In a variety of threads, Dr. Earl Geddes has presented his takes on more meaningful methods to test amplifier distortions beyond HD. This is a spin off thread to discuss the subject in detail. I am putting this in the solid state forum because it likely applies more acutely here, where high level HD is a non issue in most designs, but low level audible problems are a constant matter of discussion.
Crossover distortion is a particularly insideous form of nonlinearity because it happens at all signal levels and there is no comparable mechanism in a loudspeaker to mask it. The question was asked if I have a way of identifying crossover distortion in an amplifier.
Yes, I do.
You see the situation with crossover distortion is that the % distortion increases with falling signal level. This is exactly why it is so audible since this is directly opposite to our hearing.
One could therefor ***** crossover distortion by looking at THD as the signal level goes lower, which is a typical measurement. The problem is that virtually all of these THD versus level measurements are THD + noise. When this is the case, the rise in THD at lower signal levels is actually the noise and NOT the distortion, but it is impossible to tell which is which. SO this test actually masks the real problem. One would have to track the individual harmonics of the waveform, but then the noise floor is still an issue.
Hence the measurement problem is one of noise floor and how to measure distortion products down below this floor.
This is done by averaging. But normal averaging can only lower the noise floor so much - down to the noise power. But if I have a signal and I average this signal sychronously then I can raise the net signal to noise level. This too is common. But if the signal does not exactly fit the time base then I need to window it and the resultant spectral leakage makes this sychronous averaging less effective.
I use a signal that exactly fits into the time base of the A/D taking the data. This means that I don't have to use a window and I can sychronously average a signal to noise ratio that is about 20 dB better than a simpler test could achieve. This means for example that the input signal needs to be something like 976 Hz, not 1000 Hz, which doesn't exactly fit the window.
I actually had to generate the input wav file in FORTRAN using quad precision, special random number generators and rounding techniques, because the test signals needed to have a 120 dB dynamic range - very difficult with 16 bits.
I use a signal that starts out low and goes up in level. I plot out the results as the signal drops into the noise floor. This test shows vast differences in amps that measure identical with standard tests.
It also shows that my Pioneer amp - you know the "really crappy" one that I get crticized for using at RMAF - is an extremely good amplifier. As good as the best that I have tested with this technique.
Yes, I do.
You see the situation with crossover distortion is that the % distortion increases with falling signal level. This is exactly why it is so audible since this is directly opposite to our hearing.
One could therefor ***** crossover distortion by looking at THD as the signal level goes lower, which is a typical measurement. The problem is that virtually all of these THD versus level measurements are THD + noise. When this is the case, the rise in THD at lower signal levels is actually the noise and NOT the distortion, but it is impossible to tell which is which. SO this test actually masks the real problem. One would have to track the individual harmonics of the waveform, but then the noise floor is still an issue.
Hence the measurement problem is one of noise floor and how to measure distortion products down below this floor.
This is done by averaging. But normal averaging can only lower the noise floor so much - down to the noise power. But if I have a signal and I average this signal sychronously then I can raise the net signal to noise level. This too is common. But if the signal does not exactly fit the time base then I need to window it and the resultant spectral leakage makes this sychronous averaging less effective.
I use a signal that exactly fits into the time base of the A/D taking the data. This means that I don't have to use a window and I can sychronously average a signal to noise ratio that is about 20 dB better than a simpler test could achieve. This means for example that the input signal needs to be something like 976 Hz, not 1000 Hz, which doesn't exactly fit the window.
I actually had to generate the input wav file in FORTRAN using quad precision, special random number generators and rounding techniques, because the test signals needed to have a 120 dB dynamic range - very difficult with 16 bits.
I use a signal that starts out low and goes up in level. I plot out the results as the signal drops into the noise floor. This test shows vast differences in amps that measure identical with standard tests.
It also shows that my Pioneer amp - you know the "really crappy" one that I get crticized for using at RMAF - is an extremely good amplifier. As good as the best that I have tested with this technique.
I know the presentation, reasoning and the files. When I explored them in Cool Edit about 2 years ago, I found parts of the signals with very low level missing, just "zero". Of course , THD would be still low. If this was the case, I would consider such a test very very unfair. Could you explain??
gedlee said:
I am guessing that he is saying that early versions (or even newer versions) of Cool Edit decided to approximate low absolute level signals as "zero" rather than try to encode them at their true levels. Or alternately they did not write at 16bits and truncated at maybe 14 bits??
_-_-bear
taj said:
Personally, I don't have the time to do anything with it - amps are not my interest. I developed this simple test to answer a very specific question. It did that very well. We used it again to look at some amps that we were using and it prove to be enlightening there again. Beyond that I have done no further developement and don't see doing so myself.
But I really do think that this is an important area to those who believe that there are differences in amps that we don't see in the measurements.
I am fully willing to allow the use of the technique and to help others develop it, but I am not willing to give it up to the public domain as I have done this before and others have taken credit for my work.
So as long as I get credit for what I did, I am willing to share in anything about this test - including the software and source files (wav files).
4fun said:
No hardly - I don't "favor it", but I was severly chastized for using it at RMAF when, in fact, no one really knew if it was any good or not. It works just fine as my measurements show. I would not use this amp for many applications, but it suited my point at the time, which was that loudspeakers account for 99% (well you could argue 98%, but you get my point) of the audio systems sound quality.
The amp is a Pioneer DSX-V912 - a receiver. The point is that it was on sale at Costco for $150.00. I bought several of them for home theater use. I used my test to measure the amps and they were quite good actually. Especially for chip amps. I was measuring a lot of chip amps (a survey of capability) and most were pretty bad. As a chip amp this unit deffinately stands out. It compared quite favorably to a very well engineered discrete amp that I also use.
I also tested several other receivers and they were almost universally bad.
Hi
Earl, well this triggers my interest as I also love SS ( at least some of them that don't have a certain logo on it
),
Would be nice to do some " sawdust " at in this thread as long as we are awaiting for first steps of realisation in the other one .
- Maybe some would use that revealing test and share the results here ?
- Maybe we would need a little standardisation for what to post as results once the measurement technique is clear.
Would like to receive this WAV. Could you please send it to my PM, Earl ?
-------------------------
PMA, your speach is ctyptic to me. Could you share some of your measurements please and comment about the constraints ?
Greetings
Michael
gedlee said:
Earl, well this triggers my interest as I also love SS ( at least some of them that don't have a certain logo on it
),Would be nice to do some " sawdust " at in this thread as long as we are awaiting for first steps of realisation in the other one .
- Maybe some would use that revealing test and share the results here ?
- Maybe we would need a little standardisation for what to post as results once the measurement technique is clear.
Would like to receive this WAV. Could you please send it to my PM, Earl ?
-------------------------
PMA, your speach is ctyptic to me. Could you share some of your measurements please and comment about the constraints ?
Greetings
Michael
mige0 said:
I'm still trying to fgure out how to handle this. I don't want to get into the role of being an open source for this stuff and I am a little concerned about my intellectual rights.
I want to minimize my time and protect my ideas while making them available. Any suggestions how to handle this?
Attached are some examples of the output. The signals are scalled upand right to offset them for better observation. The only one on the correct scale is the lowest one - red.
Look for no change in harmonic structure of the waveform as the signal level drops. These should not be any very high harmonics.
The Sanyo is clearly a disaster, the Sony is not very good as the harmonics are higher order and are pretty predominate at mid levels. The Pioneer(my amp) shows a very clean response all the way down in signal level.
Gordy said:
It's not an arguement with concrete data, but some typical experiments make the essence of the statement clear.
If you have ever heard a good binaural recording then you know that the allusion of a perfect reproduction is certainly possible - over headphones, of course. The realism is beyond question with only very minor locations and situations lacking complete auditory perception reproduction.
But over loudspeakers this fails - why? The first is that there is now crosstalk, which can be cancelled at one location with some difficulty, but even when this is done, it fails to meet the headphone reality. The loudspeakers simply are not as good as required for the illusion to come through. But clearly the recording medium, the electronics and the headphones are. This result - changing only the loudspeakers - clearly implies that the signal integrity prior to the loudspeakers is very close to perfect. The loudspeakers aren't.
As I say we can argue the percentages (but I won't) but there is little doubt (in my mind at least) that the vast majority of the audio reproduction problem is the loudspeaker. Experience says this, the data says this and most other researchers (like Floyd Toole) say this.
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