Toole and why I like Tone Controls

mijostyn: "The system will follow the appropriate loudness curve depending on the actual volume so no matter what the volume the music sounds exactly the same. Various curves are stored in memory to cover specific problems."

Hello mijostyn,

     As I stated earlier in this thread, it’s my experience that good quality class D amps (Currently very noticeably present on my pair of class D DSonic M3-600-M monoblocks), do this naturally with all source content. I use a Levinson 326S ss balanced preamp with no tone controls.
     I have no idea how technically this happens but I’m suspecting it’s been incorporated into the comparator circuitry of the class D power modules, that are constantly comparing the inputted signal to the outputted post amplified signals to make sure they match, then making any necessary adjustments to the signals prior to releasing them to the speakers. It seems to me that this would be the logical circuitry to insert some sort of automatic, algorithm based loudness control.
     I’ve noticed this very unusual but obvious characteristic on all of the good quality class D amps I’ve owned or listened to thus far. I’m not stating this as factual, rather as a possible explanation for what I know I perceive.  I haven’t heard any other class D amp users discuss this trait but I’d still be very interested in asking Bruno Putzeys and other class D amp power module designers about this phenomenon.

Tim

I have dynamic loudness compensation. The system will follow the appropriate loudness curve depending on the actual volume so no matter what the volume the music sounds exactly the same.

Very nice! Reminds me of the Yamaha/Denon loudness dials.  Loved those too.

  As I stated earlier in this thread, it’s my experience that good quality class D amps (Currently very noticeably present on my pair of class D DSonic M3-600-M monoblocks), do this naturally with all source content. I use a Levinson 326S ss balanced preamp with no tone controls.

I'm generally pro Class D, but while I haven't heard this particular effect be unique to Class D, I will say that the amps, regardless of class, I like tend to have more extension in the treble and bass.

Perhaps low output impedance that extends to the tweeter helps this?

Tim, it would be nice but doing it requires complicated programming of several DSPs running in concert and in control of the entire frequency range at 1/2 Hz intervals in a 48 bit system. You need to have the native frequency response of the system to start which means you have to do impulse testing with a microphone. Then you create filters that make the system's frequency response perfectly flat and this is the canvas you work from now overlying the inverse of the Fletcher - Munson curves and programming the computer to hop from one to the other at the right volume. 
Now if the front end of the Class D amp were digital the appropriate electronics could be added. I think the company that is most likely to do this at a reasonable price would be Anthem. They have room control built in to some of their units so the basic computing power may be there.
Trinnov has it all except the programming for Dynamic Loudness. You can link your PC to the Trinnov and program target curves into memory so you could create inverse F-M curves and switch to them manually but the programming to do it automatically is not there yet.
Erik, yes, just way more sophisticated. You have to see the computer programming in action. The problem with the Tact system is that it is complicated and consumers had a lot of trouble using it correctly. Tact decided to go direct marketing which was a big mistake. They did not have a trained dealer network to help them out and eventually it all just crashed. The current crop of manufacturers learned from this and have simplified their systems but they also removed a lot of the functionality that made the Tact system so effective like the dynamic loudness and the most amazing bass management which has yet to be matched by anything I have seen. I can change high and low pass filters independently 1 Hz at a time and choose any slope for either filter independently from 6 dB to 80 dB per octave. I can do all this on the fly with my laptop on my lap. The computer also has total control over time and phasing between all the individual drivers and sets the delays automatically when the system is impulse tested. The improvement in sound quality is such that I now digitize my phono amp through a Benchmark ADC. May sound like heresy but there is no turning back. 

In my experience the best tone control is no tone control.

Ideally, the best tone control you can have is a properly calibrated room and system. If this is your situation, you'll have as close to an ideal response curve as is practical. No tone control necessary. Though, with some terrible recordings you may wish you had one. Unfortunately, that's our problem, and the recording engineer's fault. I've learned to live with that, but I understand the desire to make a bad recording sound good...
When I first started out, I used to think flat response curve was my goal, and couldn't figure out why it looked great on the screen and sounded like crap in the room. I didn't trust my hearing, as I hadn't yet learned to listen, and I hadn't learned to respect the 50/50 balance between measurements and listening as I do now. After 13 years of calibrating I'm 100% sure that a dead flat response curve is not even close to the holy grail. No room here to discuss what is ideal, but using DSP, EQ or tone controls to get as close to a flat response curve as possible is almost always a fool's errand.

I've been called in to "fix" many systems that were calibrated using Trinnov, Audyssey, and manufacturer's proprietary auto EQ systems including Krell and JL Audio. None of those systems approach the quality of sound you get from properly, physically, calibrating your system - the room's acoustically acceptable, the speakers and listening position are properly coupled to the room, all participants in the system are respecting and working synergistically with physics. Step 1. Always!

After that, DSP, EQ, and tone controls can be used like a pinch of salt - not a fistful or a backhoe full of salt - to knock the tops off the most egregious problems. That's it! No more! And no more than 5dB of attenuation...preferably no more than 2dB. If your setup requires that you mash the tone control or max out the DSP it's time to set everything at "0" and start over. Parametric EQ is best, with a "Q" of 0.7 being ideal, and only nudge what needs to be nudged. If you have a 20dB hump in your response curve at 32Hz, don't EQ it! Find out what the hell is causing that nightmare and fix it. It's likely a loose sheet of drywall or a floor resonance.

The more tone control or EQ you use the more mangled the phase becomes, and although it's hard to pickup on a meter, your ears and brain know that something's not right. This is a problem in the time domain which affects frequency. That's when your brain starts running its' natural hardware and software, trying to reconcile what it hears with what it knows that should sound like. Our own internal EQ. And that's when listener fatigue starts to kick in, and you may find that after 30 minutes or so you can't wipe that look off your face, you know the look that you get when you bite into a lemon. Not the look that most of us are after...

Go easy, and fix the problems at their source. Then, light on the tone control if necessary. And if you absolutely must, have a tone setting (+bass/-treble) for that rare occasion when you're listening to an 80's CD of Mike and the Mechanics. Good luck!