Adaptive signal processing
Digital filter design
Digital signal processing (3rd ed.): principles, algorithms, and applications
Digital signal processing (3rd ed.): principles, algorithms, and applications
Adaptive filter theory (3rd ed.)
Adaptive filter theory (3rd ed.)
Digital Signal Processing: A Computer Based Approach
Digital Signal Processing: A Computer Based Approach
Statistical Digital Signal Processing and Modeling
Statistical Digital Signal Processing and Modeling
Discrete Time Processing of Speech Signals
Discrete Time Processing of Speech Signals
Non-intrusive single-ended speech quality assessment in VoIP
Speech Communication
Computers in Biology and Medicine
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An increasing part of our daily personal communication takes place in various noisy environments. Ever since the broad introduction of cellular phones, we tend to communicate using these phones in cars, streets and other noisy places. Noise has a negative effect on both speech intelligibility and quality; a poor signal-to-noise ratio (SNR) may indeed result in a complete lack of speech intelligibility. This paper presents a speech enhancement method for personal communication, where the input signal is divided into a number of subbands that are individually and adaptively weighted in time domain according to a short term SNR estimate in each subband at every time instant. Hence the name adaptive gain equalizer. The signal disassembly into narrow subbands is performed using computationally effective infinite impulse response filters with low group delay. The method is focused on speech enhancement, acting as a speech booster, and remains idle when the SNR in a particular subband is low. Hence, background artifacts are eliminated. In addition, the method has proven to be advantageous since it offers low complexity and low delay. It is stand-alone and works regardless of speech coding schemes and other surrounding adaptive systems.