SPEECH ENHANCEMENT USING MICROPHONE ARRAY SIGNAL PROCESSING

SYNOPSIS
The main objective of this project is to design the single dimensional(1D),two dimensional(2D) and three dimensional (3D ) microphone arrays . Multiple microphone array speech enhancement algorithms have been simulated using the matlab. The array output has been used to signal processing techniques using beamforming to obtain frequency invariance and adaptive beamforming in which the input speech signal under the influence of noisy conditions. It has been verified that output has been analyzed using subjective and objective methods. Observations on the signal obtained is clear compared to the input noisy signal

INTRODUCTION
1.1 Introduction
Speech enhancement systems aim to improve the quality and intelligibility of speech and reduce communication fatigue due to noise, loss of speech packets or limited bandwidth. Speech enhancement benefits a wide range of applications such as wifi and cellular mobile phones, VoIP, hands-free phones, teleconferencing, in-vehicle cabin communication, hearing aids, automated voice services based on speech recognition and synthesis. Hands-free audio communication is now a major feature in mobile communication systems as well as audio and video conferencing systems. Single-input noise reduction systems strive to suppress the audibility of the noise by utilising the temporalspectral structures of signal and noise processes. The speech enhancement is usually achieved through multiplication of the signal frequency components by a gain factor derived from the estimates of the prior and the posterior signal-to-noise ratios. Multiple-input systems strive to separate the noise from the noisy speech signal. Eg adaptive noise

cancellation, microphone array beam formers and independent component analysis. Conventional telephony speech is narrowband as it is limited to a bandwidth of 300 Hz to 3400 Hz whereas wideband broadcast speech have a bandwidth of 20 Hz to 20 000 Hz. Several methods for extrapolation of bandwidth of narrowband telephony speech to wideband broadcast.

1.2 Beamforming
In many applications, the desired information to be extracted from an array of sensors is the content of a spatially propagating signal from a certain direction. The content may be a message contained in the signal, such as in communications applications, or merely the existence of the signal, as in radar and sonar. To this end, we want to linearly combine the signals from all the sensors in a manner, that is, with a certain weighting, so as to examine signals arriving from a specific angle. This operation is known as beamforming because the weighting process emphasizes signals from a particular direction while attenuating those from other directions and can be thought of as casting or forming a beam. Beamforming algorithms have shown great promise in noise reduction, through utilizing the spatial information of the noise and primary source signals. As the number of microphones in an array increases, increasing the aperture size, the ability of beamforming algorithms to extract the primary source using spatial information improves. The fixed beamformers used initially have narrowband, far-field assumptions. The spacing of the microphones as related to the distance to the sources is chosen appropriately for the far-field assumption. Despite the fact that speech signals are broadband signals, narrowband assumptions can be approximated with the use of filter banks applied to each microphone input.

1.3 Microphone Arrays

Microphone array provides an effective means of hands-free speech acquisition, finding application in large-room video conferencing, desktop teleconferencing, and hands-free high spatial

telephones. Unlike a single omni-directional microphone, the array provides

directivity, allowing it to attenuate room reverberation and other unwanted noise. In this thesis, we give an overview of the techniques available for speech acquisition using microphone arrays. In particular we will focus on methods of frequency- invariant beamforming, in which the array maintains the same spatial response over a wide frequency range. The directivity of the broadband microphone array is optimized by adjusting the spatial transducer positions and the impulse response of the beam former. The target beam pattern for optimization is defined in terms of the desired source signal directions mainlobe and the angles for background noise attenuation.

1.4 In This Project

Design and simulation of microphone arrays and beam forming techniques to obtain optimum results in delay sum beam forming. Multiple microphone array speech enhancement algorithms have been simulated using the matlab software. Under various noisy conditions the performances of the delay and sum and Griffiths-Jim beam forming methods have been tested by the subjective and objective analysis. From the results, more signal to noise ratio (SNR)

available for Multichannel beam forming methods compared to single channel methods. Speech enhancement in noisy environments improves the quality and intelligibility of speech and reduces communication fatigue. This research addresses the problem of primary source enhancement in a multiple source environment. It is important to note that enhancement techniques addressing speech signals contaminated with nonstationary speech as noise are not yet fully developed. To improve the quality and recognition of the speech signal of interest, a microphone array along with beamforming and speech enhancement algorithms can be used to separate the primary speech signal from the interfering speech signals. The novel approach of using multiple beamformers to estimate each source signal and using those estimates in traditional speech enhancement algorithms adapted to a multiple source problem is implemented in this research. Thus, it is the goal of this research to enhance the quality of the primary speech signal of interest through the development and implementation of multiple source beamforming and enhancement algorithms. The ability to separate or enhance a primary speech signal in an environment with many speakers, the so called cocktail party effect is an important issue especially in recent years with the number of people with hearing damage dramatically on the rise and with the expansion of global businesses requiring the use of more sophisticated video and teleconferencing equipment.

1.5 In This Thesis

In this first chapter brief description is given about microphone array signal processing.In the second chapter speech proceesing background and how speech is produced are discussed. In the third chapter various speech enhancement methods like single sensor and multisensor

methods are explained. In fourth and fifth chapter microphone array beamforming and beampatterns and frequency responses of microphone arrays and various beamforming techniques are implemented and results are shown. In the sixth chapter independent component analysis is discussed and cocktail party effect simulations and results are explained to acquire intelligible speech. Finally discussion about conclusion and future scope.