Why Let an Amplifier Sound Good when You can Force it to?

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It has nothing to do with the amp or the audio system, but everything with your ears. The amp (if it is a decent one) will have a flat response down to low frequencies no matter what the level, and certainly at low levels.

The point is that your ears get less sensitive to low frequencies when the level drops. Google 'Fletcher and Munson'.

That's the reason for loudness control. The idea is to put a hump in the lf response at low levels to compensate the loss of hearing sensitivity in your ears.

Jan

It is completely unacceptable.

1. your amp doesn't know how sensitive your speakers are or the distance of the listener.

Result: the loudness boosts the bass at the price of distortion and result in bommm sound on decent speakers.

2. The 'contour network' with the volume control contribute to phase changes and sound degradation.

I am very against it, it is not a flat sounding amplifier anymore, its a gimmick.
 
A loudness control needs to have more settings than "on" and "off" to be usable.

The required amount of compensation depends on the difference (in dB) between the equal loudness curves at the mastering or recording volume and at the playback volume.

Traditional loudness controls only give a flat response at maximum volume and apply bass boost depending on how far you are below maximum volume. Hence, they only get the compensation right for music styles that are meant to be played at maximum volume. For other styles they overcompensate.

With a simple bass control that approximates the difference between equal loudness contours, you can set the compensation yourself depending on the music style and volume.

Of course you always get the most realistic playback when you use the volume for which the recording is meant, and in that case you don't need any loudness compensation. When you like hardrock, that also gives you tinnitus before you know it, and probably conflicts with your neighbours.
 
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Wavewhipper said:
That goblty-goop translates to an amplifier that is able to slew it's output to reproduce the complex composition of many harmonic rich instruments all at once.
That was amazing to me when I first studied amps, how in the world does one path to the speaker make all those different vibrations at once ?
The amp and the speaker do not know they have been asked to reproduce " all those different vibrations at once"; all they know is that they have been given a voltage signal which changes with time fairly slowly, and they may have some limited knowledge of what it was in the very recent past. Anything else requires either a Maxwell demon or Fourier denial.

I love FFT. I'm simply too worn out to do much with it. And yes THAT is the explanation, but how about the person in fascination finding this out too soon to enjoy the find ?
But still that does point to the importance of an amp's slewing ability. And it's ability to eat back EMF without mutilating it's feedback system.
FFT is merely a fast way of doing Fourier. As the input signal is bandlimited only limited slewing ability is needed. Eating back emf is what low output impedance amps do for breakfast.

I don't see anything irrational about that consideration. I suppose these modern semiconductor output stages win by the brute force and ignorance of being an ultra low impedance current source.
They can be low impedance or a current source; they cannot be both because the two are direct opposites. All amplifiers are ignorant; that is exactly what we want them to be. Sadly, some amp designers are ignorant too.

The situation I was talking about last page is the possibility that in a high power transient, say so loud the woofer is at Xlim, the drive signal is suddenly off, leaving the bias impedance, the woofer snaps back to neutral position, there is a possibility of the back EMF and generated EMF pushing current into the feedback resistor network feeding one side of the input differential amp, and now the diff amp would be improperly balanced for some sudden new signal.
That would only happen if the amp was clipping, and the output stage somehow lost control. Once clipping takes place you have departed from fidelity, but unlikely that the output would suddenly go open circuit too. Woofers do not snap.

Crazy complicated isn't it ?
A good amplifier is a form of servo system, but motor drives and audio amplifiers are not the same thing so you cannot necessarily read across problems from one to the other.
 
It is completely unacceptable.

Fletcher-Munson effect is real. It quantifies how our ears really hear.

Audio designers have been attempting to address it since at least the 1950s. There have been some interesting circuits. Most circuits in consumer grade equipment are terrible though, and not really Fletcher-Munson circuits.

1. your amp doesn't know how sensitive your speakers are or the distance of the listener.

Result: the loudness boosts the bass at the price of distortion and result in bommm sound on decent speakers

Accurate observation. Most "loudness" controls match speakers that nobody seems to use. They seem to be compensated for inefficient speakers.

2. The 'contour network' with the volume control contribute to phase changes and sound degradation.

That seems to be the case. We need an adjustable control made with quality components and careful design. I've seen switched passive networks that were pretty good, mainly because "less is more" with tone controls and these circuits provided that. I've seen adjustable circuits using a potentiometer that were pretty good. In fact, I've been working on one off and on for months. I keep changing components to get it right.

I am very against it, it is not a flat sounding amplifier anymore, its a gimmick.

The place for a flat amplifier is the power amplifier. I fail to see how any practical audio system can work without some kind of tone controls. The only way around this that I see is to always play your music at the same spl that the producer mixed it at, which is typically 85 dB and louder. Some rock music is mixed at more like 95 dB.

So if you can always play your music at 85-95 dB, then maybe you can forgo tone controls. Or maybe your speakers are voiced towards the Fletcher-Munson curve; people would probably prefer that over truly flat speakers. Measurements would tell.
 
Can someone provide some good links and discussion to “the best of” loudness control designs?

My understanding also is that the relationship between preamp gain, power amp, speaker sensitivity and room size / general listening levels is crucial and generally poorly comprehended by hobbyists (while gain structure seems to be a central consideration in pro audio discussions).

This might have more to do with why different equipment sounds different than the topologies or quality.

Can someone expand upon this and provide general guidelines?

If I remember correctly Salas laid out some interesting rules of thumb in the dcb1 thread.
 
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I see the loudness button as a crude equalizer that gives me two preselected equalization profiles.
It is there to boost low and high frequency sounds at low volume settings.
I just copied from the manual of my Rotel RX-402 from 1976 technology.
That is fair enough to me, I do not expect miracles from this. If I were to ask more from the loudness button, the bass control and the treble control pots...well, I would add a full blown equaliser and spend hours and days fiddling its settings.
 
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In earlier times your vol control would have a tap somewhere, with an R/C thing to ground. When you turned down the level you'd get closer to the tap and that would cause the bass level to drop less than the rest of the audio band. This automagically corrected for decreasing low freq ear sensitivity with lower volume. Similar for treble.

But I don't think these are used in these modern times ;-)

Jan
 
In earlier times your vol control would have a tap somewhere, with an R/C thing to ground. When you turned down the level you'd get closer to the tap and that would cause the bass level to drop less than the rest of the audio band. This automagically corrected for decreasing low freq ear sensitivity with lower volume. Similar for treble.

But I don't think these are used in these modern times ;-)

Jan

I had a Braun valve amplifier a long time ago with two volume knobs, one with and one without a tap with RC series network. That way you could adjust the amount of compensation depending on the music style, loudspeaker sensitivity, room size and so on. If you didn't want any loudness compensation at all, you left the tapped volume control at maximum and used the other to control the volume.
 
I don't know how the Yamaha control works.

I also have a DIY preamplifier where the unfiltered signal goes through two ganged volume potmeters per channel and a signal with much enhanced bass goes through a separate potmeter and one of the ganged volume potmeters. Hence, if you reduce volume by 20 dB, the deep bass is only reduced by 10 dB.

One disadvantage is that you need a four-channel potmeter for stereo. I used two mechanically coupled faders for that (mounted vertically on the photo, the potmeter above it is the bass control and the potmeter below it controls channel balance). Another disadvantage is that it doesn't match the equal-loudness contours very well anymore since the 2003 revision of ISO-226 (it is really strange that something that was first measured in 1933 still changed a lot in 2003, but it did).
 

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