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ELI5, please, why so many fancier mics use BJTs or JFETs instead of op-amps? I would think the op-amp is the right tool for amplification circuits. What is this aging EE forgetting?
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Another Annoying Beginner Question: OPAs vs BJT/JFET Mics
- Thread starter Andy Froncioni
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That's... uh... that's it? #cough
rogs
Well-known member
I am one of those who agrees with that sentiment. ... not everyone does.
It you are looking to create a linear, low distortion, large headroom impedance converter for a condenser mic capsule you'd find it difficult to beat an op-amp circuit - with a suitably chosen op-amp of course.
A discrete JFET front end is only likely to offer a slightly better specification when it somes to one characteristic - the noise floor.
Although op- amps - like the OPA164* series - can offer pretty good self noise figures, it should be possible to design a JFET input stage with a lower noise floor. The distortion and headroom figures are likely to be inferior to the op-amp figures, but not the noise figure.
In reality of course, it is ambient noise that is likely to dominate the generated noise in most real word scenarios, but the JFET noise level can actually be a little lower than an op-amp design.
When it comes to designing pre-amps that don't have linear properties, all the rules change of course. Some specific mic projects will deliberately introduce non linear characteristics into the signal path, for a variety of 'special' reasons.
But that's a whole different subject, often much less concerned with the laws of physics !
Also most famous mics that are copied over and over again use fets. Fancy op-amps that are suitable for mics (low current, low noise...) are relatively new thing.
There's also unjustified bias against opamps in whole industry (probably because of early, poorly performing ones). So you might hear things like cold, sterile, harsh, slow when op-amps are described. Many high end manufacturers avoid them for marketing reasons, discrete circuits sound fancier. Some even make discrete opamps (made from individual components) to give the "VIBE" regular op-amps lack. Mostly marketing crap, or delusion.
On the other hand sound sources like snare drum, have initial very sharp and high in level transient. It is for the most part useless. It can randomly trigger compression where you don't want it, drum buss master buss...
Many engineers count on specific mic, or more often mic preamp to soft clip part of that transient and act as a limiter (neve 1073 is often used for this), so it does part of the job for them. Many don't even know what it does but they like it. Op-amp clipping sounds very bad for this kind of job, so you don't want any clipping coming from an op-amp in this case.
Having that said, i wanted a condenser with it's capsule properties but with high headroom op-amp circuit, so I can get that transient into my DAW without any clipping. So i built one. I like the flexibility of dealing with these transients in post.
Many engineers count on specific mic, or more often mic preamp to soft clip part of that transient and act as a limiter (neve 1073 is often used for this), so it does part of the job for them. Many don't even know what it does but they like it. Op-amp clipping sounds very bad for this kind of job, so you don't want any clipping coming from an op-amp in this case.
Having that said, i wanted a condenser with it's capsule properties but with high headroom op-amp circuit, so I can get that transient into my DAW without any clipping. So i built one. I like the flexibility of dealing with these transients in post.
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rogs
Well-known member
With the (gradual) move towards 32 bit float recording, digital clipping will slowly become much less of a problem.
The line input of my Zoom F3 recorder can accept a +24dBu input before clipping ( That's 34.7v p-p !)
Now, whether there will now be a whole new generation of 'compression' VSTs to try and copy the famous 'soft limiting' effects of analogue devices - like the Neve 1073 - we shall see.
The dreaded hard digital clipping should become a thing of the past, once 32 bit float becomes the dominant format...
How long that will be?.....
This thread seems to be opening up a whole new can of worms I'd never thought about, with respect to if, when, and where to use EQ in the signal chain.
Suppose you have a square wave (low-passed to avoid aliasing but not phase shifted at all).
The rising and falling edges have a tremendous slew rate because those edges are still nearly vertical. (For other noobs, "slew rate" is how fast the voltage can change. For a true square wave, the voltage changes from high to low or low to high instantaneously, but real amplifiers don't do that; the voltage changes with some maximum slope.)
Now suppose you've EQ'd it and the important frequencies aren't in phase anymore, so you don't have those big nearly vertical bits. The slew rate limit of the amp may not matter anymore, because the slope is never anything it can't handle. Or if does matter, it matters over much smaller ranges.
I went looking for illustrations of this and found the attached pictures, of a square wave filtered down to just three (odd) partials, and the same waveform with those partials shifted 30 degrees (but different absolute amounts dependent on frequency). In the latter picture you still get a darned steep rising bit, but not quite as steep or rising as far, and less steep falling bits, given the now-asymmrical waveform.
If you're counting on a slew rate limited head amp or preamp to shape your sound, it seems like it COULD matter what EQ you put ahead of that amp.
OTOH it seems that the natural phase shifts you get from e.g., the mass of a mic diaphragm resisting small fast changes will have similar effects to doing it with EQ, if I understand Ricardo correctly, so using a mic with limited HF response and using EQ to deemphasize HF may have similar effects on the waveform.
Credit where it's due: here's where I stole the pictures from.
https://www.edn.com/for-signal-distortion-phase-matters/
Hmm... that makes me wonder what would happen if I played my electric guitar through an amp where I could control the slew rate limit independently of the usual kind of harmonic distortion. What does pure slew-rate-limited distortion sound like?
Suppose you have a square wave (low-passed to avoid aliasing but not phase shifted at all).
The rising and falling edges have a tremendous slew rate because those edges are still nearly vertical. (For other noobs, "slew rate" is how fast the voltage can change. For a true square wave, the voltage changes from high to low or low to high instantaneously, but real amplifiers don't do that; the voltage changes with some maximum slope.)
Now suppose you've EQ'd it and the important frequencies aren't in phase anymore, so you don't have those big nearly vertical bits. The slew rate limit of the amp may not matter anymore, because the slope is never anything it can't handle. Or if does matter, it matters over much smaller ranges.
I went looking for illustrations of this and found the attached pictures, of a square wave filtered down to just three (odd) partials, and the same waveform with those partials shifted 30 degrees (but different absolute amounts dependent on frequency). In the latter picture you still get a darned steep rising bit, but not quite as steep or rising as far, and less steep falling bits, given the now-asymmrical waveform.
If you're counting on a slew rate limited head amp or preamp to shape your sound, it seems like it COULD matter what EQ you put ahead of that amp.
OTOH it seems that the natural phase shifts you get from e.g., the mass of a mic diaphragm resisting small fast changes will have similar effects to doing it with EQ, if I understand Ricardo correctly, so using a mic with limited HF response and using EQ to deemphasize HF may have similar effects on the waveform.
Credit where it's due: here's where I stole the pictures from.
https://www.edn.com/for-signal-distortion-phase-matters/
Hmm... that makes me wonder what would happen if I played my electric guitar through an amp where I could control the slew rate limit independently of the usual kind of harmonic distortion. What does pure slew-rate-limited distortion sound like?
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probably sounds like intermodulation distortion...
Slow active circuits often rectify RF radio station interference.
JR
discrete fets often have lower gains and require less feedback, so you can get nice-sounding audible lower order harmonics in the signal. this contributes to the "vintage sound" by some amount that i'd love to see someone measure. maybe i'll do it. really, it's all about tone shaping. you can get the same effect with an OPA mic using a really good subtle distortion plugin, like fabfilter saturn. same with transformers. it's really just for the fun of it. there's nothing wrong with OPA headamps or, for that matter, electret capsules either.
rogs
Well-known member
As I have mentioned in other threads, I'm a fan of dealing with all things 'non linear' (EQ, compression etc...) outside the microphone.
Introducing extra circuitry- especially passive EQ - at the start of the signal path inside the mic will almost certainly add some noise, as well as the intended modifications to the sound.
Adding audio processing to the mic signal after it has been amplfied to a more manageable line level poses a lot less noise problems.
For live situations - which are often more likely to be speech based - then using a channel strip like the dbx 286s - or even the budget Sub Zero RS100 - can deal with most signal modification requirements.
When it comes to recordings, then we are talking about dealing with signal processing in a DAW.
As I mentioned earlier, the trend towards 32 bit float recording, especially when used in conjunction with a high sample rate for increased bandwidth, should allow for high resolution 'unclippable' signals with an upper bandwidth limit up to 2 octaves higher than the conventional 20KHz limit.
Once you've captured this 'pristine' high slew rate, high dynamic range signal into a digital format, your post processing options are much greater than they've ever been.
At the other end of the scale, it might be thought more desireable to add some 'colour' into the mic preamp itself.
That might be either by trying to copy an exisiting design, or by intentionally introducing 'distortion' of one type or another, to make a 'nicer' sound.
Delving into that world however really is a 'whole new can of worms'!
Matt Syson
Well-known member
24 bit A/D and D/A is still unachievable due to real world physics when you consider NOISE over a 20 to 20K bandwidth because the signal is either too quiet or needing excessive voltage rails at the 'other' end of the range, so although 32 bit is undoubtedly good for PROCESSING a signal with unrestricted heqdroom, ultimar=tely you are stuck at the analogue interface. Yes 24 bit is possible if you restrict the bandwidth but by the time tou have put all the tricks together to make it work it becomes impractical. Arguing for a bandwidth limit above 20KHz has some merit but it largely makes things DIFFERENT but not necessarily technically correct AND can introduce problems like the recording of spurious HF interference (mobile phones and switchmode supplies that we are surrounded by) and the need to filter these signqls in the 'correct' place. Having to work in a Faraday caqge and have practically no gear using power or signql frequencies above the audio band (20KHz) so where do you put the restrictions? Marketing and the internet have a lot to answer for.
thor.zmt
Well-known member
What you state is true, for example, for a measurement microphone. There we want an absence of "character".
For most recording tasks a truly characterless microphone is not what recording engineers want. Which is why rarely see "measurement" type microphones used for recording.
As 'ideal' microphone we best use a large array of completely digital MEMS micro sized microphones (High SPL version) to get a flat response, no (read very low) distortion and a complete absence of analogue electronics, then we add desired directivity, frequency response shaping and distortion in DSP.
That and not Op-Amps is the right tool for microphones in the year 2567 (Buddhist Calendar).
Thor
One thing to add is that many engineers are operating under a time-crunch in the studio (due to artist budgets). In those situations, “fixing it in the mix” often isn’t feasible. There are many situations where I have grabbed a mic that I knew would soft-clip or band pass the source I was recording so that I would get what I needed to begin with. I don’t always have the time to go though and high-pass or transient-shape all my tracks, and in those cases, microphones that behave in a known (but not linear) way are invaluable.
Other situations may call for absolute linearity, but in my experience, those are fewer and farther between.
FET's are STILL better. I use them myself... e.g. 2SK117's and 170's (Real Toshiba ones!)
Question: Aesthetics aside, most OPAs need 32 v rail to rail. At that voltage the most current available from phantom pwr would be 16/v drop across 3400 ohms (2x 6.8k phantom pwr Rs in parallel) =4.7 ma.
Is that enough to drive a 600 ohm mic input w/o clipping?
Is that enough to drive a 600 ohm mic input w/o clipping?
Cqwet Dbdfte
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FET based mic circuits have very little gain. The typical Schoeps circuit has the FET in no gain phase splitter followed by PNP emitter followers. So no gain at at all.
The transformer based mikes has the FET with gain, maybe 5x, followed by a step down transformer, maybe 10:1. Capacitive feedback on the FET reduces gain more. So overall not much gain in the mike circuit.
An OP amp has much gain to offer, relies on feedback to set gain, and get more linearity, and if no gain is needed, and noise is not improved, and if the OP is set up as a unity gain buffer, what advantages does it offer?
The FET is merely a charge to voltage converter, followed by some impedance buffer. It has been kicked around for 50+ years with little differences in circuit topologies (excluding the variable capacitance to RF based circuits).
As a mic is merely a transducer, not much else is asked of it.
Gain and signal processing is handled downstream, so there is not much need for gain.
Even antique tube mikes are treated with veneration and may cost half a fortune.
Evidently OP's do not offer anything that resembles an advantage. Predictability is also a time saving factor in a production setting.
The transformer based mikes has the FET with gain, maybe 5x, followed by a step down transformer, maybe 10:1. Capacitive feedback on the FET reduces gain more. So overall not much gain in the mike circuit.
An OP amp has much gain to offer, relies on feedback to set gain, and get more linearity, and if no gain is needed, and noise is not improved, and if the OP is set up as a unity gain buffer, what advantages does it offer?
The FET is merely a charge to voltage converter, followed by some impedance buffer. It has been kicked around for 50+ years with little differences in circuit topologies (excluding the variable capacitance to RF based circuits).
As a mic is merely a transducer, not much else is asked of it.
Gain and signal processing is handled downstream, so there is not much need for gain.
Even antique tube mikes are treated with veneration and may cost half a fortune.
Evidently OP's do not offer anything that resembles an advantage. Predictability is also a time saving factor in a production setting.
Back last century (for Peavey/AMR) I designed an op amp based direct box (EDB-1) powered by phantom voltage. I was able to extract enough current and voltage to supply a BiFet op amp, with volts of output swing.
A low noise mic preamp IC probably needs more current.
That wheel is already pretty round. Not much reason IMO to reinvent that wheel.
JR
With modern opas, that's way more than enough. Low noise true jfet opas need 300 nanoamps per channel on the low end and under 2 milliamps per channel on the very high end. Maybe a couple years ago there wasn't enough power but now you could easily run two, four, even potentially 8 channels on phantom power alone now, and that's with THD <0.0001% and negligible noise. These opas are designed to take microphone capsules straight to line level impedance too, so you don't even need additional amplification afterwards.
Technology has advanced to the point where you could power an entire u67, tube and all, with no schematic changes, and DA convert it with power from a single a single usb-c cable without ever interfacing with mains power. USB tube mics are coming.
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