Myth Busted – The Too Sensitive Condenser Microphone

Wednesday, December 30th, 2009

I’ve read many times on the internet that condensers are too sensitive, they pick up the mouse in the next room, the refrigerator downstairs, the arm hairs brushing on the top of the guitar. People have suggested that a dynamic mic is better when there’s ambient noise, clumsy technique, or a bad sounding room. Have you heard this myth? Do you believe it?

My understanding of microphones and audio says this myth is not true. I’ve been told by knowledgeable experts that mic sensitivity is linear below physical clipping regardless of the transducer technology. I actually asked this question some time ago over on rec.pro.audio – a usenet discussion forum where professional audio engineers hang out. Some of their responses were rude, but they basically established that preamp gain is all that is required to match sensitivity between a dynamic and a condenser mic. Here’s a link to that thread.

Now I believe those guys. I thought it might be interesting to devise a demonstration of sorts, by plugging in a dynamic and condenser mic, playing a reference tone through a speaker in front of the mics, then adjusting preamp gains so the levels match. Then I could generate a quieter and quieter test signal by walking away from the mics making noise. Recording this diminishing sound with both mics would tell me if one could “hear things” the other could not.

Oh No!! Is the Myth True?

To my amazement, when I did conduct a demonstration for myself, I could hear more distant, quieter sounds on the condenser mic. It seemed as if the myth was true.

I knew the error was not the physics or engineering of mics, but rather something I had overlooked in setting up my test. After a day or two of research and pondering the light came on. I realized that the different pickup patterns of the two mics made my calibration procedure wrong. I was calibrating for direct, on-axis sound, but I was measuring diffuse off-axis sound. I needed to do the calibration using only the diffuse sound field, which meant moving the speaker some distance from the mics during the calibration.

A Better Test Design

At this point I also figured out that I didn’t have to move the noise source to reduce the level of the test signal, I could create test tones that got lower and lower in level and play them back from the same spot as the calibration (well, duhhhh).

Here’s the result. For the best fidelity, here’s 45db-66db-xt.wav. Or if your connection is a bit slow, the compressed version is 45db-66db-xt.mp3.

You’ll need to download one of the files and pull it into a player that can select only the right or left channel. The mic in the left channel is the condenser, a Shure KSM141. The mic in the right channel is a dynamic, the Shure SM57. This is a little excerpt from the demonstration recording. I selected the area from -45 to -66 dBFS, which is where the test tone slipped into inaudibility. I raised the level of these files substantially and there is plenty of broadband noise, so be careful not to play them too loudly. The condenser mic is in the left channel, the dynamic in the right. Listen to first one side, then the other, and see if you can hear tones at lower levels from one mic or the other.

Conduct Your Own Demonstration

If you’d like to conduct this demonstration with your own mics and room, all you need is a calibration tone and a test tone series. You can generate the calibration tone in most audio editors, and you can download my test tone series.

If you can’t figure out how to create a tone in your favorite audio workstation software, download Audacity and install it (I recommend the Beta 1.3.xx or later version). Start Audacity and choose

Generate | Tone

then fill out the Tone Generator form:

Waveform: Sine
Frequency (Hz): 1000 (a 1 Khz test tone is the normal industry standard for basic testing)
Amplitude: .6 (a very loud long 1 Khz tone can damage your speakers and possibly your ears)
Duration: 600 seconds (10 minutes should be enough)

Click OK and you’ll see a strange solid waveform. That’s your calibration tone. Just export it from Audacity:

File | Export
Save as type: (either MP3 or WAV Microsoft signed 16 bit PCM)
(Choose a directory and file name)

You can download my test tone file here. The test tone file contains volume level announcements and 1000 Hz tones starting at -9 dBFS and going down to -90 dbFS in 3 dB increments.

Here’s a screen shot of the test tone file:

Mic sensitivity test tone in Adobe Audition 3

Mic sensitivity test tone in Adobe Audition 3

and a sample of the tones, starting at -9 dBFS and going to some of the lower level tones:
[audio:test_tone_xt.mp3]

To conduct your own demonstration, connect two mics to your recording system. Place the mics at least 6 feet from the speaker, then play the calibration tone. Adjust the preamp gain so the two mics show the same input levels. Do this very very carefully, this is the most critical step in the process.

Next, simply play the test tone file while recording the two microphones. You’ll want to wait until a quiet part of the day, and be prepared to sit very quietly while the test file plays. When you’ve completed recording the test tone sequence, listen to one of the tracks you just recorded. When you can no longer hear the test tone, switch to the other track (other mic) and listen again. If your experience is like mine, the test tones will fall into inaudibility at the same level for both mics.

Better Recording By Knowing Our Tools

Mics are fascinating devices, but they’re engineered objects in the physical world. We can make better use of them if we have a better understanding of the way they really work instead of relying on incorrect assumptions and erroneous analogies. In the past audio testing required lots of expensive dedicated equipment, but now with our computer audio systems we can easily perform simple but fairly sophisticated evaluations of our audio gear, and learn to make better recordings in the process.



This entry was posted on Wednesday, December 30th, 2009 at 5:30 pm and is filed under Audio, Tutorials. You can follow any responses to this entry through the RSS 2.0 feed. You can leave a response, or trackback from your own site.


5 Responses to ' Myth Busted – The Too Sensitive Condenser Microphone '

Subscribe to comments with RSS or TrackBack to ' Myth Busted – The Too Sensitive Condenser Microphone '.

  1. Dwight Powell said in post # 1,

    on February 19th, 2010 at 3:02 am

    Hello,

    I’m trying to decide between the Shure SM27 and the Rode Nt1a. I will use the mic for vocals mostly. Do you have an opinion based on your test results.
    Thanks

  2. Fran Guidry said in post # 2,

    on February 19th, 2010 at 10:14 am

    No, I’ve never used either mic and have no opinion about either one. I would suggest that a switchable multi-pattern mic like an NT2a is more useful for most recording tasks, but both the mics you mention are well respected.

    Fran

  3. Joe W. said in post # 3,

    on January 29th, 2011 at 11:42 am

    Thanks very much for going over this topic in such detail. Your tests have verified something I have long suspected, but didn’t have the resources or full knowledge to try to test for myself. I also took the time time to enquire about this with a technician at Shure, who also seemed to suggest similar as to your findings for the most part, but also noted that a dynamic, due to a thicker diaphram and being more “mechanical” in it’s method of turning sound pressure into electrical energy, does take slightly more force to respond to lighter sound pressures than a condenser diaphragm, which they explained mostly affects the higher frequency range, as to which they said some people will “perceive” this as being more sensitive than it actually is, as was explained in the thread you linked to, due to the non-linearity of human hearing.

    So now it is easy to understand that the misconception that a condenser will amplify background sounds at a higher ratio than what it actually is (I guess an environmental signal to noise ratio would be a good enough description), somehow making them louder than what they are in reference to the source material, or even reach and grab sounds that a dynamic of the same configuration cannot, is simply ludicrous. A microphone can only reproduce sound waves that reach it’s diaphragm, with enough energy to cause the diaphragm to react, which is all just basic physics, once one understands exactly how a microphone works.

    Again, your explanation and demonstration have proven to be invaluable. I hope more people read your article and become educated enough to dispel the silly myth of a too sensitive condenser microphone once and for all.

  4. Fran Guidry said in post # 4,

    on January 29th, 2011 at 12:10 pm

    Thanks very much for your comment. I have never had any success persuading anyone that any of their audio myths were untrue, at least so far. But I continue to hope.

    Probably the key issue in all this is that we seriously mistake our ability to recognize volume difference without measurement tools. It seems like common sense that we can tell if one recording is louder than the other, but in fact our ear/brain auditory system is not designed to be an SPL meter.

    This results in “microphone shootouts” done with different performances and no level matching and people believing that the differences they hear are due to the microphone. Amazing, but nearly universal.

    Fran

  5. Micael said in post # 5,

    on December 26th, 2016 at 6:54 pm

    I have conducted a similar test myself to try and bust the myth too. My conclusions were similar to yours but I still am not sure about one thing… which can be summarized by what the Sure technician told to Joe W. in the comment above. There’s still one piece missing in the puzzle: the heavier moving coil does seem to require more energy to be moved.

    How does one explain this? I have though about this a lot and my conclusion is one of 2:

    1. The info about the moving coil needing more minimum SPL to move is wrong (I have seen someone arguing in favor of this);
    2. Or it is true, but the minimum SPL required to move it is so infinitesimally small that it has no real consequence in the real world…

Leave a reply






About the Blog

    Howdy, my name is Fran Guidry and this is my Homebrewed Music blog.

    I play Hawaiian slack key guitar and recorded my solo acoustic CD at home. Most of the recording information I find on the internet seems focused on bands, drums, multitracking, and so on but my main focus is recording solo acoustic guitar. Lately I’ve been enjoying video recording along with audio, so that shows up in the blog as well.

    I’m also a guitar nut. I love big ones and little ones, handmades and factory guitars, cheap ones and expensive ones. So I’ll be sharing the fun of exploring guitars as well, along with the challenges of amplifying acoustic guitars for live performance.

    Welcome!

Philosophy

    My recording philosophy is pragmatic, skeptical, not super critical. After all, the performance is by far the most important component of a track, and every aspect of any recording is a matter of taste.

    But I do like to know “about stuff.” Back in hifi days I learned about double blind testing. I learned that we humans can easily hear differences that don’t really exist. The more I’ve learned about our human auditory system, the more I’m skeptical of what people say they hear, especially if they claim that a particular microphone or preamp or cable has some magical property.

    I’ve only been recording since 2001, and when I started I found the usual places on the internet. I sought advice and accepted it, thought I would improve my recordings by using more expensive equipment. It didn’t work.

    Two things that did seem to lead to better recordings were experience and room treatment. Getting an appealing sound is the combination of many small details, and learning those details only comes from experience. Amd the sound of the recording space is obviously a big factor.

    I’ve only recorded seriously using digital technology, but I remember trying to record rehearsals and gigs back in analog days. I don’t have any nostalgia for analog recording and playback systems at all. I think even low end digital systems can capture marvelous recordings. So when I look at gear, I look for good specs: low noise, broad flat frequency response, wide dynamic range, low distortion. I’m not interested in colorful components, mics and preamps with a sound, I want the sound to be the sound of my guitar.

    But the last word is that I’m just learning and I hope you find something useful in my posts.