Op Amps

[Eric D]

Doc .... every op-amp manufactures or types have "Signature" sound (for lacking a better word). I used both (AD825 & AD826) long ago; they are laid back. BB is more lively and upfront (not in negative way i.e. harsh). LM6172 is also very very nice if you can prevent it from oscillating. I guess as long as your op-amp G = 1, then it is a bit harder to oscillate.
Many better op-amps are counterfeited. Maybe that is one of many answers that you do not hear the difference.

If you can hear the difference between even two kinds of high end op-amps, I am guessing you don't even use PG amps? The overall circuit on a PG amp has room for improvement, much more so than I feel an op-amp upgrade can fix. If you are talking a car stereo, there are many items in the signal path. I really cannot believe the op-amps in your amp would be the "weak link" you can pick out in the whole chain. I cannot believe you without some numbers to back things up.

[Eric D]

I was just thinking of another related tidbit as well. PG amps are filled to the brim with 2N4401 and 2N4403 transistors. These items are "jelly bean" parts which cost nearly nothing and are used all over the signal path. If I can believe any of you guys can actually hear a difference in your op-amps, then you pretty much HAVE to go through your amps and replace all of the above mentioned transistors, as they truly are lacking in audio ability. I would be 90% confident you will hear a difference in replacement of these parts long before you would hear a difference in an op-amp change.

[dBincognito]

The obvious answer is to just buy amplifiers without op-amps. Discreet components for the win!

I'd argue the point that upconverting a song that was recorded in 41k to anything higher will buy you anything/nothing. It's not like video where you can fill in the holes and questimate the picture details.

You're actually introducing distortion to the original track if you encode it in any way, be it up or down.

I produce music.....and I will argue the sample rate all day.......the sound quality is doubled every time the sample rate is......there probably is not 1 home audio receiver that samples at 41,000bps.....there is a reason for that......I will also go as far as to say....b/c I know for a fact......that a HU that will play a sample rate of 96,000bps......will sound better than a HU that has burr-brown's in it, that only plays at 41,000bps...You are not introducing distortion.....you are oversampling.....which cheap little HU's are not capable of doing.....I can render songs out at a higher sample rate b/c I have the technology to do it with( it can't be done correctly with any free software)....I do it all the time....if you try playing a higher sample rate than what your HU is rated for....nothing happens....no sound is produced.....b/c your HU cannot process the information fast enough.....and producing a higher sample rate has nothing to do with the coding....that would be changing the file type.....i.e...MP3...Wav...MP4...JPEG.....you are not encoding the file in any way by increasing the sample rate. The sampling rate, sample rate, or sampling frequency defines the number of samples per second (or per other unit) taken from a continuous signal to make a discrete signal. For time-domain signals, it can be measured in hertz (Hz). The inverse of the sampling frequency is the sampling period or sampling interval, which is the time between samples. The Nyquist–Shannon sampling theorem states that perfect reconstruction of a signal is possible when the sampling frequency is greater than twice the maximum frequency of the signal being sampled, or equivalently, when the Nyquist frequency (half the sample rate) exceeds the highest frequency of the signal being sampled. If lower sampling rates are used, the original signal's information may not be completely recoverable from the sampled signal.

For example, if a signal has an upper band limit of 100 Hz, a sampling frequency greater than 200 Hz will avoid aliasing and allow theoretically perfect reconstruction.

In some cases, it is desirable to have a sampling frequency more than twice the desired system bandwidth so that a digital filter can be used in exchange for a weaker analog anti-aliasing filter. This process is known as oversampling.


Sample rate conversion is the process of converting a (usually digital) signal from one sampling rate to another, while changing the information carried by the signal as little as possible. When applied to an image, this process is sometimes called image scaling.

Sample rate conversion is needed because different systems use different sampling rates, for engineering, economic, or historical reasons. The physics of sampling merely sets minimum sampling rate (an analog signal can be sampled at any rate above twice the highest frequency contained in the signal, so other factors determine the actual rates used. For example, different audio systems use different rates of 44.1, 48, and 96 kHz. As another example, American television, European television, and movies all use different numbers of frames per second. Users would like to transfer source material between these systems. Just replaying the existing data at the new rate will not normally work — it introduces large changes in pitch (for audio) and movement as well (for video), plus it cannot be done in real time. Hence sample rate conversion is required.

Two basic approaches are:
Convert to analog, then re-sample at the new rate.
Digital signal processing – compute the values of the new samples from the old samples.

Modern systems almost all use the latter since this method introduces less noise and distortion. Though the calculations needed can be quite complex, they are entirely practical given today’s modern processing power.

CDs are sampled at 44.1 kHz, but a Digital Audio Tape, or DAT is usually sampled at 48 kHz. How can material be converted from one sample rate to the other? First, note that 44.1 and 48 are in the ratio 147/160. Therefore to convert from 44.1 to 48, for example, the process is (conceptually):


If the original audio signal had been recorded at 7.056 MHz sampling rate, the process would be simple. Since 7.056 MHz is 160 x 44.1 kHz, and also 147 x 48 kHz, all we would need to do is take every 160th sample to get a 44.1 kHz sampling rate, and every 147th sample to get a 48 kHz sampling rate. Taking every Nth sample like this preserves the content provided the information (the audio signal) does not have any content above half the lowest sampling rate used (22.05 kHz) in this case.

So now the problem is how to generate the 7.056 MHz sampled signal, given that the original has only 1/160 of the samples needed. A first thought might be to interpolate between the existing points, but that turns out to have two problems. First, the frequency response will not be flat, and second, this will create some higher frequency content. The high frequency content can (and must) be removed with a digital filter (basically a complicated average over many points) but the frequency response problem remains.

The somewhat surprising answer is to replace the missing samples with zeros. So if the original audio samples were ..,a,b,c,.., then the 7.056 MHz sequence is ..,a,0,0,0,...0,0,b,0,0...0,0,c,.., with 159 zeros between each original sample. This too will create extra high frequency content (in fact it is worse in this respect than linear interpolation) but at least the frequency response is flat. Then the digital filter removes the unwanted high frequency content. The work of this digital filter is also much easier if zeros are inserted, since the filter is basically an average and almost all of the samples are known to be zero.

So inserting the zeros, then running the digital filter, gives the needed signal - sampled at 7.056 MHz, but with no content above 24 kHz. Then just taking every 147th sample gives the desired output. Which sample to start with does not matter - any set will work as long as they are 147 samples apart.


Insert 159 zeros between every input sample. This raises the data rate to 7.056 MHz, the least common multiple of 44.1 and 48 kHz. Since this operation is equivalent to reconstructing with Dirac delta functions, it also creates images of frequency f at 44.1−f, 44.1+f, 88.2−f, 88.2+f, ...
Remove the images with a digital filter, leaving a signal containing only 0–20 kHz information, but still sampled at a rate of 7.056 MHz.
Discard 146 of every 147 output samples. It does not hurt to do so since the signal now has no significant content above 24 kHz.

(In practice, of course, there is no reason to compute the values of the samples that will be discarded, and for the samples you still need to compute, you can take advantage of the fact that most of the inputs are 0. This is called polyphase decomposition, and drastically reduces the computation effort, without affecting the conversion quality.)

This process requires a digital filter (almost always an FIR filter since these can be designed to have no phase distortion) that is flat to 20 kHz, and down at least x dB at 24 kHz. How big does x need to be? A first impression might be about 100 dB, since the maximum signal size is roughly ±32767, and the input quantization ±1/2, so the input had a signal to broadband noise ratio of 98 dB at most. However, the noise in the stopband (20 kHz to 3.5 MHz) is all folded into the passband by the decimation in the third step, so another 22 dB (that's a ratio of 160:1 expressed in dB) of stopband rejection is required to account for the noise folding. Thus 120 dB rejection yields a broadband noise roughly equal to the original quantizing noise.

There is no requirement that the resampling in the ratio 160:147 all be done in one step. Using the same example, we could re-sample the original at a ratio of 10:7, then 8:7, then 2:3 (or do these in any order that does not reduce the sample rate below the initial or final rates, or use any other factorization of the ratios). There may be various technical reasons for using a single step or multi-step process — typically the single step process involves less total computation but requires more coefficient storage.

[dBincognito]

justonemoreamp and Eric D
How do you feel about the RCA's being on the same side as the power and speaker connections ?


I personally think it was a mistake to do....all the high end amps that I have seen....they go way out of their way to avoid the RCA's ever coming near the power and speaker connections. Everything I have read says it's not good to have those things near one another....any thoughts ?

1moreamp

As for black gate caps and such all capacitors can influence SQ

I'd like to talk about that....I am under the impression that only the Rail Capacitors influence the SQ....and not the Power Input caps....any clarification on that subject ?

[stipud]

Way to bring out Nyquist Randy! I haven't heard that name since studying Computer Science

[1moreamp]

The obvious answer is to just buy amplifiers without op-amps. Discreet components for the win!

This is why PG made the ZPA series I believe I still have two left that I begrudgingly refuse to sell at any cost. And I know where 6 new ones are, but alas my funding will not allow such exuberance at this moment...

Perhaps PG also knew something about this highly controversial subject matter. All discrete amps were all the audiophile rage years ago. I feel that op-amps are as they appear to be a circuitry simplification that allows for fast and easy design to reach a desired complexity goal of the final product...Whew that was a mouth full...

[1moreamp]

justonemoreamp and Eric D
How do you feel about the RCA's being on the same side as the power and speaker connections ?


I personally think it was a mistake to do....all the high end amps that I have seen....they go way out of their way to avoid the RCA's ever coming near the power and speaker connections. Everything I have read says it's not good to have those things near one another....any thoughts ?

1moreamp

As for black gate caps and such all capacitors can influence SQ

I'd like to talk about that....I am under the impression that only the Rail Capacitors influence the SQ....and not the Power Input caps....any clarification on that subject ?

It is always WISE to separate low level signals away from any power source. But many car amp companies in a attempt to make the amp more accessible to the owner/installer have looked the other way for a long time.
Take JL slash amps Installer's just love them, and owners that twiddle with knobs also love them. And I must admit it is convenient to have placed all the controls and input / outputs so centrally located. But it does invite induced noise headaches. Induced noise can the worst to resolve also, but JL made a small fortune off their design idea most from those that just do not know or understand the basic's of proper engineering.
With the properly selected noise canceling input circuitry the proximity induced noise issue can be filtered out.



As for Black gate caps and the such ALL interstage coupling on ALL car amps are using wet electrolytic caps, with the exception of possibly TRU, and one other European name that eludes me for the moment. < They offer mylar cap interstage coupling options like older tube amps used >

Every time there is a major connection jump say the input RCAs and in between the op-amp input stages and the main amp input they use electrolytic caps.
These caps are used primarily to block DC offset transfer. One thing higher grade Op-amps can do is reduce this DC offset issue to levels that allow for Direct coupling without wet caps. < possibly , this is also highly subjective material >
Anytime a cap is insert inline with the audio path your inviting "cap coloration" into the path with the music.
The old rule of simple is better applies in many cases in audio. But Op-amp's are for a lack of a better simple description just a micro sized power amp.
In fact standard Op-amp design is the basis for the majority of all class AB power amp design.
Consider it like this A Op-amp is mini version of one of the main discrete channels of the power amp, only in IC micro form, and lower power levels of course.
If you give it feedback it can filter or it can oscillate so can main amp stages if they are not engineered properly.

So armed with that ugly picture I just painted for you please understand that capacitors located in between stages and in the audio path are possibly a big issue, and standard designs mandate there usage to solve several stage connection issues but DC offset blocking is one I see on top all the time.
Any power amp can have DC offset. I have posted about this long ago right here on this forum. I even brought in outside links to support my rant so other intelligent information could be gleaned from the post and readers could formulate their own opinions based on their common sense.

Well Op-amps can also have DC offset also. I see it all the time. Just reference ground with a DC voltmeter and read in front of the wet cap and after the wet cap in between the op-amps and the main amp input . You will see DC offset voltage in front of the wet cap but not after.
In tube circuitry it blocked the huge plate voltages and allowed only music to pass to the next stage... In theory..

Anyway, in a perfect world, there would be no need for coupling caps IMO.
But we live in the world we have and there are coupling caps and their imparted coloration to the audio path, Which can be altered by using caps made by different methods, by different cap makers.
Hence the huge aftermarket upgrade cap arena that is out there on the market.
I like to use the old "threshold" trick of applying a 0.1ufd Wima across these electrolytic's just to restore high frequency response, and to clean up the corners of a square wave test signal. The Wima across the cap restores the high frequency response time just enough to improve the overall performance both visually on a O-scope and i think possibly sonically.
You can also see this exact same trick done in the Ti -Elite passive crossovers that Morel made for PG. These passive crossovers have this "threshold" mod across the big caps. And Yes I researched the mod till I found it titled this way buy other engineers. So please don't beat me up to bad if you disagree with the technology or the name, as I stole it from other peoples minds and I openly admit that fact.

To digress for a moment yes I agree that main rail supply caps are very important. They decide weather or not the amp has a "stiff" and well filtered DC voltage supply. By "Stiff" I mean a ample stored energy that prevent rail sag fro occurring when the amp sees high demand drive levels.
All amps suffer from main Rail sag. The question is how much at what power level and does it affect the amps distortion levels and the sonic quality of the output signal < Again we are treading into a highly subjective area here with a even deeper question about yet another dark and not well discussed area of amp design >

I personally have always ADDED more energy storage to most of my amps, including my PG amps. I used to do it ZED maps and other back in the 90's.
Caps are expensive, and engineering says the power supply if properly design can and will compensate for sag by pumping itself up and of course drawing a TON of Current off the power source. At home its the 110 AC line in the car it your poor little 70 amp hour lead acid car battery and stock alternator.

If you stiffen the secondary rail supply you can smooth out this high speed power draw and lower the impact on the 12 volt charging system. And I found in many cases you can lower the operating temperature of the car amp. Big Caps being expensive are usually used minimally to hold product costs inline with production quotas, and to keep size and weight down.

Back in the early 90's I fiddled with car amps and I on more then one occasion I added more main rail caps. Like by a factor of three to five times the original values the factory used.
The results were startling and well accepted by many of my local installer/owners. I did this to one Autotek 7300 for one of my installer buddy's at the Autophile here in Santa Clara and he managed to destroy 5 sets of four 10 inch kicker subs, and the amp never got above warm to the touch.
I of course was politely asked to remove the mod or Kicker would invalidate all warranty obligation to the Autophile if I did not. So in order to maintain the peace I was forced to undo the mod

So to recap my thoughts as this is getting way the hell too long to absorb, Yes I feel that rail caps are important, but I am inclined to research the cap value and viable increases to subjectively compensate for rail sag. Anything more is just a waste and can lead to quickly blown outputs...

[1moreamp]

On a somewhat related side note, Nelson Pass is my hero, my home amps are one of his designs, but as for Steve Mantz I have no respect for him or his work. Some of his amps are the most unreliable in history (we had countless problems with them back when I worked at Sound Buggy), and his latest complete disregard for the shipping of Vin's amps took any personal respect I had for him and flushed it down the drain…

Yeah I hear you about Steven Mantz. I have talked with him from time to time just to pick his brain. He's not stupid about audio technology. But I have never had any business dealings with him.
I did send one of my clients his way on a four channel Zed amp that needed part that I just did not have , and he did him Ok, abit pricey but OK.
But my client also sent him a US Amps 2000X as I recall and one of his now departed super tech's did a beef up on it and when it failed Steve did decline to cover it warranty wise. stating that "so and so had left the company and he was not inclined to clean up any of his mess's"

But Steve is still one of the big people in the industry and he has made bunch's money and history with some of his products. it's always nice to to be a talk with these types to glean information on the picture from a different point of view.
Just like John Yi of TRU , he is a very nice person, and is willing to spend lot so f his time talking about the technology and the industry as a whole. I found him to also be very insightful but from a different point of view, thus opening my mind up to other things I would have never thought about or questioned...

And you see Eric we do share somethings in common, I also hold Nelson Pass In very high regard. That guy does some fairly brilliant work. I started up on DIY just trying to follow his ideas about Mosfet amps and balanced amp circuitry. He is the reason why I even own Adcom amps as they are the closest thing to ever hit the car audio market that even resembles any of his work in Mosfet amplifier design.....

[Eric D]

Wow, there is definitely a lot of in depth and useful discussion going on here!

I agree with Cecil it is best to separate the terminals on amplifiers and isolate portions of them.

However, I am getting the impression Cecil and others differ with me in the area of practicality. I try to keep things as practical as I can, although I am confident my wife would disagree with me completely. I used to feel I was a more of an idealist, but I really think that was before I gained much engineering experience and a better understanding of the KISS method (keep it simple stupid).

If you look at an amp as an isolated item and strive for perfection, things have to be viewed differently. If you assume perfection on the input to the amp (both power and signal), and perfection on the output (flawless speakers), then the tweaks discussed in this thread become more useful to a degree.

I believe a change in a capacitor can cause a change in the output of the amp, but I also believe you could measure this. A cap upgrade may very well show up in the frequency response of the amp, which is really a pretty high level test of an amp with little depth to the information gained. I don't believe a change in op-amps would be measurable, and therefore I don't believe it would be detectable in our hearing. To actually detect a change in the output with the use of better op-amps would likely require starting with a very poor op-amp. In the case of these PG amps, the parts used are cheap, but they are not junk.

Now back to where I was going with all this. Since an amp is not used by itself, the overall experience is relying on your deck, your power source, any processors you use, your amp, your speakers, your speaker placement and your car. When you look at the whole system, the amp is a minor influence compared to the other parts. If you put all of the connections on one side of the amp, yes this is not desirable, but will you hear it, I think not. The remaining portions of the system are not good enough to hear a difference in the locations of terminals on an amp, or the sound of a better op-amp.

Now I present a challenge. I firmly do not believe a speaker exists which is revealing enough to expose the improvement in an amp by upgrading its op-amps, let alone human ears good enough of picking it up once the speakers produce it. I challenge anyone to prove me wrong. I don't know how to go about doing it, but I do know I worked long enough in the engineering department of a speaker company to have a very firm belief no current speaker technology exists with this resolution.

[1moreamp]

Yeah well you are correct about transducer induced distortion. You will be hell bent to get less then 2.5% distortion at reasonable listening levels out of any driver ever made by human's... but we are always on the grail quest of sonic godlyness now aren't we ???

As far as challenges, well I will leave that to younger folks. Its a bunch easier just to do ones best and leave it at that knowing every possible avenue was exhausted trying to achieve the goal... After all we can only do our best and hope for the best I feel...

[dedlyjedly]

okay...I know I asked for more technical info...but can I get the concise version! i nodded off a few times on the second page.

i kid, i kid. thanks for taking the time to share your input and opinions on these complex topics .

[neverman]

OK, I concede on the sampling rate, if done correctly part. I was actually talking about bit rates and blew it on reading comprehension, but in my defense it was late! I've seen friends take 192kbps songs and try to change them to 256kbps MP3's and think they were doing things for the better.

My Home receiver has 96khz sampling @24bits I believe (I don't want to look it up is all). Yes, it's doing that for a reason even if the source doesn't present that info as often as it'd like. It's ready and "smoothing" out what it does receive. To over-simplify, it's making a jagged sinewave more complete except many, many times over across a wide band of frequencies all at the same time.

Thanks for all the other info in here as well. I'd love to know how well the ZPA MPS show down would turn out SQ-wise.

[BigBrent]

Wow the conjecture and discussion has been great.. I am glad that I had the question regarding my M50. Thanks for the well thought out responses and the assistance.