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Additional resources for Advanced Digital Signal Processing and Noise Reduction, Third Edition
At the transmitter, speech is segmented into blocks about 30 ms long, during which speech parameters can be assumed to be stationary. Each block of speech samples is analysed to extract and transmit a set of excitation and filter parameters that can be used to synthesise the speech. At the receiver, the model parameters and the excitation are used to reconstruct the speech. 11. The aim of transformation is to convert the signal into a form that lends itself to more convenient and useful interpretation and manipulation.
1991) Adaptive Filter Theory. Prentice-Hall, Englewood Cliffs, NJ. Kailath T. (1980) Linear Systems. Prentice Hall, Englewood Cliffs, NJ. E. (1960) A new approach to linear filtering and prediction problems. Trans. ASME Ser. D J. Basic Engng, 82: 35–45. M. (1993) Fundamentals of Statistical Signal Processing, Estimation Theory. Prentice-Hall, Englewood Cliffs, NJ. S. (1983) Speech Enhancement. Prentice Hall, Englewood Cliffs, NJ. , Salz J. J. (1968) Principles of Data Communications. McGraw-Hill, New York.
In this system a directional microphone takes as input the noisy signal x m + n m , and a second directional microphone, positioned some distance away, measures the noise n m + . The attenuation factor, , and the time delay, , provide a rather over-simplified model of the effects of propagation of the noise to different positions in the space where the microphones are placed. The noise from the second microphone is processed by an adaptive digital filter to make it equal to the noise contaminating the speech signal, and then subtracted from the noisy signal to cancel out the noise.