When do measurements correlate with subjective impressions

[pkane2001]

2 hours ago, 4est said:

Realize this is a question about objective measurements seeking an objective outcome to subjective experiences.

 

That's doable. In fact, you can even do some of this for yourself, if curious. Quite a bit of content regarding this is available on ASR, but not much here, on AS. I can share some of the more relevant links, or just go there and search.

 

[pkane2001]

9 minutes ago, 4est said:

I was hoping to do this here on AS tbh, but feel free to link their pages. I personally don't know how to measure exactly what I seem to hear between components. When I hear various DACs referred to as transparent it gives me pause. IME I hear trends and differences in them, both perhaps related to the analog aspects. I am looking to be brought up to date in order to understand, whilst kick starting the ob-fi side of things.

 

Here's a good start:  https://www.audiosciencereview.com/forum/index.php?threads/audibility-thresholds-of-amp-and-dac-measurements.5734/post-127757

 

Don't go too deep into the thread, but the initial few pages should help you get going.

 

If you want to figure out levels of SINAD, THD, IMD, etc. that are audible to you, then try my DISTORT app:

 

https://www.audiosciencereview.com/forum/index.php?threads/beta-test-distort-audibility-of-distortions.10163/post-277173

 

Or you can go for Archimago's THD audibility blind test. In fact, do that first

 

 

[pkane2001]

17 minutes ago, fas42 said:

What the truth is, that the majority of people on both sides of the "battle" don't want to hear, is that not enough is measured about the performance of the system overall - and that the insertion of a different component normally alters the 'balance' of any higher resolution rig, because the engineering is always too inadequate to ensure that this doesn't happen. The fantasy is that all you have to do, using one of Paul;s beloved analogies, is to just stick a bigger engine under the hood - and your performance vehicle will always be better in every possible way ... any automotive engineer who believes this will be out the door so fast that his feet won't ... . Because, the car is a system , and every subsystem highly likely will have to be touched to ensure that the full potential of the greater power can be fully realised.

 

Until people learn to measure "properly", the debacle will continue - the answer is that the system has to be measured, not the component.

Can always count on you, Frank, to set the record straight! And to use my favorite car analogy in the process

[pkane2001]

2 hours ago, Miska said:

 

Lot of the objective reviews are just like amplifier measurements in 70s. Companies made amplifiers that had very low THD and IMD figures, yet they sounded bad. And engineers were puzzled as why because everything was measuring perfectly. Then one Finnish guy, Tapio Köykkä wrote about distorted discontinuous sounds, that were working fine on older style tube amps (for example his Circlotron-type tube bridge amp), although those had notably worse THD and IMD figures. Then other people found out about the same too. This was picked up by team of three on Technical Research Centre of Finland; Eero Leinonen, Matti Otala and John Curl, who devised a measurement method for this new type of distortion called TIM (Transient Intermodulation Distortion).

 

Then Otala & Lohstroh also designed a solid state amplifier that didn't exhibit this problem. Based on this amplifier design, a new Norwegian amplifier company actually appeared on the market, called Electrocompaniet. Later on, Otala moved on for example to work with Harman/Kardon to design the Citation XX power amplifier.

 

Matti Otala is truly someone in terms of electronics in Finland, and I would say also globally.

 

Nowadays most amplifiers behave better in this respect, but I can also see that the important measurement is not being used much these days, so who knows. But I've started using it on DACs, since these days it is possible to perform on DACs too to a reasonable extent.

 

There are also many aspects of a DAC that traditional THD+N and IMD measurements don't cover. Jitter measurements were devised later, but there are many more aspects as well. So it is not possible to conclude about "transparency" based on couple of measured figures, it is much more complex matter, especially for something as squeezed as RedBook. In addition, RedBook content itself has common problems, some that can be fixed by the DAC.

 

 

There aren't that many things that an audio device can mess up. It's either timing (phase), amplitude, or frequency.

 

THD+N captures some of that, especially when plotted against frequency and level. Jitter plots and impulse response cover timing. Spectrum analysis can show frequency anomalies. Ok, so you may not want to do averages over time, so maybe capture peak values for all of these. What else is there?

 

SINAD as a single number, jitter as a number, IMD as a number, etc. are all just very big averages, and therefore can't accurately represent the true nature of a DUT. They are simplifications. A nice THD+N/frequency or THD+N/level chart can provide a ton more information, just like a jitter spectrum or a phase plot.

 

 

[pkane2001]

3 minutes ago, jabbr said:

All I can say is that I’ve tried a few of the ASR recommended DACs, including the Topping and they leave me flat. The measurements don’t correlate with great sound for me.

 

On the other hand @Miska posted some measurements here of certain DACs two of which I love including the Pro-ject  S2D (which benefits from a good power supply) and the IFi iMicro — which has terrific input isolation. 
 

I think the differences I hear — assuming a good basic product, are mostly to do with the output stages. 
 

For me, the measurements need to have a predictive value in terms of what I hear. 

 

Haven't tried Topping DACs.  I do have a range of others, from pro to R2R. In most cases, I can't really hear the differences between properly level-matched, modern DACs with good measurements.

 

Some older ones I have from the 90's do generate more noise and more obvious distortions. A more recent (but still old) Emotiva DAC is one where I can hear the differences easily. It doesn't measure well. It also, has the curious quality of being sensitive to the USB cable I use with it. 

[pkane2001]

4 minutes ago, Jud said:

 

Measured differences? In what measures? (Curious as to what if any measurable differences a cable might make as part of a system.) Are there any differences between cables that meet spec?

 

Cable I was testing was Lush^2. It made a small but measurable difference in noise level compared to a no-name USB cable. It actually slightly increased the level of noise at the output of the Emotiva DAC. I didn't find Lush^2 to make any difference with other DACs.

 

I assume the increase in noise had to do with shielding/grounding configuration of Lush^2 picking up EMI or introducing a ground loop. (I was using the stock Lush^2 configuration it was shipped with).

[pkane2001]

54 minutes ago, Bill Brown said:

After lots of years of listening (32! Oh my) and owning tons of gear I have come to believe a few things and still have some doubts about others.

 

Tube amps (which I have owned): they can sound subjectively glorious, but is it because of a euphonic distortion spectrum?  I think that a pattern of decreasing levels with harmonic order is important and that this has been demonstrated.  I also think that 2nd harmonic distortion (more prominent in tube amps) is not easily heard, and when it is, it tends to be euphonic (see Nelson Pass' experiments in this with his SE SS designs and second-harmonic distortion generator).  I also think that higher-order distortions, even at low levels, can be perceived and are unpleasant.  Finally, with regard to tube amps, many have a higher output impedance that interacts with the varying impedance of loudspeakers and produce frequency-response changes that are likely perceptible, and again, may cause subjectively pleasing FR changes.

 

I think that modern SS amps have decreased in distortion (especially higher-order) so much that it is imperceptible.  I own a Benchmark amp.  John Curl and Mattie Otala did significant work in this area and with TIM distortion.

 

But...are there "second-order" effects in amplifiers (or audio gear in general) that can't be measured that are important?  I struggle with this and am simply not sure.

 

I believe the contributions of Toole and Olive have been very important in loudspeaker design, directivity in particular.  I own JBL professional monitors.  But....are there effects with using drivers out of the frequency band in which they are radiating pistonically (Ka > 1)?  Is this affected by cone materials, the way in which they "ring."  Maybe there is.  Charles Hansen was convinced of this and it lead to his design for Avalon back in the day.  It also is a feature of Laurence Dickie's designs (the BMW Nautilus, and now for Vivid Audio) where all operate pistonically.

 

Earl Geddes, the definition of objectivism, has produced wonderful data re. distortion perception (mainly the lack thereof) in loudspeakers, but also developed his own metric for the perception of distortion in SS amplifiers.

 

I have found JA's measurements in Stereophile invaluable in all of the above.  It is clear to me that many of the perceptions of the subjective reviewers must be thought about carefully, though there are some which I have come to know well over the years, whose perceptions correlate with mine, and whom I trust....

 

I have also blind-tested digital filters as I thought I heard a significant difference and believe my thoughts were borne out....

 

I'll point you to my own software, DISTORT, if you want to explore audibility of certain kinds of distortions. For example, you can try out only even harmonics or only odd. Or create a specific custom harmonic mix. Or, add a compressor-type effect. Or change noise floor, etc. I'm adding support for different filter types and phase controls to it, but that's still work in progress

 

[pkane2001]

9 minutes ago, CG said:

I've given an unreasonable amount of thought to this very topic over the years.  I don't have an answer yet - sorry!

 

But, here's a Gedanken experiment to consider...

 

Imagine that you just received the very latest Audio Precision test system, with all the options.  It truly is the state of the art.  You practice using it by making back to back measurements of the internal test generator.  With all the signal averaging on and everything set right, you can see that your AP unit has a residual distortion level of better than -120 dB.  Pretty great!

 

So, you measure your existing power amplifier and find that the 2nd harmonic is only 70 dB below a 1 watt 1000 Hz tone.  That doesn't seem like what you want, so you go buy a new amplifier that claims much lower harmonic distortion.  

 

Before connecting it up, you test the new amplifier with your new test gear.  Yup - the 2nd harmonic on this amplifier is -110 dB at the same power.  40 dB better.

 

Now you connect it into your system in place of the old amp.

 

The first thing you play sounds not so good.  There's a kind of crackling sound every now and then.

 

In engineering terms, WTF?  The amp sure tested pretty well.

 

So, which is wrong?  

 

The AP test system?

 

Or, your ears?

 

It turns out that neither are wrong.

 

Back to the test bench, you take the amplifier cover off and eventually find a wonky connection.  After tightening a screw, you retest the amp.  No difference.

 

Back to the system, you listen again and it's fine.

 

Again, WTF?

 

The AP system was doing exactly what you told it to do.  It applied a single tone and read the amplitude of the amplifier output spectrum over and over.  Loads of sweeps.  Because of the averaging function, you could see distortion way into the noise.  That's what averaging does - it assumes that the fundamental tone and the distortion tones are constant in amplitude, which probably is an ok assumption (maybe) and then assumes that the rest are just random chaotic events that don't repeat either in amplitude or frequency over all those sweeps.  So, much of the noise voltage just gets averaged away.

 

The crud you heard was the result of a bad connection that was random in nature and chaotic.  Some sweeps there was nothing.  Others, something.  But, the spectral content varied all over the place.  Overall, this may have increased the averaged noise floor by a dB or so.  Who can tell?

 

Now, if you had looked at the distortion with averaging turned off, the distortion likely would've been buried in the noise.  But, you'd probably have seen the crackling.  If there was a "peak" or "max hold" function activated, you would have captured the noise spikes.

 

So, it's a matter of what you're looking for.

 

How often have you seen a test result with averaging turned off?

 

Just one example of how incomplete the data set might be.

 

Turn off averaging, and use peak-hold. Problem solved? 

[pkane2001]

7 minutes ago, Miska said:

 

With TIM, the problem was that the measurements being used were operating on static sine waveforms like in THD and IMD. While problem was with transient signals. Thus TIM is measured with mixture of square and sine waves (15 kHz sine + 3.18 kHz square wave at 4 : 1 level ratio, -3 dB at 100 kHz).

 

In these cases it was due to the amplifier going to internal saturation, due to internal bandwidth issues, during transients only.

 

And this happened already with vinyl and open reel tape recordings, for example in crescendos. Problem was found because some engineers believed their ears.

 

 

Do you have any evidence that TIM is a concern in any modern DAC implementations? I thought about adding TIM measurement to DeltaWave (and actually had a test version of this working) but found no useful information in those results.

[pkane2001]

1 minute ago, Jud said:

 

Of course this raises what's somewhat the flip side of good measurements correlating with subjective good impressions - what types and levels of distortion and noise are euphonic for at least some people? (Beyond the usual mention of tube electronics euphony, there are things like the Aphex Aural Exciter, literally a piece of electronics to produce noise and hash, used in the production of Born to Run, for example.) 

 

That's precisely why I wrote DISTORT. Measurements are great, but they are just an engineering result, showing how well a device is working. They don't tell the whole story of what's audible, or what distortions I may find pleasant.

 

DISTORT lets me apply controlled levels of various distortions (in any combination) to any music track so I can determine if it rises to audible level while listening on my "perfect" system. Anyone can do the same with this app, with their music, in their system.

 

@Archimago's latest internet blind test is an attempt to do this on a larger scale, gathering input on audibility of various levels of THD. 

[pkane2001]

1 minute ago, Miska said:

 

Time will show when I have collected enough data around it. But there are certainly differences between DACs in this respect and the components are in the audio band.

 

Highest figures I've measured have components around -65 dB, and it is not unusual to have them around -100 dB. So within range I know people can hear.

 

 

Would be curious to know which DACs show such a high level of TIM. Might be interesting to see how (or if) it affects other measurements and metrics.

[pkane2001]

3 minutes ago, CG said:

Maybe.  I think it would help in this one example, but that's just my opinion.  Doing this would not only obscure distortion measurements in many cases, but it would also freak a lot of people out.

 

Have you ever seen the results of that published anywhere?

 

And, just imagine the ensuing arguments!

I’ve certainly used that mode with oscilloscope, and it does look a lot more ugly than the average, but it does represent reality. Maybe it should be used in more cases and with some published results.

[pkane2001]

1 hour ago, Jud said:

@pkane2001, I was wondering three things:

 

1. Are you reasonably sure from measurements and/or listening that the jitter in DISTORT sounds like the jitter in actual DACs?

 

2. What (other) forms of distortion does jitter cause?

 

3. Can you provide a subjective description of what jitter and the distortion(s) it causes sound like?

 

First question is pretty much for Paul; if others have reasonably *precise* answers, feel free to jump in on the other two.

 

Simulation is the simplest way to reproduce an effect for study. Before anyone claims that it's not reasonable to do so, one would at least need to understand what is being simulated and how, and then have a coherent explanation for why the simulation is invalid

(@4est)

 

Jitter is a fairly simple concept. How it affects a DAC analog output is known and can be measured. DISTORT provides a display for what jitter does to the test signal and lets you apply variable amounts of different types of jitter to it. Compare it to measurements published, and you'll see that it can reproduce a really wide variety of jitter found in the wild.

 

DISTORT produces jitter by introducing timing errors into the signal. These errors are very precisely controlled because they are computed at a high numerical precision. In fact, you can easily apply below femto-second jitter using DISTORT, or use a 0.0001Hz frequency to modulate the clock.

 

Here's the deal: I'm not going to tell anyone what jitter sounds like or if and how much I can hear. That's not very interesting to anyone. Try it yourself and you tell me.

 

All I can say is that it becomes an audible problem in my testing only when the amount is really large. Femto-second precision isn't necessary in a clock, but maybe your ears and your equipment are so much better than mine that you'll hear it. So, try it! 

 

[pkane2001]

3 minutes ago, Jud said:

 

It's interesting to me, and I'll tell you why.

 

You ask someone new to astronomy to watch through a telescope for an asteroid. They ask "What does one look like through a telescope of this power? What should I watch for?" Your reply is "That's not very interesting to anyone. Try it yourself and you tell me." What are their chances of spotting an asteroid?

 

Replay that last conversation. Your response now is "Let me show you several examples so you get a sense of what to watch for." What are their chances of spotting an asteroid now?

 

So Paul, are you hearing the effects of jitter only at higher levels because it's only ever audible to humans at those levels, or because no one has trained you to notice what lower levels of jitter sound like?

 

What's the training effect? It seems to me this becomes important in determining whether we can be subconsciously affected by levels of distortion we haven't been trained to consciously notice.

 

Ha! You had to bring astronomy into it!  With the right telescope, an asteroid can be very easy to spot, even for a novice. Think of DISTORT as a very good telescope

 

Different types of jitter sound different, different amounts also sound different. But how can I tell if I can't hear low-level jitter because of lack of training or because it's not audible to me, or inaudible to everyone? How could I possibly know the answer to that? 

[pkane2001]

42 minutes ago, CG said:

Of course, I have no idea what the severity of that might be on the sound.  Lots and lots of variables.  So, no claims from me.  But, I know guys who made experiments and have found that very low frequency phase noise on the clocks seems to have an audible effect. By very low, I mean sub Hertz.  

 

That's where DISTORT can help. You can apply sub-Hertz modulation to a clock and see what sounds like. Apply it to a simple sine wave or to a full orchestra recording. You decide how much of each type of jitter to apply, from simple sine-wave, to random, to 1/f noise, to correlated and then listen, see if you can tell the effect.

[pkane2001]

Just now, jabbr said:

You are really trying to get me to install Windows aren’t you?

 

I heard others had good luck using it with Wine on Linux and Mac.

[pkane2001]

14 minutes ago, Bill Brown said:

Uh-oh...didn't realize it was Windows-based.  I am a Windows (and computers in general) moron

 

How would drinking wine help me on a Mac.  Just kidding.

Wine helps in all cases

[pkane2001]

14 minutes ago, jabbr said:

Hmm loaded wine, got the app to show up on screen (Ubuntu 18.04 LTS) but can’t get it to easily show results... wine is a weird environment (as expected) ...

 

It would be cool to load a file containing a phase noise plot and run that against the music file.

 

So it's just not displaying results or nothing is working? I'll spin up an Ubuntu VM and see if I can get it to work.

[pkane2001]

6 hours ago, jabbr said:

Hmm loaded wine, got the app to show up on screen (Ubuntu 18.04 LTS) but can’t get it to easily show results... wine is a weird environment (as expected) ...

 

It would be cool to load a file containing a phase noise plot and run that against the music file.

 

So I installed Ubuntu 18.04 and put Wine 5 on it, then installed Distort. Seems to work OK, or did you also get this far with yours?

 

[pkane2001]

Just now, jabbr said:

I got that far, then tried to load a music file.

Ok, didn’t spend any time testing on Wine, let me see if I can figure out what’s not working.

[pkane2001]

3 hours ago, barrows said:

next answer to the question of the thread:

 

When the component is the Mola Mola Tambaqui DAC.  See measurements here:

 

https://www.audiosciencereview.com/forum/index.php?threads/mola-mola-tambaqui-dac-and-streamer-review.10770/

 

That -110dBFS peak at 50kHz needs to be worked on! I'm not going to pay that much money and still have distortion at that level!

Well, I probably wouldn't pay that much money for a DAC, period, but that really is one well-measuring DAC.

 

[pkane2001]

6 hours ago, jabbr said:

I got that far, then tried to load a music file.

 

I can load and process a file (click Save...) I can't play a file yet, because the audio library I'm using requires .NET 4.0, which isn't part of Wine 5 install. What happens when you try to generate a file? Just pick a WAV file you want to process, then pick the desired distortion, then click Save... button to generate the distorted file. I used Audacity to play it on Ubuntu.

 

I'll see if I can change the target .NET framework for the library, or install .NET 4.0 on Wine -- using it for the first time, so not remotely an expert 😄

 

[pkane2001]

2 minutes ago, barrows said:

Come on man, you are smarter than that right?  That peak is due to the slightly relaxed roll off of the digital filter, and would not ever be present with a music signal. But if you are still concerned you could always over sample to a 2x rate in software with steep filter to eliminate it.

 

That was a joke, sorry. I thought the smirking smiley face would give it away, but apparently I need to find better emojis.

 

[pkane2001]

45 minutes ago, cat6man said:

 

I have to disagree with much of the above.

 

1.  i have spent 30+ years simulating complex digital telecommunications systems and, while often causing much consternation among my colleagues over this time (everyone like easy answers), i have found that MOST simulations oversimplify the problem and answer a question or problem, but via oversimplification or by not incorporating something (known or unknown), they do not solve THE question or problem.

do you have a realistic jitter spectrum?  what is the PDF (probability density function of the jitter)? what is the reconstruction filter?  it is technically obvious from the math (don't panic, not included here) that the combination and interaction of the jitter with the reconstruction filter are intimately related, so how is this handled?  some filters are not, in fact, even linear as some have lookup tables for filter coefficients that are data sequence dependent.

 

you may use your program to see if your particular waveform and jitter stimulus is/isn't audible to an individual (and there is value in that) but abstracting that to DACs in general is highly questionable.

 

2.  "Jitter is a fairly simple concept"?  sorry, but that just isn't correct.  jitter has a spectrum, it has a pdf, it may have various forms of correlation, the clock may pick up noise of various sorts which could be random, impulsive, RFI, etc.

 

i will try out your program, assuming it runs under WINE, as i'm curious to see what i can/cannot hear but i know enough about simulations to be suspicious of claiming relevance of a level below which jitter not not matter.

 

let's try a gedanken experiment.  assume i have a 50ps rms jitter clock and there is an impulsive noise (power line spike for example......A/C is pretty crappy) impacting that low level signal 1% of the time and causing the clock off instantaneously by 10x the RMS jitter (500ps).  i'm guessing the average RMS jitter measurement would not change at all but that the DAC would see a 10x timing offset 410 times a second (i.e. 1% of the time)...........maybe that could be audible?  i don't know but it certainly seems possible and something that would likely (?) not be measured with averaging turned on.

 

[technical analogy--feel free to skip]

an example i know intimately from simulation:  in a 4G LTE system, high speed data is controlled by a 'scheduler' that assigns time slots at the mobile and base station to different users.  in addition to assigning time slots to each mobile user, the scheduler must assign from a limited set of control channel slots in order to tell each mobile when its data time slot is coming up.  if there are not enough control channel time slots available to serve all the mobile users at a specific time, that mobile user is temporarily blocked and cannot send or receive data.

the metric commonly used in simulations was average control channel utilization, and it was thought that keeping that value below 70-80% on average was sufficient for good system operation.  however, simulation of many mobile data users with many different data usage profiles showed interesting results.  even with say 50% control channel utilization, the blocking could be as high as 5-20% as the distribution of the number of instantanous users could be highly skewed.  therefore, the solution turned out to be much more complex than expected as the tail of the distribution of #users needing control channel slots dominated overall performance and the average utilization was essentially irrelevant.

just a little non-audio example to show you where i'm coming from.

there is always a simple answer to complex problems, but it will often be wrong.

unfortunately, many people (and upper management) prefer a simple answer to a complex "well it depends" answer.

 

as always, YMMV

 

Sounds like you have some background to help contribute to DISTORT development (it is still work in progress, and will be for a while!)

 

1. MOST simulations oversimplify the problem

 

Yes, that's true. But are you saying simulations are never useful? A simulation is a way to study an otherwise hard to reproduce problem under controlled conditions.

 

DISTORT wasn't designed as a device simulation tool. It's designed as a distortion simulation tool. The purpose is to allow me and others to test the audibility (and visualization) of various types of distortions, in different combination, at different levels. I want an approximation that lets me study this complex interaction of jitter, non-linear transfer function, dither, feedback, phase errors, filters, etc., in the comfort of my own home, without investing hundreds of thousands of dollars into professional measurement equipment. If you have a specific area where you think DISTORT simulation can be made better, I'm all ears

 

2.  "Jitter is a fairly simple concept"?  sorry, but that just isn't correct.  jitter has a spectrum, it has a pdf, it may have various forms of correlation, the clock may pick up noise of various sorts which could be random, impulsive, RFI, etc.

 

Jitter as a concept is the error in the timing of a digital sample. That's simple enough, no? We can talk about complex sources of jitter and the effect in the frequency domain, along with many other measurements all we want, but the basic concept is simple. 

 

Again, I'm not simulating any specific device in DISTORT, but I am giving everyone a chance to play with different types of jitter, from random (white noise), to correlated, to 1/f noise, to arbitrary sine-waves in configurable amounts. If you think there are other signals with other PDFs that should be included, tell me which, and I'll add them. 

 

In fact, if you have a recording of the noise/signal modulating the clock, I can inject the exact jitter signature into any music file.

 

..........maybe that could be audible?  i don't know but it certainly seems possible and something that would likely (?) not be measured with averaging turned on.

 

DISTORT doesn't do any averaging. It injects arbitrary jitter signal directly into a WAV file, adjusting each sample individually. RMS jitter is not something that's used or computed by the software.

 

 

 

[pkane2001]

34 minutes ago, cat6man said:

maybe i could have been a little clearer.  the realistic modelling of jitter in a real system is not simple to me.  applying a different instantaneous time offset to each instance of the chosen reconstruction filter is mathematically simple but the interaction is not.  how does your program implement a digital reconstruction filter?  i do not have any measurements or data.  i'm a theory, not hardware, geek

 

DISTORT uses a polyphase interpolation filter. The final reconstruction filter is in the DAC playing the 'jittered' waveform, and so is not under DISTORT control.

 

34 minutes ago, cat6man said:

i understand your program doesn't measure or average jitter.  however, i assume you specify the jitter you want to add in some average or static sense?

 

Each jitter source is specified individually and can be combined for a total effect. Each is controlled by amplitude and/or frequency. Here's what the configuration looks like:

 

 

34 minutes ago, cat6man said:

could you please post a link here (i'm lazy) to your program?  thanks.  i'd like to see what/if i can hear, plus then i can look at the program instead of asking dumb questions here about what it does.

 

Sure. It's in my signature and here: https://distortaudio.org/

 

It should work under WINE, although I'm not sure about sound playback through WINE device stack. Certainly you should be able to apply distortions to a WAV file, save it, and then play it back using your favorite player/device combination.