Results 41 to 50 of about 217 (166)
We propose an alternative approach to the Balanced Model Truncation method (standard method). This approach reduces substantially the order of minimum‐phase inverse filters for equalizing room acoustics. This method is based on a property of the filter z transform function, which modifies the corresponding FIR coefficients before the application of the
Maamar Ahfir +4 more
wiley +1 more source
Dereverberation by Using Time-Variant Nature of Speech Production System
This paper addresses the problem of blind speech dereverberation by inverse filtering of a room acoustic system. Since a speech signal can be modeled as being generated by a speech production system driven by an innovations process, a reverberant signal
Yoshioka Takuya +2 more
doaj +2 more sources
De-Noising Process in Room Impulse Response with Generalized Spectral Subtraction
The generalized spectral subtraction algorithm (GBSS), which has extraordinary ability in background noise reduction, is historically one of the first approaches used for speech enhancement and dereverberation. However, the algorithm has not been applied
Min Chen, Chang-Myung Lee
doaj +1 more source
Reverberant environment embedding using dereverberation autoencoder
We design a neural network architecture for dereverberation, combining linear prediction and dereverberation autoencoder. To improve reverberant speech recognition performance with low computational complexity, we propose a method to extract environmental embedding named DA‐embedding instead of applying dereverberation to the input of an acoustic model.
Sunchan Park, Hyung Soon Kim
wiley +1 more source
Model-Informed Speech Enhancement Using Virtual Room Acoustics and Acoustic Descriptor Optimization
Reverberation and background noise remain persistent obstacles to achieving clear and intelligible speech in enclosed environments. Conventional data-driven or purely empirical dereverberation systems often perform well only under training conditions but
Samuel Yaw Mensah +3 more
doaj +1 more source
Active noise control (ANC) algorithms have been developed within the adaptive algorithm framework. However, multichannel ANC systems, which include numerous reference sensors, control speakers, and error microphones, require a very long control filter converging time for control filter estimation.
Hakjun Lee +2 more
wiley +1 more source
Abstract Large continental transform faults are thought to cross‐cut the crust and extend into the lithospheric mantle. However, the strain distribution associated with these faults at mantle depths is not well understood. At the heart of this question is the rheology of the lithosphere: when stressed by displacement on a trans‐crustal fault, does the ...
Hannah F. Mark
wiley +1 more source
Implementation of Hybrid Speech Dereverberation Systems and Proposing Dual Microphone Farsi Database in Order to Evaluating Enhancement Systems [PDF]
In various applications, such as speech recognition and automatic teleconferencing, the recorded speech signals may be corrupted by both noise and reverberation. Reverberation causes a noticeable change in speech intelligibility and quality.
Farhad Faghani, Hamid Reza Abutalebi
doaj
Two-Microphone Dereverberation for Automatic Speech Recognition of Polish
Reverberation is a common problem for many speech technologies, such as automatic speech recognition (ASR) systems. This paper investigates the novel combination of precedence, binaural and statistical independence cues for enhancing reverberant speech ...
Mikolaj KUNDEGORSKI +2 more
doaj +1 more source
Time‐lapse applications of the Marchenko method on the Troll field
Abstract The data‐driven Marchenko method is able to redatum wavefields to arbitrary locations in the subsurface, and can, therefore, be used to isolate zones of specific interest. This creates a new reflection response of the target zone without interference from over‐ or underburden reflectors.
Johno van IJsseldijk +3 more
wiley +1 more source

