Amplifier reproduction of instrument Timbre


I have just been through 5 different amps using the same DAC, speakers and cables (Monarchy DAC24, Green Mountain Europa, anti-cables) and am struck with how differently each amp reproduces instrument timbre. The amps were Rotel, Arcam, Parasound, Simaudio and Ayre. Without a doubt, the Ayre reproduces the most natural instrument sound. I have been a classical and jazz musician (bass and cello) all my life and have heard instrument in many different settings. I am curious what properties of an amplifiers design and implementation might contribute to this quality.

This got me interested in the "timing" issue of amplifiers, ie the ability of an amp to respond and reproduce the attack portion of a note. I think that slew rate is one measure of this? Instrument timbre is primarily perceived by the attack and decay qualities of an instrument and the harmonic structure during sustain. Tests have shown that if the attack/decay portions of a note are removed (edited out), it is difficult for a listener to identify instruments based on just the sustain portion of the note (when played back at identical volumes).

Can anyone comment more technically on why one amp might be better at this than another? I am not interested in hearing amp comparisons or recommendations, I understand that other amps and speakers also sound realistic and that there are issues of "synergy".

thanks,
drewh1
Hi Drew,

I think that your question is put in an exceptionally perceptive manner.

One major factor that I think is relevant would be the Haas Effect:

http://en.wikipedia.org/wiki/Haas_Effect

Basically, our hearing mechanisms "latch on" to the leading edge of transient waveforms, and give them disproportionate emphasis relative to what may follow a few milliseconds or a few ten's of milliseconds later. We evolved that capability to aid localization of the source of sounds that may arrive at our ears via both a direct path and (slightly later) via reflections.

The Ayre's no-feedback design approach would, everything else being equal, reproduce the leading edges of transients more cleanly than an amplifier that uses significant amounts of feedback. That is because of the non-zero amount of time it takes for the signal to propagate around the feedback loop. During that amount of time the input may have changed significantly, and the correction being applied by the feedback process may be based on out-of-date information, so to speak. What is called transient intermodulation distortion is one of the adverse effects of that.

Creating an amplifier that uses little or no feedback while still performing well in other respects (linearity, low output impedance, low harmonic distortion, etc.) requires both very high quality parts, and a lot of care and intelligence in the design.

Also, for any given amount of harmonic distortion that the amplifier may produce, its structure is very significant. It's well recognized that low order even harmonics (such as 2nd and 4th) are euphonic and musical, while odd harmonics (3rd, 5th, etc.), and especially high order odd harmonics (9th, 11th, etc.) are the opposite.

Harmonic distortion is primarily the result of non-linearity in the transfer function of the amplifier, linearity meaning that a plot of output amplitude vs. input amplitude should be a straight line, within the amplifier's power range.

I'm sure others will contribute lots of other thoughts on your question, because there are obviously many factors that are involved, but in my case these are the ones that come to mind first.

Regards,
-- Al
If the Ayre has no negative feedback (as per Al) then it will likely have a higher output impedance and this could modulate the response you hear according to the impedance variation with frequency of the speaker. This will certainly affect timbre.


Regarding the Haas effect, the thing that interested me most about the amplifier differences was that they were all perceive through the same listening conditions and speakers which are presumably time/phase coherent. I would think any phase or time shift problems would be introduced by the speakers and not the amplifier. Unless of course, there is some fundamental design flaw in the amp.

Al - your explanation of transient intermodulation distortion is very interesting - I understand that this problem can also be introduced by the DAC in certain implementations or maybe that is a different issue.

Regarding even vs odd harmonic distortion, maybe this is old information but I understand that tubes introduce harmonic distortion on low-even harmonics and transistors on high-odd harmonics. This is why tubes have long been considered to be more "musical" than SS. I, however, prefer clean SS and find that the harmonic distortion is not a primary determinant in the naturalness of timbre. I must admit I have not spent a lot of time listening to very good (expensive) tube gear so I could just be naive.

Thanks for the responses, this is the kind of information I am looking for.

drew.