DBX Advanced Feedback Suppression White Paper

dbx ADVANCED FEEDBACK SUPPRESSION™ (AFS™)
Aaron Hammond
DSP Engineer, dbx Professional Products
INTRODUCTION
Feedback is the bane of almost all PA systems. It can take a great performance and turn it into a painful and embarrassing experience for the performer, audience, and the sound operator. Up until a few years ago there was little that could be done electronically about feedback except rudimentary efforts using EQ. With the advent of Digital Signal Processing (DSP), automatic feedback elimination has been made possible. Unfortunately, many of these earlier products did not maintain the sonic integrity of the audio signal because they required wide notch filters to suppress feedback. The dbx® AFS™ algorithm solves this problem by using Precision Frequency Detection minimum number of very narrow notch filters, which will stop the feedback without degrading the audio signal.
HOW ACOUSTIC FEEDBACK OCCURS
with adaptive filter bandwidth to place the
Acoustic feedback occurs in a sound re-enforcement system when the signal output from the speaker is picked up by the microphone and amplified, creating a feedback loop. What results is an audible “squealing” or “howling” of the system. Figure 1 illustrates a typical system setup with a microphone, mixer, amplifier, and speaker.
Figure 1
The loop gain consists of the gains in the mixer ( (
), as well as the losses in the system (
ampG _
) and amplifier
mixG _
), as illustrated in Figure 2.
sysL _
Figure 2
includes the losses from microphone to mixer, speaker to microphone,
sysL _
and any other losses in the loop. Therefore, the loop gain of the system can be represented mathematically as
.___ sysLampGmixGLoopGain =
This equation can be represented in dB as
.___
sysLampGmixGLoopGain ++=
dBdBdBdB
For feedback to occur, the loop gain must be greater than unity (or greater than 0 dB) and in phase at a particular frequency. When this occurs, the loop gain at the feedback frequency must be reduced below unity to remove the feedback. This gives us
,0___
dBattendBdBdB
where
LoopGain
G
represents the necessary amount of attenuation required to pull the
atten
=
dB
GsysLampGmixG <+++
loop gain at that frequency below 0 dB.
dbx ADVANCED FEEDBACK SUPPRESSION™ (AFS™)
The dbx Advanced Feedback Suppression™ (AFS™) algorithm eliminates feedback by placing a very narrow notch filter at the frequency feeding back. When the loop gain at that frequency is pushed below unity, the feedback
disappears. Using our patent pending Precision Frequency Detection™ with adaptive filter bandwidth, we are able to place the minimum number of very
narrow notch filters (Q = 116, bandwidth = 1/80 octave1). Utilizing very narrow notch filters preserves the sonic quality of the system.
Historically (before automatic feedback elimination), feedback was removed manually using a 1/3 octave graphic or parametric EQ. When feedback occurred, the sound engineer would guess where the feedback was located, and pull down a fader to decrease the gain at that frequency. This method unnecessarily cuts out
large portions of the spectrum. The dbx Advanced Feedback Suppression
(AFS) algorithm uses a very narrow notch filter to reduce the gain at the feedback frequency. Figure 3 compares a 1/3 octave graphic EQ with the dbx AFS narrow notch filter. Again,
G
represents the necessary cut required to
atten
very
guarantee that the feedback is removed. It is easy to see the limitations of the manual approach.
Figure 3
1
The bandwidth of a filter can be stated as Q or in octaves. The Q is computed by dividing the center
frequency by the bandwidth of the filter. For dbx feedback notch filters, the bandwidth is measured at the
-3dB point (from 0 dB). That means that no matter how deep the notch filter cuts, its bandwidth will be measured from –3 dB. This is important because many competitors will claim to have narrow notch filters, but they measure their bandwidth 3dB above the peak cut depth. In other words, for a cut depth of –18 dB, some of our competitors measure the bandwidth at –15 dB, which results in a significantly wider bandwidth filter (this will be explained graphically in the next section). The other way to measure filter width is in octaves. This means that the number stated (say 1/10 octave) is the bandwidth of the filter, which varies depending on the center frequency.
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