Benny Sällberg

Learn More
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)
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)
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 transformation.(More)
This contribution presents a fixed point implementation for acoustical 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)
In many applications where speech separation and enhancement is of interest, e.g. conferencing systems, mobile phones and hearing aids, accurate speaker localization is important. This paper presents an alternative criteria for the well known Steered Response Power with Phase Transform (SRPPHAT) algorithm, in which the steered response relates to peaks in(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)