Rainer Martin

Apparatus for improving disturbed speech signals

An apparatus for the transmission of speech signals reduces the effect of disturbances on the transmission quality. Two microphones are provided for picking up speech signals. Signal processing ensues in three partial frequency bands. The microphone signals are high-pass filtered in a lower frequency band, in the middle frequency band the signal is weighted with a scalar factor, so that this frequency band is damped during the speech pauses, and in the upper frequency band, an adaptive filter is used. At the beginning of the processing, a treble enhancement of the signals ensues, which is canceled by an inverse filter before the output of the improved signal.

A model-based auditory scene analysis approach and its application to speech source localization

Localization and classification of acoustic signals in a complex auditory scene is an every day task of the human auditory system. However, this problem presents a significant challenge for a computational system. In this paper, we propose a framework that allows the detection of different classes of signals (e.g. speech, music) and the localization of the signal sources. As a special application of this framework, we describe a method for localization of human speech in a complex acoustic scene and low SNR, based on vowel detection. In contrast to other works, we detect and extract parts of the signal mixture that correspond to a single class exclusively, using a prior model of the target signals. Herewith, a reliable localization of signal sources is possible.

Mobile radio terminal comprising a speech

A mobile radio terminal comprises a speech processor for processing a first and at least a further speech signal formed by noise and speech signal components and available as sample values. The sampled further speech signal is delayed by an adjustable delay value. Control means are provided which are used for forming gradient estimates. The control means are additionally used for recursively determining delay estimates from the gradient estimates. By rounding the delay estimates, the delay values are formed. Furthermore, their mutually time-shifted speech signals are added together by means of an adder device.