Nothing Special   »   [go: up one dir, main page]

Bendoumia et al., 2023 - Google Patents

Recursive adaptive filtering algorithms for sparse channel identification and acoustic noise reduction

Bendoumia et al., 2023

Document ID
14871908566891585145
Author
Bendoumia R
Hassani I
Guessoum A
Publication year
Publication venue
Analog Integrated Circuits and Signal Processing

External Links

Snippet

This study addresses impulse response identification and acoustic noise cancellation problems using recursive blind source separation (BSS) techniques based sparse adaptive filtering algorithms. The two-channel adaptive filtering feedback algorithms have been …
Continue reading at link.springer.com (other versions)

Classifications

    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10L—SPEECH ANALYSIS OR SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING; SPEECH OR AUDIO CODING OR DECODING
    • G10L21/00—Processing of the speech or voice signal to produce another audible or non-audible signal, e.g. visual or tactile, in order to modify its quality or its intelligibility
    • G10L21/02—Speech enhancement, e.g. noise reduction or echo cancellation
    • G10L21/0208—Noise filtering
    • G10L21/0216—Noise filtering characterised by the method used for estimating noise
    • G10L2021/02161—Number of inputs available containing the signal or the noise to be suppressed
    • G10L2021/02166—Microphone arrays; Beamforming
    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10L—SPEECH ANALYSIS OR SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING; SPEECH OR AUDIO CODING OR DECODING
    • G10L15/00—Speech recognition
    • G10L15/08—Speech classification or search
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04M—TELEPHONIC COMMUNICATION
    • H04M9/00—Interconnection arrangements not involving centralised switching
    • H04M9/08—Two-way loud-speaking telephone systems with means for suppressing echoes or otherwise conditioning for one or other direction of traffic
    • H04M9/082—Two-way loud-speaking telephone systems with means for suppressing echoes or otherwise conditioning for one or other direction of traffic using echo cancellers
    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10L—SPEECH ANALYSIS OR SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING; SPEECH OR AUDIO CODING OR DECODING
    • G10L19/00—Speech or audio signal analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signal, using source filter models or psychoacoustic analysis
    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10K—SOUND-PRODUCING DEVICES; ACOUSTICS NOT OTHERWISE PROVIDED FOR
    • G10K11/00—Methods or devices for transmitting, conducting or directing sound in general; Methods or devices for protecting against, or for damping, noise or other acoustic waves in general
    • G10K11/16—Methods or devices for protecting against, or damping of, acoustic waves, e.g. sound
    • G10K11/175—Methods or devices for protecting against, or damping of, acoustic waves, e.g. sound using interference effects; Masking sound
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04B—TRANSMISSION
    • H04B3/00—Line transmission systems
    • H04B3/02—Details
    • H04B3/20—Reducing echo effects or singing; Opening or closing transmitting path; Conditioning for transmission in one direction or the other
    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10L—SPEECH ANALYSIS OR SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING; SPEECH OR AUDIO CODING OR DECODING
    • G10L25/00—Speech or voice analysis techniques not restricted to a single one of groups G10L15/00-G10L21/00
    • H—ELECTRICITY
    • H03—BASIC ELECTRONIC CIRCUITRY
    • H03H—IMPEDANCE NETWORKS, e.g. RESONANT CIRCUITS; RESONATORS
    • H03H21/00—Adaptive networks
    • H03H21/0012—Digital adaptive filters

Similar Documents

Publication Publication Date Title
Xiao et al. Speech dereverberation for enhancement and recognition using dynamic features constrained deep neural networks and feature adaptation
Bendoumia et al. Two-channel variable-step-size forward-and-backward adaptive algorithms for acoustic noise reduction and speech enhancement
Park et al. Acoustic interference cancellation for a voice-driven interface in smart TVs
Cheng et al. Deep learning-based stereophonic acoustic echo suppression without decorrelation
Djendi et al. A new efficient two-channel backward algorithm for speech intelligibility enhancement: A subband approach
Gupta et al. Analysis of optimized spectral subtraction method for single channel speech enhancement
Djendi et al. New automatic forward and backward blind sources separation algorithms for noise reduction and speech enhancement
Cohen et al. An online algorithm for echo cancellation, dereverberation and noise reduction based on a Kalman-EM Method
Bendoumia et al. Recursive adaptive filtering algorithms for sparse channel identification and acoustic noise reduction
Bendoumia Two-channel forward NLMS algorithm combined with simple variable step-sizes for speech quality enhancement
Sayoud et al. A new speech enhancement adaptive algorithm based on fullband–subband MSE switching
Djendi et al. A new dual subband fast NLMS adaptive filtering algorithm for blind speech quality enhancement and acoustic noise reduction
Darazirar et al. A two-sensor Gauss–Seidel fast affine projection algorithm for speech enhancement and acoustic noise reduction
Bendoumia et al. Acoustic noise reduction by new two-channel proportionate forward symmetric adaptive decorrelating algorithms in sparse systems
Djendi et al. A new adaptive solution based on joint acoustic noise and echo cancellation for hands-free systems
US11902757B2 (en) Techniques for unified acoustic echo suppression using a recurrent neural network
Kamarudin et al. Acoustic echo cancellation using adaptive filtering algorithms for Quranic accents (Qiraat) identification
Park et al. Integrated acoustic echo and background noise suppression technique based on soft decision
Zoulikha et al. A new robust forward BSS adaptive algorithm based on automatic voice activity detector for speech quality enhancement
Bendoumia et al. New simplified sub-band sparse adaptive algorithm for speech enhancement in communication systems
Tedjani et al. A novel cost-effective sparsity-aware algorithm with Kalman-based gain for the identification of long acoustic impulse responses
KR101537653B1 (en) Method and system for noise reduction based on spectral and temporal correlations
Özen Acarbay et al. Performance analysis of the speech enhancement application with wavelet transform domain adaptive filters
Sundaradhas et al. Upgraded NLMS algorithm for speech enhancement with sparse and dispersive impulse responses
Bendoumia New two-microphone simplified sub-band forward algorithm based on separated variable step-sizes for acoustic noise reduction