[RazorJack]

Hi there fellow music lovers,
 
I am a big fan of Meier Audio because of the no-nonsense, neutral and dead silent amplifiers that dr. Jan Meier is known to make for well over a decade. Also the crossfeed feature is great to have. I use the (now obsolete) Stagedac and Concerto daily.
 
Recently he introduced an upgrade for existing devices which is called Frequency-Adaptive Feedback. The website I linked to describes the upgrade, which can be built into his headphone amplifiers and dac, if you send your devices to Meier Audio.
 
There is very little information about exactly what this added circuitry does, at the very end of the article. The comparison between amplifying a signal from a vinyl record which requires equalization, and an opamp makes no sense to me. I don't think opamps these days have any problems dealing with signals in the human hearing spectrum.
 
This upgrade will supposedly improve the definition of the middle and higher frequencies, and the sound will become more fluent and better defined, and enhance micro detail. But for me personally alarm bells already went off at the halfway point where he states that 24 bit recordings sound better than 16 bit recordings.
 
Honestly I am really sceptical, but response has been very positive. So I was wondering what you veteran Head-Fi scientists think about this upgrade. Much appreciated!

 

[wnmnkh]

I think he completely misunderstood why RIAA curve was implemented to vinyl in the first place.

 

[arnyk]

  Hi there fellow music lovers,
 
I am a big fan of Meier Audio because of the no-nonsense, neutral and dead silent amplifiers that dr. Jan Meier is known to make for well over a decade. Also the crossfeed feature is great to have. I use the (now obsolete) Stagedac and Concerto daily.
 
Recently he introduced an upgrade for existing devices which is called Frequency-Adaptive Feedback. The website I linked to describes the upgrade, which can be built into his headphone amplifiers and dac, if you send your devices to Meier Audio.
 
There is very little information about exactly what this added circuitry does, at the very end of the article. The comparison between amplifying a signal from a vinyl record which requires equalization, and an opamp makes no sense to me. I don't think opamps these days have any problems dealing with signals in the human hearing spectrum.
 
This upgrade will supposedly improve the definition of the middle and higher frequencies, and the sound will become more fluent and better defined, and enhance micro detail. But for me personally alarm bells already went off at the halfway point where he states that 24 bit recordings sound better than 16 bit recordings.
 
Honestly I am really sceptical, but response has been very positive. So I was wondering what you veteran Head-Fi scientists think about this upgrade. Much appreciated!

 
According to his diagrams, the basic idea is to predistort the input signal with a high pass filter, and then use an amplifier stage that compensates for the frequency response problems caused by the high pass filter.
 
This seems to make as much sense to me as peeing in a pot of soup and then running the soup through a sewage treatment plant to sanitize it.

 

[jcx]

I agree that its wrong headed but the logic probably follows from accepting the flawed rubic of "flat loop gain over audio" - a prescription put forward by Otala in his Transient/Phase Intermodulation Modulation work
 
if Meier is using "high loop gain" monolithic op amps then the open loop gain normally is dominant pole compensated internally
with flat gain set by external negative feedback the "excess gain" or "feedback" is rolling off at 6 dB/octave from low audio frequencies on up
few monolithic op amps are made today with external compensation pins so he has hit on a method to manipulate the feedback vs frequency totally external to the op amp
 
Otala was however wrong on the flat loop gain prescription
Bob Cordell showed his flawed assumption, then built amp and specialized phase selective measurement hardware that separated PIM/TIM from the usual IMD that it is a component of and showed his high loop gain audio power amp design had sub nanosecond PIM/TIM in Otala's own terms
 
"phase distortion" was revisited recently by Ron Quan using 3 tone measurement derived for RF practice, translated for audio in custom hardware - he had to measure 30 year old monolithic op amps worse than the TL07x to get numbers above his measurement hardware noise floor
 
despite the counter evidence Otala's "flat loop gain over audio" has proven a popular rubic for those starting from the proposition that "something's wrong" with global negative feedback in audio

 

[Jan Meier]

Dear headfellows,
 
At low frequencies in a quasi-static approach of an amplifier we have the relation: Vout = A1.Vin + A2.Vin.Vin + A3.Vin.Vin.Vin + ....
 
A1 is the amplification factor and for an ideal, linear amplifier the factors A2, A3, ..... are all zero. Unfortunately such an amplifier does not exist.
 
If Vin is a pure sinus-signal the output signal Vout will be a distorted sinus. The non-linear factors A2, A3, .... add components with frequencies that are a multiple of the original input signals. Harmonic distortion!
 
At least equally important in real life audio signals, however, is the intermodulation distortion.
 
The effective amplification factor for small, high frequency signal component is the derivative of Vout as a function of Vin:
 
Aef = A1 + 2.A2.Vin + 3.A3.Vin.Vin + ..........
 
This value thus is not a constant but changes continuously, especially in the present of large, low frequency signal components. The amplitude of the high frequency output signal thus is not a constant but "beats"/"pulses". This is called intermodulation distortion. The larger the range of Vin, the larger the range of Aef, the larger the inmodulation distortion of the small, high frequency signal components.
 
The original idea behind the FF-technique was, to reduce the range of Vin at the nodal points of feedback, without lowering the amplitude of the high frequency components. This would lower the range of Aef and thus the inmodulation distortion measured at the high frequencies. When this idea was still very fresh I simply did build an amplifier that allowed me to switch between "classic" mode and FF-mode by pressing a single button. Personally I did find the sonic effect quite astonishing and my impressions agree well with what is said by other people in this Forum.
 
One of my problems right now is, that for myself I do not fully understand why it sounds so much better. Giving the concept more thought afterwards my original ideas seem to be somehow flawed. I fully admit (but won't go into detail)!! That's why I never advertized it as a method to reduce IMD.
 
Another problem is, that I have not been able to prove anything with my measurement system yet. The distortion of this system is higher than the distortion of my amps! At its very best my measurement setup has 0.003% intermodulation distortion!
 
Fact is, that all my friends who were able to make a direct AB-blindtest between the "classic" amplification scheme and the FF-technique strongly preferred the latter.
 
Fact also is, that for various reasons I have strong doubts that the effect can be measured with the standard distortion-tests. These tests all dissect the output signal by Fourier-analysis and I'm not sure whether that is the right method here.
 
So at the end I have at least as many questions as you guys have!
 
The only thing I know for sure is, that the new approach does sound better!!!  
 
Cheers
 
Jan

 

[jcx]

hi Jan, I bought my HD600 from you long ago when that was still possible for us in the US
 
 
I am harsh on those still claiming Otala was right with his TIM/"flat loop gain" prescription given the quality, thoroughness of the refutations
 
claiming you hear a difference is to me quite reasonable as long as its not presented as proof of Otala's claims since they are invalid in EE terms, therefore can't be explanations of (undemonstrated) audible differences re TIM/PID "FM" IMD vs "ordinary" IMD products
 
I'm still skeptical overall - while allowing for "maybe something's there" with circuit changes
 
I'd carefully look at frequency and level differences when trying ABX - hard to buy capacitors in even 2% tolerance
 
 

 
Clark, David L., "High-Resolution Subjective Testing Using a Double-Blind Comparator", Journal of the Audio Engineering Society, Vol. 30 No. 5, May 1982, pp. 330-338
http://home.provide.net/~djcarlst/abx_crit.htm]ABX Amplitude vs. Frequency Matching Criteria
 
 
"prosumer" soundcards are fine to verify to the required frequency response, most are quite fine for IMD,IMD sweeps, Mutlitone...
 
https://www.google.com/#q=cabot+audio+distortion+measurement goes into audio distortion measurements: 19+20 kHz 1:1 is quite "stressfull" compared to most music
 
good soundcards should capture audible differences - allowing the ABX to be done with foobar2000 ABX plugin and .wav of what ever signal or music you think shows up the difference after passing thru 2 otherwise matched amps
 
with free .wav editors, Audacity and  Audio DiffMaker you can actually look at the difference waveforms, verify your frequency, level matching, I'd also look at noise levels, frequency variation of the noise above hearing thresholds

 

[Jan Meier]

Dear JCX,
 
"claiming you hear a difference is to me quite reasonable as long as its not presented as proof of Otala's claims since they are invalid in EE terms"
 
I guess it's a good thing that I'm not an electrical engineer!  
 
" I'm still skeptical overall "
 
Very understandable, especially as I haven't been able to provide any scientific proof whatsoever (yet).
 
>" I'd carefully look at frequency and level differences when trying ABX - hard to buy capacitors in even 2% tolerance"
 
I'm very well aware of the differences that changed frequency responses or a different playback level can produce. Especially the latter is an important factor often overlooked. However, the differences heard between the classic-mode and the FF-mode are clearly of a different nature.
 
"audio distortion measurements: 19+20 kHz 1:1 is quite "stressfull" compared to most music"
 
Yes, but the problem is that the mixture of two high amplitude high frequency signals in this ITU-R test has no resemblance at all to real life audio signals. In my measurements I almost exclusive use the SMPTE test with a large low frequency component and a smaller high frequency component (60 Hz and 7 kHz combined in a 12 dB ratio (4:1)).
 
" good soundcards should capture audible differences"
 
I use a professional RME external sound card.
 
"with free .wav editors, Audacity and  Audio DiffMaker you can actually look at the difference waveforms"
 
Only if the waveforms are sampled at exactly the same points of time. However, even a multichannel soundcard normally only has one single AD-converter and there is a multiplexer in front that switches between input signals. Since different signals are sampled at different points of time there always will be an effective difference waveform, even if waveforms are exactly the same!
 
> verify your frequency, level matching
 
Of course!
 
> I'd also look at noise levels, frequency variation of the noise above hearing thresholds
 
CORDA amplifiers are very "black". I always take much care of that. At normal volume settings these amps do not produce any audible noise!
 
Actually, due to the lower feedback (higher effective feedback impedance) at lower frequencies the FF-mode does produce a little bit more background noise. But it's still very inaudible!
 
I guess for me the effect of the FF-technique is like a beautiful woman. Lovely to hear/watch but very difficult to understand. But the fact that I don't understand doesn't mean she isn't real.
 

 
Jan

 

[RazorJack]

Hi Jan, great to see you here! Thank you very much for providing those details. It's starting to make a bit more sense to me. Please forgive me if it seemed distrustful of me to start a thread about this in the Science forum without informing you. For what it's worth, I never thought any of the claims weren't true from people who believe this upgrade is beneficial. If it's made by Meier, it's just got to be good, I sure have been a satisfied customer for long enough to blindly believe that
 
Really though, the first thing that crossed my mind when reading about this upgrade was exactly as arnyk put it:
 

  This seems to make as much sense to me as peeing in a pot of soup and then running the soup through a sewage treatment plant to sanitize it.

 
But apparently it's a bit more complex than that!
 
Why is this upgrade is also available for the Daccord? Could the intensity of this effect be variable?
 
Has something similar ever been done in the world of hi-fi?
 
It's weird, but first with crossfeed and tonal balance, and now this ff circuit, it seems that your devices "mess around" with the audio signal's frequency response so much that it's hard to believe that Meier gear has the reputation for being so neutral! 

 

 

[Speedskater]

It seems strange, now some 40 years later, to bring Otala and TIM into a conversation. There were so many problems with his papers and his ideas.

 

[jcx]

please tell that to John Curl and a dozen other frequent posters at diyAudio - a quick search of my posts shows I have pointed out the problems with Otala's formulation more frequently there in recent years
 
however flawed it seems to be an enduring meme still popular as an "explanation" with the "there's something wrong with (global, high) feedback" crowd

 

[Jan Meier]

Dear headfellows,
 
"Why is this upgrade is also available for the Daccord?"
 
The DACCORD uses some amplification stages to desymmetrize and to buffer the signal of the DAC-chip. However, as I do not have access to the amplification stage of the DAC-chip itself (WM8741) the effect of the FF-technique is less noticable as with the amps.
 
* Has something similar ever been done in the world of hi-fi? "
 
Not to my kowledge, but then there are many things done in the past that I haven't heard off.
 
"with crossfeed and tonal balance, and now this ff circuit, it seems that your devices "mess around" with the audio signal's frequency response so much that it's hard to believe that Meier gear has the reputation for being so neutral!"
 
The purpose of Crossfeed is to make sound more natural, which doesn't always imply more neutral. The disadvantage of many crossfeed filter versions is, that the frequency response of mono-signals is also effected (comb filter effect). With the natural crossfeed filters this is prevented. There are no changes in frequency response, so I guess we can say that the crossfeed filters on CORDA amplifiers are the most neutral ones on the market right now.
 
There is however a psychoacoustic effect of crossfeed that gives many people the feeling that bass becomes slightly weakened. For this the tonal balance switch was introduced. Be aware that the sonic characteristics of headphones by their designers were optimized for plain stereo listening. Using crossfeed slightly changes the perception of the balance of low and high frequencies. This is compensated by tonal balance.
 
As for the FF-technique, its use in my opinion does not contradict my design goals for neutrality. The vinyl record uses strong emphasis/deemphasis to optimize the quality of sound and  yet one can well strive for neutral sounding record players.
 
" It seems strange, now some 40 years later, to bring Otala and TIM into a conversation."
 
There are good reasons I didn't !!
 

 
Jan

 

[arnyk]

  please tell that to John Curl and a dozen other frequent posters at diyAudio - a quick search of my posts shows I have pointed out the problems with Otala's formulation more frequently there in recent years
 

 
There are people around here whose knowledge of these things is far beyond mere science. I know only of explanations based on science. Those explanations fail with them.
 
 
Quote:

Quote:

  however flawed it seems to be an enduring meme still popular as an "explanation" with the "there's something wrong with (global, high) feedback" crowd

 

 
Two words: Sighted Evaluations.