But, sadly, that's exactly the FR curve that many listeners think of as "real" in a singer (which is where that mic is often found). It represents the-sound-of-audio, at least for female singers and has those sizzling breathy sibilants many listeners take as realistic (and sexy) when played back at home. If the studio engineer used a flat condenser mic, the engineer at the board would have to dial-in EQ to mimic that curve or, sadly again, customers would complain.Speaking of nasty-sounding resonances within the audio frequency range, the attached image shows a famous audio transducer that is the worlds best selling model in its category. As you'd expect from the multiple nasty, poorly controlled resonances seen in its frequency response, it sounds utterly horrid.
Astonishingly, hardly anybody seems to notice how bad it sounds, and it continues to be a best-seller, literally the most popular device of its type.
Now, when Toole uses his favourite female singers as test sounds for group tests or a certain other DIYer with big horns and 15kw power, just what is going on? Perhaps people like "flat" speakers with proper dispersion in order to play their cooked recordings.
Footnote: strictly speaking, may not be "sad" if it makes listeners happy. Not for me to judge if people prefer some cosmetic treatments on their audio. But that is quite different from extolling what is actually the arbitrary virtues of something ill-defined called "flat".
B.
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Uh, OK, it's hard to argue with that.
Perhaps he/we listen to recordings where the producer knows what he is doing.
If the game is to edit my post until all context is lost so that you can mock it, I can play too:
-Gnobuddy
A rather extreme conclusion, methinks?
Good help is hard to find, lucky tight!
-Gnobuddy
I don't want to drag your thread off-topic, but in just the last few years, I've converted four women singers (three altos, and one who sings both alto and soprano) away from their precious SM57s simply by having them listen to themselves singing first through their SM57, then immediately switching to a Nady SPC 25 ( Amazon.com: Nady SPC-25 Condenser Vocal Microphone - High sensitivity, cardioid pickup pattern, powered with AA battery or phantom power: Musical Instruments )
Once they had heard the simultaneously muddy, muffled, and screechy sound the SM57 produced, and the astonishing improvement through the condenser mic, they all wanted to know where to get the SPC 25, and how much it cost.
On the other hand, I have never converted any of the male singers I know away from their SM57. The men all "know" that the SM57 is the best and most popular handheld vocal mic in the world, where none of the women knew or cared. So the men won't even try an A/B test with the Nady - they already know they have the best-selling hand held stage mic, so why would they bother listening to anything else?
To be fair, I have heard one (and only one) female singer whose voice sounded great through an SM57. She also sounded great with no microphone. I don't know what characteristic of her voice allowed it to run the gauntlet of an SM57 frequency response unscathed - but she had a narrow alto range, which I suspect fit in between the boomy bass and screechy midrange/treble of the SM57.
Also to be fair, the SPC 25 is not suited for singers who bellow - you can overload the microphone if you're the type who screams, roars, shrieks, and bellows their way through a song.
The Shure, on the other hand, has a diaphragm made of old shoe-leather or maybe chunks of truck-tyre or something. It sounds like crap at all volume levels, but you will not manage to overload the capsule itself. So it's well suited to belters, screamers, and shriekers. I am told (by almost every reviewerr) that it is also good for driving nails into things.
-Gnobuddy
Point taken, but I still think that the phrase that I quoted, even in context,
is kind of superfluous.
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If the context is still unclear, this was in response to a comment you made earlier: "A big resonance inside a band which has hundreds or thousands of acoustic resonances - what's one more?"
My point was that we can hear the difference between resonances in the loudspeaker, and acoustic resonances in the room. That's why you can tell the bad speaker from the good one, even in a live room with hundreds of acoustic resonances.
Our ear/brain mechanism doesn't work the same way as a single measurement microphone. Our ears can process the hundreds of room acoustic resonances / reflections / associated time delays & phase shifts, and make an estimate of the *room* properties (size, reverberation time). Those room resonances do not have the same effect on the sound as poorly controlled resonances in the speaker itself do.
I first encountered this effect nearly twenty years ago, when I was part of a team attempting to flatten the frequency responses of high quality studio monitor speakers using DSP in the audio chain. Initially we attempted to flatten the frequency response as measured in-room with a measurement mic - and the results sounded bad, particularly if we allowed the corrections to extend up into the midrange and treble regions of the spectrum.
If, on the other hand, we flattened the anechoic response of the same speakers, and then moved them into the listening room, and switched in and out the corrective filter, the speakers sounded better with the corrections.
The measurement mic couldn't tell the difference between room modes and speaker errors - but our ears could.
-Gnobuddy
Gnobuddy,
The SM58 is the most popular handheld vocal mic in this part of the world (USA), the SM57 is a lower cost popular "instrument" mic. The SM57 wind screen is not appropriate for most hand held use, though is sometimes chosen because it's proximity effect is greater than the SM58 due to the diaphragm being closer to the windscreen.
Vocal mics are an artistic decision, you are free to choose whatever you or the artists you record use, or attempt to convince others that their choices could be "improved".
Art
Sorry, silly slip of the mental cogs on my part. I have never been able to keep the two numbers straight in my head.
But the appearances are quite distinctive, and the nearly spherical metal grille of the SM58 hard to mistake. It is the SM58 that sounds like garbage. I have no experience with the SM57.
The official Shure frequency response for the SM58 shows the same sort of poorly controlled, high-Q resonances in the mid/treble range. The microphone has the "shrieky" sound to match.
Of course. But I find it curious when the most popular choice is also the one that has some of the worst technical specifications, and sound that is so remarkably bad that the first thing the person at the mixing desk does when she/he spots an SM58, is dial in a massive bass cut - often as much as 10-15 dB.
I will add that this is not the only respect in which I fail to understand many of the very popular choices made by my own species. At least choosing to use an awful microphone is relatively benign in the big scheme of things.
-Gnobuddy
You make the "context" clearer, but I would take excepts to your explanation. However, we are already way off topic in this thread and I am concerned about taking it further off-track.
I'd love to carry this one somewhere more appropriate however.
Wow. What a thread. It took me two days to get to get here.
I'm very late to the party but I have also dabbled with MFB. Interesting reading!
In some ways my experience is eerily similar to Gnobody's. I nearly fell off my chair when I read that he (I'm pretty sure it is he and not she) started with an LED, phototransistor and light blocker. I started with an incandescent bulb, light-dependant resistor and "calibrated" curved cardboard light blocker glued to the back of a woofer cone. The displacement feedback worked - but only at infrasonic frequencies well below system resonance. Then came the discoveries why this was so limited etc., etc.
Gnobody - you have an excellent practical understanding of the acceleration feedback topic and explain things very clearly. Hat's off for sharing your experience and knowledge. I think when I get some more time it might be worth starting a new thread to share information on the acceleration feedback system I played with way back when. The good thing about that is I still have all the circuit design and test result information I can share FWIW. I'm also interested because want to try MFB again at some point.
For anyone reading this thread who might be struggling a bit with what is happening to a loudspeaker in a closed box and all this talk about how it goes though a 180 Deg. phase shift and the response falls off at 12dB/Oct, perhaps the thought process I went through back then can help others to visualize it.
Consider the two regions the LF (Low Frequency) speaker needs to operate in. Unlike the midrange and tweeter we want the LF driver to function through its natural frequency. It is helpful to consider these regions the "mass dependent region" above Fr where F=ma rules and the "compliance dependant region" below Fr where F=kx rules.
First, with regard to amplitude, as Gnobody has so clearly pointed out, at low frequencies (the compliance dependant region) the displacement is proportional to the force exerted by the voice coil (F=kx). Displacement will eventually be a constant regardless of frequency as the frequency is lowered. We are conveniently ignoring the error in the air spring contribution to the constant k at very low frequencies due to leakage but that's okay - we are just trying to visualize what us going on.
How does this change the response? Well, above resonance, the amplitude has been increasing as frequency drops to keep the SPL constant. Below resonance, this isn't happening so the SPL is falling off. In mathematical terms, the RMS pressure is proportional to the square of the frequency and if you do the math this reveals the characteristic 12dB/Oct decay where displacement is a constant. You don't need to know the math, just that the reason for the decaying response can now be easily visualized due to the constant displacement characteristic.
Now with regard to phase, also at these low frequencies below resonance where F=kx, maximum displacement is going to coincide with the time the force is greatest (at the peak of the voice coil current for a sine wave). I think this is pretty easy to visualize. But above Fr where F=ma, the maximum force F (and maximum VC current) is not coinciding with maximum displacement. Rather it is coinciding with maximum acceleration. Now we have a phase shift that is easy to visualize when transitioning between the mass and compliance dependant regions.
My apologies that this ended up being kinda wordy.
I'm very late to the party but I have also dabbled with MFB. Interesting reading!
In some ways my experience is eerily similar to Gnobody's. I nearly fell off my chair when I read that he (I'm pretty sure it is he and not she) started with an LED, phototransistor and light blocker. I started with an incandescent bulb, light-dependant resistor and "calibrated" curved cardboard light blocker glued to the back of a woofer cone. The displacement feedback worked - but only at infrasonic frequencies well below system resonance. Then came the discoveries why this was so limited etc., etc.
Gnobody - you have an excellent practical understanding of the acceleration feedback topic and explain things very clearly. Hat's off for sharing your experience and knowledge. I think when I get some more time it might be worth starting a new thread to share information on the acceleration feedback system I played with way back when. The good thing about that is I still have all the circuit design and test result information I can share FWIW. I'm also interested because want to try MFB again at some point.
For anyone reading this thread who might be struggling a bit with what is happening to a loudspeaker in a closed box and all this talk about how it goes though a 180 Deg. phase shift and the response falls off at 12dB/Oct, perhaps the thought process I went through back then can help others to visualize it.
Consider the two regions the LF (Low Frequency) speaker needs to operate in. Unlike the midrange and tweeter we want the LF driver to function through its natural frequency. It is helpful to consider these regions the "mass dependent region" above Fr where F=ma rules and the "compliance dependant region" below Fr where F=kx rules.
First, with regard to amplitude, as Gnobody has so clearly pointed out, at low frequencies (the compliance dependant region) the displacement is proportional to the force exerted by the voice coil (F=kx). Displacement will eventually be a constant regardless of frequency as the frequency is lowered. We are conveniently ignoring the error in the air spring contribution to the constant k at very low frequencies due to leakage but that's okay - we are just trying to visualize what us going on.
How does this change the response? Well, above resonance, the amplitude has been increasing as frequency drops to keep the SPL constant. Below resonance, this isn't happening so the SPL is falling off. In mathematical terms, the RMS pressure is proportional to the square of the frequency and if you do the math this reveals the characteristic 12dB/Oct decay where displacement is a constant. You don't need to know the math, just that the reason for the decaying response can now be easily visualized due to the constant displacement characteristic.
Now with regard to phase, also at these low frequencies below resonance where F=kx, maximum displacement is going to coincide with the time the force is greatest (at the peak of the voice coil current for a sine wave). I think this is pretty easy to visualize. But above Fr where F=ma, the maximum force F (and maximum VC current) is not coinciding with maximum displacement. Rather it is coinciding with maximum acceleration. Now we have a phase shift that is easy to visualize when transitioning between the mass and compliance dependant regions.
My apologies that this ended up being kinda wordy.
Then given the choice of displacement, velocity, or acceleration feedback, acceleration would be the correct criterion for flat sound output level? But above system resonance, acceleration feedback shouldn't have much gross cone motion to correct and it would act only on emergent noise and cone distortions?
I think the enterprise of T/S simulation has hidden from view the amount of judgment and compromise that goes into non-feedback driver design needed to balance a variety of physical factors (mass, diameter, room location loading, etc.) in what gets propagated into a room from a cone.
rscamp - sure would be nice to learn of your experiments, new thread or not. One more experimenter impressed by MFB sound quality?
B.
I think the enterprise of T/S simulation has hidden from view the amount of judgment and compromise that goes into non-feedback driver design needed to balance a variety of physical factors (mass, diameter, room location loading, etc.) in what gets propagated into a room from a cone.
rscamp - sure would be nice to learn of your experiments, new thread or not. One more experimenter impressed by MFB sound quality?
B.
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Thanks to all who continue to contribute their MFB knowledge and experience.
My MFB circuit (preamp and sensor integrator part) is working. I have been trying to get it to work with IRS2092, TPA3255EVM TI amplifier, and my Genesis Loudspeakers. This is why taking so long...I have tried to make it amenable to use with any of the amps / power supplies so a bit tricky. For example the transformer in the Genesis is 46.9V gives about 70V DC; the TPA3255EVM max is 53V DC. I wired the transformer for "230V" but still only feed it 115; this is giving me about 32V at the TPA3255 amp - good enough. Of course, now I am only getting about +/-7.5 VDC to power my circuit instead of +/- 15 V but it is working. Pics coming soon.
My biggest problem is understaning the mounting of the piezo in various other drivers - from the sony 10", the 12" Kenwood carbon fiber KFC-1200F, various aluminum cone 12" woofers - counter to one's expection, if I mount the driver as rigid as possible for example mimicking the way the piezo is mounted in the Genesis woofers, when I mount the piezo similar to this on other woofers, I get huffing and puffing and wooshing, on the other hand if the piezo is mounted with a softer more compliant connection, it seems functional. Somewhat of a consternation.....
My MFB circuit (preamp and sensor integrator part) is working. I have been trying to get it to work with IRS2092, TPA3255EVM TI amplifier, and my Genesis Loudspeakers. This is why taking so long...I have tried to make it amenable to use with any of the amps / power supplies so a bit tricky. For example the transformer in the Genesis is 46.9V gives about 70V DC; the TPA3255EVM max is 53V DC. I wired the transformer for "230V" but still only feed it 115; this is giving me about 32V at the TPA3255 amp - good enough. Of course, now I am only getting about +/-7.5 VDC to power my circuit instead of +/- 15 V but it is working. Pics coming soon.
My biggest problem is understaning the mounting of the piezo in various other drivers - from the sony 10", the 12" Kenwood carbon fiber KFC-1200F, various aluminum cone 12" woofers - counter to one's expection, if I mount the driver as rigid as possible for example mimicking the way the piezo is mounted in the Genesis woofers, when I mount the piezo similar to this on other woofers, I get huffing and puffing and wooshing, on the other hand if the piezo is mounted with a softer more compliant connection, it seems functional. Somewhat of a consternation.....
I'm pretty sure of that as well.
Thank you! Much appreciated.
I have to understand something fully before I can work on it, not just at a mathematical level (complex polynomial fractions, aka transfer functions), but at a more intuitive gut level as well. The math is the bones, but I want to see the flesh on the bones as well, otherwise, how do you know what the beast actually looks like?
And once you do know, it's relatively easy to communicate that to other people, I think.
-Gnobuddy
Well, frequency response was very flat and the reduction in measured harmonic distortion at low frequencies with large excursions was huge. I think someone touched on this in an earlier post, but the human ear is much more sensitive to, say, a 90Hz third harmonic than to the corresponding 30Hz fundamental - especially at lower SPLs. When the third harmonic is drastically reduced in a situation like this, you bet it is audible.
The third harmonic is not musical (not the right note an octave up) so if there is a lot of third order it gets hard to tell what note what the fundamental is supposed to be. Also as previously pointed out, at 20 Hz with low distortion, the sound is really barely audible and more of a sensation. So when the 60Hz third harmonic for a 20Hz tone disappears, the speaker is still working very hard but there isn't nearly as much to hear. This is fidelity to the original though and although some would argue the pure 20 Hz isn't that important, in some cases the sensation it produces is part of reality - like maybe the guitar player on stage thumps his foot and this produces very low frequency vibrations in the floor. Correctly reproducing this from a recording adds to the realism.
Having said all this, the caveats are the driver needs to be pushed to see such large benefits and I can't imagine the linear motors/suspension for drivers were up to today's standards so I would imagine the benefits are somewhat lesser today. But still when properly implemented it results in increased fidelity which is a noble pursuit in my view.
I'm pretty sure of that as well.
Thank you! Much appreciated.
I have to understand something fully before I can work on it, not just at a mathematical level (complex polynomial fractions, aka transfer functions), but at a more intuitive gut level as well. The math is the bones, but I want to see the flesh on the bones as well, otherwise, how do you know what the beast actually looks like?
And once you do know, it's relatively easy to communicate that to other people, I think.
-Gnobuddy
I'm completely lost if I can't visualize something. Sometimes I think this limits what I can do.
I don't think the work I did was as thorough as yours and I didn't have to solve as many fundamental problems to start. For example, I used Philips AD8067/MFB drivers. I'm also somewhat "electronics challenged" these days having worked on this so long ago and hardly touched anything electronic since then.
On the plus side, I worked with underwater hydrophones and particle velocity sensors using piezo elements and with commercial accelerometers for transducer calibration and as input to resolving issues with shock and vibration so I have some practical experience there. From this I understand why you were gently trying to convince hombre of the importance of considering potential misbehaviour of the accelerometer mounting arrangement at higher frequencies. I would add that depending on the quality of the design, there can also be parasitic (unwanted) accelerometer signal output resulting from inadvertent strain on the element from structural or other mechanical inputs.
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Hitting your foot produces a very low frequency ?
I imagine that against the floor, or hit the microphone ?
if it is music with electronic help, not even the guitarist on stage will feel the sound of his foot, only an internal physical sensation. (like when you kick a soccer ball) Those who attend the artistic show do not find out about that sound.
And if it's music without the help of electronics (increasingly difficult to find, almost an impossible mission) then we should talk about a classical guitar performer. Well ... and how many meters away from it do you think any attendee at the event should be to hear that detail?
Come on, only those who are very close to him will listen to him, unless he is a performer of flamenco music ......
Now, if you want to say that this detail can be recorded in a recording of excellence, you should take note and correct that "very low frequency"
I agree.
To me, the benefit of MFB is not that it extends frequency response down to 20 Hz or 10 Hz, which, in my opinion, is essentially meaningless for several reasons. (For starters, those frequencies are too low to hear, and a reasonable-sized woofer cannot generate enough SPL to matter at such low frequencies in any case.)
The benefit of MFB is something else: it's the ruler-flat frequency response - in the frequency range where it matters - and the crisp transient response that goes with it. The tightest, purest bass I've ever heard came from the MFB system I prototyped.
In fact, on some music, the MFB woofer sounded too tight (overdamped.) I believe this is because the mix engineer who made that music did not have MFB woofers to listen to when making the mix. The speakers he did have were a bit boomy in the bass, so he ended up cutting too much bass in the mix (maybe also putting blankets in the drum kit, etc). This would not be audible with ordinary (boomy) woofer systems, but it was revealed by the extremely neutral MFB woofer.
-Gnobuddy
Hitting your foot produces a very low frequency ?
I imagine that against the floor, or hit the microphone ?
if it is music with electronic help, not even the guitarist on stage will feel the sound of his foot, only an internal physical sensation. (like when you kick a soccer ball) Those who attend the artistic show do not find out about that sound.
And if it's music without the help of electronics (increasingly difficult to find, almost an impossible mission) then we should talk about a classical guitar performer. Well ... and how many meters away from it do you think any attendee at the event should be to hear that detail?
Come on, only those who are very close to him will listen to him, unless he is a performer of flamenco music ......
Now, if you want to say that this detail can be recorded in a recording of excellence, you should take note and correct that "very low frequency"
Wow. I didn't expect to be taken to task on such a minor point.
My example would be totally dependant on the structure of the stage and I could easily picture a suspended structure where the heel of the guitarist would produce a very low, audible transient tone. But okay, forget about that. The venue is a small club located next to a subway in New York and the low frequency energy is coming from the rumbling of the transit vehicles that pass by every 5 minutes. It doesn't matter what it is, the point is there are very low frequencies and infrasonics that exist in some recordings that when reproduced can add realism to the experience.
Wow. I didn't expect to be taken to task on such a minor point.
Be cautious when posting as some people have more time than sense. (General statement, not pointed at anyone particular) They have nothing to do but argue on the internet. My time is way to important to waste on non-productive discussions.
As a side note I have put down many a mat on "old" elevated wooden stages in aditoriums/gyms because the spot where the musician wanted to be fell right between stage supports. And his foot tapping could be heard as a ~55-65 Hz beat clear as day on the mixer or in the back of the venue. Yes I know you mentioned 20 cycles and I have not heard it that low but was amazed the first time I heard 50Hz in the back of a packed auditorium from a foot tap. I actually crawled under the stage looking for some type of stage/boundary mic no one knew about.