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This article proposes several improvements to an adaptive fingerprint enhancement method that is based on contextual filtering. The term adaptive implies that parameters of the method are automatically adjusted based on the input fingerprint image. Five processing blocks comprise the adaptive fingerprint enhancement method, where four of these blocks are(More)
— A real-time Digital Signal Processor (DSP) based implementation of a subband beamforming algorithm and its evaluation for dual microphone speech enhancement is presented. The algorithm, a calibrated constrained beamformer, is described theoretically and a real-time structure is proposed, including an efficient approach for multichannel data(More)
This contribution presents a fixed point implementation for acous-tical active noise control in hearing defenders. The well known filtered-x least mean squares structure is conformed to fixed point arithmetic and evaluated in real time. The measured performance of the implementation is 20dB to 30dB attenuation of broad band noise and ca 60dB for sinusoidal(More)
— This paper presents and evaluates a hybrid implementation of a low complexity algorithm for speech enhancement, the Adaptive Gain Equalizer (AGE). The AGE is a subband based time domain method for instantaneous boosting of speech. By combination of digital analysis and analog synthesis, main advantages of the digital domain and the analog domain are(More)
— Human speech is the main method for personal communication. However, interfering noise could degrade the intelligibility of speech, eventually resulting in errors. Thus, efficient speech enhancement algorithms are needed for example in hand held battery powered hearing aids. This paper presents an implementation of a time domain method for speech(More)
This paper presents a theoretical analysis of a certain criterion for complex-valued independent component analysis (ICA) with a focus on blind speech extraction (BSE) of a spatio-temporally nonstationary speech source. In the paper, the proposed criteria denoted KSICA is related to the well-known FastICA method with the Kurtosis contrast function. The(More)
The quality of speech, or important speech parameters such as the intelligibility, clearness or naturalness of speech, can be emphasized by signal processing. Such processing for improving speech quality can be found in telecommunication applications, e.g. mobile telephony, internet telephony or personal intercom. Blind methods are preferable over(More)
This paper presents a real time implementation of a blind beam-former for subband speech enhancement. The beamformer adap-tively maximizes the statistical kurtosis measure of the beam-former's output signal. Speech carries high kurtosis and noise often exhibit lower kurtosis. Hence, maximization of the output signal's kurtosis enhances speech, in general.(More)
— The general quality of speech or important speech parameters such as the intelligibility, clearness or naturalness of speech can be emphasized by signal processing. Such processing for improving speech quality can be found in telecommunication applications, e.g. mobile telephony, internet telephony or personal intercom. By careful selection of domain for(More)