Atma-Sphere Class D Mono blocks

No, I dont like clinical or synthetic. I can live with clinical, but not with syntethetic.
At the same time I have to defend the current development of the audio industry. Every improvement I have had in my system has been on the "clinical" side at first. Better cables sound cleaner and less "fat". A better DAC differentiates better and creates less "overhead". After a while I realize Im closer to the music with less equipment in between.
Synthetic is the worse sound signature I can think of. Something is seriously wrong if it sounds the opposite of natural.
I havent heard Aavik except in shows, so I will not comment on its sound. Im very familiar with Atma-sphere and use it in my system.
Aavik and Atma-sphere are completely different comanies with completely different products. Aavik amps are a products with a capital P -- with a very sleek design in contemporary style -- intended to boost ownership pride. Aavik looks different and feels different from any other product and will give you top performance of the industry of today. "There was little to criticize" sums it up. This company simply know how to produce and market a good commercial product.
Atma-sphere is all about research, invention and good sound in my opinion. Its products are not intended to seduce, at least not by the looks. They are "utilitarian" in design, mostly. The OTLs have a touch of vintage and nostaligia in the design, but thats it. Atma-sphere takes a different technical approach to the previous OTLs and class D amplifiers, as I understand it, and drives if to a very interesting conclusion. It bring something fresh to the product line, but not unless there is a real improvement. I suppose that is how a smaller company should be working to survive and Atma-sphere is not unique in this respect in the audio industry. It develops trust.
Atma-sphere upgrades old products and Aavik develops new and more dashing products. Also a difference.
Both strategies are OK, for different kinds of people.
Aavik (and similar companies) can and does bring steady incremental improvements to audio equipment and we should appreciate and take advantage of it. And so does Atma-sphere (and similar companies), maybe not incrementally, but more fundamentally and in giant-steps, when it happens.
Im biased, I own Atma-sphere amps.
 
At the same show (after the official closing time) we attended a semi-private demo of the new Kii SEVEN speakers. The demo was textbook: long fragments of various tracks. While we were listening, Bruno Putzeys walked into the room and afterwards we had a fairly long conversation about several topics, primarily of a technical nature.

Although Bruno clearly represents the 'scientific/technical approach' to audio R&D (he hardly listens to his 'brainchildren'), this doesn't rule out the occasional 'element of natural sound' ending up in his products, such as this midrange:

1698100866348.png



Moreover, he also appreciates some tube amplifiers, stating: "I hear the result of an (empirical) optimization process and admire the efforts".
 
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I Think the Atma-Sphere's, when setup with the right ancillaries are very musical. They are absolutely not bleached, lean, analytical or mechanical sounding. If that's a help. I heard the Avid systems at recent shows and my reaction was similar to yours. Though I felt it was more about the speakers than the electronics. But who knows?
 
In order not to go into too much detail, in my view the 'purity' (naturalness) of audio reproduction is largely determined by what I refer to as 'signal integrity'. This obviously applies - in different ways - to the electrical (front-end) and electro-mechano-acoustical (transducers) domain.

An audio waveform contains many concurrent frequencies, which represents a complex time-domain signal (Joseph Fourier).
So there are two ways of representation:
1. periodic time function = time domain
2. single finite time pulse = frequency domain

"It can be shown in the lab that a periodic waveform (for example a square wave) can be formed by summing a series of harmonically related sine waves of appropriate amplitude and phase.

Because it is a composite of all the component frequencies. Just the same as we hear multiple frequencies in air, a microphone will respond to the sound pressure level changes as our ears do, converting this into a single waveform. We don’t hear sound as individual frequencies, we hear it as composite. This is what happens electronically although you could read this as positive and negative going air pressure as well:"

20 Hz tone:
1698162198322.png

200 Hz tone:
1698162233858.png

summed into this single waveform:
1698162279328.png


This is the absolute basic level and therefore the fundamental starting point for audio (reproduction).
The implications of these basic principles on all components in a chain are numerous and as you zoom in on (sub-sub... etc.) system level, the complexity increases exponentially.

To give an example: even well-known audio physicists often ignore the fundamental principle of energy storage in woofers. Energy storage (= loss factor) is usually associated exclusively with resonances, while this phenomenon actually encompasses more than just that.

"What is the loss factor ??
The loss factor is a measure of the inherent damping in a material when it is dynamically loaded. It is typically defined as the ratio of energy dissipated in unit volume per radian of oscillation to the maximum strain energy per unit volume. For a nonlinear elastic material, this applies to the tangential stiffness."

Simply put: loss factor is related to damping and therefore affects the signal (transmission).
 
Agree on Squarewave response, cant wait for Ralph to post up a few on his class D amps ..!


Regards
 
In order not to go into too much detail, in my view the 'purity' (naturalness) of audio reproduction is largely determined by what I refer to as 'signal integrity'. This obviously applies - in different ways - to the electrical (front-end) and electro-mechano-acoustical (transducers) domain.

An audio waveform contains many concurrent frequencies, which represents a complex time-domain signal (Joseph Fourier).
So there are two ways of representation:
1. periodic time function = time domain
2. single finite time pulse = frequency domain

"It can be shown in the lab that a periodic waveform (for example a square wave) can be formed by summing a series of harmonically related sine waves of appropriate amplitude and phase.

Because it is a composite of all the component frequencies. Just the same as we hear multiple frequencies in air, a microphone will respond to the sound pressure level changes as our ears do, converting this into a single waveform. We don’t hear sound as individual frequencies, we hear it as composite. This is what happens electronically although you could read this as positive and negative going air pressure as well:"

20 Hz tone:
View attachment 118665

200 Hz tone:
View attachment 118666

summed into this single waveform:
View attachment 118667


This is the absolute basic level and therefore the fundamental starting point for audio (reproduction).
The implications of these basic principles on all components in a chain are numerous and as you zoom in on (sub-sub... etc.) system level, the complexity increases exponentially.

To give an example: even well-known audio physicists often ignore the fundamental principle of energy storage in woofers. Energy storage (= loss factor) is usually associated exclusively with resonances, while this phenomenon actually encompasses more than just that.

"What is the loss factor ??
The loss factor is a measure of the inherent damping in a material when it is dynamically loaded. It is typically defined as the ratio of energy dissipated in unit volume per radian of oscillation to the maximum strain energy per unit volume. For a nonlinear elastic material, this applies to the tangential stiffness."

Simply put: loss factor is related to damping and therefore affects the signal (transmission).
Most drivers have low efficiency (<1%) and most of that is lost as heat dissipation in the voice coil. This is also where dynamics of most drivers suffer because of thermal compression. Heat goes up, resistance goes up, current goes down and SPL doesn’t increase like it should…the driver “flattens” out and this can be easily heard.
 
Here are two square waves, both 10KHz. I tried to keep them at the same level.
The first one is our M-60 OTL:10KOTL.jpg The second one is our class D:
10KClassD.jpg
The main difference you see is the presence of the 'residual', which is what is left of the switching frequency. Usually when measuring the amp a filter is used to knock it out, which, being at about 500KHz, isn't audible but it bothers people when they see it, although its harmless. You can see that the class D has a bit less gain than the OTL; the lower square in each photo is the input signal.
 
Impressive for class D, cant say i have seen better on a class D , how is the PSU noise not audible thou ..?


Regards
 
Impressive for class D, cant say i have seen better on a class D , how is the PSU noise not audible thou ..?


Regards
If by PSU you mean Power Supply Unit, the amp employs a toroidal power transformer and conventional rectifiers. We found we had to snub them for low noise as they were the main source of noise the amp radiated. We added an input AC power filter as well, since any noise on the AC line can mess with digital equipment elsewhere in the system, even if the circuit otherwise meets EU Directives (CE mark). So we are well below the limits for noise.

As a result the amp is really quiet. My speakers are 98dB and I have to have my head in the midrange horn to hear anything at all. Most of the noise it makes comes from the opamp input buffer.
 
Most drivers have low efficiency (<1%) and most of that is lost as heat dissipation in the voice coil. This is also where dynamics of most drivers suffer because of thermal compression. Heat goes up, resistance goes up, current goes down and SPL doesn’t increase like it should…the driver “flattens” out and this can be easily heard.
True, although thermal compression is mainly an issue in PA environments where sustained high output levels are required.

The (system) sensitivity, which correlates with efficiency, is absolutely of primary importance. It's leading in my opinion.
 
If by PSU you mean Power Supply Unit, the amp employs a toroidal power transformer and conventional rectifiers. We found we had to snub them for low noise as they were the main source of noise the amp radiated. We added an input AC power filter as well, since any noise on the AC line can mess with digital equipment elsewhere in the system, even if the circuit otherwise meets EU Directives (CE mark). So we are well below the limits for noise.

As a result the amp is really quiet. My speakers are 98dB and I have to have my head in the midrange horn to hear anything at all. Most of the noise it makes comes from the opamp input buffer.
A basic C snubber?
 
True, although thermal compression is mainly an issue in PA environments where sustained high output levels are required.

The (system) sensitivity, which correlates with efficiency, is absolutely of primary importance. It's leading in my opinion.
No, it is definitely an issue with home audio. The drivers are not design to withstand impulses that heat the coils quickly and cool down slowly. You can lose several dB on peaks…just where you need it most is where flattens out. This is why horns and other high sensitivity speakers give greater dynamics…they still compress but at much higher than typical domestic volumes. They are usually playing with mW (mA) rather than heating up with Watts (A).
 
No, it is definitely an issue with home audio. The drivers are not design to withstand impulses that heat the coils quickly and cool down slowly. You can lose several dB on peaks…just where you need it most is where flattens out. This is why horns and other high sensitivity speakers give greater dynamics…they still compress but at much higher than typical domestic volumes. They are usually playing with mW (mA) rather than heating up with Watts (A).
There are so many factors involved in power/thermal compression that it's very difficult to draw simple/unambiguous conclusions. A simple remedy to avoid this phenomenon or at least significantly reduce it (for hi-fi at home) is indeed to opt for efficient pro drivers, which is what I do.
 
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True, although thermal compression is mainly an issue in PA environments where sustained high output levels are required.

The (system) sensitivity, which correlates with efficiency, is absolutely of primary importance. It's leading in my opinion.

Thou not mutually exclusive , they represent different metrics, agree thermal compression has very little correlation to home audio where typically 1-2 watt RMS is not an issue ..!
 
Thou not mutually exclusive , they represent different metrics, agree thermal compression has very little correlation to home audio where typically 1-2 watt RMS is not an issue ..!
It’s not the RMS power that’s the issue. You can’t look at this like a static system, which is what an average power is, the power is constantly jumping around. Heating up is fast…cooling down is slow…that hysteresis is a relevant factor.
 
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It’s not the RMS power that’s the issue. You can’t look at this like a static system, which is what an average power is, the power is constantly jumping around. Heating up is fast…cooling down is slow…that hysteresis is a relevant factor.
Why OTLs, field-coils, or D class might take high fidelity back to the future?
 
It’s not the RMS power that’s the issue. You can’t look at this like a static system, which is what an average power is, the power is constantly jumping around. Heating up is fast…cooling down is slow…that hysteresis is a relevant factor.

Beg to Differ the RMS is what add the heat and not dynamic peaks in the ms range, as it does not bring much to the table. At domestic levels there is no thermal issues whatso ever in drivers or xovers operating at 1-2 watt RMS ..

Well unless using really poor quality drivers , but its academic we are discussing only quality drivers ..!

As to VC rise in dynamic speakers you can do the experiment yourself measure Vc temp before and after 30 mins of standard listening..!

Also use pink noise and measure Db before and after with 2.83 V


Please let us know your findings ..!
 
Beg to Differ the RMS is what add the heat and not dynamic peaks in the ms range, as it does not bring much to the table. At domestic levels there is no thermal issues whatso ever in drivers or xovers operating at 1-2 watt RMS ..

Well unless using really poor quality drivers , but its academic we are discussing only quality drivers ..!

As to VC rise in dynamic speakers you can do the experiment yourself measure Vc temp before and after 30 mins of standard listening..!

Also use pink noise and measure Db before and after with 2.83 V


Please let us know your findings ..!
No, that has nothing to do with instantaneous heating. You are talking about temperature rise that has to go through lots of material (magnets etc.). You won't be able to accurately measure the real-time fluctuations.
 
Some background info.
 

Attachments

  • Thermal simulation of loudspeakers, P. J. Chapman (AES Paper 1998).pdf
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  • Heat Dissipation & Power Compression in Loudspeakers (Douglas J. Button).pdf
    933.6 KB · Views: 3
  • Measuring system for the derivation of thermal loudspeaker parameters (Gottfried K. Behler; Ar...pdf
    159.3 KB · Views: 1

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