Acoustic Echo and Adaptive Noise Cancellation

Microphone System
Farrukh Tanveer
Electrical Engineering Department
Fast Nuces Lahore Campus
farrukh_1823@yahoo.com

AbstractAcoustic echo is a common phenomenon in

todays telecommunication systems. It occurs when an
audio source and sink run in full-duplex mode. An
example of this is a hands free loud speaker telephone. An
effective technique of reducing acoustic echoes in a hands
free telephone system is the adaptive echo cancellation
microphone system [1, 2]. In this case, the received signal
through the telephone loud speaker is the output (audio
source). This audio signal is then resonated through the
physical environment and picked up by the systems
microphone (audio sink). Unwanted echo reduces by using
adaptive filtering techniques which results in improving
communication quality. The adaptive noise and echo
cancellation microphone system is more efficient and less
complex than the combination of the traditional adaptive
noise cancellation and acoustic echo cancellation
techniques.

1. INTRODUCTION
A common problem met in hands-free telephones is the
presence of echoes. These echoes generated acoustically by
the coupling between the loudspeaker and the microphone
via impulse response of a room. In recent years, there has
been great interest in the use of adaptive filters as acoustic
echo cancellers to remove echoes.
An adaptive filter can be characterized by its structure
and adaptive filtering algorithm. The traditional acoustic
echo canceller [3] uses one adaptive FIR filter with the
least mean- square (LMS) algorithm [4] to model the echo
path between the loudspeaker and the microphone.
In acoustic echo cancellation applications, the input signal
of the adaptive filter is highly correlated and the impulse
response of the acoustic echo path is very long [8].
Unfortunately,this results in slow convergence, large excess
mean-square and numerical errors [5, 6].
One
technique to solve this problem is sub band adaptive
filtering. In conventional sub band adaptive filters, each
sub-band uses an individual adaptive sub filter in its own

adaptation loop, which decreases the convergence rate of

SAFs because of the aliasing and band- edge effects.
Recently, an adaptive combination of full band adaptive
filters has been proposed. The benefit of this combination
is that it can obtain both fast convergence rate and small
steady state mean square error without estimate of the
system noise power. This problem can be solved by placing
the reference sensor next to the noise source and placing
the primary sensor close to the signal source that is located
far away from the reference sensor [8]. However, this
arrangement also reduces the correlation between the
noise components in the two signals. However, it is not
always possible to place the reference sensor far away from
the signal source in practical applications such as cellular
communications. The order of adaptive filter required for
noise cancellation is another practical aspect.
In order to achieve satisfactory noise cancellation, an FIR
filter of order N = 1500 is required for a separation of a
few yards between the two sensors [8]. Similar to the
adaptive echo canceller, high order filter results in high
complexity, slow convergence, and high errors.
In this paper, we propose an active noise and echo
cancellation microphone system. This system overcomes
the problems in the conventional system. Section 2
presents the principle and algorithm of the active noise
and echo cancellation system. The steady-state analysis of
the noise and echo cancellation system and comparison of
its performance and the traditional adaptive echo
canceller are also presented in this section. The
experimental setup and simulation results are presented in
Section 3.

2.

PRINCIPLE AND ALGORITHM

The structure of the adaptive noise and echo cancellation

system is shown in Fig. It consists of two directional
microphones and two adaptive filters. The primary
microphone Mp is pointed at the near-end taker and the
reference microphone Mr is pointed at the opposite
direction.

The LMS algorithm is used to update the coefficients, b(n).

It is expressed in the following equation.

Xp (n) and Xr (n) are outputs from the primary and

reference microphones Mp and Mr, respectively. These
outputs are also called the signals of the system. A (n) is the
adaptive filter of order n and is used for noise cancellation.
B(z) is an adaptive FIR filter of order M and is used to
cancellation the acoustic echo, Ya(n) and Yb(n) are the
outputs of filters A(z) and B(z) respectively, e(n) is the error
signal used to adjust the coefficients of the adaptive filters
and also the system output. The delay is used to guarantee
the causality of A(z) [4]. The operation mode detector is
developed in [2].

2.1

IDLE MODE

In this mode, only the background noise from the noise

source is present. Therefore only A(z) is used. The output
signal e(n) can be expressed as;

Where, the step size u determines the convergence and

stability of the algorithm.
Thus, a low-order adaptive filter B(z) can be used to
achieve significant echo cancellation.

2.3 Transmit Mode

In this mode, only the near-end speech and the noise are
present. The fixed filter A*(z) with coefficients obtained
from the previous IDLE mode is used to cancel the noise in
the primary signal. The filter B (z) is not used in this case
since there is no acoustic echo. The system output can be
expressed as;

The near-end speech leaking into Mr can be ignored

because Mr is a directional microphone pointing at the
opposite direction of the near-end talker. Thus, the nearend speech picked up by the primary microphone will be
sent out with minor distortion.

2.4 Double-Talk Mode

The LMS algorithm is used to update the coefficients, a(n).

It is expressed in the following equation;

Where, the step size u determines the convergence and

stability of the algorithm.

2.2 Receive Mode

In this mode, the near-end speech is absent, but both the
far-end speech from the loudspeaker and the noise from the
noise source are present. Assuming that the noise source
locates at the same place as in the idle mode and receive
mode, the optimal filter A*(z) obtained from the previous
idle mode is able to cancel the noise components in Xp (n).
The adaptive filter B (z) is updated and performs the
acoustic echo cancellation to minimize the residual error.
The system output becomes;

All signals, the near-end speech from the near- end talker,
the far-end speech (echo) from the loudspeaker and noise,
are present in this mode. Therefore, there is no update
performed for neither of the filters. The system output can
be expressed as;

The fixed filter A*(z) from the previous idle mode is used to
cancel the noise components in the primary signal, while
the fixed filter B*(z) from the previous receive mode is used
to cancel the acoustic echo.

3. Signal Model of Acoustic Echo Canceller

A block diagram for acoustic echo canceller is illustrated in
Fig. When the far-end signal u(n) goes through the echo
pathW0 (z) of a room, the acoustic echo is produced. The
acoustic echo is picked up by the microphone together with
the near-end signal (n), resulting in the microphone signal
d(n).The near-end signal may contain the system noise v(n)
and near-end speech s(n).
The goal of the adaptive filter w(k) is to produce a replica
of the echo signal, y(n), by adaptively adjusting the tapweights of W(z).Then y(n) can be used to cancel the echo
by subtracting it from the microphone signal d(n).

12 feet)] from the noise source to the sensors were

chosen. For the traditional adaptive noise cancellation
system, the reference microphone was placed next to the
noise source and the primary microphone was placed at
12 feet away from the noise source. For both systems,
the directional microphones were used for the
experiments.

4. Experiments
Different experiments were performed in a typical
conference room. The experimental setup for the
adaptive noise cancellation microphone system is
illustrated in Fig. The Ariel DSP- 16 system was used for
data acquisition. A sampling rate of 10 kHz was used for
all the experiments. The system performance was
estimated by the equation:

Where, M is the averaged period in samples and k is the

time at which the adaptive filter is assumed to have
converged.

First, the performance of adaptive noise and echo

cancellation microphone system is evaluated. For each
noise source location, the same filter order (N = 128)
was used and the optimal step size was chosen to
achieve the best result. When the distance is 2 feet the
noise reduction is 35.39 dB and the step size is 0.005.
When the distance is 3 feet the noise reduction is 27.04
dB and the step size is 0.007.when, the distance is 6 feet
the noise reduction is 20.93 dB and the step size is 0.01.
Finally when the distance is 12 feet the noise reduction
is 16.07 dB and the step size is 0.1. The results show that
the system achieved best performance when noise
source is placed 2 feet away From the sensors, which is
realistic for most practical applications.
Secondly, the performance of both adaptive noise and
echo cancellation microphone system and adaptive noise
cancellation systems are evaluated under the same filter
length. For each filter order, a different step size was
chosen to achieve the best performance. The results
show that the adaptive noise and echo cancellation
microphone system (when noise source is 12 feet away
from sensors) achieved -16 dB noise reductions by using
a filter of order 128. However, the traditional adaptive
noise cancellation system has to use a much higher
order filter (over 1500) to obtain only -14 dB noise
reduction. Thus the adaptive noise and echo
cancellation system has a much higher performance for
the entire range of filter orders.

4.2 Echo Cancellation

4.1 Noise Cancellation

The performance of adaptive noise and echo
cancellation system was first compared with the
performance of the traditional adaptive noise
cancellation system. For the adaptive noise and echo
cancellation system, four different distances 2, 3, 6 and

The adaptive noise and echo cancellation system and

traditional adaptive echo cancellation systems were
compared by using the same speech segment under the
same conditions. The performance of both systems was
evaluated with different filter order N. The optimal
convergence factor, u for each filter order was found
and used in the final result [1]. The adaptive noise and
echo cancellation system has much higher performance
over a long range until the filter order is increased to
around N = 600. The system has a very good
performance with a low order filter of (N = 32, p = 22.34
dB).

If the two microphones can be positioned even closer,

higher performance may be achieved. Experiments
show that the performance of the adaptive noise and
echo cancellation system is always superior to the
traditional adaptive echo cancellation system when the
filter order is less than 50. Also, the adaptive noise and
echo cancellation microphone system has a satisfactory
performance with a much lower order adaptive filter.
Therefore, the system is very promising for low-cost
applications. It also has other advantages such as fast
convergence, low excess mean-square error, and small
the use of low order adaptive filter.

5 Conclusion
This paper presents the integration of an acoustic noise
canceller with the adaptive acoustic echo cancellation
microphone system. By using two highly correlated
microphone signals, the adaptive noise and echo
cancellation
system
demonstrated
satisfactory
performance using much lower order filters. The system
was broadly tested using computer simulations with real
statistics collected from the conference room. The
experimental results show that the adaptive noise and
echo cancellation microphone system can provide
significant noise and echo reduction with much lower
filter orders than the combination of the conventional
adaptive noise cancellation and adaptive echo
cancellation systems. Thus, the adaptive noise and echo
cancellation microphone system provides flexibility and
high performance at a low cost for teleconferencing,
cellular communications, hands-free telephones,
desktop multi-media and public address systems.
To verify the effectiveness of the proposed scheme,
experiments using different distances as well as for
different step sizes were performed. The experimental
results demonstrated that the proposed scheme can
obtain improved performance as compared to the

adaptive noise
cancellation.

cancellation

or

adaptive

echo

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