Everyone knows what audio feedback is, but what exactly causes it.
Take a simplified diagram of a microphone feeding an amplifier, which in turn feeds a speaker.
In operation, the audio signal (Red Sinewave ), enters the microphone and is then amplified. The output of the amplifier is then fed to the speaker. The gain of the system is calculated by the amplitude of the final signal divided by the amplitude of the original signal.
Calculation of the System Gain must take into account:
the Microphone output level
the Amplifier Gain (can be easily measured)
the Speaker efficiency
A "Sound Meter" will help you calculate the gain.
The following discussion all relate to the audio volumes at the microphone position
If the speaker is in the same room as the Microphone, some of the amplified audio from the speaker will combine with the original audio at the microphone.
(For this situation, assume that both audio sources are in phase at the microphone.)
Amplified audio level from the speaker at this location:
if less than the original signal.
No of feedback
The total system amplification will be less than unity.
if equal to the original signal
Verge of Feedback
The system amplification will be unity (1x)
if equal to or greater than the original signal (up to a maximum of +6 db)
The system amplification will be more than unity, thus the amplified signal will keep increasing in volume (Feedback). This requires the presence of the original signal in the early stages of the feedback, thus removing the original signal can sometimes stop the feedback. In actual situations, this can cause ringing effects.
if greater than the original signal by 6 db or more
Feedback increases to maximum power of the amplifier
Once the the feedback starts (it needs some audio to start), losing the original signal will not stop it. The level of the returning signal is enough to continue building higher until the limit of the amplifier is reached.
Example .... if the total equipment gain is 55%
(each 'Pass' is equal to the delay time thru the system)
1st pass thru .... the return audio level will be 110% of the original audio
2nd pass thru .... the return audio level will be 115.5% of the original audio
3rd pass thru .... the return audio level will be 118.5% of the original audio
4th pass thru .... the return audio level will be 120.2% of the original audio
5th pass thru .... the return audio level will be 121.1% of the original audio
6th pass thru .... the return audio level will be 121.6% of the original audio
The Real World
The previous situation was with a single tone and with the two sources in perfect phase. In an actual situation, the source would be various audio frequencies and various phases. System response (and EQ) and room acoustics would alter the level of different frequencies returning from the amplifier. One frequency would be closest to being in phase and have the greatest level and thus the feedback would start at that frequency. Moving a short distance with the microphone may completely alter this resonant frequency (many other variables can also change this response).
Feebback is Caused by the Same Audio Signal being reamplified many times
Methods used to correct Feedback
Adjust the EQ of the system to the room (this can eliminate hot frequencies)
reduce amplifier gain
reduce the number of open microphones to only those currently needed
use direction microphones
reduce amount of monitor volume (or use headphone/earbud monitoring instead)
(obviously some of these methods won´t be useful in some situations)
A delay is put in the audio path. A tone would have to be sustained for the duration of the delay to add to itself (return audio) and cause feedback. This can increase the Gain−Before−Feedback. Often used in large arena situations.
Audio Frequency Shift
The frequency of all of the audio is shifted slightly higher (or lower). The return audio would be the wrong frequency to add to the source and cause feedback. This can also increase the Gain−Before−Feedback. Useful only for voice announcements (definitely not music of any kind).
A popular method today. The audio is fed thru equipment which has numerous very sharp (ie narrow) notch filters. To set this equipment up, the system gain would be increased to the point of feedback. The frequency of the first notch would be adjusted to notch out the frequency. The system gain would then be increased to the new feedback point (should be a new frequency). The next notch filter would be adjusted similarly. Continuing with other filters would null out the worst offending frequencies.
Often this equipment would automatically set the notches for you when you follow their setup procedure. Some notch filters are usually left for dynamic operation. These would automatically be set by the equipment when feedback occurs in changing situations (ie stage movement), and can be reused for other frequencies when the situation changes.
OK now you can´t hear feedback
If you are marginally below the feedback point, you still have a problem. .......
" Phase distortion "