摘要:
This invention describes a method for generating noise references for adaptive interference cancellation filters for applications in generalized sidelobe canceling systems. More specifically the present invention relates to a multi-microphone beamforming system similar to a generalized sidelobe canceller (GSC) structure, but the difference with the GSC is that the present invention creates noise references (from 25-1) to the adaptive interference canceller (AIC) (21-1) filters using steerable beams (from 22) that block out the desired signal when the beam is steered away from the desired signal source location.
摘要:
This invention describes a method for generating noise references for adaptive interference cancellation filters for applications in generalized sidelobe canceling systems. More specifically the present invention relates to a multi-microphone beamforming system similar to a generalized sidelobe canceller (GSC) structure, but the difference with the GSC is that the present invention creates noise references (from 25-1) to the adaptive interference canceller (AIC) (21-1) filters using steerable beams (from 22) that block out the desired signal when the beam is steered away from the desired signal source location.
摘要:
A method for temporal adjustment of adaptation control of an adaptive interference canceller (AIC) 21 based on spatially weighted beamforming pre-processing. Spatial blocking performance is enhanced while generating noise references for the AIC 21-N by a beamformer, by introducing dynamic adjustment to the AIC filter adaptation control (46).
摘要:
A method and device for improving the quality of speech signals transmitted using an audio bandwidth between 300 Hz and 3.4 kHz. After the received speech signal is divided into frames, zeros are inserted between samples to double the sampling frequency. The level of these aliased frequency components is adjusted using an adaptive algorithm based on the classification of the speech frame. Sound can be classified into sibilants and non-sibilants, and a non-sibilant sound can be further classified into a voiced sound and a stop consonant. The adjustment is based on parameters, such as the number of zero-crossings and energy distribution, computed from the spectrum of the up-sampled speech signal between 300 Hz and 3.4kHz. A new sound with a bandwidth between 300 Hz and 7.7kHz is obtained by inverse Fourier transforming the spectrum of the adjusted, up-sampled sound.
摘要:
Focused error correction and/or focused error detection is used in the information coding system according to the invention. The purpose of the present invention is to present a speech encoding method, in which the number of speech parameter bits on which error correction coding and/or error detection coding focuses is automatically adjusted in relation to the number of total speech parameter bits as the function of the quality of the information transfer connection. In the information encoding system according to the invention there is no need to reduce the number of bits used for speech encoding, due to which the voice quality of the speech remains high. In the information encoding system according to the invention the error correction and/or error detection is focused on the bits most important for the voice quality e.g. as the function of the C/I (Channel to Interference) - parameter describing the quality of the information transfer connection. The muting of speech synthesizing occurring in prior known systems on poor information transfer connection is in the information encoding method according to the invention reduced by using focused error detection.