|6th December 2001, 06:25 AM||#21|
Join Date: Sep 2001
Location: Fairbanks, Alaska
I would love to do that since I was planning to do some design work shortly anyway that would include the Baxandall circuit.
|6th December 2001, 07:18 AM||#22|
Join Date: Mar 2001
Location: Tallinn, Estonia
The Baxandall circuit has a varing input ipmedance with different control settings... not so good, especially if one would like to do the circuit all passive (volume pot + tone controls)
I have heard though that Sony uses a very good tone control circuit in their better amps. The best about those is that the input impedance for any control setting is always the same. Does anyone know more about this and may be have a part of the schematic. I think it would be of interest to many of you
PS. I'm currently listening to 'no tone control system' but I would like to include them if I can find a good enough solution that I feel does the job well. There have been occasions when I have felt the need for them.
|9th December 2001, 10:49 PM||#23|
Join Date: Feb 2001
re: the APTcircuit, as promised
OK, here we go (longish message follows):
Earlier in the "who uses tone controls" discussion, I mentioned I would share
some info on the APT preamp from the early 80's. There is a patent on the
design, #4,220,817 by Frank S. Kampmann, assigned to APT in 1980. The summary
is this approach eliminates the undesireable frequency bumps, etc. at mid and
higher frequencies as a result of adjusting the bass control. The patent has
some nice frequency response curves of the original baxandall design that show
this effect clearly and some curves of the improved design that show how it
eliminates this problem (sorry, no scanner - visit http://www.delphion.com).
The trick seems to be adding a unity gain buffer for the bass cap. See the APT
tone control schematic here: http://www.enteract.com/~mlloyd1/aud...e-controls.gif
I did the math a LONG time ago; it works quite well. They talk about it in the
patent description. I used a good op amp in place of the emitter follower;
made the math easier :-) Of course, you can simulate the circuit and plot
responses that way.
You will notice a switch for the bass cap in the schematic. You can select
shelving and non-shelving bass modes. From the manual:
"Bass Mode: With the switch down, bass control affects mainly deep bass
frequencies with minimal impact on mid-bass. With switch up, bass control
affects entire bass range uniformly." From the product description: "In the
shelving mode, the Bass control can be used for a variety of program and
speaker equalization purposes; more emphasis is placed on adjusting higher
bass frequencies than in the non-shelving mode." So, basically, you could do
conventional bass control or low frequency loudness compensation with this
By the way, as I thumb through this, I remember that APT wrote great manuals
and product descriptions. Good informative reading, if you get a chance.
Another "by the way": similarly to the conventional baxandall design, this
circuit inverts polarity. APT used an inverting line stage with the volume
control in the feedback loop (Doug Self uses a similar, but slightly different
design for volume control in his preamp designs to take advantage of tracking
improvements in dual linear pots vs. that in dual log pots - ratio of
resistance controls the volume instead of absolute resistance values). So,
what do you do if you build this without the inverting stage and you don't
want to listen to your music in inverted phase? Easy, flip the wiring to your
Remember this design is from the late 70's: There are electrolytic coupling
and the Op AMps are 1/2 of TL072CP. Nobody on this list uses that stuff these
days, do they? :-)
All NPN transistors are 2SC1345E. Diodes are 1N4148 or 1N914.
Another nice touch is the 100K trimpots around the BASS and TREBLE control
pots. Why you ask? Well, you can adjust these trimpots such that when the
panel controls are centered, the boost/cut is zeroed.
The emitter follower + current sink buffer in front of the tone controls
includes a switchable 40KHz minimum phase, 2-pole filter to minimize Slew
And yes, the designers included a switch to remove the tone controls from the
circuit entirely :-)
Let the comments begin ............
[Edited by mlloyd1 on 12-09-2001 at 11:27 PM]
|10th December 2001, 12:23 AM||#24|
Join Date: Apr 2001
Location: Northern Noricum, near the Limes
I would consider the Pultec type (passive) equalizer !
I did not make the thing yet, but I just had the inductors made by somebody.
I also will not use a tube circuit, I think I will rather take one of my Naimclone buffer and gain stages.
This circuit has a great reputation over the years in one incarnation or another (expensive like TubeTech) as a highend mastering tool in the studio. Here it is mainly used for very small last corrections of the stereo signal.
Many audio engineers (and among them there a famous names) rate it high and love the sound (yes, they love the sound they are creating...)
If something is good for the studio, why should it not be good for the (occasional) use at home?
I agree with Grey that if a recording is not made and mastered properly there is little we can do to reverse the job. You can only go on adding something, if it is more gear or more effects.
But there are those special cases where the music is so much better than the recording, so that a LITTLE EQ here and there can help to work things out better somehow. This will also apply to the treatment of room acoustics, but it is so easy to make matters worse.
But this is the territory of studio work, remastering etc., we are leaving the hifi/highend areas here.
I have listened to so many equalizers in so many different settings and setups, but only very few could actually make the sound of the equipment or the music sound better. Most of the time you just could gain long faces by just switching the stuff into bypass mode or pulling it out of the chain.
As Rod Elliot also mentioned in one of his articles (I forgot which one), one gets acustomed to different sounding environments quite easy.
Maybe that is also one of the reasons why we can stand the sound of all the loudspeakers, although they are so tremendously different from each other !
Applying EQ changes things but does not make things better here. Each of you also know that amplifiers can sound quite different from one another and that is surely not a question of frequency linearity, they are all flat (exept when the design is badly faulty).
mlloyd1, I shrinked your schematic as requested (11k). You can download it here (I do not have your email adress).
|10th December 2001, 05:35 AM||#26|
Join Date: Sep 2001
Location: Fairbanks, Alaska
Different phase shifts for different folks
My entry here is mostly in reponse to a post by GRollins a while back in this tread in regards to time aligned loudspeakers.
I have built a number of time-aligned loudspeakers and have sufficient loudspeaker test equipment to monitor the results. In my experience these are only time aligned at a specific location. I you move your head away from the sweet time aligned spot (up, down or side to side) you are no longer time aligned.
Many of the time aligned speakers have a sloped front so that the main beam of the tweeter is actually aimed at the ceiling. This appears to be primarily done in a attempt to reduce the effect of reflections from the front baffle that would occur if the tweeters were set back. Thus much of the HF signal actually bounces of the ceiing and may arrive at the listeners ears out of phase due to differences in ceiling height.
Moving your head away from the sweet spot is like inducing phase shift. None of the signals will arrive exactly at your ears at the same time anymore. This of course primarily affects the upper midrange and high end. Then of course there are also those room reflections that add or subtract. These usually have a phase relation ship that is totally unknown.
So although it is nice to discuss and reduce phase shift in equipment to a minimum lets also think about what happens when the signal starts moving air. Especially when this is done with different drivers for different frequency ranges.
The game of sound reproduction is a weird ball game and we should not concentrate to much on phase shift that is induced by the desire to modify frequency response. Until such time that all equalization is done strictly with math in the digital domain then the phase shift done by analog networks inside equipment will remain a fact of life. This is really nothing to worry about since other phase shifts are much more dominant in the average listening environment. Who knows, adding a little phase shift electronically may just help in reducing phase shift caused by sound bouncing around in a listening room.
[Edited by alaskanaudio on 12-10-2001 at 12:51 AM]
|10th December 2001, 04:23 PM||#27|
Join Date: Feb 2001
Location: Columbia, SC
What I found over the years wasn't that I was 'correcting for deficiencies in the recording,' I was correcting for deficiencies in my system. I had speakers that claimed to be flat to--I forget--something like 30 or 35Hz, but in reality they were dropping off pretty rapidly below about 50Hz. I couldn't figure out why I wasn't getting any bass. I bought a sub (my highs weren't any good, either, but I stopped short of buying super tweeters). It was a joke. The sub had response about equal to what I was already listening to. I added a graphic equalizer (this in addition to the tone controls I already had in my preamp), and another gizmo, and another, and another.
To make a long story short, my system sounded like a table radio on steroids.
I finally got smart and started buying decent equipment, paying less attention to what the specs claimed, and more to how it actually sounded. Guess what? I finally achieved a system that had low bass...without a single tone control in sight. Imaging improved by several thousand percent because I got rid of all the phase shift and nasties that muddy that up. Detail retrieval, ditto. Etc.
People who say that the phase shift inherent in circuitry (whether tone controls or otherwise) isn't a big deal need to get out and listen to live, unamplified music more often.
Digital correction is already here. There are at least a half-dozen units on the market. I'll reserve judgement as to how effective they are until later, as I haven't heard one.
Saying that room reflections render phase shift in the system moot is the same argument as saying that there are already so many (fill in the blank: opamps, tone controls, gain stages, whatever) in the recording chain that it won't hurt to use another. It's muddy thinking. It leads to mid-fi sounding gear. Don't get me wrong, there's a place for mid-fi, but mid-fi masquerading as pseudo high-end is intellectually dishonest. I've been quite a ways down that road. I finally saw through the illogic, but...egad, I wish I had back the money and time that I wasted.
Besides...room reflections are easily controlled.
|11th December 2001, 06:15 PM||#28|
Join Date: Sep 2001
Location: Fairbanks, Alaska
I can see all the points you make very clearly and can appreciate exactly what you are saying.
I do not use tone controls but I would like to have such a tool available to me if and when I see the need to use it. So one of my projects someday will be for a tone control circuit that I find most suitable. It is naturally understood that phase shift changes will occur when adding any equipment or changing listening position.
Unfortunatly for all of us there is no way to eliminate all phase shift. A typical power amplifier for example has a 4.5 to 5 degree phase shift at 20 Khz. Lower quality amplifiers lacking good high frequency response well past 20 Khz can have considerably more phase shift on the high end than 5 degrees. Perhaps even 12 to 15 degrees.
On the low frequency end flat amplfiers do not induce enough phase shift even bother measuring.
So we can all say that it is absolutely true that adding more equipment will increase phase shift in one way or another. But if adding a piece of equipment causes enjoyment to go up for the user then the equipments goal has been reached. Such a system with its added equipment can still provide extremely high quality audio. It should not be assumed that it is now all of the sudden low or mid fidelity.
Choosing equipment has always been a ball of worms. I have never been satisfied with any equipment I purchased. Thus for the most part always designed and built my own. I have done this on and off since the early 1960's. And of coarse started with tubes and I'am still totally comfortable working with them now.
Being a electronics technician by trade for almost all my life does cause me to think more into the technical aspects for things. I don't believe in the absolute minimalist way of designing audio equipment. I will attempt to explain.
For examply when a minimalist will use two transistors for a simple amplifier I may use seven. The extra five would be used to put the two main devices in a environment were I can get the absolute best performance out of the two that really count. Thus the seven is the minimal level of parts required to achive my goal in this particular example to meet my technical.requirements.
Let put this in a slighly different way. A unicycle has only one wheel. It will get you were you want to go but not with great comfort. So a second wheel is added and we call it a bicycle. It is faster and also more comfortable, but since we only used eight spokes in the wheels they wobble. So we have wobble distortion. To fix this we add more spokes in the wheels to control and reduce the wobble distortion.
Now after this is done we have a fairly well behaved bicycle. It no longes shakes with wobble distortion and we have a smooth riding machine.
The logic is quite simple more parts can indeed make things better. There is of course a limit. It is simply this, use the least amount of parts required to reach your goal. Beware however that there may be others out there who have different standards. Some may need a smoother ride yet and demand more parts. Others may want a rougher ride are remove a few.
This helps make life so interesting.
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