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General debate thread about audio measurements

SIY

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But the standard measurements we make tend to be measuring single tones.

Who is "we"? Have you bothered to actually look at the measurements that people do in the 21st century?

And is Fourier denial a thing again?

This is a tired and inaccurate objection.
 

Jakob1863

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That the difference between blue and red lines is due to differing understanding of the brain function.
Ah..

If you meant it wrt to the function of the whole brain function then i´ll withdraw my comment. :)

I read it more as related to the (still quite complex) interaction between the physiological hearing apparatus and the higher brain functions.
The difference between the blue (which was in the old standard) and the red line is profound.
These graphs should be accompanied by a caution line expressing that the lines in it are based on certain specific experimental conditions and (usually) a quite small sample size. not to mention additional influence coming from (sometimes very subjective) statistical considerations.

If we would really know about the "how and why" our results weren´t depending on these conditions as we would know .... within the boundaries of precision.
 

Frank Dernie

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Thanks for your response. I visited England once. I saw the Buckingham palace from outside. :)
I see what you are saying. That's a concept that's given me pause too. We have signals that are already beyond the threshhold of audibility and even if they aren't can impact the ear very little. So if the signals are below audbilty what does it matter. For me that's a strong argument. However even very cheap gear can reach this level. So even though I can't prove it, it's never proved true for me that things sounded the same.

I'm willing to venture something that contradicts what I usually say about audbility. That is technically the two signals aren't the same. They are different. It's hard for me to imagine that would make a difference though due to audibility.

But the standard measurements we make tend to be measuring single tones. Except perhaps the IMD test. Perhaps something changes during music playback. I know the DACs/preamp impedance also can matter. The Yggy first generation didn't do too well with 600 ohm impedance. A lot of dacs now are used as preamps to power amps as well.
I don't see a whole lot about timing in standard measurements. I think maybe we see impulse response.

I had a question since you seem to have been involved in record players. I notice that LP's do not have ringing or a digital filter they have to go through while (most) digital audio does. The LP players do have an RIAA filter and other distortions but they don't have a digital filter and ringing. Also unlike DS DACs they don't have things like taps and ways to approximate the waveform produced. What are your thoughts on that?

What two signals aren't the same?

In a linear system, which is what we are trying to achieve, testing with standard tones should be fine, as long as Fourier is right. Such tests may fall down a bit testing something non-linear, but that is junk anyway.

RIAA is an equalisation curve rather than a filter (though the 1976 curve does sensibly include a high pass filter to remove overload due to spurious subsonic cartridge output).
The standard RIAA equalisation curve was made available in 1954 (IIRC), up until then different recording companies used (slightly) different equalisations to each other.
I don't know if the ringing seen from standard reconstruction filters is a problem/audible but I doubt it based on its frequency and duration. The Shannon Theorem shows that the digital system precisely reproduces the waveform, not approximately, as long as the frequency content doesn't exceed half the sampling frequency, which is easily achieved.

LPs have MUCH bigger shortcomings than that which are well and truly in the audible frequency range and duration!
They have been well understood for decades but are conveniently ignored or glossed over frequently nowadays. Several of the shortcomings are because of the compromises necessary to manufacture the disc, which means even if you could make a perfect record player some shortcomings would still be there from the LP itself.
All 4 of my record players do sound very nice (if different to each other) but when I had a range of DACs to listen to here a few years ago any difference between them was vanishingly small and inconsequential or non-existant and they were all from well known makers and the most expensive was 10x the price of the least.

In terms of listening to things I would say that anybody who makes sound recordings will know that digital is audibly more accurate than analogue. I started with a valve mono reel to reel recorder in the 1960s and worked up to a Revox B77, which I still have, but however carefully I set levels I could always tell the off-tape sound from the microphone feed. The first digital recorder I used (StellaDAT) produced recordings indistinguishable to the microphone feed in comparison.
Mind you, a bit of tape saturation sounds quite nice and a plugin simulating it is popular with some recording engineers today, which is a bit of a laugh!
 

Blumlein 88

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Ah..

If you meant it wrt to the function of the whole brain function then i´ll withdraw my comment. :)

I read it more as related to the (still quite complex) interaction between the physiological hearing apparatus and the higher brain functions.
The difference between the blue (which was in the old standard) and the red line is profound.
These graphs should be accompanied by a caution line expressing that the lines in it are based on certain specific experimental conditions and (usually) a quite small sample size. not to mention additional influence coming from (sometimes very subjective) statistical considerations.

If we would really know about the "how and why" our results weren´t depending on these conditions as we would know .... within the boundaries of precision.
http://www.sengpielaudio.com/Fletcher-MunsonIsNotRobinson-Dadson.pdf

I believe that blue line is from the 1950's work of Robinson and Dadson. The above page shows those vs Fletcher-Munson. The red lines are of a recent ISO standard. They all differ somewhat.

Fletcher-Munson used tones over headphones. Later work by Robinson-Dadson used a centered loudspeaker in an anechoic chamber.

Recent ISO curves altered the curves based upon more recent testing. Surprisingly the recent work also with loudspeakers is closer to Fletcher_Munson results which were from headphones. Of course those curves will vary if sound isn't from straight ahead and vary for each person depending upon the angle.

I fail to see the relevance in the recent discussion on measurements beyond knowing conditions of testing and using them appropriately. The general trends across frequency and how our sensitivity varies with level are all very similar and the part of this worth knowing. Such curves aren't for making .1 db adjustments to some signal or another.

https://www.iso.org/standard/34222.html
 

garbulky

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Who is "we"? Have you bothered to actually look at the measurements that people do in the 21st century?

And is Fourier denial a thing again?

This is a tired and inaccurate objection.
I only look at measurements made in the16th century. If it's not done by abacus it's not worth it.
 

garbulky

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What two signals aren't the same?

In a linear system, which is what we are trying to achieve, testing with standard tones should be fine, as long as Fourier is right. Such tests may fall down a bit testing something non-linear, but that is junk anyway.

RIAA is an equalisation curve rather than a filter (though the 1976 curve does sensibly include a high pass filter to remove overload due to spurious subsonic cartridge output).
The standard RIAA equalisation curve was made available in 1954 (IIRC), up until then different recording companies used (slightly) different equalisations to each other.
I don't know if the ringing seen from standard reconstruction filters is a problem/audible but I doubt it based on its frequency and duration. The Shannon Theorem shows that the digital system precisely reproduces the waveform, not approximately, as long as the frequency content doesn't exceed half the sampling frequency, which is easily achieved.

LPs have MUCH bigger shortcomings than that which are well and truly in the audible frequency range and duration!
They have been well understood for decades but are conveniently ignored or glossed over frequently nowadays. Several of the shortcomings are because of the compromises necessary to manufacture the disc, which means even if you could make a perfect record player some shortcomings would still be there from the LP itself.
All 4 of my record players do sound very nice (if different to each other) but when I had a range of DACs to listen to here a few years ago any difference between them was vanishingly small and inconsequential or non-existant and they were all from well known makers and the most expensive was 10x the price of the least.

In terms of listening to things I would say that anybody who makes sound recordings will know that digital is audibly more accurate than analogue. I started with a valve mono reel to reel recorder in the 1960s and worked up to a Revox B77, which I still have, but however carefully I set levels I could always tell the off-tape sound from the microphone feed. The first digital recorder I used (StellaDAT) produced recordings indistinguishable to the microphone feed in comparison.
Mind you, a bit of tape saturation sounds quite nice and a plugin simulating it is popular with some recording engineers today, which is a bit of a laugh!
Very nice! Glad you are enjoying it. It probably doesn't matter as you've addressed it, but I was talking about comparing two digital devices.
,Re: StellaDAT that was back in the 80's right and used digital audio tape? Was it a 16 bit or 14 bit unit? Any info you care to share about it? I see it also has balanced inputs.
Looks like a pretty TOTL machine back then.
used_item176a.jpg

I actually had not too long ago a nice DAT recorder. Unfortunately it was more cumbersome than I wanted to deal with, I had to buy DAT tape etc so I don't have it anymore. It was 16 bit 44.1 khz.
 

Jakob1863

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http://www.sengpielaudio.com/Fletcher-MunsonIsNotRobinson-Dadson.pdf

I believe that blue line is from the 1950's work of Robinson and Dadson. The above page shows those vs Fletcher-Munson. The red lines are of a recent ISO standard. They all differ somewhat.

Yes, the former standard was the Robinson/Dadson family of curves, the new ones are neither Fletcher/Munson nor Robinson/Dadson, that was afair what Sengpiel pointed out.

Fletcher-Munson used tones over headphones. Later work by Robinson-Dadson used a centered loudspeaker in an anechoic chamber.

Recent ISO curves altered the curves based upon more recent testing. Surprisingly the recent work also with loudspeakers is closer to Fletcher_Munson results which were from headphones. Of course those curves will vary if sound isn't from straight ahead and vary for each person depending upon the angle.

And there are newer publications where the authors argue that good reasons exists to revise the current ISO standard (i.e. ISO 226:2003), but the current one was confirmed a couple of years ago.

I fail to see the relevance in the recent discussion on measurements beyond knowing conditions of testing and using them appropriately. The general trends across frequency and how our sensitivity varies with level are all very similar and the part of this worth knowing. Such curves aren't for making .1 db adjustments to some signal or another.

https://www.iso.org/standard/34222.html

Amirm posted the ISO case as support for his argument and i thought that it illustrates too why it would be so important to know exactly "why and how" our hearing sense/Brain combination processes the sound fields.

Ah, you´ve linked the ISO page and there are some conditions listed for better understanding/interpretation of the curves. The wikipedia article now also links to a very informative paper by Suzuki et al. about the work leading to the new standard and about where our current understanding is still lacking:
Precise and Full-range Determination of Two-dimensional Equal Loudness Contours

In a broader sense, as seen in this discussions about the argument "if it´s below the thresholds of audibility it can´t be heard" , the specific conditions are routinely not mentioned (often not even noticed) but categorical conclusions were drawn.
As said before that argument is tautological, if the presenter really knows about the audilitiy thresholds of a specific human, but usually he only refers to the known thresholds of audiblity in general, which by definition are most likely not meeting those of that specific listener.
So it should be used with care. :)
 

Frank Dernie

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Very nice! Glad you are enjoying it. It probably doesn't matter as you've addressed it, but I was talking about comparing two digital devices.
,Re: StellaDAT that was back in the 80's right and used digital audio tape? Was it a 16 bit or 14 bit unit? Any info you care to share about it? I see it also has balanced inputs.
Looks like a pretty TOTL machine back then.
used_item176a.jpg

I actually had not too long ago a nice DAT recorder. Unfortunately it was more cumbersome than I wanted to deal with, I had to buy DAT tape etc so I don't have it anymore. It was 16 bit 44.1 khz.
It was the standard 16/48 of early DAT. I do have a Pioneer too which can have 16/96 at twice tape speed.
The StellaDAT was very obviously better than a reel to reel tape recorder so I didn't do extensive comparisons I was so delighted. I still have it but the control EPROM seems to have lost its programme :( I use the PIONEER D-C88 when I replay DATs now.
On the type of music I record the digital recorders are transparent. I can't tell the difference between the microphone feed and the recorder output so comparing 2 digital devices just reveals they are both transparent and the same. The DAC in my recorder sounds exactly the same as the Linn Klimax DS level matched. There -may- be a slight difference on the mechanical pedal sound on one piano recording I have, but that was the only minute difference I heard and am pretty sure I couldn't tell which was playing if I had not just listened to the other.
Maybe I have cloth ears :)
OTOH I do hear the differences between speakers, cartridges and turntables easily enough.
 
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garbulky

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This is a reply to "amirm" about his comments on Schiit Ygg vs. Benchmark DAC3. Disclaimer: I do not have any commercial interest in Schiit Audio or Benchmark; I am just a music lover (classical music only), do not even qualify as an audiophile (because machines for me are just means to the end of listening to music; have no interest in constant changes to audio equipment). Current setup: only CDs (redbook): Yggy (DAC) -> Mutec MC-3 USB reclocker + Mutec Ref 10 Master Clock -> Rogue Audio RP-7 Preamp (tube)-> Cary specially made stereo tube amp (75W/channel) -> Revel F208 speakers. Amirm: thanks for your strong views about the superiority of Benchmark DAC3 over Yggy (not upgrade to Analog 2 yet), I ordered a Benchmark DAC3 DX. I was in a hurry because I would not send my Yggy for a $550 Analog 2 upgrade (plus shipping) IF Yggy is in fact inferior to Benchmark DAC3. Merely 20 minutes of comparison, I can tell you that your measurements do not mean anything to me because I rely on my own ears. DAC3 cannot even hold a candle to Yggy even if it may have the best measurements in the world. I find it barely listenable - the sound is strained and edgy; switching to Ygg, the sound opens up enormously and becomes musical. Mind you, I treasure transparent sound and do not love the traditional tuby sound at all (the Rogue Audio RP-7 pre-amp and the Cary amp are quite transparent, not those of warm or lush tuby kind). After I remove the RP-7 preamp and let DAC3 drive the Cary amp directly, the sound gets significantly better, but still not in the same league of my Yggy/PR-7 setup - part of the reason may be that DAC3 does not have enough juice to drive a relatively low-powered Cary (the Revel F208 needs more power than 75W/channel - I am waiting for a pair of good transparent tube monoblocks). After 20 minutes of comparison, the difference was so startling obvious that I promptly shipped the Yggy for Analog 2 upgrade today. Will return DAC3. Interesting comparision: Benchmark DAC3 weighs about 4-5 pounds and about 1/6 of the size of Yggy; Yggy weighs about 20 pounds. Size does matter here. Do not know or care anything about technology involved. I am not interested in having a debate and am not partisan to any brand or company. I have much more interest in talking about classical music (I listen mostly to Bach - classical music is my religion so to speak).

Hi Ray, I hope it's okay I post here so that I don't start a back and forth debate in a thread not meant for that.
I find your post interesting. I am subjective in the way I evaluate my gear which sounds similar to how you do it. I think it's neat that you actually put your money where your mouth was and ordered the Benchmark DAC3 DX. Anyway, now we have a subjective comparison. Most on this forum would discount it because it was not double blind, but those people are not me.

In case you are wondering about the value of measurements to correlate with what you hear, I found an eye opening thread here.
This one: https://www.audiosciencereview.com/...eed-an-external-dac-headphone-amplifier.4116/
Here Amir did some measurements and found that the Ygdrassil didn't measure as well as his laptop onboard soundcard.

index.php

So my thoughts were wait does that mean it sounds about as good as a laptop headphone jack?
Then I went further and noticed that the Oppo-205 had beat the Benchmark DAC3! Holey moley! In fact it's the best measuring unit out there!
I just happened to have spent quite a lot of time with the Oppo-205's (using its analog output) both with and without external preamps. And well needless to say I thought it was one of the more mediocre dacs I've heard. So at that point, I think I realized there's only so much these measurements are going to tell me about what satisfies me in audio. Now it still provides important information but expecting to ascribe a better performing dac to better sound or a worse performing dac to worse sound is futile.
 

amirm

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So my thoughts were wait does that mean it sounds about as good as a laptop headphone jack?
Then I went further and noticed that the Oppo-205 had beat the Benchmark DAC3! Holey moley! In fact it's the best measuring unit out there!
I just happened to have spent quite a lot of time with the Oppo-205's (using its analog output) both with and without external preamps. And well needless to say I thought it was one of the more mediocre dacs I've heard. So at that point, I think I realized there's only so much these measurements are going to tell me about what satisfies me in audio. Now it still provides important information but expecting to ascribe a better performing dac to better sound or a worse performing dac to worse sound is futile.
You can't keep swearing on a bible that God exists to an atheist.

You keep using evaluation methodology that is NOT recognized in audio science as a counter for data generated under audio science. You can't rationalize these two. It is like demanding that your doctor explain why it is that people say drinking lots of carrot juice cures their cancer. He is just going to throw you out of his office.

If you want to share your subjective experience, it needs to be done under controlled listening experiments. Anything else is a waste of bits in our forum. Match levels, have a loved one switch the sources without you knowing 10 times in a row. Then compare your results with their log of which is which. Only then you should bring that result to us.

Anything else is living in the Matrix and thinking that is steak you are eating. :)

I have sat through a AB test of Yggdrasil to Oppo 105 and there is no way, no how you would be able tell the two apart let alone have a preference for one.
 

garbulky

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You can't keep swearing on a bible that God exists to an atheist.

You keep using evaluation methodology that is NOT recognized in audio science as a counter for data generated under audio science. You can't rationalize these two. It is like demanding that your doctor explain why it is that people say drinking lots of carrot juice cures their cancer. He is just going to throw you out of his office.

If you want to share your subjective experience, it needs to be done under controlled listening experiments. Anything else is a waste of bits in our forum. Match levels, have a loved one switch the sources without you knowing 10 times in a row. Then compare your results with their log of which is which. Only then you should bring that result to us.



I have sat through a AB test of Yggdrasil to Oppo 105 and there is no way, no how you would be able tell the two apart let alone have a preference for one.
I don't disagree with you, Amir. I'm not bringing any kind of (scientific) results or claiming anything scientific here. Subjective listening in and of itself isn't really scientific.

I was directing my comments to this other guy who happens to listen subjectively too, so maybe it makes more sense to him.

Anything else is living in the Matrix and thinking that is steak you are eating. :)
Sorry but gotta disagree with you there. You don't have to do DBT experiments to enjoy the sound or like a particular device. In fact people do it very well and every time they claim they like something better, they are 100% right (to them).
 

amirm

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You don't have to do DBT experiments to enjoy the sound or like a particular device.
The discussion is never about enjoying a device. People enjoy listening to their phones far more than people enjoy listening to high-end audio.

The discussion is enjoying additional fidelity. That additional fidelity as with the Matrix, is not there in reality. It can be demonstrated anytime you are willing.
 

garbulky

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The discussion is never about enjoying a device. People enjoy listening to their phones far more than people enjoy listening to high-end audio.

The discussion is enjoying additional fidelity. That additional fidelity as with the Matrix, is not there in reality. It can be demonstrated anytime you are willing.
It's already been demonstrated to me... I fail even sighted level matched AB tests. I highly doubt I'll do any better on blind AB tests
 

trl

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It’s all about the type of distortions we like more (https://www.electronics-tutorials.ws/amplifier/amp_4.html) and their level, about the level of 2nd and 3rd harmonics, the level of odd and even harmonics, channel imbalance (yes, 0.5dB of channel imbalance may “improve” the soundstage because simply changes the song).

Analog/TI engineers wrote some real facts here http://www.ti.com/lit/ds/symlink/opa604.pdf:
“[...] Op amp distortion can be considered an internal error source which can be referred to the input.” - again, it’s about the type and level of distortions we like most.

“[...]
The sound quality of an op amp is often the crucial selection criteria—even when a data sheet claims exceptional distortion performance. By its nature, sound quality is subjective. Furthermore, results of listening tests can vary depending on application and circuit configuration. Even experienced listeners in controlled tests often reach different conclusions.
Many audio experts believe that the sound quality of a high performance FET op amp is superior to that of bipolar op amps. A possible reason for this is that bipolar designs generate greater odd-order harmonics than FETs. To the human ear, odd-order harmonics have long been identified as sounding more unpleas- ant than even-order harmonics. FETs, like vacuum tubes, have a square-law I-V transfer function which is more linear than the exponential transfer function of a bipolar transistor. As a direct result of this square-law characteristic, FETs produce predominantly even-or- der harmonics”
- so, of course tubes will produce a more lovely sound, same applies for the FET-input opamps.

It’s clear that we love added harmonics, sometimes phase shift too, this is happening on the Opera stage, in rock concerts...because it’s normal for the sound to bounce from the wall’s caves and to please our ears more than the neutral sound of an isolated studio.

Now what would be better: to feed our ears & brains with distortions and added harmonics or to listen to the clinical sound of a studio recording (read it: good sounding vs. perfect measuring equipment)? :)
 

Sal1950

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Now what would be better: to feed our ears & brains with distortions and added harmonics or to listen to the clinical sound of a studio recording (read it: good sounding vs. perfect measuring equipment)?
That's up to you.
Do you want to listen to a SOTA High Fidelity rig, or to something along the lines of a 1930 triode tube amp? Come by and I'll let you check out the glorious tone of my 1925 Atwater Kent.
 

trl

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No, thank you, I preffer to keep out of “drugs” and stick to the clinical sound of...Objective2 perhaps.

Tubes sound, full of harmonics, is like living inside a Matrix and getting out of the there needs help from AA (Audiophilus Anonymus). :)

However, if I’ll ever stick to a tube amp, then I’ll pick some from Garage1217...kinda picky and subjective oppinion perhaps.
 

Cosmik

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Could we put to bed the idea that it is possible to create harmonic distortion? If I play three tones simultaneously and add second harmonic distortion, what should I get? Presumably three pairs of tones - is the fantasy. But in order to do this, I would need to separate the three tones, individually synthesise the harmonics, then add them all back together again.

What would it mean to add second harmonic distortion to a violin section playing twenty different notes, or all the same note but ever-so-slightly different? While the trombones are sliding up and down at the same time?

The reality is that it is only possible to generate harmonic distortion for a single tone, and this is done using a 'bent transfer function'. The same bent transfer function on two tones, or twenty tones, produces garbage.

With DSP it might in some circumstance be possible to analyse the signal and split it into its components - but a valve isn't doing this.

In the recording studio, the producers have access to individual feeds and so 'harmonic distortion' might be more viable - on a solo voice, for example. The consumer can only do the same trick by choosing simple recordings - which explains much of the audiophile canon perhaps?
 

pkane

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Could we put to bed the idea that it is possible to create harmonic distortion?

I'm a bit confused by this. Harmonic distortion is produced by a non-linear transfer function. It's an effect and not the cause. It is possible to simulate various non-linear transfer functions, symmetric or asymmetric, in DSP. This can then be applied to any complex piece of sound or music you'd like, not just simple sine waves. If you do feed it a single tone, you'll see even and/or odd and higher order harmonics being added in an FFT plot. The result will be much more complex when used on a complex waveform. I don't see anything strange or impossible here. Are you saying this can't be done?
 

Cosmik

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I'm a bit confused by this. Harmonic distortion is produced by a non-linear transfer function. It's an effect and not the cause. It is possible to simulate various non-linear transfer functions, symmetric or asymmetric, in DSP. This can then be applied to any complex piece of sound or music you'd like, not just simple sine waves. If you do feed it a single tone, you'll see even and/or odd and higher order harmonics being added in an FFT plot. The result will be much more complex when used on a complex waveform. I don't see anything strange or impossible here. Are you saying this can't be done?
I am saying that a nonlinear transfer function generates intermodulation distortion when applied to complex signals - not harmonic distortion. For sure, you can have any transfer function you want using DSP, but it won't be 'harmonic' when applied to a complex signal.
Intermodulation (IM) or intermodulation distortion (IMD) is the amplitude modulation of signals containing two or more different frequencies, caused by nonlinearities or time variance in a system. The intermodulation between frequency components will form additional components at frequencies that are not just at harmonic frequencies (integer multiples) of either, like harmonic distortion, but also at the sum and difference frequencies of the original frequencies and at sums and differences of multiples of those frequencies.
 
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