International Journal of Scientific & Engineering Research, Volume 4, Issue 4, April-2013 699

ISSN 2229-5518

Active Noise Cancellation System Using

DSP Prosessor
G.U.Priyanga, T.Sangeetha, P.Saranya, Mr.B.Prasad

Abstract---This paper presents an integrated approach in designing a noise reduction headset for the audio and communication applications.
Conventional passive headsets give good attenuation of ambient noise in the upper frequency range, while most of these devices fail below 500 Hz.
Unlike the feed forward method, the adaptive feedback active noise control technique provides a more accurate noise cancellation since the microphone
is placed inside the ear-cup of the headset. Furthermore, the system uses single microphone per ear cup, thus produces a more compact, lower power
consumption, cheaper solution, and ease of integration with existing audio and communication devices to form an integrated feedback active noise
control headsets. Simulation results will be done to show that the integrated approach can remove the disturbing noise and at the same time, allow the
desired speech or audio signal to pass through without cancellation.

Keywords— Active Noise Control, Adaptive feedback, least mean square, active noise control, passive noise control, feedback ANC, filter length

1. Introduction

Acoustic noise problem becomes more serious, because

of increasing numbers of industrial equipment such as
engines, blowers, fans, transformers, and compressors are
in use. The traditional passive earmuffs are valued for their
high attenuation over a broad frequency range. However,
they are relatively large, costly, and ineffective at low
frequencies. Active noise control (ANC) system cancels the
unwanted noise based on the principle of superposition.
Specifically, an anti noise of equal amplitude and opposite
phase is generated and combined with the primary noise,
thus resulting in the cancellation of both noises[3]. The
ANC system efficiently attenuates low frequency noise
where passive methods are either ineffective or tend to be
very expensive or bulky. ANC is developing rapidly Fig.1 Integrated audio and ANC system
because it permits improvement in noise reduction, which
results in potential benefits in weight, volume, and cost[1].
Current research in ANC for communication and 1.1 Comparison of algorithms
entertainment headsets focuses on using feed forward
technology . In practice, Impulse noises exist in many real applications, such as
stamping machines in manufacturing plants, IV pump
sounds in the hospital. LMS algorithm is best to avoid such
however, the feed forward ANC systems for headsets have noises. A new modified FXLMM algorithm is also proposed
significant stability and performance deficiencies caused by to achieve better performance in controlling impulsive
non-stationary reference inputs, measurement noise, and noise[8]. Computer simulations are carried out for all the
acoustic feedback[4]. In response to these issues, an three algorithms (FXLMS, FXLMM and modified FXLMM)
adaptive feedback and the results are analysed[8].
ANC system is developed here for an integrated audio and
1.2 Lms filter
communication headsets as shown in fig.1
--------------------------------------------------------------------------
The LMS Filter block can implement an adaptive filter
GU Priyanga, T.Sangeetha, P.Saranya-UG Students, using five different algorithms. The block estimates the
K.S.Rangasamy college of technology, Tiruchengode, Namakkal
filter weights, or coefficients, needed to minimize the error,
[email protected]
e(n), between the output signal y(n) and the desired signal,
Mr.B.Prasad- Guide, Assistant Professor, K.S.Rangasamy college d(n).The signal that we want to filter is connected to the
of technology, Tiruchengode, Namakkal
input port. The input signal can be a scalar or a column
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International Journal of Scientific & Engineering Research, Volume 4, Issue 4, April-2013 700
ISSN 2229-5518

vector. The desired signal is connected to the Desired port. communication headsets, the microphone placed inside the
The desired signal must have the same data type, ear cup picks up both the residual noise and the desired
complexity, and dimensions as the input signal. The Output audio signal as shown in Fig.3. The audio components will
port outputs the filtered input signal, which is the estimate also become the interference to the IFBANC algorithm and
of the desired signal. The Error port outputs the result of a method is devised to neatly combine the audio and the
subtracting the output signal from the desired signal. ANC systems.

When we select LMS for the Algorithm parameter, the

block calculates the filter weights using the least mean-
square (LMS) algorithm. This algorithm is defined by the
following equations.

Fig.3 Integrated feedback ANC system

2. Placement of error microphone

Fig.2 Process undergoing in LMS filter
The spectral filtering of a sound source that is caused
1.3 Feedback ANC system primarily by the outer ear (pinna) will affect the secondary-
path transfer function . The asymmetrical, complex
In an ANC application, the primary noise is not construction of the outer ears cause a unique set of delays,
available during the operation of ANC because it was resonances, and diffractions that collectively translate into a
cancelled by the secondary noise. Therefore, the basic idea unique transfer function for each microphone position .
of an adaptive feedback ANC is to estimate the primary Intensive experiments were conducted to find the optimum
noise and use it as a reference signal for the ANC filter[4]. location for the design of the AFANC headphone. The
An adaptive feedback ANC system is required for sampling frequency chosen for the measurements is 8 kHz,
applications include spatially incoherent noise generated and the frequency range of interest being between 0–3.38
from turbulence, noise generated from many sources and kHz. As illustrated in Fig. 3, the secondary path estimation
propagation paths, and induced resonance where no is conducted for eight different error microphone locations
coherent reference signal is available. on the median plane around the ear-pad of the headphone.
The positions #6, #7, and #8 are clearly identified.
1.4 Integrated feedback ANC audio system

We can integrate the above feedback ANC with the

receiving audio input to form an Integrated Feedback
Active Noise Control (IFBANC) system. In IFBANC, the
residual noise picked up by the error microphone is used to
synthesize the primary noise for updating the adaptive
filter coefficients using the FXLMS algorithm. Since the
IFBANC is integrated with the existing audio playback
systems (such as walkman or Mp3 players) or
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3.1 Filter Length

The filter length of the adaptive system is inherently
tied to many of the other performance measures. The length
of the filter specifies how accurately a given system can be
modeled by the adaptive filter. In addition, the filter length
affects the convergence rate, by increasing or decreasing
computation time, it can affect the stability of the system, at
certain step sizes, and it affects the minimum MSE. If the
filter length of the system is increased, the number of
computations will increase, decreasing the maximum
convergence rate. Conversely, if the filter length is
decreased, the number of computations will decrease,
increasing the maximum convergence rate. For stability,
due to an increase in length of the filter for a given system,
you may add additional poles or zeroes that may be smaller
than those that already exist. In this case the maximum step
Fig.4 Different positions for the placement of error
size, or maximum convergence rate, will have to be
microphone
decrease to maintain stability. Finally, if the system is under
It was observed that the system showed the flattest specified, meaning there is not enough pole and/or zeroes
frequency response with least number of dips and peaks at to model the system, the mean square error will converge
the microphone locations #8. Since position #8 is near the to a nonzero constant. If the system is over specified,
external auditory meatus, the microphone could fit in more meaning it has too many poles and/or zeroes for the system
easily inside the hollow in that location. model, it will have the potential to converge to zero, but
increased calculations will affect the maximum
3. Experiments carried using MATLAB convergence rate possible.

4. Results
The noise reduction headset is implemented using the
Adaptive feedback active noise control mechanism. The MATLAB was used to simulate the hardware portion of
system is more compact, consume less power and ease of the study. In order to simulate properly, the same
integration with existing audio and communication system. microphone used in hardware must be used to record
It works on the Superposition principle. Anti noise that has various noises. The first consisted of simulating the
the equal magnitude as that of the original signal but inversion of noise and adding two signals destructively.
opposite in phase is generated and combined with primary The second test consisted of creating filters so retain or
noise that result in the cancellation of noise. ANC system remove specific frequencies. The test involved adaptive
attenuates low frequency noise that cannot be eliminated filters using the least mean squares methods.
using the passive headsets. 4.1 Simulink model for noise cancellation

Fig.5 Basic block diagram of proposed system

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Fig.8 Waveform of exterior mic input to LMS filter

Fig.6 Simulink model of proposed system

The Waterfall window displays the behavior of the

adaptive filter's filter coefficients. It displays multiple
vectors of data at one time. These vectors represent the
values of the filter's coefficients of a normalized LMS
adaptive filter, and are the input data at consecutive sample
times. The data is displayed in a three-dimensional axis in
the Waterfall window.

Fig.9 Output waveform

The model was executed and the above mentioned output

waveforms were obtained. The upper port (exterior mic) of
the Acoustic Environment subsystem is white noise. The
signal output at the lower port (pilot's mic) is composed of
colored noise and a signal from a .wav file. This demo
model uses an adaptive filter to remove the noise from the
signal output at the lower port. When we run the
simulation, we hear both noise and the music in .wav
format. Over time, the adaptive filter in the model filters
out the noise so we only hear the music without noise.
Fig.7 Waveform of pilot mic input to LMS filter
5. Conclusion

An integrated feedback ANC algorithm has been

presented in this paper to reduce the interference without
cancelling out the desired audio signal. The integrated
structure of the IFBANC system, enable a cheaper solution

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International Journal of Scientific & Engineering Research, Volume 4, Issue 4, April-2013 703
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in a digital signal processing based platform. In addition, [7] F.Li and Ji Ming,” Robust Speaker Recognition in Noisy
only a single microphone per head cup can produce a more Conditions,” IEEE trans. on audio, speech, and language
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system can be implemented using the hardware DSP Acoustic Noise Using a Self-Tuned Filter,” IEEE trans. on
TMS320c54x processor. circuits and systems—I, vol. 51, pp. 822-836, November
2004.

6. Future work

DSP modules can be purchased and the codings can be

downloaded . Then the module can be fixed to the
commercial headsets. The error microphone is attached to
the ear muff and the two microphone inputs can be given
to the module so that the compact active noise control
headset can be built.

7. Reference

[1] Cheng-Yuan Chang and Sheng-Ting Li,” Active noise

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