia reference). The input signal is applied to a comparator (C) that switches between high and low depending on if the input is above or below the triangle wave. If it was just a DC (static) 0-V input instead of a triangle wave, you'd just get a square wave at the output at the signal frequency. If the DC (bottom) input was moved up and down, you'd get rectangular waves, still at the same frequency, with with different pulse width. The triangle generator means the pulse width varies with (modulates) the signal and thus you get a PWM output signal. The switching controller ensures the top and bottom output devices do not turn on at the same time -- that would short the +/- power rails with all the problems that implies. That also means there is a small "dead zone" when neither device is on during the switching phase. How much that matters depends upon how fast you switch, how large the dead zone (hysteresis), etc. Usually not a big deal in practice for audio amps now that switching frequencies have gotten well above the audio band.

View attachment 21623
https://en.wikipedia.org/wiki/Class-D_amplifier

This is what happens if we replace the triangle generator with a DC voltage above, at, and below the center point of the input signal (assumed 0 V):
View attachment 21614
The frequency is the same but the pulse width changes (assuming a fixed input frequency).

Feedback can be (and is) applied at several places. You can treat the blocks from the input to the speaker as an analog power amp (it is, technically) and then stick an input buffer (op-amp) in front and take the feedback from the output just like any other amplifier:

View attachment 21622
Making this stable means keeping the delay through the amp low enough and switching frequency high enough that the amp circuitry and output low-pass filter does not add too much phase shift. That was one of the biggest problems with early designs; the switching frequency was fairly low, so the output filter had to roll off just above the audio band, and that caused a lot of phase shift in the audio band itself. Too much phase shift and the feedback is in phase with the input so now it adds instead of subtracts and you have built an oscillator. A very powerful, speaker-eating oscillator. Switching frequencies have gotten much higher so this circuit can work well and is, I suspect, still the main compensation network for most of today's amplifiers.

An interesting approach is to take the feedback from point 2 back to point 1. That does a comparison of the output pulse before the filter to the digital input signal. It will not compensate for nonlinearity (distortion) in the input buffer, comparator, or output filter, but solves the phase lag problem of the output filter. I have only seen that used on low-frequency motor controllers and such where great linearity is not required (and sometimes the motor itself is the output filter).

Adding feedforward compensation is more prevalent today, at least in the very few schematics I have seen (remember this is not my day job). In this scheme some of the input signal is "fed forward" to the output to help bypass the switching stages and output filter. The output buffer is often a low-power class A or AB amplifier that handles not only error correction but also provides most of the output for low-level signals. You have to align the phase of the feed-forward circuit to the main signal path, of course, to add the right amount of feedforward compensation at the right time. Again, I do not know, but suspect many modern class D audio amplifiers are using this sort of approach. Some call it a hybrid design due to the class A/AB output driver.

View attachment 21621

Other schemes put switching output stages in parallel so the effective switching rate is much higher, use more complex generators to create PWM/PFM and other output signals, and so forth. Too complicated for one post.

HTH - Don
OP was asking about PWM throughout not Class D. The input to Class D amplifier is analog, isn't it? In fully digital converters, it is
PCM -> PWM -> low pass filter -> analog output. The only such DAC+amp combo in existence that I am aware of is Sony's S-Master. Apparently, TI and Philips have/had such devices in the past?
 

Willem

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I have a little Ava Maestro 50 amplifers that apparently works in just this way. It is tiny, elegant, and has auto on/off. The latter was the reason I bought it. Power output is about 25 watt ch, enough for a modest bedroom system but not enough to properly drive the Harbeth P3ESRs in my study. I have never seen measurements.
 
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