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. / Avhandlingen behandlar specifikation och konstruktion av mikrofon-arrayer för att extrahera talinformation. En ny adaptiv delbands beamforming-algoritm föreslås där många av nackdelarna hos konventionella adaptiva beamformers är eliminerade. En utvärdering i en bil med ett frihands-system bekräftar fördelarna med den föreslagna metoden. Blind signal-separation diskuteras och en ny struktur föreslås, baserad på en inverterande kanalidentifiering utförd i frekvensdomän med en kontinuerlig separation utförd i tidsdomän. Filterbanks-egenskaper och designmetoder diskuteras tillsammans med begränsningar som finns i beamforming-strukturer utförda i delband.
| Identifer | oai:union.ndltd.org:UPSALLA1/oai:DiVA.org:bth-00206 |
| Date | January 2001 |
| Creators | Grbic, Nedelko |
| Publisher | Ronneby : Blekinge Institute of Technology |
| Source Sets | DiVA Archive at Upsalla University |
| Language | English |
| Detected Language | English |
| Type | Doctoral thesis, comprehensive summary, info:eu-repo/semantics/doctoralThesis, text |
| Format | application/pdf |
| Rights | info:eu-repo/semantics/openAccess |
| Relation | Blekinge Institute of Technology Dissertation Series, 1650-2159 ; 1 |
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