This paper presents a method to simultaneously perform 20 dB acoustic echo cancellation and 15-20 dB speech enhancement using an adaptive microphone array combined with spectral subtraction. Primarily intended for handsfree telephones in automobiles, the microphone array system simultaneously emphasizes the near-end talker and suppresses the handsfree loudspeaker and the broadband car noise. The array system is based on a fast and efficient on-site calibration and can be used in other situations such as conventional speaker phones.

Simultaneous echo cancellation and car noise suppression employing a microphone array

This paper introduces a new method for the estimation of sound source distance and direction using at least three microphone sensors in indoor environments. Unlike the other methods that normally use approximations in obtaining the time difference between sensors, this method exploits the existed geometrical relationships of the sensors to form an exact solution to estimating the source position. To overcome reverberation, an enhancing pre-process has been used for different sound sources with different spectra, e.g., single frequency, multiple frequencies and different noise shapes. Source direction and distances are estimated from time of sound wave travel and distances of acoustic sensors. Using the method described in this paper a level of 1° accuracy is obtained. Several experimental tests have been undertaken that verify the results. Conclusions and future work are also described.

https://iopscience.iop.org/article/10.1088/0957-0233/19/2/024003/pdf

Accurate sound source localization in a reverberant environment using multiple acoustic sensors

This paper presents a new method to simultaneously perform 20 dB acoustic echo cancellation and 10 dB speech enhancement utilizing an adaptive microphone array. The system is based on a fast and efficient on-site calibration. Primarily intended for handsfree telephones in automobiles, the microphone array system simultaneously emphasizes the near-end talker and suppresses the handsfree loudspeaker and car noise. The method can also be used in other situations such as conventional speaker phones.

Acoustic Echo and Noise Cancelling Using Microphone Arrays

The increased use of personal communication devices, personal computers and wireless cellular telephones enables the development of new inter-personal communication systems. The merge between computers and telephony technologies brings up the demand for convenient hands-free communications. In such systems the users wish to lead a conversation in much the same way as in a normal person-to-person conversation. The advantages of hands-free telephones are safety, convenience and greater flexibility. In many countries and regions, hand held telephony in cars is prohibited by legislation. By placing the microphone far away from the user a number of disadvantages are introduced, which results in substantial speech distortion and poor sound quality. These disturbances are mainly caused by room reverberation and background noise. Furthermore, acoustic feedback generated at the near-end side is a problem for the far-end side talker, who will hear his/her own voice echoed with 100-200 ms delay, making speech conversation substantially more difficult. Digital filtering may be used to obtain a similar sound quality as for hand held telephony. Three major tasks must be addressed in order to improve the quality of hands-free communication systems; noise suppression, room reverberation suppression, and acoustic feedback cancellation of the hands-free loudspeaker. The filtering operation must perform the above mentioned tasks without causing severe near-end speech distortion. A properly designed broad-band microphone array is able to perform all the given tasks, i.e. speech enhancement, echo cancellation and reverberation suppression, in a concise and effective manner. This is due to the fact that the spatial domain may be utilized as well as the temporal domain. This thesis deals with the problem of specification and design of beamformers used to extract the source signal information. A new subband adaptive beamforming algorithm is proposed, where many of the drawbacks embedded in conventional adaptive beamforming are eliminated. Evaluation in a car hands-free situation show the benefits of the proposed method. Blind signal separation is discussed and a new structure based on frequency domain inverse channel identification and time domain separation, is proposed. Further, filter-bank properties and design are discussed together with performance limitations in subband beamforming structures.

Optimal and Adaptive Subband Beamforming

This paper describes the results of a new approach devoted to the localization of ground borne acoustic sources. It is demonstrated that an array made of at least three unidirectional microphones can be exploited to identify the position of the source. Sound features extracted either in the time domain or in the frequency domain are used to localize the direction of the incoming sound. This information is then fed into a semi-analytical algorithm aimed at identifying the source location. The novelty of the method presented here consists in the use of unidirectional microphones rather than omnidirectional microphones and in the ability to extract the sound direction by considering features like sound amplitude rather than the time of arrival. Experimental tests have been undertaken in a closed environment and have demonstrated the feasibility of the proposed approach. It is believed that this method may pave the road toward a new generation of reduced-size sound detectors and localizers, and future work is described in the conclusions.

/pdf/localization-of-sound-sources-by-means-of-unidirectional-1h1i6vnpp1.pdf

Localization of sound sources by means of unidirectional microphones

The quality of audio teleconferencing in large rooms and noisy environments can be increased with the use of steerable directional microphone arrays. A minimum bandwidth of 4 oct is required to faithfully transmit the speech signal. In a typical teleconferencing arrangement, only discrete angular directions are of interest and therefore the microphone steering directions are quantized. A standard delay‐sum beamformer can result in noticeable frequency response changes as the talker moves between these steering locations. In an effort to mitigate this problem, a broadband constant‐directivity beamformer has been designed and constructed. A few of the algorithms developed in this work will be discussed and compared to existing techniques. Basically, the solution revolves around the design of FIR filters that are inserted in the delay‐sum beamformer after each element. A constant‐beamwidth 4 oct steerable linear array microphone using directional elements will be described. A real‐time implementation utilizing multiple AT&T DSP3210 digital signal processors is also described.

Thomas C. Chou

Papers

A constant‐directivity beamforming microphone array