Consider the mic amp

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TedF

Well-known member
Joined
Aug 19, 2005
Messages
181
Location
UK
Almost every day I read postings on this and other forums, about the relative merits of different commercial microphone amplifiers, together with opinions about why they sound different. Everyone seems to have an opinion, and more often than not it?s a simplistic view of various parts of the circuit; whether a transformer adds ?colour?, or the relative merits of different chips. It seems that almost no-one stands back from the subject and sees the whole picture.

I?m not trying to give answers here, I?m just trying to put things into perspective and stimulate debate and original thinking.

One of our biggest problems is the inability to understand the refinement of our own ears. Yes, we can measure amplitude frequency response and ?distortion? down to what we think are very fine limits, but our understanding of physics does not sit well with the real world of perception; for example, how important is frequency response? As a sales tool it used to be paramount, yet the reality is that it is very much less important to our ears than relative phase response (one reason why the few good EQs can enhance a sound while most destroy it!).
It?s so obvious, yet so overlooked and misunderstood that the most important factors in ?quality? of sound are accuracy of transient response, and phase relationships.

Looking at basic microphone preamp design, and the factors that I have read about recently in the forums?..
Yes, I?m sure that the Jensen transformer is as good as they get. The use of a transformer on the plus side brings galvanic isolation and the ability to ?match? the input impedance to the microphone. (to achieve an optimum bridging load), but the downside is that it will introduce phase distortions at the extremes, although these can be thought of as beneficial.

It?s good to get rid of the input coupling capacitors! While I include comments of ?smearing sound by electrolytics as snake oil, input caps often add limitations to LF performance, and can prove unreliable.

The amplifier itself is probably the biggest source of political opinion.
Modern op-amps, when used correctly are truly transparent. BUT the devil is in the detail! The amp will only operate perfectly if the power supplies are truly zero impedance and the current paths in the components around the amp are exactly as envisaged by the designer. This is never the case, so small performance changes occur between layouts and designs.

I believe that many ?front end? operational amplifiers suffer from stability problems during transients, and that while such ?distortions? are extremely difficult to measure, they can be clearly heard and described as ?brittle? or ?cold? sounds, that is why in my own designs I restrict the gain on ?front end? amplifiers and tend to use balanced circuit forms where there is at least a chance for some of these momentary distortions to cancel themselves.
Another remedy is to adopt low gain discrete circuits; less feedback means less chance of instability in this case.

The use of class A discrete circuitry of course has another major advantage, that is that non linearities in the amplifier show themselves as 2nd order distortion?. Acceptable to our ears; but that?s a favourite hobby horse of mine so leave it at that.

The last subject that has a gross effect on performance, yet is rarely mentioned, is overload margin. I learned the importance of overload margin designing preamps for broadcasting. It used to be a compromise between overload margin and noise, but that hardly applies now. A good mic amp set up with a ?normal? overload margin of say 20dB is going to sound distinctly average?. It will sound clean but uninteresting because of how it is treating transients; they will be hitting the power rails and causing those momentary instabilities. Increase the overload margin to 30dB and suddenly you have air and space in the sound reproduced; note, this is not an attribute of a great mic amp?. It?s what should have been there all along!

So, to sum up, it?s not the inherent quality of the SSM2017 or the new chips from ?Thats? that are important, it?s the way the mic amp deals with the unmeasurable, accidentally or on purpose!
:guinness:
 
Nice post..... you are right, some people love transformers.... including this dude. http://www.gearslutz.com/board/showthread.php?t=86261

But it is the sum of all the parts, and the implementation by the designer, that give the whole character. Your post has helped stretch my brain a little more.... thank you.

I'd like to learn more about overload margin.... more reading... :thumb:
 
[quote author="TedF"]
A good mic amp set up with a ?normal? overload margin of say 20dB is going to sound distinctly average?. It will sound clean but uninteresting because of how it is treating transients; they will be hitting the power rails and causing those momentary instabilities. Increase the overload margin to 30dB and suddenly you have air and space in the sound reproduced; note, this is not an attribute of a great mic amp?. It?s what should have been there all along!
:guinness:[/quote]

When you say 'overload margin' do you mean the range of cutoff compared to the intended maximum input signal?

I love transformers myself, I've gotten very good results with an average circuit and an overbuilt transformer. I don't know why all these mics nowadays have to skimp on the iron!!!! :mad: Maybe OK for broadcast, but not good for recording...

Good post!
 
Simply put, he means headroom before clipping.

Hey Ted, don't forget to talk about polarity inversion and other such nasties that lots of opamps have when they slam into the rails!

There are some really really nice opamps out there but few explore them. Most are content with the same old 5532/5534 setup because they are stable and cheap and offer decent performance. Everyone here has seen the $$L schemos and how they goosed great performance out of a middle of the road IC. Still, if you study their schemos you will see where they found these ICs to lack and replaced them with something else..

Again back to the opamp drama, people get put off with cost and with stability and tend to fall back onto the simple designs using the same old excuse of "it's good enough". It may be good enough but a few more parts could make it GREAT or even spending 5$ or more on a totally different opamp could be the improvement needed. I've been experimenting with the whole "air" concept. I'm trying to get my audio to sound like I'm standing in the room. This involves using good mics, good preamps and a lot of super high speed opamps that give me an extended freq response, not simply 20-20k that people accept as "good enough". The harmonics are important even if we cannot hear them.

I'm using high slew video opamps in a lot of my prototypes along with some other ICs that people ignore on a daily basis due to the herd mentality of "it's just an IC, I want transformers and discrete", something that Ted touched on in his post. He also turned me onto currentfeedback opamps some time ago too which I am finding to be delightful as well.

I guess it's the tone of the forums these days. We all started by cloning and building kits and such, some of us have moved on to designing, some just moved on. Lets talk about something truly new to the forum for once.

:thumb:
 
how important is frequency response?
that is a good point , I did an investigation on psicoacustic of a couple of month for part of my tesis work. we all want our preamps be very flat on frequency but our ears are not flat at all. on guitar amps we want a not flat frequency , maybe on mic preamps it should be on the same way.
 
[quote author="12afael"]
how important is frequency response?
that is a good point , I did an investigation on psicoacustic of a couple of month for part of my tesis work. we all want our preamps be very flat on frequency but our ears are not flat at all. on guitar amps we want a not flat frequency , maybe on mic preamps it should be on the same way.[/quote]

How would you want to capture a signal if not in a completely accurate way?

At least point me to some keywords for some further reading please!
 
[quote author="12afael"]we all want our preamps be very flat on frequency but our ears are not flat at all. on guitar amps we want a not flat frequency , maybe on mic preamps it should be on the same way.[/quote]

Maybe we want flat, but since two different preamps that both measure flat on the bench (and maybe even have identical THD and what else) still sound different there seems to be something beyond 'flat' or 'distortion' or 'headroom' we are not measuring.
For example, if you sweep a sine from 20Hz to 20kHz to figure out whether it's flat it will never tell you whether the level at 8kHz will still be the same when there is an additional signal at, say, 100Hz and close to the headroom limit. After all, we are listening to music which is in techical terms a lot of sine tones at the same time (with different freq. and level), not just one as on the test bench.

Olaf
 
I don't care about flatness, I have a rack full of flat preamps and mics that are supposed to be flat. make any combination of the above and they all sound different and very UNflat. Having a preamp that has an unflat freq response could be great as long as you knew what that response looked liked and could apply it.

Lets say my room has a node at 2.2khz which rises about 4 db, which it does, and all my preamps are flat. None of those preamps are going to help me keep from EQing that bump out. now if one of my preamps has a dip at 2k it would help me greatly.
 
Nice read ted :thumb:

I see it this way before anyone can judge a pre or a mic or any gear one must know their ears and the room they are in. It's been my expierence that I can record in the worst conditions and come out with good recordings because I know and trust my ears and because my ears can tell me how fooked a room may or may not be. When I was working at westlake and amorris would remember this, We were tune the main speakers with the owner of the company in one of our studio rooms. There had been a problem and after a while of teching it, things were not comming out as they should. The owner designed them so we brought him in to help in the matter. Aside from a blown super tweeter( I think it was a super might just been a tweeter) anyway aside from some blown componants due to a hip-hop session, The room had a funny bump in it. Where as the average decay time in the control room was something like 500 MS @ a given frequency give or take around 200 HZ it was decaying at something like 1.5 seconds. Far greater. After pulling to room apart it turns out there was an open/not sealed off conduit pipe. This was acting in a much similar fashion of blowing on a bottle and getting a tone. this had been going on for quite some time. How many engineers did mixes in their and didn't notice. As long winded as that was my point is without good ears no gear can help you. If you don't know what your listening for then what good is it?

As for flatness nothing is truely flat even a mic in Omni which yields the flatest frequency responce on that mic is not truely flat. Our ears do not work that way the world doesn't work that way. IMO
 
not simply 20-20k that people accept as "good enough". The harmonics are important even if we cannot hear them.

I thought most records were deliberately rolled off at around 15 khz anyway.....on my frequency chart the upper harmonics of the cymbal and violin only go to 16khz ......
 
I've noticed that both my analog and digital albums are often rolled off -3db at @30hz-15kHz. I believe broadcast audio is roughly -3dB @80Hz-12kHz.

Doesn't this imply a phase change leading up to the rolloff points also?

I'm wondering if this phase change doesn't apply to most digital processing :?:

What if one could compensate the phase angle to accomodate a broadcast pre-emphasis. I'm sure it's already being done - I think that the higher end broadcasters have phase rotators before transmission.

I'd love to read that paper on supersonic harmonics.

Either way, I personally believe in clinical accuracy. As good a reason as any to enjoy this thread...
 
Transformer specs are pretty useless. Take a non symetrical wave form, maybe in the shape of a square wave, and put it through a transformer.
All sine wave calculations get thrown out the window when that happens.
And with modern music, there is plenty of square wave around.
 
A few thoughts, at random:

[quote author="TedF"]Yes, I?m sure that the Jensen transformer is as good as they get. The use of a transformer on the plus side brings galvanic isolation and the ability to ?match? the input impedance to the microphone. (to achieve an optimum bridging load), but the downside is that it will introduce phase distortions at the extremes, although these can be thought of as beneficial.[/quote]

Not a whole lot, though, in the case of a good transformer. Looking at deviation from linear phase, by Jensen's specs the JT-115K-E is off by 3.5 degrees at 20Hz, while the JT13K7 is off by 1 degree at 20Hz. They're both dead zero at 20kHz. Within the audible spectrum, at least, that's pretty flat phase response.

The amplifier itself is probably the biggest source of political opinion.
Modern op-amps, when used correctly are truly transparent. BUT the devil is in the detail! The amp will only operate perfectly if the power supplies are truly zero impedance and the current paths in the components around the amp are exactly as envisaged by the designer. This is never the case, so small performance changes occur between layouts and designs.

Yes and double yes. A lot of opamps get a bad rap because they're run from crappy power supplies and behave accordingly. Properly fed, they can sound very good.

I believe that many ?front end? operational amplifiers suffer from stability problems during transients, and that while such ?distortions? are extremely difficult to measure, they can be clearly heard and described as ?brittle? or ?cold? sounds, that is why in my own designs I restrict the gain on ?front end? amplifiers and tend to use balanced circuit forms where there is at least a chance for some of these momentary distortions to cancel themselves.

It would be interesting to see some examples. I'm not expressing doubt, and I understand the difficulty of measuring this, but I suspect that if it's happening it's detectable, perhaps with a good storage scope.

The use of class A discrete circuitry of course has another major advantage, that is that non linearities in the amplifier show themselves as 2nd order distortion?. Acceptable to our ears; but that?s a favourite hobby horse of mine so leave it at that.

Well, only if the class A discrete circuitry is unbalanced! A push-pull output stage will still produce mainly odd-order distortions whether it's integrated or discrete.

The last subject that has a gross effect on performance, yet is rarely mentioned, is overload margin. I learned the importance of overload margin designing preamps for broadcasting. It used to be a compromise between overload margin and noise, but that hardly applies now. A good mic amp set up with a ?normal? overload margin of say 20dB is going to sound distinctly average?. It will sound clean but uninteresting because of how it is treating transients; they will be hitting the power rails and causing those momentary instabilities. Increase the overload margin to 30dB and suddenly you have air and space in the sound reproduced; note, this is not an attribute of a great mic amp?. It?s what should have been there all along!

Yeah, it should never overload. Note that raising the overload voltage of the circuit is one way to do this; the other is to lower the operating level of the signal. Lowering the river rather than raising the bridge.

But an op-amp-based circuit should have a clip light on it, and it should have one that reacts to a single transient with a visible flash. That means something a little more complex than the usual indicator, like a comparator which triggers a timer or a peak-hold circuit so the clip light stays on longer than the transient itself.

So, to sum up, it?s not the inherent quality of the SSM2017 or the new chips from ?Thats? that are important, it?s the way the mic amp deals with the unmeasurable, accidentally or on purpose!
:guinness:

Rather than "unmeasurable" how about "not usually measured"? I suspect the subtle stuff is measurable if we choose the right measurement techniques.

Oh, and let me throw it my own pet hobbyhorse: I strongly suspect the ear is differentiating between different sorts of distortion behavior at very low levels, even though it does not hear the signal as overtly distorted. Instead, my suspicion is that it maps the differences in distortion behavior as differences in timbre, or in more obscure descriptives such as "soundstaging".

Peace,
Paul
 
Thanks pstamler for spotting one of my over-simplifications..... :sad: yes, even order distortion is only predominant where the class A circuitry is unbalanced; as soon as you look at balanced circuitry, you need to be sure that the normal types of non-linearity are very low indeed, to avoid 3rd order and higher nastiness.

Paul (pstamler) has that air of confidence about him that shows his keenness and appreciation of audio; he believes completely in our ability to get to the bottom of all these areas of debate, that we will be able to measure and quantify things like those momentary instabilities, and I'm sure he is right, but I'm equally sure that it's an economic impossibility!
Even a quick look at these sorts of problems throws up the peripheral contributing factors that work to confuse.
But that's why we are here, on this forum!

I agree very much with Paul where he says that the ear is diferentiating between different sorts of distortion behavior at low levels (although I think he means 'the brain' rather than the ear).
This particular debate needs to take notice that odd order distortions are not found in nature :wink: Even order 'distortions', or more correctly 'deviations' abound in all we hear.

Thank you Paul for your random thoughts, that turned out to be not so random!
 
[quote author="TedF"]Paul (pstamler) has that air of confidence about him that shows his keenness and appreciation of audio; he believes completely in our ability to get to the bottom of all these areas of debate, that we will be able to measure and quantify things like those momentary instabilities, and I'm sure he is right, but I'm equally sure that it's an economic impossibility![/quote]

Well, for professionals who have cost centers and bean counters and time contraints and all that. I suspect some of the work could be done by talented amateurs...or academics.

I agree very much with Paul where he says that the ear is diferentiating between different sorts of distortion behavior at low levels (although I think he means 'the brain' rather than the ear).

Yes -- that's my over-simplification. I meant the ear-brain system.

This particular debate needs to take notice that odd order distortions are not found in nature :wink:

Clarinets? (Which reminds me -- I gotta call our clarinet player.)

Peace,
Paul
 
Of course, there's nothing natural about a clarinet. Then again, the clarinet bird, found in a dwindling habitat in central Alabama...okay, just kidding.

Peace,
Paul
 
I've been reading this thread over and over again... Good stuff.

I'm more of a synth DIY-er but this discussion is just as valid in that field! Everyone seems to want to build a DIY Moog filter clone. Sure I've built one myself but at some point it gets boring to see the same circuit being discussed for the 1000th time. Especially because that type of design sounds rather plain by itself. The Moog sound (if that's what you're after) has just as much to do with the sound of the oscillators and especially the VCA and the way they interact with that particular filter design.

But now I'm beginning to get into audio DIY, I've come to a better understanding of the 'newbe' situation. The fact of the matter is that most people are insecure about their DIY-ing. So they put in a transformer that some famous designer once used (even though half of the components in the rest of his design are chosen to compensate for the crap that same transformer probably gave him) just so they can feel more secure. They stick with tried and tested designs, put in a few tubes, preferably matched and of certain brands, etc. It's a straw to hold on to in a field of work that they don't fully understand. They need a way to make sure that the box they've built sounds good.

I say 'they' but I'm actually one of them! So be patient with us like I'll be patient with the next newbe who wants help with his ladder filter thingy. It's all just a lack of confidence.

Anyway I want to thank you for this thread. I was planning on doing a slowblow srpp micpre (is that thing component junky heaven or what?) because I have a vibraphone which is proving difficult to record. But now I think I'm going to pick up the challenge and do an opamp pre.

The problem with the vibraphone is as follows:

- you can clearly hear the mallet hitting the bar (I like that)
- then there is a small silence (that's the problem)
- then the resonant part of the sound starts (sounds fine again)

So the mallet and bar sound don't glue together like in real life. Is that a typical problem like the transient response mentioned above? Or not enough headroom? There's no audible clipping though...
 
Hi Mickey,
That's an interesting problem! I have to admit that the last time I recorded vibes seriously was back in the early 80s. I used a valve micamp and a Neumann KM84; I remember quite clearly because of the way I had to set the micamp gain well down to allow for that thumping transient as the hammer hits the note. I think I got over the problems by using a High-pass filter to get rid of the LF in the initial thump, and an Altec 436C compressor to even out the sound. The vibes is a strange instrument; it's very difficult to record what you hear; mic it too close and you get serious hammer noise problems, mic it too far back and the acoustics of the room and crosstalk become the problem.
In terms of actual sound, I don't think it's too much of a problem, your main concern should be dynamic range and overload margin; like the piano it's easy to get transient overloads and not realise it.
 
I've noticed the same problem with certain cymbals too. I use sabian aax and you get the same initial attack with a signifigant drop in sound then an overwhelming mid range resonance. It seems to do that regardless of the mic used and the preamp used. It's likely my recorder since that is the only item in the chain that I haven't changed out in an attempt to figure it out.

I record the cymbals as hot as possible to get as much range as possible with the 24 bits, then I compress pretty heavily with a varimu. attack/ release times and compression level can sort of work the missing pieces back in but unfortunately it tends to make the initial attack and subsequent sustain sound a little strange, at least when comparing to the real thing. Most folks who haven't heard the source say it sounds fine though..

Maybe it's a problem with the components of the initial attack causing some kind of phasing at the mic diaphragm itself? If I stand certain places in the room I can hear the cymbals swish in and out but to my ears I can still hear the reverb components and other reflections "fill in" the missing sounds.. with close mic'ing maybe those are getting left out?

I'd try moving the mics further away and moving them around to see what happens to that sound and let us know what happens.
 
Good points both, thanks.

I don't have the kind of resources Ted mentions though. I use a pair of oktava small diaphragm mics because that's the only pair I own and I like the sound better with a little stereo spread. I tried a tube mic and a ribbon also. The ribbon mic giving the best results but I only own one of them and the difference isn't spectacular so I decided to stick with the oktavas. Best results with the mics 60cm (2 feet) away from the instrument and using the omni capsules.

So that seems to prove you right, Svart. The room sound fills in the gap pretty nicely. It's an overdub so no bleed through from other instruments and the room itself is decent sounding. So luck is on my side this time!

Still I feel a good mic pre would improve the sound. The mics go straight into the desk at the moment because I only have one decent mono pre (the stereo problem again). The desk is a soundtracs mxr so it's not a disaster but still far from ideal.

Anyway... this isn't what this thread was about so I'll stop right here.
 
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