Why so few direct digital amps?

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I find that thinking of Class D as if it were a single-ended triode (SET) amp helps me get better sound.

With an SET amp, 50% of the sound quality is the PSU, because there is very little PSRR.
30% of the sound quality is the output transformer. In Class D, this is the output filter...

Feedback can improve the PSRR, yes. However not many SET amps like to use feedback...

And I don't see that digital volume control is a disadvantage. By the way, you could adjust volume at the speaker with a speaker-level autoformer volume control.

Or move your chair!
 
The TAS chips do high order noise shaping in order to achieve sufficient bandwidth/dynamic range, but they run open loop - the power stage needs a very low power supply impedance in order to achieve the audio performance that they give in the datasheet. If the power supply voltage is wandering around with the audio envelope, the THD/IMD numbers go through the roof.

Core Audio Technology claims that what makes their Kratos amp sound so good and separates it from other direct digital amps is the power supplies it uses. They make linear power supplies which are used in their amps. The Kratos III is a direct digital amp marketed to the high end audio market (in other words, it's expensive.)
 
I use the DDX320 in a tri-amped setup and I'm very happy with it. I'm preparing a new project for a digital PWM amplifier based on the TI processors.
Take a look at the TAS56xx amps. Those are closed loop amps, combined with the PWM modulator (TAS5558) you get power-supply-volume-control (PSVC), which promises to improve dynamic range at lower power level. You also get some very nice features with this, like a DAP (Digital Audio Processor).

I cannot comment on sound quality. Just try it out. They have evaluation boards.
 
On the surface (read, to a novice) the benefits of direct digital amplification over class D seem obvious. By direct digital amplification I mean an amplifier that takes a digital signal (SPDIF/PCM - a square wave) converts it to PWM (another square wave), amplifies it, and converts it to analog at the output. This as opposed to a Class D amp which takes an analog signal, converts it to PWM, amplifies it, and converts back to analog at the output. (Yes I know this is an over simplification, but I hope it will do for what I want to ask.) The obvious point being that the direct digital amp eliminates the DAC, which has got to be a huge advantage, especially considering how much people will invest in a good quality DAC.

Now, there are a handful of direct digital amps out there. Some very inexpensive and some very expensive, and I know of one (and only one) amp module available to the DIYer. This is opposed by a huge number of class D amps available, both as commercial amplifiers and DIY modules. So, my question is, why are there so few direct digital amps? It doesn't make sense to me, given what seems to me to be the obvious advantage that they possess. I am sure there must be some technical reason why the direct digital technology has not been pursued any more than it has been. But what? I would love to hear people's take on this.

Interesting question. Maybe Wadia's PowerDac is such an approach, but I don't know exactly.
http://www.diyaudio.com/forums/class-d/108571-power-dac-like-wadia-full-digital-amp.html
 
I've just been playing around with power supplies on the I.AM.D V200 amplifier and so far, the best sound is with a Connexelectronic SMPS300R followed by a Lundahl LL1694 choke in serial connection. Very happy. Love CLC supplies!

Interesting. I know nothing about switching supplies in general, or the 300R in particular, but wouldn't have guessed you could just hang a choke and more caps on the output. Did you do a sim first, or just go for it? It's an intriguing possibility.
 
On the surface (read, to a novice) the benefits of direct digital amplification over class D seem obvious. By direct digital amplification I mean an amplifier that takes a digital signal (SPDIF/PCM - a square wave) converts it to PWM (another square wave), amplifies it, and converts it to analog at the output. This as opposed to a Class D amp which takes an analog signal, converts it to PWM, amplifies it, and converts back to analog at the output. (Yes I know this is an over simplification, but I hope it will do for what I want to ask.) The obvious point being that the direct digital amp eliminates the DAC, which has got to be a huge advantage, especially considering how much people will invest in a good quality DAC.

Now, there are a handful of direct digital amps out there. Some very inexpensive and some very expensive, and I know of one (and only one) amp module available to the DIYer. This is opposed by a huge number of class D amps available, both as commercial amplifiers and DIY modules. So, my question is, why are there so few direct digital amps? It doesn't make sense to me, given what seems to me to be the obvious advantage that they possess. I am sure there must be some technical reason why the direct digital technology has not been pursued any more than it has been. But what? I would love to hear people's take on this.

I think there will be a lot of those Digital PCM-PWM amps in automotive, HT and professional gear. The small form factor, simple design and ultra high efficiency results in very small amps with lots of functionality that can be build into the loudspeaker, do digital crossover and EQ and even run from batteries and made wireless.
I guess the high end audio industry will be very reluctant to implement this technology. Small and cheap amps, no DAC's needed, no preamp, no fancy cables.

The only pcb's I know of are from HiFiMeDIY, Cirrus Logic and Texas Instruments.
 
Just remember that all the closed loop direct digital amp, have analog feedback. It's only the open loop direct digital amps, that are something special - but they all require really good power supplies.

When that is said, I do like the sound from direct digital amps in the lower cost segment, but the only one I can truely say is superior is the Tact Millenium, and that one is not cheap. But, Panasonic XR-55 do a pretty good job with spdif input - just don't push it. It's horrible with low impedance, and when cranking the volume to max, everything sound bad.
 
Interesting. I know nothing about switching supplies in general, or the 300R in particular, but wouldn't have guessed you could just hang a choke and more caps on the output. Did you do a sim first, or just go for it? It's an intriguing possibility.

Just tried it, no simulation.
Got even better sound now with a transformer followed by LCLC, simulated it in PSUD. Much better than a regulated supply, unfortunately expensive and heavy ��
Even playing around with snubbers on the power supply makes a noticeable difference to the sound.
Cheers,
Mike
 
Digital and class D amplifiers are both fruitful generators of noise. In class D this is
attenuated by the output low pass filter.
Can anyone devise a class D differential output stage that would further reduce audio spectrum noise by common mode rejection.
This would also allow less negative feedback to be applied. When we achieve the noise
levels of the best analogue amps, then they will rule in our increasingly digital world.
 
A while ago we designed such a "direct digital amp" (electronic) for a customer using I2S audio data interface. System includes one board with TI's PWM modulator DSP and up to 8 power stage boards - this gives flexibility to use any number of channels (up to 8) for stereo/multichannel/bridged mono/active loudspeakers... with one interface board.

Power stages from TI are available with different ratings and due to the modularity it is easy to modify for high-end or multichannel, for small form factor, high or low power or for any other specific request. Hand wiring the demodulators or custom made demodulator inductors or industrial coils offer a huge range of flexibility there.

Audio relevant electronic was galvanically isolated (also I2C) and we used a beagleboard black to control the unit through simple software (which gives a lot of flexibility and can be replaced by whatever you want).

System audio interface accepts I2S which can be the output of any TOSLINK/Spdif/DAC converter (we never used the BBB's real time PRUS to generate the I2S but this potentially allows a perfect I2S output, externally clocked)

If there is interest to set up such a family of electronic boards for the DIY market post here. The DSP has a lot of SMD so we need a minimum quantity to produce it with little cost, the power stages can be small boards with SMD components populated and the - audio relevant - demodulator can be done by everyone's own flavor.
 
TADIGITAL I would be very interested in a unit or building blocks that had toslink in and say 50 really hi-end watts out. If it also included say 12 biquads + delay it can serve both as a driver for a conventional speaker with speaker and from comp. (think Devialet) or as a driver for one speaker/way in a multi-way system.

The "unit" also needs a control interface and in the situation where one would use say 6 of them to drive a 2 channel 3-way system, a controller. I would use the biquads to do filter and eq per driver. One could use 2 for a traditional set of stereo speakers. Toslink distribute one and the same L+R signal to all units so the unit need to be configured to be L or R.

I want to use opto in 2016 and get the analogue signal paths as short as absolute possible (this is the central concept here) so only one channel per unit. Each unit shall carry it's own 230 mains input power supply. If this unit can fit into a 20x10x5 cm stainless box and cost less than 100€ it would be good 🙂 I would build myself 8 of them for my 4-way open baffle system + the controller/opto distributor.

Why not use TI stuff only?

Is this possible? Time sync issues? System level control (aka volume).

Anyone else interested in such a "building block"?

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Hi TNT,

>I would be very interested in a unit or building blocks that
>had toslink in and say 50 really hi-end watts out. If it also
>included say 12 biquads + delay it can serve both as a driver
>for a conventional speaker with speaker and from comp.
>(think Devialet) or as a driver for one speaker/way in a
>multi-way system.
The proposed TI components provides biquads (7/ch) and limited delay. What you mean with "speaker with speaker and from comp."?

>The "unit" also needs a control interface and in the situation
>where one would use say 6 of them to drive a 2 channel
>3-way system, a controller. I would use the biquads to do
>filter and eq per driver. One could use 2 for a traditional set
>of stereo speakers. Toslink distribute one and the same L+R
>signal to all units so the unit need to be configured to be L or
>R.
Right. Use one pwm generator with two 50W amp modules and you get a traditional stereo. Or use 6 25W modules and generate two 3-way speakers without crossover network. Or use two pwm generators with 3 300W modules for two loudspeakers eliminating any analog wiring.
Or ..

>I want to use opto in 2016
what kind of opto? TOSLINK?

>and get the analogue signal paths as short as absolute
>possible (this is the central concept here) so only one
>channel per unit.
There is no analog wiring at all anymore. Not even an analog signal - if you want the audio power supply is the analog input to the speakers, switched on and off extemely fast and according to the digital signal.

>Each unit shall carry it's own 230 mains input power supply.
Any particular in mind? You may want to add voltage control for hardware volume setting.

>If this unit can fit into a 20x10x5 cm stainless box
stainless box is something to spend some thoughts to it. Reason is that the demodulator coils may need space - but this depends a lot on the signal quality requirments.

>and cost less than 100€ it would be good 🙂
Amplifier itself is far less than 100€, again depending on quality requirements, complete system depends on the amount of building blocks!

>I would build myself 8 of them for my 4-way open baffle
>system + the controller/opto distributor.

>Why not use TI stuff only?
We made not too bad experience

>Is this possible? Time sync issues? System level control (aka
>volume).
All controls are done through software, max. volume may also be controlled by power supply voltage. TI offers an incredible amount of software volume control without loss of information as they expand the digital signal before creating the PWM.

>
>Anyone else interested in such a "building block"?

>//[/QUOTE]
Your pic looks like you want to connect to the loudspeaker itself? If so, you may be able to use the loudspeaker coil itself as demodulator coil - saving lots of space in the amplifier
 
What you mean with "speaker with speaker and from comp."?

I wonder myself 🙂 ... they unit could be used to drive a full range speaker including compensation/eq or used to power individual drivers (in this case also with filter) in a multi-way speaker.

Your pic looks like you want to connect to the loudspeaker itself? If so, you may be able to use the loudspeaker coil itself as demodulator coil - saving lots of space in the amplifier[/QUOTE]

As I wrote, shortening the analogue signals paths to a minimum is the goal with the topology. So yes, this means that the unit ends up at the terminals of the speaker which in turn means that you need 2 for a stereo set. Each unit only do 1 channel.

Looking at your comments I think I was not able to convey my idea completely correct - hope the below picture clarifies my intention.

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