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Axign/AX5689.... Fresh From the Bench: Improving GaN with Digital Control

Thank you @pma, the following is the suggested layout of the Sabaj A30a, for comparison. The A30a uses STA516BE(Mosfet)/PFFB where the GFB/analog appears to feed directly to the AX5689 chip, unlike the the Axign Class-D Audio Amplifier PWM Controller Demonstration Board which uses AX5689S/LMG1210/GaN/PFFB and the GFB/analog is feed directly to the ADCs.

1756535883589.png


This is an earlyer article, Amplifier Series: Axign of Things to Come (2018).... Axign's new AX5689 chip lays a very high-speed feedback loop around a Class-D amplifier, transforming its performance to almost theoretical levels.

What are your thoughts of the differences?
 
Thank you @pma, the following is the suggested layout of the Sabaj A30a, for comparison. The A30a uses STA516BE(Mosfet)/PFFB where the GFB/analog appears to feed directly to the AX5689 chip, unlike the the Axign Class-D Audio Amplifier PWM Controller Demonstration Board which uses AX5689S/LMG1210/GaN/PFFB and the GFB/analog is feed directly to the ADCs.
I think this might just be incorrect. None of the schematics on the page show external ADCs. They are in the chip, and highly specialized for the task.

Interestingly, AX5689S, is nowhere to be found on the internet, except for the AudioXpress page.
 
AX5689 General features:

- 4-channel BTL digital Class-D amplifier
controller with digital inputs
- 8 differential low-latency ADCs with a 115 dB
dynamic range
- 8 programmable digital loop-filter slices
- 8 CMOS level PWM outputs, configurable as
4 BTL channels
- Feedback loop possible after the output filter,
across the loudspeaker nodes
- Configurable interconnections between slices
and ADCs for versatility and MIMO control
- Volume control and soft mute
- Dynamic loop control with programmable
ramp enabling pop-free mode transitions

Technical limitations, as per datasheets:

1756540255916.png

1756540289055.png



These are parameters as the technical status of now.

P.S.: And sorry I am an engineer, manufacturer's semi-marketing pseudo-technical info (like that of Sabaj below) is not enough to me. They even did not write the part number properly.

1756540377334.png


 
I think this might just be incorrect. None of the schematics on the page show external ADCs. They are in the chip, and highly specialized for the task.

Interestingly, AX5689S, is nowhere to be found on the internet, except for the AudioXpress page.

1756542036851.png
 
All I see is a square.. I think that’s the buffer they talk about. It doesn’t make sense to have the ADCs externally. The special ADCs in the chip are a critical part of the whole thing.
Thank you, yes (that is very possible/likely), especially as the article does not name the ADCs as a chip/external chip?
This diagram also appears to indicate this likelyhood, doesn't it?

1756543511594.webp
 
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Thank you, yes (that is very possible/likely), especially as the article does not name the ADCs as a chip/external chip?
This diagram also appears to indicate this likelyhood, doesn't it?

View attachment 473079
It's not "very possible/likely", the system just can't work without a special ADC that has high sampling speed, low holdup time and is very accurate. These simply do not exist as stand alone chips.
 
It's not "very possible/likely", the system just can't work without a special ADC that has high sampling speed, low holdup time and is very accurate. These simply do not exist as stand alone chips.
Thank you, by stand alone, do you mean external
 
AX5869:

AX5869C:

This is where to start the thoughts.

Thank you @pma, yes, effectively the AX5689/AX5689C can be used as an Analogue/Digital PWM/Preamp and as an Analogue/Digital Integrated Amp with PFFB, which the Axign Class-D Audio Amplifier PWM Controller Demonstration Board demonstrates, doesn't it?

@KL.... , ADC is a part of the AX5689 chip, as clearly stated in the datasheet. No need to try to find different explanations or own theories.

Thank you @pma, yes no need as you have suggested, although posts asked and thought it best that they reach/come to same conclusion (that you have posted and the datasheet clearly states).... apologys, hope that is ok :=)


A little more information.
The AX5689C
  • is a 4-channel bridge-tied load (BTL, 8 CMOS-Level Pulse-Width Modulation (PWM) Outputs, Configurable as 4 BTL Channels) audio amplifier controller IC with digital inputs and CMOS-level pulse-width modulation (PWM) outputs. The device enables high-order digital control loops with feedback for Class-D output filters.
  • Embedded low-latency analog-to-digital converters (ADCs, 8 Differential Low-Latency Analog-to-Digital Converters (ADCs) with a 115dB Dynamic Range) are used to close the loop behind the output filter in the digital domain (Feedback (FB) Loop Possible after the Output Filter, across the Loudspeaker Nodes). The digital control loop has a high loop gain for all audio frequencies.
  • The AX5689C suppresses all errors caused by the power supply, power stage, and output filter within the audio band (20Hz to 20kHz). It can achieve superb performance levels while enabling lower-cost options.
  • The low-latency ADCs, PWM outputs, and configurable interconnects (configureable Interconnections Between Slices and ADCs for Versatility and Multiple-Input/Multiple-Output (MIMO) Control) allow for custom configurations, including analog audio inputs, analog line outputs, power supply control, current-sensing, temperature-sensing, and more. Volume Control and Soft Mute. Dynamic Loop Control with Configurable Ramp Enabling Pop-Free Mode Transitions
  • The AX5688C is compatible with power stages from various vendors, and can read and respond to power stage diagnostic signals. The maximum output power (POUT) is dependent on the user application and is scalable based on the number of channels, selected power stages, loads, and supply.
The AX5689
  • is a 4-channel BTL (8 CMOS level PWM outputs, configurable as 4 BTL channels) audio amplifier controller IC with digital inputs and CMOS level PWM outputs. It enables high order digital control loops (Control interface.... SPI, fast mode I2C control interface with selectable address for multi-chip systems), with feedback after Class-D output filters. Serial audio input/output.... I2S/TDM Serial Audio Interface with 16 downstream and 8 upstream channels, Input sample rate: 32 to 768 kHz, 16 to 32-bit supported audio formats.
  • Embedded low-latency ADCs (8 differential low-latency ADCs with a 115 dB dynamic range) are used to close the loop behind the output filter (8 programmable digital loop-filter slices) in the digital domain. The digital control loop has a high loop gain for all audio frequencies.
  • The AX5689 suppresses all errors caused by the power supply, power stage and output filter within the audio band (20 Hz - 20 kHz). The AX5689 reaches superb performance levels while enabling cost down options.
  • The low-latency ADCs, PWM outputs and programmable interconnects (Feedback loop possible after the output filter,across the loudspeaker nodes) allow custom configurations (Configurable interconnections between slices and ADCs for versatility and MIMO control, Volume control and soft mute, Dynamic loop control with programmable ramp enabling pop-free mode transitions), like analog audio inputs/analog line outputs, power supply control, current sensing, temperature sensing, etc.
  • The AX5689 is compatible with power stages from various vendors and can read and respond to power stage diagnostic signals. The maximum output power is dependent on the user application and is scalable with the number of channels, selected power stages, loads and supply.
 
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A suggestion to consider the following measurements, PR+TI (Power Rating + Transparentcy/Impedance), PR+TC (Power Rating + Transparentcy/Capacitance)….
  • Transparentcy/Transparency…. it can be suggested that Transparentcy means NO Feedback, NO Distortion, NO Noise, that is reasonable, isn’t it? There do not seem to be any Amplifyers that would meet this Definition/Requirement. ASR indicates that THD+N of -95db and/or better, is sufficient/enough to be considered as Transparent, doesn’t it? External aspects such as Speaker+Cable Load Impedance/Capacitance also affect Amplifyer Transparentcy, don’t they? Also as Power/Current increases, THD and Noise tend to decrease/increase, don’t they?
  • The Axign Class-D Audio Amplifier PWM Controller Demonstration Board, currently, falls a little short, 92 SINAD, of the -95db/THD+N Transparentcy Threshold, doesn’t it? It also suffers, quite badly, with the utilised 2uF Speaker Capacitance test regardless of Load Impedance, which even though (considered) extreme is (apparently) commonly exhibited by Electrostatic Speakers, which suggests is not extreme and needs to be considered. It does pass the 2~10R/ohm Speaker Impedance test, easyly/invariant/without dependence.
    • PR+TI, what would it mean? It could be suggested that the measurement could be simply a Power Rateing, for 2~10R/ohm Speaker Impedance and 20~20Khz Frequency Range, where the Transparentcy is a THD+N of -95db and/or better. For the Axign Class-D Audio Amplifier PWM Controller Demonstration Board the PR+TI would be 0, because it does not meet the Transparentcy measurement of THD+N of -95db and/or better.
    • PR+TC, what would it mean? It could be suggested that the measurement could be simply a Power Rateing, for 2uF Speaker Capacitance and 20~20Khz Frequency Range, where the Transparentcy is a THD+N of -95db and/or better. For the Axign Class-D Audio Amplifier PWM Controller Demonstration Board the PR+TC would be 0, because it does not meet the Transparentcy measurement of THD+N of -95db and/or better.
  • To be concise (and precise), First let us consider Power Rateing + Transparentcy Impedance/Capacitance.
    • For convenience, let us say that its THD+N of -96db, for 2~10R/ohm Speaker Impedance and 20~20Khz Frequency Range, is 20 Watts then its PR+TI would be 20.
    • For convenience, let us say that its THD+N of -96db, for 2uF Speaker Capacitance and 20~20Khz Frequency Range, is 20 Watts then its PR+TC would be 20.
    • Note that the connecting Cables Impedance/Capacitance, also needs to be considered.
    • How would this be Helpful? Let us look at the questions/statements commonly posted….
      • What Speakers to go with my Amplifyer?, What Amplifyer to go with my Speakers?
      • You won’t hear any difference because the Amplifyer is Transparent?
  • For 20~20Khz Frequency Range, if we know that an Amplifyers PR+TI = 20 and PR+TC = 20, Speaker Impedance/Capacitance is 2~10R/ohms and <=2uF, then, knowing that 1 Watt is more than sufficient for 110db/SPL Speakers, Extrapolate as follows for an Amplifyer with PR+TI = 20 and PR+TC = 20….
    • 110db/SPL Speakers, Amplifyer with PR+TI = 20 and PR+TC = 20, Transparent/1Watt required
    • 107db/SPL Speakers, Amplifyer with PR+TI = 20 and PR+TC = 20, Transparent/2Watt required
    • 104db/SPL Speakers, Amplifyer with PR+TI = 20 and PR+TC = 20, Transparent/4Watt required
    • 101db/SPL Speakers, Amplifyer with PR+TI = 20 and PR+TC = 20, Transparent/8Watt required
    • 98db/SPL Speakers, Amplifyer with PR+TI = 20 and PR+TC = 20, Transparent/16Watt required
    • 95db/SPL Speakers, Amplifyer with PR+TI = 20 and PR+TC = 20, Nolonger Transparent/32Watt required which means another Transparent Amplifyer would need to be chosen, for the chosen Speakers.
  • The above is for consideration and discussion….
Back to the Axign Class-D Audio Amplifier PWM Controller Demonstration Board, which cannot be considered as Transparent for/via the abovementioned, and Test Report….
  • Test Report summarised….
    • Unlike conventional Class-D controllers, the inputs are natively digital and analogue/AX5689 ADCs, where nothing within the AX5689 chip is analog until the (PWM/Analogue) output. The chip’s input is addressed via the usual I2S protocol. The open loop gain versus frequency is also implemented digitally, allowing a high open loop gain, with a fifth-order roll-off. This configuration allows the distortion to be consistent across the audio band, but also leads to high suppression of the carrier.
    • PFFB Analogue/Feedback is converted by the AX5689 ADCs to digital feedback and processed by the AX5689 chip/digitally. The ADC is required to have high bandwidth and low latency for stability. ΔΣ ADCs can have issues with high quantization noise. Digitally filtering it introduces latency and phase shift, which complicate stability. The PFFB allows compensation of the PWM/Carrier filtering and greatly reduced noise in the audio band. In the implementation tested here, the sampling is done every 20ns, which is more than fast enough to achieve a stable control loop well beyond the audible range. The AX5689/ADCs are doubled up to get a 3dB increase in signal-to-noise. Surprisingly, there is no dissipation penalty for the ADCs which reduces their power use to 15mW each.
    • The PFFB/GFB processing more effectively reduces output/LC filters nonlinearities (Multiphase/EMI/etc).
    • GaN really shines with low resistance (and thus low I2R heating) to drive very low impedance Speaker loads.
    • The AX5689 controller chips could be used with other output transistors (e.g., silicon MOSFETs, SiC, other GaN devices) just as easily. Although details of this particular board may vary a bit (Sabaj A30a) for future designs, the basic concept has a lot of flexibility in application.
    • The demo board requires a PS of 50V nominal but the Test utilised a (relatively) inexpensive 48V supply rated at 20A.
    • Fig3 noise spectrum shows noise-shaping, which removes any issues of audible hiss, even with extremely sensitive speakers. The low 1/f corner is icing on the cake.
    • Fig4 shows changing the Load from 10R to 2R caused NO change in the output voltage! This was true at several spot frequencies (50Hz, 200Hz, 1kHz, 5kHz, 15kHz) and is a remarkable result, predicting that the amp’s sound will remain controlled/uncolored even using speakers with wildly swinging impedance.
    • Fig5/6 shows 1% THD point at about 140W/8R and 280W/4R. Note that the curves look almost identical but scaled for power by the load resistances.
    • Fig7 shows the THD for a 4R load with 220W!
    • Fig8 shows 10V signals at 10kHz and 11kHz across an 8R load where the largest error is a sideband at -96dB which is quite an excellent result.
    • Fig9/10 shows distortion versus frequency at 4V out for 8/4R, and Load 8R paralleled with the 2µF. With this capacitive load, the rise in Noise dominates and the result is less good/terrible.
    • Listen…. Swapping the Axign board back and forth with the Nilai DIY amps currently in my system, the sound remained unchanged — a high compliment (note a (repeated/repeatable) subjective conclusion)!
    • Conclusion…. the AX5689/Demo Board reference design delivers good distortion performance, very low noise, excellent carrier suppression, exceptionally low source impedance and drive capability. It took major effort to make the amplifier misbehave, all while running absolutely cool with no external heatsinking!
  • Is there a full/complete circuit diagram of the Axign Class-D Audio Amplifier PWM Controller Demonstration Board?.... please post.
  • Can the Axign Class-D Audio Amplifier PWM Controller Demonstration Board/Integrated Amplifyer be improved to achieve a better THD+N? Let us know?
 
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Until a circuit diagram, of the Axign Class-D Audio Amplifier PWM Controller Demonstration Board, can be located/posted, to discuss....

A suggestion, for discussion....
Gm Metric (Geddes and Lee Metric)…. Article/Paper No1, Auditory Perception of Nonlinear Distortion - Theory
Gm Metric (Geddes and Lee Metric)/Article/Paper No1.pdf

Historically…. distortion has been measured using specific signals sent through a system and quantified by the degree to which the signal is modified by the system. The human hearing system has not been taken into account in these metrics. Combining nonlinear systems theory with the theory of hearing, a new paradigm for quantifying
distortion is proposed.

Audio has viewed distortion metrics in the context of the nonlinear systems response to a sinusoid or sometimes, two or more sinusoids - basically a signal based metric. The current metrics that are used for distortion are THD, IM/IMD, Multi-tone inter-modulation, etc, which are all usually expressed as a percentage or number (the ratio of the distortion by-products to the total system output. In an absolute sense this view of distortion is satisfactory. If our goal were to eliminate all distortion then clearly any measure of its level is adequate. It is neither reasonable nor desirable to set as our goal the complete elimination of all distortion. From a cost effective standpoint, reducing distortion below perceivable levels is a complete waste of time and money. It may also be that we might want a scale by which to compare two levels of distortion in order to make tradeoff decisions. In this context, we will show that the signal-based metrics fall far short of the mark, for they fail to correlate with, or even consider, subjective impression. Therefore, it seems intuitively obvious to question the belief that a distortion measurement which is based purely on the mathematical difference between specific input and output waveforms, without any regard for the human hearing system, would yield a reliable metric? We think not!, but the need for a reliable distortion metric is obvious!

To be useful the metric must be consistent and reliable. The same number must mean the same thing in every context and there must be a close correlation between the metric and the response that it is intended to scale which is where the signal-based distortion metrics fail. For example, 01% THD of one type of nonlinear system can be perceived as unacceptable, while 10% THD in another type of nonlinear system is perceived as inaudible. Even one of these simple examples is sufficient to invalidate THD as a viable/relyable metric for discussion of the perception of distortion. Furthermore, 1% THD is not at all the same as 1% IM, but this article we will show that neither correlates with subjective perception, thus falling short of what of what is necessary/required.

Hearing Mechanism/Ears…. is not linear and taken as a System is also quite complex. The attribute of hearing that will overwhelmingly dominate the perception of distortion is masking. Masking is also the principal effect used in the creation of all modern techniques of perceptual coders (MP3, AAC, etc.). Masking has no analog in linear systems theory, and it is not very intuitive since it does not occur in systems other than the ear. Masking is predominately upward toward higher frequencies although masking does occur in both directions. The masking effect increases, masking occurs further away from the masker, at a substantial rate with excitation level.
  • The masking effect of the human ear will tend to make higher order nonlinearitys more audible than lower order ones.
  • Nonlinear by-products that increase with level can be completely masked if the order of the nonlinearity is low.
  • Nonlinearitys that occur at low signal levels will be more audible than those that occur at higher signal levels.
Metric…. is a value (or it could be a function or multi-valued but a single value is usually desirable) which is given as an attribute of a relationship to indicate its scaling in some predefined context. For instance, temperature can be a metric in the context of human perception heat content. We can describe the perception of temperature in words like hot, warm, cool, or cold and since temperature also has an exact scientific scaling (conveniently), it is a simple matter to map from the subjective metric to the physical one. We must always remember however, that the subjective terms are relative and a precise mapping is often difficult to obtain. Whenever human perception is involved, metrics can only ever be statistically relevant.

Background…. a system that does not pass a signal through to its output that is indistinguishable from the input signal is said to distort it. Some signal modifications are desirable, like equalization and frequency response changes to improve the sound quality, but some Distortion is undesirable, like most Speaker resonances or the Nonlinear distortion of a signal. A system is said to be nonlinear if its input and output are not linearly related in a mathematical sense. Such systems do not obey the principle of superposition and can have frequency responses that are signal dependent. In fact, even the concept of frequency response is a linear one and its application to a nonlinear system must be done with care.

A single tone input to such a system does not produce a single tone at its output, but a multiplicity of tones. Nonlinear systems cannot be analysed with classical linear systems theory and as such they pose a significant impediment to systems analysis. Even though, in a good audio system linearity is usually one of the design goals, there are very real situations where one might want the system to be nonlinear, but and nonetheless, the Intent/Desire of Focus/Application is and will be, restricted to those systems where linearity and low distortion is a goal/focus, noteing that there are a multitude of mechanisms that can create distortion.

Consider, for consideration, in the Digital domain, the received Digital Signal could simply be made larger (via digital DSP and/or equivalent) then transcoded/transmuted/translated (digitally enlarged) then passed/delivered to a nonAmplifying/Unity Gain Analogue Power/Buffer (for PWM, carrier removed then passed/delivered directly to the output Power/Buffer section) which passes/delivers it to the Speakers (with very, very low output impedance). This is essentially being done/attempted/performed via the AX5689 chip but with PWM amplifycation, (still) very Novel.

Gm Metric…. the article proposes a metric that is based on the shape of the nonlinearity curve that has the following features….
  • It should be more sensitive to higher order nonlinearitys than lower order ones. To meet this objective they propose useing the second derivative of the nonlinear transfer function since this function increases in value according to the square of the order. To alleviate a sign problem with this value we propose squaring this term. This function also addresses the third requirement above.
  • It should be weighted towards greater values for nonlinearitys at lower signal levels. To meet this requirement they propose using a cosine-squared function which is unity for small values of the signal and zero for the largest ones.
  • It must be immune to changes in offset and gain (first order slope) since these are inaudible effects. To meet this requirement they propose integrating this function to yield a single number which will be call Gm (Geddes/Lee metric), a proposed metric for distortion which is based in the general theory of nonlinear systems and takes into account the characteristics of human hearing.
  • 1756724360459.png
    Pg6/7, this exact equation represents the central hypothesis. It is important to note that Eq.(5) is actually a property of the system, not of a signal sent through the system. It is completely independent of the actual input signal sent through the subject system and is thus valid for any signal. They suggest the Equation is to be applied, in the form shown, requires a knowledge of the shape of the nonlinear transfer function T(x), as defined in Eq.(1). As shown it is basically a frequency independent measure. There is no ambiguity in performing the calculations at a particular frequency, but in a real system T(x) can be a frequency dependent, i.e. T(x, f), in which case G m(f) will also be frequency dependent. Firstly it is a new approach to evaluating a nonlinear system, formulated to be consistent with the psychoacoustic criteria that the classical measures do not account for. Secondly the article shows the results of clinical tests which use each of these metrics on simulated nonlinear systems.
This is Geddes and Lee Article/Paper No1 and Article/Paper No2 shows the relationship of the signal based metrics to subjective impression is virtually nonexistent. It is our hope that the audio community will give the outdated notion of THD, IM/IMD, signal types, etc. (signal-based concepts) as these are all just symptoms of the real problem – nonlinearity.
 

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At this point and in the context of the limitations of human aural perception, doesn't obsessing over better and better distortion/noise numbers amount to chasing solutions to a virtually non-existent problem? Beyond gear marketing goals, doesn't it make more sense to focus on still-extant, clearly audible issues like transducer shortcomings?
 
That's fine. This digital PWM feedback is a logical next development step and the future is in this approach. I only do not want to slide to semi-marketing appraisals.

Regarding that HF noise, Stuart measures with AP and much higher resolution than I have. But, with lower bandwidth.

Axign digital loop FB idle spectrum (measured with AP) looks like this:

View attachment 472850

Purifi output idle spectrum (measured with DSO) looks like this:

View attachment 472851

The red line illustrates DSO method noise limit, however extended to 6MHz.

What is evident:

- Purifi has an individual carrier frequency line at 499kHz/-14dBV and its odd harmonics. But the HF noise floor, except for the carrier residuals, sits at the -80dBV which is the measurement limit.

- Axign seems to have carrier at 1.03MHz, very well suppressed. But the noise floor, noise shaped, is higher up to 600kHz, starting at 40kHz/-55dBV.

Which is better? I do not know. Both inaudible. Purely audio band parameters are better with Purifi.
Hi @pma, re Axign seems to have carrier at 1.03MHz, very well suppressed. But the noise floor, noise shaped, is higher up to 600kHz, starting at 40kHz/-55dBV.
PWM Dead Time will be much shorter, won't it? but the AX5689 Controller appears to handle this very well, doesn't it? Would you suggest that this is the case?

This thread, ASR/Peak Amplification AM-400C2G GAN amp module with THD 0.0008%, prompted a consideration of.... how would the Axign Class-D Audio Amplifier PWM Controller perform with the Infineon CoolGaN/GaNFETs and its PFFB configuration? Would you suggest that utiliseing the Infineon CoolGaN/GaNFETs is worth considering, a beneficial consideration, with the/this implementation of the Axign Class-D Audio Amplifier PWM Controller Demonstration Board?

So, you might wonder how feedback is done, since that usually involves taking the difference between input and a fraction of the output signal, usually an analog task. The answer involves another interesting innovation — digital feedback.

This technique was described in a series of papers and patents from Daniel Schinkel and Fred Mostert (see Resources). Instead of a conventional feedback loop, the differential-mode amplifier output is digitized by a high-speed low latency Delta Sigma (ΔΣ) ADC, then input and output are subtracted in the digital domain.

Why hasn’t this conceptually simple approach been done before? This path is not an easy one. The ADC is required to have high bandwidth and low latency for stability. ΔΣ ADCs can have issues with high quantization noise. Digitally filtering it introduces latency and phase shift, which complicate stability. So, almost on-theme here, the ADC derives feedback via a DAC that has a low-pass filter function. This allows compensation for digital filtering and greatly reduced noise in the audio band. In the implementation tested here, the sampling is done every 20ns, which is more than fast enough to achieve a stable control loop well beyond the audible range. In the Axign chip, the ADCs are doubled up to get a 3dB increase in signal-to-noise. Surprisingly, there is no dissipation penalty for the ADCs. Axign’s design reduces their power use to 15mW each.

The output drive is configured for multiphase, which greatly reduces electromagnetic interference (EMI) and demands on the output filters, and the feedback is taken following the output filters. It’s well known that nonlinearities in output inductors are often the limiting factor for Class-D amplifier performance. This can be overcome at significant money, space, and heat costs with inductor design, but even more effectively with post-filter feedback. I’ve seen this used in several other premium Class-D amplifiers (e.g., Orchard, Hypex, Purifi), but in combination with multiphase drive, efficiencies, and performance can be maximized at low cost.

Speaking of efficiency, the demo board uses GaN output devices. As I’ve discussed in a couple other articles, GaN brings a lot of benefits, specifically from the higher charge carrier mobility and thus lower channel resistance. In theory, one can achieve higher switching speeds with GaN compared to silicon, but this goes well past the point of diminishing returns. Where GaN really shines is exploiting the low resistance (and thus low I2R heating) to drive very low impedance loads

Axign prioritized efficiency at lower power by at least two means. First, the low power consumption of the ADCs
; and second, through the use of a novel PWM scheme, which takes advantage of the differential outputs and modulates the common-mode voltage. This also greatly reduces the stress and dissipation of the inductors used in the output filters. Because of the low power consumption resulting from the design, the dissipation of the amps at idle and low power is under 1.5W per channel, enabling low heat output and higher reliability in active speaker use. Of course, at high power, the Class-D advantages dominate, and the use of low on-resistance GaN output devices further increases the advantage.

It would be nice to have a Schematic and/or Circuit diagram, wouldn't it? There must be one available, hopefully :=)

Note that there is a Gm Metric Part2 to come but this paper/article is takeing me time to unpack and, especially, as enthusiasm permits. I thought that it had/has potential, especially as music is nonLinear (with aspects of (deliberate) N/D to consider), but Linearity/Transparency/Measurement are still very, very important, especially when the aspect of Compatability/Synergy would need to be considered, aren't they? I am/are hopeing that it can suggest/indicate that a component will/can Sound like this.... very enjoyable, enjoyable, moderately enjoyable, or not enjoyable :=)
 
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A suggestion to consider the following measurements, PR+TI (Power Rating + Transparentcy/Impedance), PR+TC (Power Rating + Transparentcy/Capacitance)….
  • Transparentcy/Transparency…. it can be suggested that Transparentcy means NO Feedback, NO Distortion, NO Noise, that is reasonable, isn’t it? There do not seem to be any Amplifyers that would meet this Definition/Requirement. ASR indicates that THD+N of -95db and/or better, is sufficient/enough to be considered as Transparent, doesn’t it? External aspects such as Speaker+Cable Load Impedance/Capacitance also affect Amplifyer Transparentcy, don’t they? Also as Power/Current increases, THD and Noise tend to decrease/increase, don’t they?
  • The Axign Class-D Audio Amplifier PWM Controller Demonstration Board, currently, falls a little short, 92 SINAD, of the -95db/THD+N Transparentcy Threshold, doesn’t it? It also suffers, quite badly, with the utilised 2uF Speaker Capacitance test regardless of Load Impedance, which even though (considered) extreme is (apparently) commonly exhibited by Electrostatic Speakers, which suggests is not extreme and needs to be considered. It does pass the 2~10R/ohm Speaker Impedance test, easyly/invariant/without dependence.
    • PR+TI, what would it mean? It could be suggested that the measurement could be simply a Power Rateing, for 2~10R/ohm Speaker Impedance and 20~20Khz Frequency Range, where the Transparentcy is a THD+N of -95db and/or better. For the Axign Class-D Audio Amplifier PWM Controller Demonstration Board the PR+TI would be 0, because it does not meet the Transparentcy measurement of THD+N of -95db and/or better.
    • PR+TC, what would it mean? It could be suggested that the measurement could be simply a Power Rateing, for 2uF Speaker Capacitance and 20~20Khz Frequency Range, where the Transparentcy is a THD+N of -95db and/or better. For the Axign Class-D Audio Amplifier PWM Controller Demonstration Board the PR+TC would be 0, because it does not meet the Transparentcy measurement of THD+N of -95db and/or better.
  • To be concise (and precise), First let us consider Power Rateing + Transparentcy Impedance/Capacitance.
    • For convenience, let us say that its THD+N of -96db, for 2~10R/ohm Speaker Impedance and 20~20Khz Frequency Range, is 20 Watts then its PR+TI would be 20.
    • For convenience, let us say that its THD+N of -96db, for 2uF Speaker Capacitance and 20~20Khz Frequency Range, is 20 Watts then its PR+TC would be 20.
    • Note that the connecting Cables Impedance/Capacitance, also needs to be considered.
    • How would this be Helpful? Let us look at the questions/statements commonly posted….
      • What Speakers to go with my Amplifyer?, What Amplifyer to go with my Speakers?
      • You won’t hear any difference because the Amplifyer is Transparent?
  • For 20~20Khz Frequency Range, if we know that an Amplifyers PR+TI = 20 and PR+TC = 20, Speaker Impedance/Capacitance is 2~10R/ohms and <=2uF, then, knowing that 1 Watt is more than sufficient for 110db/SPL Speakers, Extrapolate as follows for an Amplifyer with PR+TI = 20 and PR+TC = 20….
    • 110db/SPL Speakers, Amplifyer with PR+TI = 20 and PR+TC = 20, Transparent/1Watt required
    • 107db/SPL Speakers, Amplifyer with PR+TI = 20 and PR+TC = 20, Transparent/2Watt required
    • 104db/SPL Speakers, Amplifyer with PR+TI = 20 and PR+TC = 20, Transparent/4Watt required
    • 101db/SPL Speakers, Amplifyer with PR+TI = 20 and PR+TC = 20, Transparent/8Watt required
    • 98db/SPL Speakers, Amplifyer with PR+TI = 20 and PR+TC = 20, Transparent/16Watt required
    • 95db/SPL Speakers, Amplifyer with PR+TI = 20 and PR+TC = 20, Nolonger Transparent/32Watt required which means another Transparent Amplifyer would need to be chosen, for the chosen Speakers.
  • The above is for consideration and discussion….

At this point and in the context of the limitations of human aural perception, doesn't obsessing over better and better distortion/noise numbers amount to chasing solutions to a virtually non-existent problem? Beyond gear marketing goals, doesn't it make more sense to focus on still-extant, clearly audible issues like transducer shortcomings?
@Bruce Morgen, thank you, yes, when useing the term 'Transparent' everything needs to be considered, doesn't it? What can feed it and what it is feeding or it can feed, which is the (entire) Setup/Implementation, needs to be considered to be called 'Transparent' in a Setup/Implementation, doesn't it? If the term 'Transparent' is used it should be (very) clearly indicated why and how a Device/Component can be utilised so that it can be indicated as 'Transparent', for a Setup/Implementation or a range of Setups/Implementations.... this is reasonable, isn't it? We are seeing old tech, such as MosFets, utiliseing (very successful) old and new Implementations and now GanFets with possibletys, which is fabulous, isn't it? Do you remember when people would suggest that MosFets were only good for Industrial use and now the same is suggested for GanFets.... good to have a giggle, isn't it :=)
 
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Do you remember when people would suggest that MosFets were only good for Industrial use, where the NAD3020 quite quickly quietened those suggestions,
This is pure conjecture. In reality, NAD3020 does not contain any MOSFETs :facepalm: . NAD only started using MOSFETS in amps more than a decade later.

Early MOSFETS were particularly not very suited for linear audio applications. They had higher capacitance and non-linearities, and needed more voltage swing for the same current than BJTs. They were also much more expensive in the early days. Slow adoption of MOSFETS for audio happened in the 80s, led by Hitachi as maker of the first lateral MOSFETS that were speced good enough to be used in linear amplifiers. Only in the mid to late 90s were these amps becoming mainstream. And now with class D, almost everything is MOSFET.
 
Early MOSFETS were particularly not very suited for linear audio applications. They had higher capacitance and non-linearities, and needed more voltage swing for the same current than BJTs. They were also much more expensive in the early days. Slow adoption of MOSFETS for audio happened in the 80s, led by Hitachi as maker of the first lateral MOSFETS that were speced good enough to be used in linear amplifiers. Only in the mid to late 90s were these amps becoming mainstream. And now with class D, almost everything is MOSFET.
Thank you, Ah yes, removed that bit, good that you picked that up, and good explanation (that MosFets got there in the end). My NAD 3020 was about 1983/84 and modded by Allen Wright, even before I received it. At the time, he did not tell me what he did although he did mention MosFets (or was that FETs?) and some other aspects of the Mods that he did and it did sound quite/very nice :=)

NAD 3020 Series 20 mk-3 MOSFET Output Conversion
 
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Hello everyone!
I started developing an amplifier on AX5689C, the output stages will be built on GaN ISG3208EA
 
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