September 27, 2014

Understanding The Proximity Effect

proximity effect audio, microphone proximity, mic proximity effect
Don't get too close...!
The Proximity Effect describes how bass frequencies build up as we bring cardioid or bidirectional microphones closer to a sound source.  Many dynamic, condenser, and ribbon microphones fall into one of these categories.  Primarily affecting frequencies below 100 Hz, up to 16 dB of proximity-related boost can occur when the microphone is placed very close to the source.

This apparent bass "boost" results from a combination of two things: the deliberate decrease in microphone transformer output at high frequencies, and the inverse square relationship of acoustical energy with respect to distance traveled.

1. Deliberate Decrease in Microphone Transformer Output at High Frequencies

Believe it or not, most microphones are designed intentionally to decrease transformer output at higher frequencies.

But why?

Microphone designers decrease transformer output at high frequencies because high frequencies naturally produce greater pressure differences at a microphone's diaphragm and thus, a greater resulting electrical output.  Compensation is needed to ensure this naturally occurring effect does not disrupt the output balance between highs and lows.

To explain this in more detail, let's review how microphones work: directional microphones, also known as pressure gradient microphones, pickup sonic information by observing the difference in sound pressure at the front and rear of the microphone's diaphragm.  The pressure differences cause the diaphragm to move, and in doing so, an electrical output is created that is transmitted through a cable, allowing the captured sound to be re-amplified or recorded.

In the case of a dynamic microphone, this is normally accomplished by the attachment of a voice coil to the diaphragm, which moves through a magnetic field to generate voltage.

dynamic mic, dynamic microphone, dynamic voice coil, microphone voice coil, mic voice coil, dynamic microphone diaphragm
The voice coil inside a dynamic microphone

Similarly, in the case of a ribbon microphone, a metal ribbon is suspended between the two poles of a magnet to achieve the same effect.  (Because of their design, ribbons are more sensitive to smaller changes in SPL, making them excellent for transient response -- often better than a condenser, even -- and create less high-frequency distortion in comparison to both dynamic and condenser microphones.  Many engineers love them for their more natural, analog sound, but they are delicate and not ideal for all recording situations.  Listen to my shoot-out results to see what I mean.)

The suspended ribbon in a ribbon microphone

With condenser microphones, a thin piece of metal foil is placed closely to a back plate, forming a capacitor when connected to its power source (e.g., phantom power, 48V).  As the sound source causes the metal foil (diaphragm) to move, the resulting changes in capacitance capture and transmit an electrical representation of our sound source.

The capacitor within a condenser microphone

Remember, the important thing to realize is that in all these cases, the microphones obtain their output signals by responding to the difference in pressure as observed from the front to the rear of the diaphragm.  This is what we call the Pressure Gradient, and it is largely determined by microphone design elements that specify how a sounds arriving at the front of the diaphragm are channeled around to the rear of the diaphragm.

Soundwaves must travel around to the rear of the diaphragm

Most directional microphones use several "delay paths" to route an incoming signal around to the rear of the diaphragm, and depending on the methods used, there will be different arrival times (at the rear of the diaphragm) and a variance in resulting relative pressure differences across the frequency spectrum.  The greater the distance traveled to reach the rear of the diaphragm, the wider the gap will be in comparative pressure differences between high and low frequencies.

Regardless of these complex design elements, the key takeaway is that higher frequencies always create greater pressure differences simply because their waveforms have shorter period lengths as compared to lower frequencies at the same amplitude.

Higher Frequencies with Shorter Period Lengths

Lower Frequencies with Longer Period Lengths

As a higher frequency travels around to the back of the diaphragm, its relatively shorter period length leads to a greater swing between the peaks and troughs of its corresponding sinusoidal wave, as compared to a lower frequency signal that has a longer period length and less swing over the same travel distance (this is graphically depicted in the images above).  Ultimately, this is what causes the greater pressure difference at higher frequencies, despite amplitudes being the same, and creates the need for compensation for balancing purposes.

This is what the pressure difference will look like without compensation.

Without compensation

The compensation solution normally applied takes this naturally occurring mathematical property into account, then "cancels it out" (attempts to re-balance output levels across the frequency spectrum) by decreasing transformer output at higher frequencies based on their expected increase in sound pressure as compared to low frequencies.  Compensation curves are designed to produce essentially a flat output as shown below, but in reality are not perfect and may be a bit above or below baseline at any given point.

WITH compensation

And as we are about to learn, this compensation does not produce the proximity effect in and of itself, but turns out to be the reason it is so noticeable when it does occur.

2. Inverse Square Relationship of Acoustical Energy

Sounds waves propagate similar to a sphere of energy moving outwards.  The intensity of the sound source is distributed across the entire surface area of this "sphere," so as our sound wave travels farther out, the acoustical energy in a square meter of the sphere's surface decreases by:
1 / D2 
(D = distance from the source)

This means as our sound wave travels 2x the distance from its source, its total energy spreads over 4x the area and will only retain 1/4 of its energy.  Similarly, at 3 times the distance from its source, it will spread over 9x the area and will only retain 1/9 of its energy.

With this in mind, as a microphone moves closer to its source, you can imagine how even small adjustments to its placement can result in major mathematical differences in intensity.

This applies to all frequencies, though, not just low ones, so how is this connected to the compensation described above, and what the heck is actually causing the proximity effect?!

Remember that microphones capture and transmit sound as a result of pressure differences on each side of the diaphragm.  Normally, the primary factor affecting these pressures is the pressure gradient we defined above, which results from the fact that an incoming sound wave must travel around to the rear of the diaphragm.

As long as this pressure gradient is the primary, heaviest-weighing factor in creating a pressure difference at the diaphragm, everything should remain "normal" and balanced pursuant to the output transformer compensation built into the microphone.

Furthermore, due to the inverse square relationship of acoustical energy, we can generally expect the pressure gradient to be the primary factor in diaphragm pressure at any large distance.

However, as we bring the microphone closer to the sound source, this same inverse relationship means that the diaphragm is exposed to increasingly more sound pressure from the energy, or intensity, of the sound source (we'll call this source intensity), and eventually a point is reached close to the sound source where the pressure placed upon the diaphragm from source intensity is greater than the pressure placed upon it by the pressure gradient.

Realize: the source intensity and pressure gradient are two completely independent factors affecting pressure on the diaphragm, and this is the key to understanding.

As soon as source intensity overtakes the pressure gradient, the resulting sound captured and transmitted by the microphone will no longer operate according to the assumptions by which it was designed (which relied on the pressure gradient).

If this happens in a microphone without pressure compensation built in, the graph will look like this:

And in a microphone with compensation, we can visually see the proximity effect taking place:

Once source intensity takes over as the primary pressure factor, the compensation built into the microphone to deal with pressure gradient differences between high and low frequencies is no longer necessary.  But the compensation doesn't just disappear because it isn't needed (microphone design is not this advanced); it continues to affect the source signal by cutting highs relative to lows.

THIS is what creates the proximity effect, as you could say the microphone is being used "beyond its operating specifications," and the design methods that were used to create a "balance" at typical distances are now throwing its functionality out of balance.

Even without compensation, you can see in Figure 12 above that a proximity effect of sorts will still occur, it will just be less obvious given that the highs are still creating a higher pressure difference than the lows.  The bass will sound to be a little boosted, even if it is not as dramatic as with the true proximity effect in a microphone with pressure compensation built in.



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