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Author Topic: Classic Tube Amp Sound  (Read 7675 times)

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Solderdude

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Re: Classic Tube Amp Sound
« Reply #40 on: August 08, 2013, 09:21:19 PM »

Where does NFB come into this?

Consider the following...  (boring long read ahead, those not really interested should skip this post  :)p8 )

Lets take a non feedback design (tube amp for instance) with a single tube for voltage amplification and a second one as 'follower' to act as impedance transformer.
Say 10x gain for the voltage gain stage and 0.9x for the impedance transformer (cathode follower).

The bigger the load the lower the output voltage will become due to voltage division by the output Z of the output stage and the load.
As that output device is not linear the 'loss' in output voltage across the terminals will be greater as the load Z gets smaller and also when the amplitude get's bigger as more current is demanded.

This 'flattens' the output signal somewhat and is thus distortion.
I should add that the word 'distortion' sounds very negative and may give the impression distortion is a BAD thing by definition.
It actually is NOT and depends on WHAT type of 'distortion' it is.
With tubes the distortion may not sound as 'distortion' but may come across as 'more natural' sounding (as the 'distortion' closely resembles the harmonics that are present in the original music signal itself and may even be completely 'mask' or 'enhance' the original harmonics that are present by nature. It also depends on some other factors by the way.

You can make this 'loss' in output amplitude at higher amplitude levels smaller by not using low load impedances, or paralleling more output devices. Still there will always be some losses but they will be smaller.
The end goal is to make the loss '0' in essence and to 'unload' the voltage gain stage which is high impedance and should not be loaded at all.
Linearity in the voltage gain stage can be improved by topology changes (current sources instead of resistors or other trickery)

Another way is to 'compensate' the voltage gain stage so it has more gain at higher currents and thus compensate for the 'loss' of output voltage in the output stage. This would have to be dynamic as that 'correction' depends on the current drawn (output voltage/load Z). As the output stage always creates a 'loss' the gain would need to increase which it can't as it is fixed (non feedback).

So you need to lower the overall gain lowered or need the voltage gain to be higher by adding 1 more amplification component.

So 1 tube has about 10x gain (as an example) which may be what we need. 2 tubes would give 10x10 = 100x which would be too much or with a different topology and local feedback reduce it to 30x or so. Still too high. So we take the output signal and lower that in the reciprocal amount of the gain we need. So if we want 10x the resistor divider will have to be 0.1x

We subtract (hence the word negative in NFB) the lowered output signal from the input signal which is the feedback.
We use 1V on the input and when the gain is exactly 10x the output voltage of the divider will be 0.1x thus also be 1V. The subtraction from those signals will be 0V exact.
When the input voltage is 1V the output voltage due to the 30x gain would become 30V and the feedback divider will give 3V and as that is subtracted from 1V the resulting voltage would be -2V on the input meaning the input resultant voltage seen for the amp (it sti ll receives 1V in) will force the input voltage (inside the amp itself) lower. So effectively when 1V is applied at the input and the
'subtraction' happens the 'internal' input voltage will become about 0.33V and when this voltage is amplified 30x you get 10V which is 10x the 1V actual input voltage.

This feedback is (almost) instant and continuous so with feedback the output voltage will be more stable.
For arguments sake lets say the voltage gain = 30x and unloaded the output stage has 1x gain the total gain internal is 30x. Under load and at full voltage swing the output stage drops to 0.8x. The voltage gain stage will need to compensate that loss and thus the voltage swing on the output of the voltage gain stage will need to increase 1.2x to compensate for the loss in the output stage. As there are non-linearities in that voltage gain stage (and the output stage) there will be more distortion but as the output voltage wants to 'regulate' itself, because of the feedback, to an exact 10x higher copy of the input voltage the biggest portion of the 'compression' is undone which lowers the distortion in the actual output signal as the copy is 'closer' to the input signal but the internal voltages are higher and this increases the distortion a bit. This way NFB (negative Feed Back) does NOT completely compensate and thus distortion will still be present but there will be less of it and the harmonics spread will differ as well.

The fun part of NFB is that it works better when more 'internal gain' is present. The feedback loop becomes more accurate. So more amplification stages and thus more internal gain (called the open loop gain) will give a more accurate amplified 'copy' of the input signal. So an amplifier design with just one tube and mild feedback will have slightly less distortion in the output signal. A design with 2 tubes  and overall feedback will be more accurate and thus have less distortion. A design with more tubes will be even better in this aspect.
It appears that more gain = lower distortion so all amps must have more internal gain it seems.

Alas TOO much is never good as this presents other problems as bandwidth will drop and because each components (gain stage) adds some delay (extremely little but it is present) and the 'last amplification stage' still needs to 'deliver' the full voltage swing speed problems become problematic.

When a fast change in input voltage is there (square wave or needle or step) and there is always some 'delay' internally the output may 'hunt' slightly before it stabilizes. This can be see as a gradual sloping output voltage that 'overshoots' and drops back again (or even rings).
When the input signal is not capable of being faster than the internal BW the output can 'follow' it.
For this reason in good designs the input FR is limited by an input RC filter which slows the input signal down.

So NFB lowers distortion and how much depends on the 'internal' amount of gain and how linear the 'subtraction' circuit is made. This shows the importance of the input stage b.t.w. as that NEEDS to be good to get low distortion.

So topology (quality of the 'feedback point'), amount of internal gain (open loop gain), speed (propagation delay) of the amplifier stages, load (current drawn by it) determine distortion.

The better the circuit behaves WITHOUT feedback the less 'correction' is needed and the more accurate the (amplified) output voltage will resemble the input voltage.

I have to add that this is a rather 'crude' description and doesn't cover all the aspects and numbers are only to make an example.  ::)
« Last Edit: August 08, 2013, 09:53:06 PM by Solderdude »
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OJneg

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Re: Classic Tube Amp Sound
« Reply #41 on: August 08, 2013, 11:29:37 PM »

Thanks for taking the time to write all that out. I think I understand it now.

So the original problem stemmed from the output device (or stage I guess) having an output impedance that was large enough to create a voltage drop on the output when loaded, yes? And using NFB you're able to correct this by providing more gain to get rid of the so called "compression" that would happen without NFB. Feedback loops can be quite confusing. p:8

 And I suppose your average output triode has an Rp (plate resistance) that is larger than the the output Z you would get from a semiconductor, which is why you said valves are less linear w/o feedback.
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runeight

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Re: Classic Tube Amp Sound
« Reply #42 on: August 09, 2013, 04:16:04 AM »

Gents, maybe you'll permit me to take a sightly different approach.

It seems to me that the last question posed by Anaxilus is really not a proper question. It's kind of like asking what happens when an irresistible force pushes on an immovable object. The question is framed such that you really can't answer it. But, I think it's possible to answer sub-questions that might illuminate something.

I think the first thing is that the active devices don't stand alone in a circuit. It's not just that there are other passive components around them, it's also that they are in a particular circuit topology. This circuit topology has as much to do with the sound as do the devices themselves.

As someone pointed out BJTs are current controlled devices, FETs are voltage controlled, tubes are voltage controlled. In fact, tubes are the hollow state analog of depletion fets. In an enhancement FET the gate must be positive (for an N FET) with respect to the source. In a depletion FET the gate must (in general) be negative. In a tube the grid must be negative (for normal operation) with respect to the cathode.

Each device type, to some degree, dictates parts of the circuit design.

For the moment, let's assume that you're building an amp entirely using one type of device, BJT or FET or Triode. Greatly simplified thoughts.

For BJTs some elements of the circuit have to push current in and out of the BE junctions. In addition, if you want the devices to be turned on at the quiescent op point, then you must have some BE idle current. This dictates certain conditions and driving circuitry. For example, with very high power output BJTs a lot of base current is required. If you're pushing peak 10A and the hfe is only 100 you need peak 100mA into the bases. The drivers for these transistors have to be selected and configured for this.

Since the BE junction is a forward biased diode it typically doesn't exhibit a lot of capacitance. Thus, you don't necessarily have to account for significantly increased loading as the frequency goes up.

BJTs exhibit thermal runaway. They also operate happily at fairly low voltages.

The characteristics of the devices, to some degree, already set some conditions on the design.

Mosfets don't require gate current to bias to an operating point. But, they can have large gate/source and gate/drain capacitances. In high current or high voltage mosfets these capacitances can be fairly large, large enough to be relevant in the audio band. What this means is that whatever is driving the gate may not have to push much charge at low frequencies, but quite a lot at high freqs. The driver stage has to be designed to handle this capacitative load. And if this is a totally FET design, then there are capacitors everywhere.

If using enhancement FETs (by far the most common today) then there has to be a positive (negative for P) gate/source voltage to establish an idle current.

These characteristics dictate, to some degree, the circuit design.

Tubes are high voltage, low current devices. The grids are negative with respect to cathode. The are no P channel tubes because holes don't exist in a vacuum. We can't build complementary circuits. In general tubes exhibit fairly low capacitances because their elements are far from each other and there is no dielectric between them.

These device characteristics dictate, to some degree, the circuit design (e.g., the need for output transformers to achieve high output currents).

It is rare, in my experience, that you can take an amp circuit designed solely for one of these types of active device and simply wholesale replace all the active devices with another type. The only situation that I can think of is where you could put depletion fets in for tubes assuming they can stand the voltages and would achieve similar quiesc ent currents. Maybe someone knows of another situation.

So, why all this? Well, if we were comparing three amps, each made entirely with one type of active device, we are comparing not just the behavior of the devices but the specific circuit topologies needed to make them work as a practical amplifier. MHO is that these circuit topologies are equally as relevant to the comparison. And because there are multiple, effective topologies for each type of device, the comparisons form a set of combinations which are hard to get a handle on.

OK, so what to do? We can take the purely external, black box approach and set some conditions on performance.

Any of us could design a 10W amplifier with any of these devices with 20-30% THD. If three of us did this and we did a blind test we might actually think that one sounded better than the other two. But, all three would still be junk.

Part of this question has to do with what we might call a "well designed" amplifier. There are decades of conversation, theory, and practice on this topic, although maybe not using this exact language.  :)p6

We can make an arbitrary definition for discussion. Let's say we want an amp which can deliver 10WRMS into 4R (resistive) with .1% THD at 1kHz and -1db from 20Hz to 100kHz and a gain of 25. This is not a difficult target, but let's call this a "well designed" amp. These are entirely externally measurable characteristics and don't tell us anything about what's inside the amp.

Our first problem is that all three of these active device types are non-linear except at very small excursions around operating points. So, our job is take these non-linearities and linearize them to achieve the "well designed" result. And, being whiz designers we do this and build our amps. Then we compare them.

Regardless of the result, a dominant part of the questions as to why one "sounds" better than the other two (if there is one) is not which particular type device is used, but rather what techniques were used in the design to linearize the gain and to achieve the bandwidth?

The techniques for doing this have been around for decades in one form or another. They include local NFB, global NFB, using CCSs for high dynamic resistances, reducing interstage loading, using diff amps for common mode rejection, etc. Most of the time these techniques are used in combination and in various places around the circuit.

When we listen to these three amps we are not just listening to the "sound" of the particular active device used, but also the "sound" of all the other circuit elements put in place to linearize them sufficiently to make our "well designed" amp.

Unfortunately, it's even more complex than this. The simple example assumes that each linearizing circuit element can be made from the chosen active device for that amp design. But, we have for decades mixed and matched. Tube amps commonly use CCS loads made from BJTs or FETs. BJTs and Mosfets are often used in combination. Sometimes all three types are in the same design. In these instances, the behavior of the active devices is mixed together. What is the sound of this??

In every instance, including so-called "money is no object" amps, there are always a set of design choices and trade-offs. Always. There is always a point where the designer has to say, ok this is good enough to achieve what I intended to achieve.

So, when asking what is it about tube amps vs SS amps, it seems to me you have be more specific. Like, what is it about this particular tube amp and its internal topology that is (apparently) more pleasing than this SS amp with its particular internal topology? If we ask that question, we might have a chance.

Or, you can follow this link in this post from cnpop e on diyaudio:

"For many years (and maybe still?), an audio engineer called Richard Clark ran an "Amplifier Challenge," where he offered a prize of $10,000 to anyone who could demonstrate in double-blind testing that they could reliably discriminate between any two amplifiers, subject to certain fairly basic conditions. The "challenging" amplifier and the comparison had both to have less than 2% THD at the levels at which they were operated; the signal levels would be carefully matched to 0.05dB accuracy; and if either had a significant deviation from flat frequency response in the audio range, then an equaliser would be used on one or other amplifier (challenger's choice) to match them. Nobody ever won the prize, although of order 1000 have tried.

Typically, as I understand it, if the challenging amplifier were a tube amplifier then Clark would make a little R/C equaliser to match the solid-state comparison amplifier's frequency response to the tube amplifier, and maybe add a series resistor at the output of the SS amplifier to match the output impedance of the tube amplifier. So effectively, by this means, he would mock up the performance of the tube amplifier at the cost of about $5 worth of components, he reckoned.

Different people will place different interpretations on what this shows, I'm sure. To me, it would seem that if no one can hear the difference between the real tube amplifier and the mocked-up one in blind listening tests, then the sounds are the same.

Chris"

More details here: http://tom-morrow-land.com/tests/ampchall/rcrules.htm



Many things have been left out so this post doesn't get too long.
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OJneg

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Re: Classic Tube Amp Sound
« Reply #43 on: August 09, 2013, 06:10:51 AM »

I think we can all agree at what you're getting at. But I don't think we're any closer to answering Anax's question  :-S

It seems to me that the general consensus on this forum is that different amplifiers can sound different. Sometimes they only sound good with certain headphones. Sometimes they just sound plain bad. If we were to break down all these circuits and attempt to derive their transfer functions stage by stage, would it get us any closer to an answer? Because it seems that when we try to just treat the device as a "black box", they more or less don't reveal the differences that we're hearing. Or at least within what many consider to be audible thresholds.

Right now, it seems the best we have to go on is whatever design quirk the amplifier seems to have, specifically with regards to the output stage. SET, DHT, Class-A, Class-AB, Class-D, BJT, MOSFET, OTC, OTL, cap coupled, whatever. This makes loose sense to me I guess, because it means putting different loads on that output topology can yield different results in how the load interacts with what's driving it. But what's actually happening and how can it be characterized? I don't know. And I fully accept the possibility that we're all just hearing shit.
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ultrabike

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Re: Classic Tube Amp Sound
« Reply #44 on: August 09, 2013, 06:28:50 AM »

Because it seems that when we try to just treat the device as a "black box", they more or less don't reveal the differences that we're hearing. Or at least within what many consider to be audible thresholds.

Nah, under certain conditions I believe those "black box"-es will reveal their strengths and weaknesses... In the case of "tube sound" however, one has to be specific because more than likely different "tube" amps along with their circuit topology will likely sound different form the next... In other words, one should not generalize something so diverse.

Right now, it seems the best we have to go on is whatever design quirk the amplifier seems to have, specifically with regards to the output stage. SET, DHT, Class-A, Class-AB, Class-D, BJT, MOSFET, OTC, OTL, cap coupled, whatever. This makes loose sense to me I guess, because it means putting different loads on that output topology can yield different results in how the load interacts with what's driving it. But what's actually happening and how can it be characterized? I don't know. And I fully accept the possibility that we're all just hearing shit.

Yup, I guess that is the question.

@runeight

AWESOME!!! :)p1 :)p1 :)p1 :)p1 :)p1

It makes sense that different active elements (TUBES, BJT, FET, MOSFET... - and among them different sub-classes - and combination of all of the above) will require different circuit topology approaches given their distinctive characteristics. It may follow that depending on the approach selected, each solution will exhibit different characteristics (some more audible than others).

Among the many goals behind these designs, amplifier "transparency" (relative to the recording) has been among the most sought after by some (but not always). Based on what I've heard from a simple Valhalla + HD600, S7 + HD800, CTH + Paradox, among others, I feel tube amps can fulfill this "transparency" goal to great success.

However, like you said, there are always trade-offs (NSTAFL) and while it may be possible to design an amp that will perform great with a healthy range of headphones, it may not necessarily be the best for each and everyone. I believe this is the case for the Valhalla, the fabled O2, and pretty much everything else "tube" or not. Furthermore, other requirements may come into play (SWAP, $, manufacturability, portability, ...) which may narrow down the choices.

Back to no-feedback, non-inverting, Class-A architectures (which I think the Valhalla is a good quality example off), Solderdude mentioned that linear behavior is difficult given the lack of FB. However, the Valhalla seems to have pulled a decent job with the HD600 which has a r elatively large impedance. What is it about such architecture that seems to perform well with some high impedance/hi-fi cans, but no so with lower impedance/lo-fi ones?
« Last Edit: August 09, 2013, 06:37:13 AM by ultrabike »
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Marvey

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Re: Classic Tube Amp Sound
« Reply #45 on: August 09, 2013, 06:55:15 AM »

Since tubes are high voltage low current devices, a transformer helps them drive the more current hungry lower impedance headphones.


Vahalla I believe is single-ended OTL - transformerless cap coupled. So one needs to make sure the impedance of headphones is sufficiently matched (much higher) than the high output impedance of the amp.
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ultrabike

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Re: Classic Tube Amp Sound
« Reply #46 on: August 09, 2013, 07:58:26 AM »

Thanks Marv! :)
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Solderdude

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Re: Classic Tube Amp Sound
« Reply #47 on: August 09, 2013, 08:17:51 AM »

Back to no-feedback, non-inverting, Class-A architectures (which I think the Valhalla is a good quality example off), Solderdude mentioned that linear behavior is difficult given the lack of FB. However, the Valhalla seems to have pulled a decent job with the HD600 which has a relatively large impedance. What is it about such architecture that seems to perform well with some high impedance/hi-fi cans, but no so with lower impedance/lo-fi ones?

The answer to this question is already given, but it could easily be missed or dismissed on account of 'that can't be it'.
It has to do with the type of distortion and how our brain interprets the added harmonics (perception).

The perception bit is responsible for the complete dissonance (below certain levels) between measurements and perceived differences.
I am fully aware some will say, ah but we all hear the same (or use the argument we do not) and it reproduces so there must be something we cannot measure or do not know how to yet. 
Seems very plausible there may always be an interaction between different frequencies and amplitudes and different loads that is not logged in a number or measurement.
The description of an output signal of say an amp effectively only consists of a single voltage (2 in case of stereo) changing its value over time. The conversion of an electrical signal to sound waves is extremely complicated. The workings of the brain are unravelled bit by bit over time but only with respect to a single 'asset' NOT the whole picture, too many variables. I think it will be safe to say perception will never be put in equations / fully understood as it also involves biological aspects.

A single test tone can be analysed with ease and accuracy but if you start to do various tests at the same time you cannot tell the 'individual' test results apart any more. So to characterise an amps performance (or DAC for that matter) you need to do various tests and interpret the results as a combination. The same is even more valid for just the few measurements we can do on speakers and headphones, FR, distortion (for various types), dispersion patterns for different frequencies, time delay, interaction, room effects, as well as all the variables in headphones that are 'missed'.
For instance lets take pad compliance. Someone with a smaller head and headphone A because of the clamping force of a headphone NOT being a constant thing but varies with the width and height adjustment of the cups. This can cause a pad to compress differently on small and large heads yet the same headphone A may not seal well on a smaller head and may seal better on a larger head and the pads may be compressed more. This variable alone is NEVER taken along in any test yet has a profound influence on the sound (arguably).

For electrical signals though all the different aspects are pretty well known, voltage, current, phase, resistance, impedance, time e.t.c. can be measured extremely accurate with resolutions FAR beyond the capabilities of human hearing. I don't think an undiscovered 'parameter' is missed that will link the dissonance.
However, they cannot be measured at the same time nor show interaction IF present. We also cannot use computers to 'follow' a single instrument in a plethora of other signals BUT they can be used to look for other aspects. This feeds the notion that we need more measurements to get a better correlation between what is heard and measured so we can all live happily ever after.

I used to have the same thoughts about amps until I started to experiment with differential amps and actual music signals and started realising that what I (and those even right next to me during the same experiment) heard and measured did not usually correlate and started to look for the 'why'. A personal quest ..

« Last Edit: August 09, 2013, 08:57:06 AM by Solderdude »
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ultrabike

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Re: Classic Tube Amp Sound
« Reply #48 on: August 09, 2013, 07:19:28 PM »

The answer to this question is already given, but it could easily be missed or dismissed on account of 'that can't be it'.
It has to do with the type of distortion and how our brain interprets the added harmonics (perception).

The perception bit is responsible for the complete dissonance (below certain levels) between measurements and perceived differences.
I am fully aware some will say, ah but we all hear the same (or use the argument we do not) and it reproduces so there must be something we cannot measure or do not know how to yet. 

I agree that is sometimes perception is indeed the answer. There are such things as frequency and temporal masking that are sometimes exploited in audio compression with success. I believe there is also a limit as to how many frequencies we can perceive at the same time and so forth.

However, in the specific case of the Valhalla, the distortion data seemed to back up the high-quality audio quality perception when using a relatively high impedance headphone such as the HD600. Similarly, measurements showed that perceived audio quality degradation when using a moderate impedance headphone (HD558) correlated with a significant increase in THD distortion across the whole frequency range.

In fact, driving the HD600 with a Valhalla amp was perceived as better sounding than an HD558 driven by a lower impedance amp such as a Focusrite 2i2. Distortion results showed that the HD558 driven by the 2i2 had significantly more distortion than the HD600 + Valhalla combination as well.

So while it is true that psycho-acoustics are responsible to some degree for audio quality perception, in this case distortion numbers seemed to have correlated well with audio fidelity perception.
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Solderdude

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Re: Classic Tube Amp Sound
« Reply #49 on: August 09, 2013, 07:48:24 PM »

So while it is true that psycho-acoustics are responsible to some degree for audio quality perception, in this case distortion numbers seemed to have correlated well with audio fidelity perception.

Indeed that is very true hence the remark '(below certain levels)'  I put in that sentence
The perception bit is responsible for the complete dissonance (below certain levels) between measurements and perceived differences

I consider a THD distance of -40dB (1%) to be audible and below -60dB (0.1%) debatable.
Anything below -80 dB (0-01%) cannot be detected IMO.
It also depends on the distribution in the frequency range (spread of the higher orders).
Tube doesn't add horrible ones but IM may well have increased dramatically as well (needs to be measured to confirm) and IM does NOT sound pleasant at all.
Also one needs to bear in mind that all frequencies in a music signal will produce their harmonics and they will 'add'.
For that reason alone it pays to stay below -80dB which fortunately isn't that hard to make.

In case of the HD558 and Valhalla this -40dB border is reached so indeed audible. It is caused by non linearities in the design (including IM being high ?) and the 'muddying' is also because of the increase in bass (due to voltage division and the dramatic increase in impedance) in the 'boomy' area so triple nasties...

With the HD600 connected it all stays below the 'minimal' levels in the all important midrange area so sounds fine as the HD will only add pleasantness to the sound.
The HD558 simply is not as good a performer (THD wise) as the HD600 is, though the HD558 is still a more than decent headphone, especially FR wise.
It shows FR is important but not the ONLY important factor, THD and impedance are also important factors to be considered.
You measurements have shown this quite well which is a good thing.
« Last Edit: August 09, 2013, 08:00:26 PM by Solderdude »
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