Mixing engineer ears are only inside their studio. They know their equipment well more than others. I think he would be laughing from behind. I worked in few studios before. When we hire mixing engineer from other places, they can only guide and exchange ideas. The mix were never that good as the regular guy working there. You might have seen musicians around the world gather to perform music. Ever heard two mixing engineers from different places gather in a new studio to make better mix? Anybody can listen to music and comment. From doctors to school teachers. Can anyone say that they hear 0.5 % distortion in that system when a music is played? I doubt it. Nobody sits in front of their audio system with a distortion measurement device. There is no reference to compare distortion free music and 0.5 % distortion music. If you can listen and enjoy music in a system for half hour, then the system is good. Be it mono, stereo, 5:1 or even a bedroom radio. Many sine wave generators produce with 0.1 to 1% distortion that is not audible. No Hi-Fi distortion snake oil here. Creating music and listening to music are two different things.Regards.
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DNR circuit board found
It uses the LM1894 View attachment LM1894.pdf
Here's the listing on eBay LM1894 noise reduction circuit DNR dynamic noise reduction circuit | eBay
However, it won't fit in the MT-2 enclosure.
It uses the LM1894 View attachment LM1894.pdf
Here's the listing on eBay LM1894 noise reduction circuit DNR dynamic noise reduction circuit | eBay
However, it won't fit in the MT-2 enclosure.
Voltwide's reply covers this very well. Hearing ability will follow a normal distribution (the famous "bell curve"), just like many other statistical variables. Some audiophile shop clerks will have lousy hearing. Some schoolteachers will have outstanding hearing. Some musicians will be deaf ( Evelyn Glennie - Wikipedia ).
But none of the people you test will hear as well as a dog can - our species has limitations built into the anatomy of our ears and the wiring of our brains.
Because hearing ability in a human population follows the mathematics of the normal distribution, that means there will be a mean value (average hearing ability for the species), and there will also be a standard deviation (a measure of how much variation there is from one person to the next.)
The mathematics of the bell curve is such that approximately 68% of people will have hearing that lies within plus or minus one standard deviation of the mean value. Approximately 95% of people will have hearing that lies within plus/minus two standard deviations. Approximately 99.7% of people will have hearing that lies within plus/minus three standard deviations.
This means only 0.3% of people - roughly one in 300 - will have hearing so bad, or so good, that it isn't within three standard deviations of the mean.
If you tested 10,000 people, and 9970 of them could not hear any distortion from your amplifier, you've built yourself a virtually perfect amp. Maybe one in every three-hundred people might hear a problem with it.
It would be very difficult to do better than this - you'd have to test some 100,000 people to get any trustworthy data about 1-in-a-thousand superhumans. At some point, even a well-funded research lab would decide that adding the next decimal point would not be worth the enormous increases in cost.
Audiophiles love to believe they have superhuman hearing, though they usually have no actual evidence to support their belief. In the field of social psychology, this is called "illusory superiority". It's a delusion, a mental flaw, and a very common one - a lot of people think they're better than everyone else at everything. It's a failing of our species, linked to our fragile egos and our tendency to
This CBS article on the subject is a nice read: Everyone thinks they are above average - CBS News
Getting back to the statistics of human hearing, if you want to do good science and get exactly the right numbers, you would have to test every human on earth. That's impossible, obviously, so the best we can do is to test a large number (call it N), and the statistics of the bell curve tells you that your results will not be exact, but have some uncertainty in them, and the uncertainty is proportional to the square-root of the number of tests conducted.
This is why any researcher trying to get good data about human hearing has to conduct a *lot* of hearing tests. If a clueless magazine reviewer does 5 tests, the results are statistically worthless. If you do 50 tests, the results are a tiny bit better, but still very inaccurate, so you can't draw any worthwhile conclusions from them.
In the era when actual researchers working at large and well-funded research labs were studying these things, they did thousands of listening tests, with hundreds of individuals, in dozens of different facilities.
It requires a lot of money and manpower, and large facilities to conduct such tests, so we probably won't see them repeated in the same way. The results have already been determined anyway, and are now part of our collective body of knowledge in the field of audio - why re-invent the wheel?
Agreed. I used to have 20/15 vision when I was younger (I could see at 20 ft distance what the average person could only see at 15 ft distance).
I also had an experience very much like yours, where a colleague found a small dip in a loudspeaker frequency response curve (caused by diffraction effects caused by the bass-reflex port opening on the front of the speaker.) In this case, the big difference was the type of music my colleague liked to listen to: synthesized instrumentals consisting of long sustained sine-wave tones and frequency glides.
It drove me mad to listen to more than a few minutes of the stuff, but those swept-sine tones revealed the little dip in the frequency response which I (and several others) couldn't hear using more normal music. Of course, the measurement microphone picked it out with no difficulty.)
Here you go: Sound Impairment Monitor (SIM) - Is This The Answer?
If the amplifier is poor-quality, or you drive it to clipping, or the protection circuits kick in, then there will be enough difference between input and output signals to measure.
If the amp is really good, the difference is so small that it becomes buried in the noise floor, and cannot be measured. Or it is so far down from the fundamental that it is far below audibility.
It's insane how far amp design has been pushed. Douglas Self's "Blameless Amp", for instance, was rated at 0.0005% THD. If that amp is putting out 10 watts RMS of "good" signal, the power in the distortion signal is 0.0005% of ten watts. That calculates out to 50 millionths of a watt.
A few years ago I found out that we can indeed hear 50 millionths of a watt of power if you're in a quiet room, and you feed that signal into a very efficient guitar speaker (100 dB SPL@1W @1m). Hi-Fi speakers tend to be 20 dB less efficient, so those 50 microwatts are going to be hard to hear now - they only generate a 37 dB SPL from the speaker. This is about the background noise level in a very quiet library - much quieter than the purr of a cat, for instance.
But now we're talking about 10 watts of "good" power going into that same speaker, along with the 50 uW of distortion. The speaker is putting out 100 dB SPL of "good" power - and the distortion is only creating 37 dB SPL of distortion at the same time.
So, can we hear a very quiet 37 dB sound when it's played along with a very loud 100 dB SPL sound? Easy answer, can you hear your cat purr if you're at an airport and there's a jetliner taking off?
In fact, a purring cat is around 55 dB ( How Many Decibels is Your Cat’s Purr? - Cats ), which is much louder than the 37 dB SPL of the distortion from the 0.1% THD amp. And 100 dB SPL is roughly the equivalent of thirty household vacuum cleaners all going at the same time.
Can you hear a cat purring in a room with 30 vacuum cleaners all going at the same time?
Can you hear a sound that's nearly 20 dB quieter than the cat's purr, while the 70 vacuum cleaners are all going at once?
Easy answer, right? There's no way on earth we can be hearing that very faint sound from the 0.1% distortion in our hypothetical amplifier.
When I first started calculating numbers like these, I realized that it isn't the least bit surprising that we can't hear 0.1% THD from an amp even when it's playing at very loud SPL.
Thanks to ITPhoenix for his patience in putting up with this meandering around the actual thread topic.
-Gnobuddy
Thanks to ITPhoenix for his patience in putting up with this meandering around the actual thread topic.
-Gnobuddy
It's quite ok. I was reminded by several of you that I need to pay attention to the entire signal chain, frequently touch base with reality, and the need to study more.
Thanks for the link!! Very interesting article, and I'll be incorporating a SIM into my next build, for sure! I noted, though, that he credits Peter Baxandall for the invention of the subtraction circuit.
If your amp is "good" (< 0.1% THD), then the sum of the harmonics will be ~ 60 db below the fundamental----this is certainly ABOVE the noise floor of most modern amplifiers; I would suspect by at least 20db. Therefore the difference residual from the subtractor will NOT be "buried in the noise floor", and should definitely be able to be measured.
Your points about audibility are well taken; though I do wonder exactly how the researchers introduced the various levels of distortion to come to their conclusion. And I still do not believe that THD measurements are the ultimate test of an amp's performance; therefore two amps performing identical in THD tests can still sound different. I agree that this subtraction method should definitely be utilized for comparisons. Has no one actually ever done this? Hard to believe, as it is such an EXCELLENT idea for a meaningful test.
I first read about it in an interview with Peter Walker, and I thought it was his invention. Perhaps I got it wrong, or mis-remembered. I think I read the interview in the 1980s, some thirty five years ago now!
I think one way was to switch in a signal (pure 2nd harmonic, say) from a second signal generator, and find out if test subjects could hear it. I think this is how it was first discovered that, say, 5% 7th harmonic distortion is a lot more audible than 5% 2nd harmonic distortion.
The frequency response is also very important. Back in 1940 or so, it was very hard to get a truly flat frequency response through the entire audio frequency range, and you could hear those inevitable peaks and dips.
Nowadays, even $5 chip amps are essentially flat from DC to well beyond the highest audible frequency. But amps can misbehave if they're fed very high frequencies that we can't hear directly.
So, to compare two amps for sound quality, you need several things:
- Input signal band-limited exactly the same for both amps (say 10 Hz to 40 kHz).
- All EQ removed, both amps dead flat.
- Volume levels matched to within 0.1 dB (very important, and always omitted in every audio showroom I've ever visited.)
- Both amps switched into the exact same loudspeaker (not two nominally identical loudspeakers a foot apart, as room modes and manufacturing variations can make those sound different.) Also rarely done.
I agree, it's so simple it's brilliant. There is really no argument with whatever is revealed by this test method.
To me, the standard tests (swept sine frequency response, THD, IM, etc) are also extremely revealing, and provide information you can't get from the Baxandall / Walker test. For example, the Baxandall/Walker test doesn't tell you what the actual frequency response of the amp is.
But to appreciate how single-frequency sine-wave testing works, and what it reveals, one has to understand and trust some fairly hard-core mathematical principles (Fourier transform, superposition theorem, etc.) It's a little more abstract, and a little less accessible, than the Baxandall/ Walker test.
-Gnobuddy
Yes, this makes sense to me, and shows why the Total Harmonic Distortion figure, which lumps all harmonics together, can be a misleading measurement. That doesn't mean it's of no value, but it isn't the whole story.
I agree, and if I were able to test (in this manner) the two amps that I compared so many years ago that had identical THD measurements but sounded clearly different to me, I would have welcomed it. If it indeed showed NO DIFFERENCE WHATSOEVER, then I would have to conclude that my ears were deceiving me.
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It isn't the whole story when you have, say, 20% THD, for the reasons you mentioned.
But it is the whole story when you have 0.1% or less THD. Hundreds of carefully-conducted listening tests have shown that there is no harmonic so unpleasant that human ears can detect it when the sum of all harmonics is 0.1% or less.
An analogy: you can easily detect (taste) a pinch of salt in your glass of water. You can much more easily taste a pinch of jalapeno pepper sauce in your glass of water - like 5th harmonic distortion, it's more noticeable than the "2nd harmonic" - salt.
However, you cannot taste a gram of salt added to your Olympic-sized swimming pool full of water. Nor can you taste a gram of jalapeno sauce added to your swimming pool. At that low a dilution, it doesn't matter if its salt (2nd harmonic) or jalapeno (5th harmonic) - you can't taste either one.
Like THD, we could invent a "total pollutant" number, which lumps together the PPM (parts per million) of salt, and the PPM of jalapeno juice. If that total pollutant number was 100,000, it wouldn't tell you much - 100,000 PPM of salt is not the same as 100,000 PPM of jalapeno juice. But if the total pollutant number is lowered to 10 PPM, it tells you everything: you can't taste 10 PPM of salt, 10 PPM of jalapeno juice, or a 5 PPM + 5 PPM mix of both.
THD works the same way. When it's low enough, it means you can't hear the amps distortion. "Low enough" seems to mean about 0.5% in practice. With a hefty safety margin, the industry chose 0.1% as "low enough to be absolutely inaudible under all listening conditions."
You don't need the SIM to tell you if your ears were deceiving you. Just a series of double-blind experiments, and a proper statistical analysis of the results.
First, you call your friends Pat and Jeff to help.
Pat operates a silent switch that switches between amp A and amp B. Both amps receive the identical band-limited input signal; both amps are preset to have output levels within 0.1 dB; both amps feed the same identical pair of (stereo) speakers when switched in.
Jeff is positioned where he cannot see Pat, but he can communicate with him via switches and lights. Jeff can flip a switch that tells Pat "Please flip the switch and randomly select one amp". Pat can flip a switch that lets Jeff know "I've flipped the switch, we are go for the next listening test." And Jeff can flip another switch that tells Pat "Our test subject thinks he heard amp A", or "Our test subject thinks he heard amp B".
Now you (dotneck) are ready to go. You tell Jeff when you're ready for your next listening test; he lets Pat know to start the test; when you're done, Jeff let's Pat know this particular test is over, and which amp you think you've heard. Pat keeps a running tally of which amp he actually selected, and which amp you believe you heard.
It is very important that Jeff doesn't know which amp Pat has selected at any time, so he doesn't accidentally give anything away to you by accidental body-language (Google "clever hans counting horse" for more information as to why this is so important.)
It is also very important that you don't know which amp you are listening to in any of the tests.
The term "double blind" comes from the fact that neither Jeff, nor you, know which amp is currently playing through the speaker. This is absolutely essential to prevent human bias from taking over.
Suppose you do a 100 listening trials. If you got 50 of them right and 50 of them wrong, you obviously can't tell which amp you're listening to. If you get 60 of them right and 40 wrong, you still can't tell the two amps apart - if you flip a coin 100 times, there is a significant possibility of getting heads 60 times.
However, if you get it right 95 times, and wrong 5 times, you almost certainly are able to tell the two amps apart. There is very little chance of this result being a statistical fluke.
What if you got 65 right answers? 70? 75? What does that mean? Can you really tell the two amps apart, or not?
The answer is found using a branch of probability called "hypothesis testing". It is fairly complex stuff, as this PDF from an MIT lecture shows: http://www.mit.edu/~6.s085/notes/lecture2.pdf
To get an answer you can trust, you need to do a LOT of trials, and the results absolutely must be statistically analyzed in this way.
All of this procedure was routine decades ago, when audio listening tests were conducted by actual researchers working in actual research labs (like the truly astonishing Bell Labs.)
Today, an audio listening test is more likely to have been performed by some ignorant twit whose only qualification is that he likes his own opinions. He almost certainly hasn't conducted a double-blind test (which would expose the truth, that he does not have golden ears). He almost certainly hasn't met any of the other conditions needed for an impartial test. And he's almost surely never even heard of hypothesis testing, much less does he have any idea how it's done.
Incidentally: your ears (and mine, and everyone else's) deceive you every time you listen to a stereo system. As we all know, the entire concept of stereo is based on the fact that if you put an identical music signal in the front-left and front-right speakers, you will hear a single mono signal right in front of you. You are hearing something that isn't actually there - your ears are deceiving you!
Think about what a terrible failing of our hearing system that is. Imagine you had a hungry lioness right-front ahead of you, and her identical twin-sister right-left ahead of you - and YOU SAW A SINGLE LIONESS RIGHT IN FRONT OF YOU?
But our hearing system is just that stupid, just that incompetent! Why do we trust the silly thing at all?
-Gnobuddy
> Imagine you had a hungry lioness right-front ahead of you, and her identical twin-sister right-left ahead of you - and YOU SAW A SINGLE LIONESS RIGHT IN FRONT OF YOU? ... But our hearing system is just that stupid, just that incompetent! Why do we trust the silly thing at all?
In defense of "stereo"-- two lionesses are never *IDENTICAL*, same growl-pitch, same underfoot twig-snap.
And the ear developed for the much more likely case of single or non-identical threats.
In stereo, if something is mixed to "center", it is probably a single thing (say mouth or cowbell), and the two channels are closely matched.
This all ignores the acoustics of the lion-space and the victim-space. The sound-field inside a reflective space is very complex and can often reveal if the source is one or two lionesses however identical. Two cowbells must necessarily be offset from each other, sideways or front-back. This gives various path-delays for the first reflection.
In defense of "stereo"-- two lionesses are never *IDENTICAL*, same growl-pitch, same underfoot twig-snap.
And the ear developed for the much more likely case of single or non-identical threats.
In stereo, if something is mixed to "center", it is probably a single thing (say mouth or cowbell), and the two channels are closely matched.
This all ignores the acoustics of the lion-space and the victim-space. The sound-field inside a reflective space is very complex and can often reveal if the source is one or two lionesses however identical. Two cowbells must necessarily be offset from each other, sideways or front-back. This gives various path-delays for the first reflection.
That's why I specified identical-twin lionesses. As Mary-Kate and Ashley Olsen proved with their movie hijinks, identical twins can confuse the heck out of any situation.
More seriously, what was in the back of my mind was exactly what you alluded to - our senses didn't evolve to let us listen to stereo systems, they evolved for survival in the wild. And our brains evolved along with our eyes and ears over hundreds of millions of years of animal evolution.
So the internal signal-processing your brain applies to signals from the ear entering via the vestibulocochlear nerve is almost certainly quite different from the way your brain processes signals from the optic nerves.
When our ancestors were living on the African savannah, I would guess that probably the most important thing was to hear the lioness-growl in the first place, and get an approximate direction. Hopefully that would give you enough time to use your eyes to locate the actual source of the sound.
But we are agreed on the underlying point - our ears didn't evolve to do a good job of processing stereophonic audio signals, in fact, it is precisely because they do a terrible job that the illusion of stereo becomes possible.
Anyone remember a long-ago toy called the View-Master (TM)? It did for your eyes what stereo does for your ears. It presented each eye with a near-identical image, but taken from two locations spaced slightly apart. Once you coaxed your brain into blending the two images, you saw 3D where formerly two flat photographs (transparencies) had been.
Speaking of eyes and human vision: ever wonder why the video from a Go Pro helmet-cam is so much jerkier than what you see through your eyes? After all, your eyes are jerking around just as much as the camera.
Imagine a fast-moving animal that got motion sickness and fell over puking every time it ran? We can see why evolution would have doomed those creatures to extinction long ago.
It turns out that there is a region of our brains dedicated to applying heavy signal-processing to the signals from the eyes, very efficiently doing what many SLRs now do: removing motion blur and vibration from the signal through software manipulation.
-Gnobuddy
That is probably true with sine waves. But what about real music? You mentioned that the researchers "added" harmonic distortion to sine waves with a second generator. How did they add a specific amount of distortion to music?
Maybe not, but I'd rather know EXACTLY what the differences are between the two amps; maybe I could train my ears to actually HEAR the difference (if there were one). It would also be interesting to listen to the subtraction-amp's output (as Rod mentioned). Methinks that would also be VERY revealing.
Real music is far better at covering up harmonic distortion than pure sine wave test signals are. In other words, it's harder to detect amp distortion when listening to music; it's easier to hear amp distortion using a sine-wave test signal.
The reason is simple - real music already contains many, many harmonics and other overtones. For instance, look at the attached FFT of a single plucked (acoustic) guitar note.
In that image, I see 15 overtones (16 spikes total) strong enough to be seen above the noise. The 4th harmonic is stronger than the fundamental (a common characteristic of guitars, because we don't pick the string near its mid-point.) The 7th harmonic is quite strong, and even the 11th harmonic is surprisingly strong.
So if our amplifier now adds a little bit of its own 4th harmonic, say, it gets mixed in with the 4th harmonic already present in the guitar note itself - your ear/brain is not going to hear anything new.
If the amp produced enormous amounts of 4th harmonic distortion (25%, say), you would hear a change in timbre of the guitar note. But small amounts of amp harmonic distortion will just go unnoticed.
By contrast, a pure sine-wave test tone has no harmonics of its own. So if the amplifier generates a little THD, there is a much better chance of your ear/brain noticing it.
That spectrogram shows an instrument (acoustic guitar) which sounds pretty "clean" - it doesn't have strong harmonics, compared to many other classical musical instruments. A violin, say, or a clarinet, has far stronger overtones. And a full orchestra or band puts out so many frequencies all at the same time that it's not at all easy for the ear/brain to pick out small amounts of amplifier distortion.
As I've mentioned a few times, the threshold of detectability, found by conducting vast numbers of double-blind listening tests, using a pure sine-wave test signal, under controlled conditions, seems to be about 0.5%.
The lower number of 0.1% was chosen to have a healthy safety margin below that 0.5% threshold.
We've forgotten how good we have it now, in the era of digital music. You can't get 0.1% THD from a vinyl record; you can't get it from tape, either. And you can't get it from a loudspeaker.
Why people continue to fixate on the most perfect links in the audio chain (preamp, power amp, digital audio files), I don't know. A hundred and fifteen years after the invention of the vacuum triode (in 1905), audio electronics is perfection itself as far as human perception goes.
The failings we can still hear - usually without much effort on our part - are in the microphones, the listening room, and the loudspeakers. All the messy mechanical bits, moving parts (including moving air in a room) flopping about, obeying Newton's laws, which means they don't like to start moving, and they don't like to stop moving, and so they don't want to follow the complex fluctuations of a music signal.
-Gnobuddy
P.S. Guitar pluck FFT taken from here: Guitar Pluck
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With real music the possibilities for intermodulation products are very real, and these are non-harmonically related to the source material. Since all distortion necessarily both harmonic and intermod distortion, we use the harmonic distortion as a proxy for all distortion as its much easier to measure.
Any component with ultra-low THD must have ultra-low IMD, and be very transparent, music, sinewave, or whatever.
Any component with ultra-low THD must have ultra-low IMD, and be very transparent, music, sinewave, or whatever.
Somewhat final solution
I actually read and enjoyed all those heavy posts! It reminded me somewhat of the content in "Handbook for Sound Engineers" by Glen Ballou.
Here's what replaced the first stage FET buffer in the MT-2. All the resistors in the schematic wound up being Vishay 0.4w thin film, since they are high quality, low noise, and have the 1/8w body size which fit well in this cramped unit.
There was a noticeable noise improvement with just the resistor changes. The first version was set to G=2 in the interest of keeping the feedback resistor values low. The improvement over the resistor change was minimal.
The gain was increased to G=3 and abracadabra, the noise has been reduced to about 2/3 what it was. Surprisingly, there was no need to reduce the bypass output. In fact, with either gain version, the output was slightly attenuated. Moreover, the 220k Rf on the second stage opamp was left alone, since all the controls functioned about the same as stock. At some point maybe that will be upgraded with an OPA1611 on a DIP to SIP adapter, which I found reasonably priced. Leaving well enough alone for now.
Thanks to everyone for the myriad suggestions and comments.
I actually read and enjoyed all those heavy posts! It reminded me somewhat of the content in "Handbook for Sound Engineers" by Glen Ballou.
Here's what replaced the first stage FET buffer in the MT-2. All the resistors in the schematic wound up being Vishay 0.4w thin film, since they are high quality, low noise, and have the 1/8w body size which fit well in this cramped unit.
There was a noticeable noise improvement with just the resistor changes. The first version was set to G=2 in the interest of keeping the feedback resistor values low. The improvement over the resistor change was minimal.
The gain was increased to G=3 and abracadabra, the noise has been reduced to about 2/3 what it was. Surprisingly, there was no need to reduce the bypass output. In fact, with either gain version, the output was slightly attenuated. Moreover, the 220k Rf on the second stage opamp was left alone, since all the controls functioned about the same as stock. At some point maybe that will be upgraded with an OPA1611 on a DIP to SIP adapter, which I found reasonably priced. Leaving well enough alone for now.
Thanks to everyone for the myriad suggestions and comments.
Wonderful! Congratulations on a successful outcome!
(An aside: am I the only one who thinks it's odd that diyAudio has an emoji for celebratory clinking beer-mugs, but not one for clapping / applause? Only people who drink can celebrate?
)-Gnobuddy
It was shorted and there was no audible difference in bass roll-off with the low E. I did not bother to investigate. In fact, I should never had added it in the first place. There is no high-pass filter in the original FET buffer.
This will round the edges of the "square wave" and affect harmonics noticeably, thus changing the character of the MT-2. It will also raise the clipping voltage; whether that means anything.....Next recommended step; replace each silicon clipping diode with a series red LED and Ge diode (diy never ends)
The PCB is not DIY friendly. It is small, and packed full and tight with components. Although a through hole, it uses lead-free solder, which is more difficult to remove than tin-lead. Moreover, the traces are paper thin. If I decide to experiment further, some brass pins will be inserted into the holes involved so the components may be tacked onto them, and probably left there when done. 100pcs test point interface pin - prototype breadboard pcb buy 2 get 1 free | eBay
Agreed. My current focus is on the idle hiss, so I don't have to use a gate. There is of course a limitation to that. The original goal was to get more sounds when compared with the MXR Dist +.
The MT-2w is 150 USD and I'm not paying that. My pedal cost 50 plus 5 for the mods so far. I might replace the second opamp circuit with a quieter one, but I suspect the cost/benefit will be minimal. At least I'll have fun soldering the SOIC amp to the adapter board.
Speaking of diodes, I have seen using a sharp edge on one half of the waveform and a softer one on the other. If memory serves correct, the Fuzz Face does that with Ge. But regardless what is used to create the square wave, subsequent filtering/EQing is necessary, whether on the box, the amp, or the axe.
The speakers I have reproduce hiss noise religiously with fuzz. Jensen C10Q.
The MT-2w is 150 USD and I'm not paying that. My pedal cost 50 plus 5 for the mods so far. I might replace the second opamp circuit with a quieter one, but I suspect the cost/benefit will be minimal. At least I'll have fun soldering the SOIC amp to the adapter board.
Speaking of diodes, I have seen using a sharp edge on one half of the waveform and a softer one on the other. If memory serves correct, the Fuzz Face does that with Ge. But regardless what is used to create the square wave, subsequent filtering/EQing is necessary, whether on the box, the amp, or the axe.
The speakers I have reproduce hiss noise religiously with fuzz. Jensen C10Q.
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