Qanbar Ali Khan(2011342), Waqas Khalid(2011315) Mohammad Osama(2011),Imran Khan(2011)

Signals & Systems Lab, Faculty of Engineering Sciences, Ghulam Ishaq Khan Institute,Topi,Pakistan
Lokomotiv7@hotmail.com waqaskhalid@hotmail.com

ABSTRACT:
Acoustic echo cancellation is a common Occurrence in todays telecommunication systems. It occurs when an audio source and sink operate in full duplex mode .The signal interference caused by acoustic echo is distracting to both users and causes a reduction in the quality of the communication. This paper focuses on the use of adaptive filtering techniques to reduce this unwanted echo, thus increasing communication quality. Adaptive filters alter their parameters in order to minimize a function of the difference between a desired target output and their output. In the case of acoustic echo in telecommunications, the optimal output is an echoed signal that accurately emulates the unwanted echo signal. This is then used to negate the echo in the return signal. The better the adaptive filter emulates this echo, the more successful the cancellation will be. This paper examines various techniques and algorithms of adaptive filtering, employing discrete signal processing in MATLAB. Also noise cancellation algorithms are implemented using simulink in MATLAB.

INTRODUCTION:
Acoustic echo occurs when an audio signal is reverberated in a real environment, resulting in the original intended signal plus attenuated, time delayed images of this signal. This project will focus on the occurrence of acoustic echo in telecommunication systems. Such a system consists of coupled acoustic input and output devices, both of which are active concurrently. An example of this is a hands-free telephony system. In this scenario the system has both an active loudspeaker and microphone input operating simultaneously. The system then acts as both a receiver and transmitter in full duplex mode. When a signal is received by the system, it is output through the loudspeaker into an acoustic environment. This signal is reverberated within the environment and returned to the system via the microphone

input. These reverberated signals contain time delayed images of the original signal, which are then returned to the original sender (Figure 1, ak is the attenuation, tk is time delay). The occurrence of acoustic echo in speech transmission causes signal interference and reduced quality of communication.

FIGURE 1: Origins of acoustic echo

The method used to cancel the echo signal is known as adaptive filtering. Adaptive filters are dynamic filters which iteratively alter their characteristics in order to achieve an optimal desired output. An adaptive filter algorithmically alters its parameters in order to minimize a function of the difference between the desired output d(n) and its actual output y(n). This function is known as the cost function of the adaptive algorithm. Figure 2 shows a block diagram of the adaptive echo cancellation system implemented throughout this paper. Here the filter H(n) represents the impulse response of the acoustic environment, W(n) represents the adaptive filter used to cancel the echo signal. The adaptive filter aims to equate its output y(n) to the desired output d(n) (the signal reverberated within the acoustic environment). At each iteration the error signal, e(n)=d(n)-y(n), is fed back into the filter, where the filter characteristics are altered accordingly.

FIGURE 2: Block diagram of an adaptive echo cancellation system

This project deals with acoustic echo as applies to audio signals, although the techniques will be applicable to a variety of other disciplines. The goals of this project are as follows: To examine adaptive filtering LMS technique as they apply to acoustic echo cancellation and audio signals. To simulate LMS algorithm using Matlab. THEORY:

Echo is a phenomenon in which a delayed and distorted version of an original sound or electrical signal is reected back to the source. There are two main types of echo, namely network (or line) and acoustic echoes. As we are doing work on acoustic echo, the explanation of acoustic echo is that if a communication is between one or more hands-free telephones, thenacoustic feedback paths are set up between the telephone's loudspeaker and microphoneat each end. This acoustic coupling is due to the reection of the loudspeaker's soundwaves from walls, oor, ceiling, windows and other objects back to the microphone. Thecoupling can also be due to the direct path from the loudspeaker to the microphone,see Figure 1. Adaptive cancellation of such acoustic echo has became very importantin hands-free communication systems, e.g. tele-

conference, video-conference and PCtelephony systems. The eects of an echo depend mostly on the time delay between the initial andreected sound waves (or sound signals), and the strength of the reected sounds. In thecase of acoustic echo, if the time delay is not long, then the echo can be perceived assoft reverberation, which adds artistic quality, for example in a concert hall. However, astrong echo that arrives a few tens of milliseconds or more after the initial direct soundwill be highly undesirable and irritating.

RESULTS OF LMS ALGORITHM:

The LMS algorithm was simulated using Matlab. Figure 2 shows the input speech signal which is collected from the computer system through microphone. Figure 3 shows the desired echo signal derived from the input signal. Figure 4 shows the adaptive filter output which will reduce the echo signal from the input signal. Figure 5 shows the mean square error signal calculated from the filter output signal. Figure 6 shows the attenuation which is derived from the division of echo signal to the error signal. The adaptive filter is a 1025th order FIR filter. The step size was set to 0.02. The MSE shows that as the algorithm progresses the average value of the cost function decreases.

FIGURE 2:INPUT SIGNAL

FIGURE 3:DESIRED SIGNAL

FIGURE 4:ADAPTIVE FILTER OUTPUT

FIGURE 5:MEAN SQUARE ERROR

FIGURE 6:ATTENUATION

RESULTS OF NLMS ALGORITHM:

The NLMS algorithm was simulated using Matlab. Figure 7 shows the input signal. Figure 8 shows the desired signal. Figure 9 shows the adaptive filter output. Figure 10 shows the mean square error. Figure 11 shows the attenuation. The adaptive filter is a 1025th order FIR filter. The step size was set to 0.1.

FIGURE 7:INPUT SIGNAL

FIGURE 8:DESIRED SIGNAL

FIGURE 9:ADAPTIVE FILTER OUTPUT

Figure 10: Mean Square Error

Figure 11: Attenuation NLMS algorithm is having the advantage over the LMS algorithm incase of Mean square error and Average attenuation and its summary of the performance is presented in Table 1. Table 1. Summary of adaptive algorithms performance ALGORITHMS ITERATIONS FILTER MEAN AVERAGE COMPUTATIONS ORDER SQUARE ATTENUATIONS ERROR LMS 7500 1025 0.001 -11.2435 2N+1 NMS 7500 1025 0.0004 -13.6812 3N+1 FUTURE WORK: The future work to be done is about the same to input the audio file as a input for different algorithms namely NLMS (Normalized Least Mean Squares) Algorithm,VSSLMS (Variable Step Size Least Mean Squares) Algorithm, VSSNLMS (Variable Step Size Normalized Least Mean Squares) Algorithm and also with RLS Algorithm.Once determined the above mentioned

parameters using different algorithms a comparison is done with respect to the calculations and choosing the best for the abovementioned project and implementing the same using processor or at the gate level implementation. There are many possibilities for further development inthis discipline, some of these are as follows.The real time echo cancellation system can beimplemented using the TI TMSC6711 DSK. CONCLUSION: In the present work, the MDF adaptive filter isimplemented on Cortex-M4 processor to eliminate the acoustic echo of the far-end speaker. It requires less memorystorage, small FFT size. In performance, the MDF adaptivefilter has a smaller block delay and is faster. This is achievedby updating the weight vectors more often and reducing thetotal execution time in most of the processor. Furthermore,the total number of blocks needed can be changed dynamically without interrupting the normal operation. The MDF adaptive filter is most suitable for real-time applications implemented on the hardware. ACKNOLEDGEMENT: It is just because of Sir Shoaib, today we are able to make such a tough project. Who strive hard on us to enhance our skills peculiar in MATLAB . He taught us in such a cordial enviroment that difficult task also became easy. I also want to thanks Sir Umer Rahim for his efforts in the signal and systems subject. REFFERENCES: [1]Homana, I.; Topa, M.D.; Kirei, B.S.; Echo cancelling using adaptive algorithms, Design and Technology of Electronics Packages, (SIITME) 15th International Symposium., pp. 317321, Sept.2009. [2]. Paleologu, C.; Benesty, J.; Grant, S.L.; Osterwise, C.; Variable step-size NLMS algorithms for echo cancellation 2009 Conference Record of the forty-third Asilomar Conference on Signals, Systems and Computers., pp. 633-637, Nov 2009. [3]. Soria, E.; Calpe, J.; Chambers, J.; Martinez, M.; Camps, G.; Guerrero, J.D.M.; A novel approach to introducing adaptive filters based on the LMS algorithm and its variants, IEEE Transactions, vol. 47, pp. 127-133, Feb 2008. [4]. Tandon, A.; Ahmad, M.O.; Swamy, M.N.S.; An efficient, low-complexity, normalized LMS algorithm for echo cancellation, IEEE workshop on Circuits and Systems, 2004. NEWCAS 2004, pp. 161-164, June 2004. [5]. Eneman, K.; Moonen, M.; Iterated partitioned block frequency-domain adaptive filtering for acoustic echo cancellation, IEEE Transactions on Speech and Audio Processing, vol. 11, pp. 143-158, March 2003.