NEW GaN Amplifier : Dual ACM8816 GaN amplifier chips + AK4118AEQ [YJCO D816]

[Pausi]

(The smart person learns from everything and from everyone)

A smart person learns by learning basic knowledge of audio and electronics before making decisions. But I guess you can’t convince a flat earther with pictures and measurements of the globe.

 

[Svend P]

In Germany, there are internet forums where owners of Chinese amps exchange ideas about what you can do to make them sound even better. Since many Chinese amps have outsourced power supplies, one idea is to replace them. With stronger ones. Exceptionally, the people who have tried it all agree: If you replace the standard power supply with a stronger one, the amp sounds better. That's what everyone writes. I read that and thought to myself: Yeah, if you turn the volume up and really push it, okay. But at normal room volume? No. But I wanted to try it myself. With my new YJCO D816, bought out of curiosity, I have 2 power supplies, both in GAN technology, that I can compare. One has 48 volts 5 Ampere, the other one 48 Volts 10 Ampere.

So all amplifiers sound the same, EXCEPT when you feed them from a 10 A supply? Riiight. You're an easy target

 

[Sir Sanders Zingmore]

It has nothing to do with each other. I just didn't want to answer you because I had asked you to name devices for a price comparison. That would have been really easy. But you didn't.

oh FFS you wanna play this "I asked you first game", are we in kindergarten (also are you sure you actually did ask me first)?

 

[Michael Kohler]

Especially when they can make more money selling 10 A power supplies instead of 5 A ones.

NTK, I’d like to ask you something.
You once tried to explain to me the technology of a full digital amplifier, I only partly understood it, but since then I’ve thought a lot about how it could possibly work.
So in my words:
At the input of such an amplifier, there is a digital signal, which is called Pulse Code Modulation, PCM. The amplifier converts this signal, which isn’t audible as music to us humans, into another digital signal form, Pulse Width Modulation. We can hear that as Music for example, and this PWM signal also has the power to drive speakers directly. After this PWM signal conversion, there’s finally an analog filter, normally in the form of a coil and a capacitor, so that the high clock frequency of the amplifier doesn’t interfere with playback.
Did I roughly get it right?

 

[davmol]

The best thing is, anyone can just try it themselves.
Surely there are enough readers with with an amplifier with an external power supply.

I did give that one a test run myself a while ago, after buying and running the Fosi v3 monos with a single 48V 10A supply and the splitter cable and filter (so 5A each). I then saw the same comments you saw (about improved sound with 10A for each amp) and so I bought another 10A supply, and for me it made zero difference to the sound. I also changed op amps and it made zero difference to the sound. I did the same with the 3e Audio A7 and again heard no difference between 5A and 10A supply that I could hear.

 

[Michael Kohler]

Steinway Lyngdorf doesn't claim a "digital signal" drives a speaker directly.


Bold mine.

​

https://steinwaylyngdorf.com/fully-digital-systems/

From the product description of the new Lyngdorf TDAI 2210.

https://bestofhighend.com/lyngdorf-audio-launches-tdai-2210-digital-streaming-amplifier/

Unlike traditional amplifiers, it eliminates analog conversion until the speaker outputs, using a digital signal to drive speakers directly. This approach, powered by Lyngdorf’s Equibit technology, removes noise, hiss, crosstalk, and distortion, ensuring pristine audio at all volume levels. The innovative volume control adjusts gain in the output stage, preserving signal integrity for a dynamic, uncolored sound experience.

 

[Buckeye Amps]

From the product description of the new Lyngdorf TDAI 2210.

https://bestofhighend.com/lyngdorf-audio-launches-tdai-2210-digital-streaming-amplifier/

Unlike traditional amplifiers, it eliminates analog conversion until the speaker outputs, using a digital signal to drive speakers directly. This approach, powered by Lyngdorf’s Equibit technology, removes noise, hiss, crosstalk, and distortion, ensuring pristine audio at all volume levels. The innovative volume control adjusts gain in the output stage, preserving signal integrity for a dynamic, uncolored sound experience.

You're quoting a review article while he is quoting Lyngdorf directly. Do you ever read what others link to?

Quote from Lyngdorf

As conventionally defined, the Lyngdorf TDAI and Steinway Lyngdorf power amps do not amplify at all. Instead of a conventional D/A converter in the input stage, true digital amplifiers are digital-to-analog converters with sufficient power (current and voltage) to drive loudspeakers directly.

Interesting that they point out digital to analog conversion is still happening, eh?

 

[Michael Kohler]

I did give that one a test run myself a while ago, after buying and running the Fosi v3 monos with a single 48V 10A supply and the splitter cable and filter (so 5A each). I then saw the same comments you saw (about improved sound with 10A for each amp) and so I bought another 10A supply, and for me it made zero difference to the sound. I also changed op amps and it made zero difference to the sound. I did the same with the 3e Audio A7 and again heard no difference between 5A and 10A supply that I could hear.

Thank you! I honestly get happy when someone writes and stays factual, even if they don’t agree with me.

When I swap my 5 A power supply for the 10 A one, music sounds a bit fuller and a little more colorful. It’s a bit like pressing the loudness button on an older amplifier, but of course the effect is much smaller. At completely normal volume. I didn’t want to believe it, but I tried it out of curiosity.
I tried it on 2 devices, on my current YJCO D 816 and years ago on an Aiyima D05.
I would choose the 10 A power supply just for that reason alone, because the price doesn’t really matter. A 48 V 10 A power supply currently costs 40 euros in Germany, the 5 A one 30 euros. Plus, the YJCO D816 could actually make use of the 10 A power supply’s output, even though in reality no one listens that loud.

The current, excellent Class D amplifiers from Purifi and Hypex also have way more power than anyone could ever need. Nobody gets upset about that either.

 

[torgeirs]

input of such an amplifier, there is a digital signal, which is called Pulse Code Modulation, PCM. The amplifier converts this signal, which isn’t audible as music to us humans, into another digital signal form, Pulse Width Modulation

The problem with the PCM to PWM conversion in digital domain is it is not as accurate as doing it as PCM -> DAC -> comparator -> PWM power stage.
The PWM power stage need an analog bi level preamp with very high bandwith to be able to do the switching cleanly. To make it even cleaner the signal from the PWM power stage is fed back to before the comparator. A normal digital two level driver is not good enough and there is no input for the feedback between the PCM and the PWM digital stage.

 

[Michael Kohler]

The problem with the PCM to PWM conversion in digital domain is it is not as accurate as doing it as PCM -> DAC -> comparator -> PWM power stage.
The PWM power stage need an analog bi level preamp with very high bandwith to be able to do the switching cleanly. To make it even cleaner the signal from the PWM power stage is fed back to before the comparator. A normal digital two level driver is not good enough and there is no input for the feedback between the PCM and the PWM digital stage.

Thank you. Honestly. I will read it and think about it so many times until I hopefully understand it.

 

[torgeirs]

So for the Purifi and Hypex and the new ICE modules the analog comparator and PWM stages are very linear in themselves. End then the system has analog feedback that makes it even more linear. The problem with feedback is that the amps has to be 10 times faster than the fastest input signal for each 20 dB of THD and noise reduction. So the feedback is analog and if converted to digital in a whole digital chain, the ADC must be superfast and linear. So it gets to expencive.
Its done in motor control, but the motors don't care that much about THD. 60 dB is good enough for a motor. To get to 120dB the ADC must be 1000 times faster.

 

[torgeirs]

Its like driving a car. You must be able to steer the car faster than the curves bend compared to the speed of your car. If you are slower than the car you crash.
So if you speed up you must turn the wheel faster.
A PWM stage is like a glass of water that is either empty or full. But if the one filling the glass is not accurat it will spill. And that spill is the distortion and noise.
If we did not see our feedback would be slow and we would spill more. As we can se the feedback is much faster and we can stop in the right time. But it dont help to se if we can not control our muscles.
Feedback = see
Amp speed = control muscles
Delay = unconsentartion when pouring

 

[torgeirs]

And in all this the car and table must be stable. The faster the amp the more prone it is to be unstable. Like a formula 1 car or a glass of water on a boat. So great engeneering and cost must be put in to make the F1 car more stable. Dont help to be verstappen if the car is to
unstable.

 

[Michael Kohler]

You're quoting a review article while he is quoting Lyngdorf directly. Do you ever read what others link to?

Quote from Lyngdorf

Interesting that they point out digital to analog conversion is still happening, eh?

The text comes from the official product announcement or the press and informational materials from Lyngdorf Audio about the Lyngdorf TDAI-2210 streaming amplifier.

Why do you keep attacking me anyway? 1.) it really doesn't bother me 2.) it comes back to you. Really, I'm completely sure.

 

[Michael Kohler]

And in all this the car and table must be stable. The faster the amp the more prone it is to be unstable. Like a formula 1 car or a glass of water on a boat. So great engeneering and cost must be put in to make the F1 car more stable. Dont help to be verstappen if the car is to
unstable.

I don't -yet- understand very much of what you're writing. Maybe 40%, now?
But one thing I do understand. You REALLY know your stuff on this topic. My respect.

 

[torgeirs]

Luckily it is a whole science behind this. Control theory helps us every day. But some of it is not that intuitive, Like the speed and feedback thing.

A user manual can be helpful https://www.infineon.com/assets/row...nfineon-ref-audio-ganb-750w-usermanual-en.pdf
FB is feedback.
Qs are PWM power, fastest
IRS are comparator, superfast
IEDN are preamp for powerstage. superfast
And opamps i front for feedback and correction. fast and accurate
As long the amp in front is accurate the circutry after has to be only fast and stable. Because of the feedback
But the amp i front must be much faster than the signal change
And if you put in enough delay in the system it all blows up. (Smoke from the powertransistors)

 

[Buckeye Amps]

The text comes from the official product announcement or the press and informational materials from Lyngdorf Audio about the Lyngdorf TDAI-2210 streaming amplifier.

Why do you keep attacking me anyway? 1.) it really doesn't bother me 2.) it comes back to you. Really, I'm completely sure.

What are you talking about? You linked to an article written by a writer for Best of High End Audio detailing the product on display at Munich. Sure, it may contain wording from an official Lyngdorf press release, but it is still a second-hand source reporting it. JustJones linked to the direct product description from Lyngdorf.
It would be like me linking to the actual science publication vs. someone linking to a news article that paraphrases the original article.

I guess it comes off as attacking because you seem allergic or avoidant to anything that doesn't conform to your already self-confirmed bias.
I am starting to feel dumb that I have been trolled on a high level for nearly six pages. Shame on me

 

[NTK]

NTK, I’d like to ask you something.
You once tried to explain to me the technology of a full digital amplifier, I only partly understood it, but since then I’ve thought a lot about how it could possibly work.
So in my words:
At the input of such an amplifier, there is a digital signal, which is called Pulse Code Modulation, PCM. The amplifier converts this signal, which isn’t audible as music to us humans, into another digital signal form, Pulse Width Modulation. We can hear that as Music for example, and this PWM signal also has the power to drive speakers directly. After this PWM signal conversion, there’s finally an analog filter, normally in the form of a coil and a capacitor, so that the high clock frequency of the amplifier doesn’t interfere with playback.
Did I roughly get it right?

Yes. That's basically it. I'll give a little more detail.

This is the usual explanation for beginners on how PWM is generated. The top figure shows the signal we want to amplify. There is also high frequency triangular wave (sawtooth will work to). The frequency of the triangular wave is usually about ≥20x our highest signal frequency.

These 2 signals are fed into a comparator. The output of the comparator is shown in the middle figure. The pulses are coded as +1 when our input signal is greater than the triangular wave, and -1 when our input signal is less than the triangular wave. That is our PWM waveform. (Another method to generate a similar, but not the same, waveform is to use the sigma-delta modulator. The waveform is more correctly known as PDM instead of PWM.) There is no sampling or numerical computations involved in generating the PWM and the process can be done using analog circuits.

The pulses can now be used to drive power switching transistors to switch voltages and currents at much higher levels, and output much higher power pulses. The frequency spectrum of this PWM is shown in the bottom figure. We have out original signal, along with a lot of other byproducts that we don't want. However, the levels of those byproducts are high only when they are at around the switching frequency or higher. Therefore we can recover our desired amplified signal by filtering out the high frequency components of this high power PWM. Typical we use a cutoff frequency of about 1/10 the triangular wave frequency for this low pass filter.

For "digital" amplifiers, the PWM or PDM waveform is synthesized in the digital domain instead of using the above method.

 

[shuppatsu]

It’s possible that Michael would hear a difference in a double blind volume matched ABX test, because those “digital” amps have high output impedance. I think the differences would be subtle in the extreme with the standard 4 and 8 ohm test loads, but it may be pronounced enough to cause noticeable frequency response deviations under the complex load presented by some speakers.

Of course that’s a flaw with this particular class of amplifier, not proof that it’s better. But depending on the listener’s speakers, room, and disposition, he might actually prefer it.

Anyway, thanks @NTK, you have the patience of a saint, and I and many of others have benefitted from your clear explanations.

But I have a question: Are measurements everything or nothing?

 

[Michael Kohler]

Yes. That's basically it. I'll give a little more detail.

This is the usual explanation for beginners on how PWM is generated. The top figure shows the signal we want to amplify. There is also high frequency triangular wave (sawtooth will work to). The frequency of the triangular wave is usually about ≥20x our highest signal frequency.

These 2 signals are fed into a comparator. The output of the comparator is shown in the middle figure. The pulses are coded as +1 when our input signal is greater than the triangular wave, and -1 when our input signal is less than the triangular wave. That is our PWM waveform. (Another method to generate a similar, but not the same, waveform is to use the sigma-delta modulator. The waveform is more correctly known as PDM instead of PWM.) There is no sampling or numerical computations involved in generating the PWM and the process can be done using analog circuits.

The pulses can now be used to drive power switching transistors to switch voltages and currents at much higher levels, and output much higher power pulses. The frequency spectrum of this PWM is shown in the bottom figure. We have out original signal, along with a lot of other byproducts that we don't want. However, the levels of those byproducts are high only when they are at around the switching frequency or higher. Therefore we can recover our desired amplified signal by filtering out the high frequency components of this high power PWM. Typical we use a cutoff frequency of about 1/10 the triangular wave frequency for this low pass filter.

View attachment 558395

For "digital" amplifiers, the PWM or PDM waveform is synthesized in the digital domain instead of using the above method.

Dankeschön! (in English,Thank you!) First of all, I’m happy with myself because I’ve properly understood the way a “Full Digital Amp” works.
Then I’m also happy because others read this too and learn something from it. I think only very few people here have knowledge like “NTK” or “torgeirs.”
And now I’m going to read ntk’s post over and over and think about it until hopefully I understand it.
At this point, a special thanks also to “torgeirs.” You guys are among the good ones!