R/E/P Community
R/E/P => Klaus Heyne's Mic Lab => Topic started by: Marnay on June 02, 2013, 12:06:19 PM
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Hello to all,
Can someone confirm that the value capacitor of C7 in the AKG 451E is 3.3 mfd?
Thanks in advance
http://www.gyraf.dk/schematics/AKG_C451E.GIF
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I just received the answer: it is 33 micro farad, rated 6.3V
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6.3 volts is pretty close to the split rail at 4.5 volts. I would use 10 volt ratings. Tantalum caps can be found a bit smaller than in the past. I use a 47 uf Panasonic FR 25 volt el cap bent sideways to clear. A small .01 uf Wima MKP-2 across it lets the 'air' out. Also, the input blocking cap may be a yellow Wima polycarbonate. Replace that with the FKP-2 polyprop version, much better.
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Thank you!! good idea for Wima 0.01micro, I often use this solution in my analog console with Panasonic FC and Ero capacitors
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I just make the change and the result is very good! Thank you!!
I also made this change on my old C 414 EB and I am satisfied with the result!
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(...) A small .01 uf Wima MKP-2 across it lets the 'air' out.
Please explain in simple language what the effect is when you put a small capacitor in parallel with the large electrolytic cap.
Thanks, KH
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A simple analogy is a gate. The cap is the gate. A larger gate lets more low end in. The smaller gate lets the smaller stuff in (high frequency, low level details) that the larger gate doesn't due to it's size and construction.
A smaller value "bypass" cap lets those electrons through that are normally converted to heat and wasted in larger, less conductive capacitor formulations. Examine the frequency response of a cap and you will see different fomuations roll-off earler than others, the worst are electrolytic caps and tantalums. Polystyrene and polypropylene films are the best along with some exotics like teflon caps. Dialectric absorbtion and dissapation factors are the specs that degrade a capacitor. They are very good for those film caps.
When you audition these parts and hear new, low level high frequency audio details, that's telling you that they were always there, but the gear/parts were "chewing your food for you" so to speak. When I first heard them I felt like the gear designers were robbing me of some of my music. That no longer happens here.
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There was a time when, I, too, thought using bypass caps would bring audible progress: in the 1990s I equipped virtually every job with polystyrene-bypass 001's, after I theorized in a similar vein as Jim, and after I bought into the "MultiCap" mega-dollar-capacitor story. Then I actually heard the MultiCap, and I realized that all that supposedly better high frequency pass-through, courtesy of the bypass cap, was nothing but phase smear due to the paralleling of two diverse components with unequal time constants/arrive times.
I have since abandoned that tack, and rely instead on better capacitor material for the single coupling cap I use. Now I have my relaxed sound back without the grit.
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My experience with bypass caps is much different. All capacitor formulations have limitations. Larger value caps are inherently slower with reduced rise times and increased DA. Smaller value caps are faster and have less DA losses. Higher voltage rated caps of the same value have increased rise times.
This technique is also used in timing, power supply bypassing and other circuits, it's a common, well known and used technique in the electronics industry. Look at the innards of your audio gear and you will see larger electrolytic caps used for the power supply filtering along with smaller local ceramic bypass caps. This is done to lower the impedance of the power network, a requirement in high speed analog and digital circuit design. Other audio manufacturers have also used this technique for the audio path including Studio Technologies, Manley Labs, Dolby Labs, dbx, etc. Even some low cost Chi-com mics from MXL have film cap bypasses added.
Besides needing a bit of basic engineering to avoid crossing over capacitor impedance curves, the properly selected bypass capacitor more closely approaches the open sonics of no capacitor.
The best way to emperically select a bypass capacitor is to evaluate them in a circuit that will also allow direct coupling, the operation without any capacitor. Only then can you completely remove the DUT (device under test) from the equation. That way you can compare the sonic results of a single cap, with a bypass added and also be able to compare to the "Holy Grail", no capacitor. To not do this is to come to a false conclusion without using all the facts and data available.
If one finds increased grit from a quality film capacitor, I would suggest that the grit is a form of THD that a slower, less revealing cap filters out for you. That grit is most likely still there no matter what cap you select. I would treat the circuit first to remove the non-linearities that are adding grit to the signal before blaiming a capacitor for revealing it. If it doesn't sound good without any capacitor, it doesn't sound good. I found I need the rest of the audio system to be direct coupled, that removes any additional audio reaction from other capacitors in the chain. This includes the mic preamp, monitor amps, etc. Here there are no capacitors in the chain when evaluating any capacitor. Listening to the sound of one cap through another is misleading, especially an electrolytic capacitor. My monitoring path on the bench is free of coupling capacitors so I can focus on the DUT, not my monitoring path.
Even the best caps have a sonic signature and audible quality losses. That is why I am proponent of direct coupled audio design. Most of my condenser mics here are designed without a coupling cap off the capsule, the most offending place to use one. The diaphrams are directly coupled to the jfet gate. These mics are down to one cap in the signal path, I wish I could also remove that too but 48 volts must be tamed. It's a bypassed 1 uf mylar film cap.
I also use Bas Lim's excellent Rel-Caps and MIT MultiCaps here. If I am reading correctly, you discontinued the use of .001 polystyrene film caps as a bypass after you heard another brand of cap, the MIT?
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Very interesting discussion on caps, really scratches that brain itch...
I get Klaus' concern about varying phase shift, etc across two different caps. There's also different dialectric constants and velocity properties as well, all frequency-dependant.
But I'd think these differences would be tiny at audio frequencies.
Also, with differing phyisical sizes come differing properties such as self-inductance and wavelenght-related electrical resonances within the component itself.
I think the self-inductance of the larger cap is what necessitates the smaller "hi-freq bypass" cap.
And no matter how many caps, they still smear the signal, current leads voltage as much as 90 degrees in capacitors...
It's the continuously varying audio signal with it's dynamic history that creates a varying
phase difference.
Just a few thoughts...
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(...) But I'd think these differences would be tiny at audio frequencies.
(...) I think the self-inductance of the larger cap is what necessitates the smaller "hi-freq bypass" cap.
How about taking off the thinking cap for a moment, and putting your ears to work? Listen with bypass, then without, then report back!
Thanks,
KH
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for what it's worth, the human ear is incredibly sensitive (especially when trained) and is more than capable of detecting "tiny" differences, particularly in certain sensitive bands of hearing!
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And no matter how many caps, they still smear the signal, current leads voltage as much as 90 degrees in capacitors...
Not globally, but only on the crossover frequency of a filter arrangement.
In the stop band the phase shift is going up to 180°.
In the pass band shift aproximates 0°. The (ideal) cap behaves as if it's almost not there.
So finally it's a matter of the correct size of the cap that makes the biggest difference in sound, as it defines the x-over frequency of a filter in a given circuit.
Besides there might be other effects like:
- Losses due to internal resistance.
- Dielectric losses.
- Internal resonances caused by stray inductance (usually far out ouf the audio band).
- Microfonics.
- Nonlinearities, specially if used inproperly, e.g. polarized Elko used without DC precharge.
All of these can easily be measured, no "golden ears" necessary to detect them.
No wonder that a cap can have a "sound".
Regards
Kai
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I don't understand (maybe I actually do*) the constant dismissal of using the sensory organ for which the medium was designed as a good indicator for quality. "Golden" or otherwise perceptive ears are the measuring mark with which we build, choose, and enjoy the finest musical instruments humankind has ever invented. It also happens to be a most useful measuring mark for selecting microphones and other audio components, as well as their sub-components, intended for recording and reproducing music. We surely would start with the visually perceived quality of the picture when examining a good photograph.
When it becomes unclear which exact feature of a capacitor is responsible for smear or otherwise inaccurate and unpleasant sound transmission (see your list of at least six possible culprits above) perceptive ears remain in the end the most helpful factor for choosing or rejecting said capacitor. More scientific speculation (a valid oxymoron) becomes quickly academic and unproductive towards the goal of identifying which of the culprits is responsible. So let's stop knocking good hearing, and decision-making based on it, once and for all.
I think it's most useful to refine audio components and make improvements in their design AFTER detecting which ones sound good, or lack artifacts, but clearly in that order. Otherwise we end up with more Multi-Caps©.
* could it be a low self-regard for one's own abilities to hear well enough to judge with confidence?
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Excellent test equipment has a place as well in evaluating electronic components. The human ear is just too unpredictable to use it as a benchmark simply because everyone hears differently. Test gear is used in the electronics industry for consistancy. Like analog tape, the human hearing fades with time, test gear doesn't. Your "Golden Ears" will not remain that way forever, they will change and suffer losses.
Test gear levels that all out eliminating the euphonic variations and biases. It won't tell you when something sounds good but is unparalled in finding errors or limitations. That's if you care about that stuff.
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I agree with one of your last sentences, with the following modification:
"It (test gear) won't tell you when something sounds good, but can help in finding errors or limitations once your ears have detected that something does not sound good".
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(A glitch- one that has happened with this forum software a few times before- deleted the original post (except for a few quotes to which I responded), but nevertheless inserted the original poster's name as author. Though I cannot fix this problem, I apologize to the original poster. KH)
I'm sure we all wish we possessed 'golden ears'...
Not golden or otherwise mutated, modified, or genetically pre-selected ears are necessary to hear the difference between bad, good and better. Make the test any time, with an uninitiated member of your family: put that person in front of speakers or under a pair of headphones, and let yourself be pleasantly surprised by the results of a simple A-B test. The esoteric language associated with expertise and connoisseurship may be absent from the ensuing comments from your loved ones, but the outcome will be the same: we know when something sounds good, especially when we can compare between two samples.
...that exhibited ruler-flat response, with consistent conversion of acoustic energy to neural impulses, over a range of SPL & freq.
Our hearing is so decidedly non-linear, non-ruler-flat, that we had to adopt and adapt a logarithmic, rather than linear, scale of decibel representation: we are hyper-sensitive, both in terms of volume and frequency detail- in the vocal range (evolution dictated it), but worse than a deaf dog in the highs and very lows.
(we don't have) ears with accurate and repeatable transfer function, even under changing physiological conditions such as age, blood pressure, heart rate, blood sugar, O2 sat, infection/parasites, illness, etc.
Why give yourself such low grades as a critical listener? Over time and repetition, even the most unstable and unreliable variables inserted by human frailty or hangovers will still not outsmart our exquisite sense of hearing. As if you one day woke up, listened to an MXL China mic and declared: "Best mic ever!"
Until we evolve a bit more I think reliable test equipment and protocols will remain very useful.
I have not found, or heard of, "reliable test equipment" and "protocols" that were responsible for creating the ELA M 251, the C12, the U47 or any other desirable microphone. All I have seen is picture after picture of Messrs. Neumann and Kühnast Sr. in their simple workshop listening, with headphones on.
If anything, a compelling argument could be made that the current state of microphone development is a rather tragic example of technological de-evolution. How else to to explain the many manufacturers- from the giants to the hobby shops- which are trying to emulate or copy microphones designed more than fifty years ago?
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One accurate measurement is worth a thousand expert opinions ...
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... test gear) won't tell you when something sounds good, but can help in finding errors or limitations once your ears have detected that something does not sound good".
It works both ways, you hear something that doesn't (or even does!) sound good, then you can try to find out why that is by using proper test equipment.
You might even find that the device under test does produce artifacts that do make it sound better.
With proper meeasurements you can find out where those are generated (if you don't already know) - this brings the evaluation from trial and error (or even guesswork) to a more scientific level.
Of course, it all starts and ends with hearing, but inbetween measurements can be of great help, specially in complex constructions beyond a basic tube condenser circuit with few variables.
There's an exception - a person with great experience in a certain field - call him an expert - might get faster and even better results when concentrating on hearing and bypassing more then basic measurements.
Probably it takes many years to become such an expert who has a Zen-like feel for what works best.
Regards
Kai
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I've found the relationship between testing and hearing to be rather a one-way street. Measurements sometimes are helpful in analyzing what we perceived with our ears as unpleasant or artificial of plain
off".
But it never seems to be the other way around, where trial and error are shortcut through scientific analysis up front, getting to the good sound first, by itself, without hand-(ear) holding. Can you give an example of a stellar-sounding audio device, including microphone, that was by and large created or improved as the result of scientific measurements, rather than just analysis protocols post fact?
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Can you give an example of a stellar-sounding audio device, including microphone, that was by and large created or improved as the result of scientific measurements, rather than just analysis protocols post fact?
As I can't look behind the curtains, so I can only guess: DPA / Bruel&Kjaer mic's might be such.
Their 400x-line of mic's construction (and sound) is so close to their measurement mic's that I would bet they are developed with a close look on test results.
Nevertheless (sic) they sound great and unique.
BTW - even their measurement mic's can make beautiful recordings, I specially like their one inch capsules for that. Connection isn't straight-forward, but in a studio environment it can be handled.
Regards
Kai
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Can you give an example of a stellar-sounding audio device, including microphone, that was by and large created or improved as the result of scientific measurements, rather than just analysis protocols post fact?
Bricasti M7 Reverberator. Try and build something like that without the world's best test gear.
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The device you cite is a computer, in my view, not an audio device in the classic term. Maybe I should have been more precise in my wording. Or maybe we limit the question down to microphones.
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It's a traditional audio device in many studios, including mine. Whether built with 6 Analog Devices Blackfin processors (each outperforming your Intel PC), a sheet of stretched steel or even a treated room, reverberators have been used a lot longer than microphones have been around. Classical halls and churches are also considered reverberators and were a large part of the creation of classic music pieces. That is why those halls were built that way.
I'm with Uwe on this one. Your coveted German mics were also built with the best test gear available at that time as well. I have no doubt that the designers would have used and appreciated modern test rigs as well, if they were available. I suspect you will find TEF and Audio Precision rigs at Sennheiser's factories today. If you ask those current designers if they believe in your "design by ears" process, you may be in for a bit of a surprise.
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What do you think the ratio of modern Sennheisers to vintage Neumanns is in the condenser mic closets of the world's top studios?
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... just look at the history of Neumann, particularly in the 'Classic Era". Toward what end do you think Neumann actually pioneered various audio test equipment, including highly revered measurement microphones (MM-series) and logarithmic level recorders, and why was (still is) Neumann instrumental in the development of measurement standards for audio. Trust your ears, but verify through accurate measurements has always been the mantra for reputable and successful players in the field of electro-acoustics. In my humble opinion, it is as irresponsible to dismiss the importance of documenting technical standards through measurements as would be the failure to actually listen to the gear under investigation.
Without multiple and careful measurements along the various steps during the manufacture, the expectations for acoustic properties could never be met by listening alone.
While this fundamental insight is shared by Sennheiser and Neumann in their design and manufacture, its connection to modern Sennheiser VS classic Neumann condenser microphones in the world's top studios is a misguided red herring at best...
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...Neumann actually pioneered various audio test equipment, including highly revered measurement microphones (MM-series) and logarithmic level recorders...
Although I have quite a bit of knowledge about historic audio measuring equipment (I do collect B&K), I never heard of a level recorder from Neumann.
Might it be you mix Neumann and Bruel&Kjaer, one of the world leading companies for audio test equipment (to say the least)?
Or do you mix Neumann with Sennheiser, who built some measurement equipment, but never much beyond some useful Audio Level Meters?
Sennheiser, BTW, aquired Neumann lately, but there's still some separation between the two brands.
EDIT: I found the Neumann P2 audio level recorder from 1934.
This was 15 years ahead of B&K, who built one in 1949.
http://www.neumann.com/?lang=en&id=about_us_history_part_2
http://asadl.org/jasa/resource/1/jasman/v21/i2/p91_s1?isAuthorized=no
Neumann history:
http://www.ppvmedien.de/pdf/Neumann_S32_47_Ansicht.pdf
QUOTE:
No guesswork, just the facts
Despite the rapid growth of the firm and all the organizational work that went with
it, Georg Neumann poured all his energies into research and development. He was always looking for new solutions and was intolerant of guesswork. “It was the inexactitude of the methods of measurement in use at the time that provoked Neumann, who was a modest but determined man and something of a precisionfanatic, to develop new electrical and mechanical measuring techniques,” records the commemorative volume for the 50th
anniversary of the firm’s foundation.
Speaking in 1978, Professor Aschoff of the Technische Hochschule, Aachen, explained that “at the time of the founding of the company, Georg Neumann was already convinced that further development in the field of electro-acoustics would have to go hand in hand with improved techniques of acoustic measurement, ...
Regards
Kai
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Kai, in the interest of full disclosure, I started my employment with Sennheiser as Service manger in February 1971. My current position is that of Director - Technical Services for Sennheiser Electronic Corp. and its Neumann Division in the USA. I hope my professional affiliation and my interest in the history and heritage of our industry may have earned me some credibility. I am not surprised by your independent confirmation of the importance of proper measurements in the fields of audio and electro acoustics.
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What do you think the ratio of modern Sennheisers to vintage Neumanns is in the condenser mic closets of the world's top studios?
If I look at my studio vintage Neumann is the clear winner - I do own just one pair of Sennheiser MKH40s, but more then 10 vintage Neumann's - besides some new ones.
In average Studios the balance might be more even - vintage Neumanns are expensive - Sennheiser doesn't build a great variety of studio condensers, but offer one for almost each purpose.
BTW: I like the MKH40's if a room sounds overly bright, a situation where a Schoeps or Neumann might not give enough separartion between direct- and roomsound.
Regards
Kai
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What do you think the ratio of modern Sennheisers to vintage Neumanns is in the condenser mic closets of the world's top studios?
That can easily be confirmed using sales and serial number records. I suspect they have built quite a few more lately than 30+ years ago. I don't recall seing them in every Guitar Center in the USA back then. Like old cars some are also no longer in service.
Then again, who decides what qualifies for one of the "world's top studios" and who is counting?
The larger question is what does that have to do with this discussion?
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It has to do with the choices audio professionals at the highest level of their profession (recording studios) make. And my contention is that, while Sennheiser has single-handlely rescued a morose Neumann company from bankruptcy in the 1980s, and today builds great dynamic mics, highly intelligible airport PA systems, the best consumer-grade headphones at any pice point, the best theater hearing impaired installations in the world, etc., etc., its highly technology research-based approach to designing professional condenser microphones has not been widely accepted in our field. Which I attribute to the lack of soul these mics have.
Take that last statement for what it is meant to express: the characteristic of a microphone that really matters above all else in music recording is the ability to transmit and convert the musical event in such a way that the listener responds positively emotionally.
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Maybe for some, but the averge kid listening to 4 bit compressed MP3 audio probably misses that. Seems a bit romantic to me. Like my dad said:
"Nostalgia isn't what it used to be".
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This discussion is resonant with the 'two ears - two sounds" discussion, it's all about left brain - right brain.
Different people may seat their thinking in one or the other hemisphere, leading to a different perceptual world-view.
"According to the left-brain, right-brain dominance theory, the right side of the brain is best at expressive and creative tasks. Some of the abilities that are popularly associated with the right side of the brain include:
Recognizing faces
Expressing emotions
Music
Reading emotions
Color
Images
Intuition
Creativity"
The 'righties" rely (mostly) on their visceral reaction to the sound, test equipment comes after.
"The left-side of the brain is considered to be adept at tasks that involve logic, language and analytical thinking. The left-brain is often described as being better at:
Language
Logic
Critical thinking
Numbers
Reasoning"
The "lefties" are more concerned with measurement and analysis before subjective feeling.
I don't think either is exclusive, it's more a case of emphasis (unless your corpus collosum is cut but that's another story... ).
No right or wrong, we need to integrate both approaches in order to efficently make progress.
Many people use strategies to integrate their thinking; rocking, tapping, humming, "scat" subvocalizations, etc.
Think Glenn Gould...
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Great post. Thank you!
Now the next question becomes: what should be the relationship and interaction between the two sides of the brain when designing microphones, recording with them, or listening to music? Which part of our brains should be in service to which? How should they interact with each other?
My contention, which I have used (successfully) in my profession for thirty years, is: the analytical side shall be in service to the side of my personality that perceives music and is able to enjoy it, emotionally. If musical happiness is a right brain affair, then let it report back to the left side to make the changes that improve the microphones and the visceral perception of the music they record.
That, I contend, was the secret of all famous, desirable microphones: their often less than perfect analytical, left-brain performance specs (think high distortion levels of, say, a U47!) is nevertheless perceived not as deficient, but as being closer to what we need to hear to make us happy listeners.
And that may be, why many modern microphone designers (MG, Sennheiser, and AKG included) are, in my opinion, on a path that led them to mediocre microphones, despite their increasingly technological sophistication.
So, who is in service to what? should be examined further: What should be the ultimate goal in a recording microphone? We often gloss over this most fundamental question.
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What should be the ultimate goal in a recording microphone?
To convert an acoustic signal to an electrical waveform?
The rest is window dressing, expect a thousand opinions on that subject. Otherwise, everyone would be in agreement on a single microphone choice.
Many times I have "auditioned" microphones in front of very capable singers. Sometimes the microphone selected had more to do with how the singer felt looking at it than the sonic results.
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What should be the ultimate goal in a recording microphone?
There is a simple answer - musicality.
But what is that - for me it's if the mic helps the performance to sound more like music, less then isolated notes.
But that's not done with the one, single, perfect mic, which, IMO, doesn't exist.
Sometimes I need several different ones to choose from to use the one that fits best.
This might be the mic with the strong midrange to boost the voices special dramatic character.
Or the one with the boosted "airy" hights that makes me listen to the sexy, breathy voice sound of the girl singer.
Same applies, of course, to any instrument, e.g. the mic that started this discussion is one of my favorites for natural skinned percussion instruments, because it gives a special rounded but present attack in the full freq. range.
It sounds like the performer plays with a high intensity.
Compared e.g. to a Schoeps or Neumann it sounds more "dramatic", although in the measurement these mic's aren't much different.
And luckily often there is a performer where it doesn't matter what mic I use - out comes music anyway.
Regards
Kai
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Just saw "20 Feet From Stardom" a fascinating documentary about and with seminal background singers of the 1950s through today. Highly recommended. Not the least because studio mics used for the live recording portions of the movie were entirely one of three models: U47 (galore) C24 and U87. Go figure. The final scene features Darlene Love in front of a U47. There was not a dry eye in the movie theatre when she was done.
Let's stop the endless equivocation and pretense that there are all these choices in microphones. There are really (and unfortunately, given the service headaches) very few contenders at this highest level of vocal performance. Once you have such a vocal talent in front of you, you obviously pull out the absolute best match for the job. And there are really very few to choose from. Just a handful. That's all. Producers and engineers in that league keep coming back to the same five. Or six.
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Producers and engineers in that league keep coming back to the same five. Or six.
U47
U67
U87
C12
251
...
what else?
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M49
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M49
oh yeah!
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and I realized that all that supposedly better high frequency pass-through, courtesy of the bypass cap, was nothing but phase smear due to the paralleling of two diverse components with unequal time constants/arrive times.
Phase, especially at high frequencies, is something that is more easily measured than heard. Do you have any kind of measurement or documentation to support your thesis that parallel capacitors produce audible summed phase differences at high frequencies?
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Do you have any kind of measurement or documentation to support your thesis that parallel capacitors produce audible summed phase differences at high frequencies?
No, I don't. I hear it, and select the capacitors I use in my work accordingly (and continue to remove the bypass caps I used to install in my mods, for a while, in the 1980s and 1990s, once my hearing ability matured enough to recognize the shortcoming of that approach).
I could ask in return: Do you have any kind of measurement or documentation to support your thesis that "phase, especially at high frequencies, is something that is more easily measured than heard"?
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Decades of studies show the ear is not conductive to hearing phase, only phase summed errors. No need to re-examine that after all these years unless someone here thinks all those studies are inconclusive. They would also need more evidence than what they believe they hear, like actual measurements.
As for measurements, the Audio Precision rigs offer a great phase vs frequency measurement ability that allows the user to see the results of phase without guesswork. I strive to maintain a 2 hz to 200k hz operational bandwidth, that eliminates any phase shift from the audio band of 20~20k hz. A pair of capacitors wired together will show no phase errors in the audio band down to 2 ppm. (parts per million for those that skipped math).
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Decades of studies show the ear is not conductive to hearing phase, only phase summed errors.
My point. Besides: anytime you press a signal (electricity) through a component like a resistor or capacitor, you will get time delay, compared to the same signal passed through a straight piece of wire. Is there any argument about that?
If you now press that same signal through two paralleled devices of vastly different storage capacity, you inevitably incur phase shift, i.e. different time delays generated by the two capacitors. Any argument about that?
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"anytime you press a signal (electricity) through a component like a resistor or capacitor, you will get time delay, compared to the same signal passed through a straight piece of wire. Is there any argument about that?"
Well, I do disagree, a purely resistive load should exhibit zero phase shift at any freq.
I also disagree with Kai's much earlier statement that capacitors only exhibit phase shift near the X-over freq.
All caps smear the phase relationship between voltage and current, regardless of freq.
"When capacitors or inductors are involved in an AC circuit, the current and voltage do not peak at the same time. The fraction of a period difference between the peaks expressed in degrees is said to be the phase difference. The phase difference is <= 90 degrees. It is customary to use the angle by which the voltage leads the current. This leads to a positive phase for inductive circuits since current lags the voltage in an inductive circuit. The phase is negative for a capacitive circuit since the current leads the voltage. The useful mnemonic ELI the ICE man helps to remember the sign of the phase. The phase relation is often depicted graphically in a phasor diagram."
This 'smearing' is why coupling caps are not as clear as a DC-coupled circuit.
We're really talking about two kinds of phase; phase of one freq vs. another in the signal and phase of voltage vs. current in the signal.
Both easily measured.
Neither sounds good...
I think in better vacuum tube circuits the coupling cap phase errors may be balanced by the output xfmer inductance, providing the load with a signal that has voltage and current back in phase.
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To be honest, unlike the phase smear of capacitors, which I hear, I have not been able to hear that of resistors, but made a (false?) assumption that ANY component in the path of a straight wire line will display audible artifacts.
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Some resisters do exhibit inductive or capacitive reactance, wirewound resisters can be the worst.
I think old-style carbom comps may be the best for audio but they exhibit thermal/vibration noise.
I wonder about metal film resisters as I think the metal film may be etched in such a manner to cause inductance, though I've never measured them. I do like how quiet they are.
This whole idea of voltage/current phase is a real problem in AC power sistribution.
Most loads are inductive (because of power xfmrs and motor windings), it's known as 'power factor' (PF) and power companies go to great lengths to correct the imbalance.
You may notice banks of very large oil-filled capacitors high on the power poles near large consumers of electricity; it's the only available way to correct this E/I phase problem because it can cause tremendous currents to be reflected back down the line and destroy xfmrs and even the generators.
I theorize that in audio circuits this also happens, causing a sort of 'comb filtering'.
Old tube amps used coupling caps, balanced by interstage xfmrs and output xfmrs, leading to a coherent signal path that just sounds better and less cluttered.
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...it can cause tremendous currents to be reflected back down the line ... I theorize that in audio circuits this also happens, causing a sort of 'comb filtering'.
No, reflection only appears if the line is long compared to the wavelenght, then you have a "transmisson line". In these, BTW, signal speed is much lesser then speed of light.
This happens, e.g. on a long analog telephone line (cross country) not properly terminated - you get an echo.
For shorter cables this cannot happen, as the signal is progressing close to the speed of light, so it's the same (in phase) all over the whole cable or element.
You cannot find a "wave" riding on the cable.
Commenting on delay in analog circuitry (a myth):
Analog circuits (or elements) that have almost linear frequency response do not have any delay besides minor phase shifts at both ends of the transmission range.
In fact it's hard to build a delayline even for some microseconds of delay using analog circuitry.
It's done for special purposes, e.g for color tv sets: http://en.wikipedia.org/wiki/Analog_delay_line
http://de.wikipedia.org/wiki/Verz%C3%B6gerungsleitung
Regards
Kai
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Speaking of inductance in resistors-- It does exist in film types, but in minuscule amounts only. Some carbon film and metal film resistors have the resistive element spiral-wound against the substrate, which causes some inductance. I was always told that this is almost always too small to have great significance at audio frequencies, but can have some impact in other applications-- for this reason, manufacturers like Caddock and Vishay make metal film resistors with almost zero inductance.
But I'm sure someone with better, more first-hand information can set us straight.
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Kai, we're not talking about phase shift of one freq vs. another, it's the voltage vs. currentphase shift that causes distorted waveforms.
The AC power distribution circuits operate at 60 Hz, well within audio bandwidth.
"Capacitance has the property of delaying changes in voltage as described in Module 4.3. That is, the applied voltage reaches steady state only after a time dictated by the time constant. In AC circuits voltage and current are changing continuously, and in a purely capacitive AC circuit the peak value of the voltage waveform occurs a quarter of a cycle after the peak value of the current. Therefore a phase shift is occurring in the capacitor, the amount of phase shift between voltage and current is +90° for a purely capacitive circuit, with the current LEADING the voltage. The opposite phase shift to an inductive circuit."
It's this phase shift that I theorize causes audible 'smearing'.
And yes, massive currents can reflect back into the source, causing the proverbial 'standing wave' in the audio bandwidth.
In AC power circuits the current can be large enough to burn down a skyscraper, as in "The Towering Inferno", a real occurrence caused by this phase imbalance.
"The level of interference created when a three-phase wye system is split up and used as three single-phase circuits is truly something to behold. For example, as much as 20% (or more) of the power used by fluorescent ballasts is reflected back onto the power grid in the form of reactive or harmonic currents -- now that’s a lot of distortion. In the late 80’s, a 40-plus-story office building in Los Angeles actually burst into flames because of these reactive currents. Incredibly, the origin of the fire was determined to be from excessive harmonic distortion in fluorescent lighting circuits which created a high-frequency current overload and literally a meltdown of the electrical wiring system. The First Interstate Bank fire in Los Angeles in May of 1988 was the event dubbed by the media as "The Towering Inferno" a la the Hollywood movie. Codes were adapted to remedy the fire danger, but the noise problem itself was never completely resolved."
http://www.equitech.com/articles/enigma.html
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My point. Besides: anytime you press a signal (electricity) through a component like a resistor or capacitor, you will get time delay, compared to the same signal passed through a straight piece of wire. Is there any argument about that?
If you now press that same signal through two paralleled devices of vastly different storage capacity, you inevitably incur phase shift, i.e. different time delays generated by the two capacitors. Any argument about that?
Even if you can hear to 100 mhz, you will not find those errors in the audio band. If you ever get a hold of a nework analyzer like Agilent or HP you will find no measurable errors up to 20k hz. Capacitor dialectric absorbtion and dissapation factors will eat up your audio far before those timing errors will.
The time delays through a metal film resistor are approaching the propagation delays of electrons through solid materials. That is usually limited to about 70% of the speed of light. That's about 103,000 miles per second of operational speed though your passive components. If you can hear that the NSA has a cushy job waiting for you.
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Kai, we're not talking about phase shift of one freq vs. another, it's the voltage vs. currentphase shift that causes distorted waveforms.
It depends on the way the capacitor is used.
If a coupling cap (series capacitor) is made big enough, it behaves almost like a piece of wire in the audio band.
This means there is almost no phase shift in the waveform, neither relative between different frequencies nor absolute nor as "voltage vs. current" shift.
Here the voltage at the output of the cap follows the input without any delay or phase shift.
The mentioned phase shift between current and voltage isn't happening in this case.
Why is that?
Because we have an almost purely resistive circuit, the capacitic influence is almost neglectable. The bigger the coupling cap, the lesser is it's influence.
Phase shift only occurs if you use a cap in a parallel circuit, something that is not done this way in audio circuitry except if you want to filter out high frequencies.
Practically phase shift appears only at very low frequencies, e.g. the coupling cap loading the output x-former of a tube mic.
Usually this pairing is intentionally tuned to about 30 Hz x-over as low cut filter.
Here the (voltage) phase shift would be 90° for 30 Hz, approaching 0° for higher f's.
Remember: what our mic amp well see is only the voltage (voltage coupling), any currents flowing around do not affect what ends up in the recording.
Don't mix switched DC voltages, power lines and audio transmission, these are different animals and different laws of physics are relevant for those.
I hope this clearifies some things.
Regards
Kai
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Phase shift also occurs in the audio band at the higher end of the audio bandwidth if there is a roll-off applied below 200k hz. This is because phase shift developes a decade above/below the roll-off point. Set the bandwidth from 2 hz to 200k hz and you will see a flat phase vs frequency plot on the Audio Precision analyzer.
The shift isn't audible but the roll-offs are if allowed to enter the audio band. Once I tested a Mitsubishi 32 track digital recorder with an all passive 10th order 48k hz low pass filter to avoid Nyquist frequencies. That rig had not phase shift but group delay of about 2500 degrees. Of course, no one complained about Madonna's hits cut with it. If you had a chance to sum that output with the direct source, it created quite a comb filter from all that group delay. Once all the tracks cut were played together, they did play well together. The original "black face" ADAT also had 2500 degrees of phase shift, that didn't stop millions from buying "Jagged Little Pill".
Compare to an analog recorder with about 90~100 degrees of phase shift at 20k hz due to the record/play equalization and roll-offs.
Anyone with a bit of curiosity can play with phase shift by building/using a tunable all pass filter with a couple of opamps and a dual pot. Commerial units are also available from Little Labs. All pass filters are the basis of the proverbial phase shifter pedals used by 1970's guitarists. They sum the phase shifted signal with the dry signal to create the comb filtering. If you get one of those and disconnect the dry signal, you won't hear anything even though the phase is shifting from zero to 720 degrees or so. That is the nature of phase, you can't hear it alone.
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...Once I tested a Mitsubishi 32 track digital recorder with an all passive 10th order 48k hz low pass filter to avoid Nyquist frequencies.
I used this machine several years here and I really liked (and still do like) it's sound and reliability.
I still have it in working condition, although it's stocked and not in use of course.
Regards
Kai