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415results about How to "Improved noise suppression" patented technology

Method for enhancing microphone array voice based on combined inhibition

The invention provides a method for enhancing microphone array voice based on combined inhibition. The method comprises the following steps of: structuring a microphone array for receiving external signals; analyzing the signals and obtaining time delays of different array signals relative to benchmark array signals in the microphone arrays opposite to a target voice source; respectively performing time delay compensation on digital signals corresponding to the two microphones, obtaining the compensated signals; respectively performing subband decomposition on the compensated array signals, and then forming fixed beams on each subband; meanwhile, respectively using blocking matrixes on each subband to obtain noise reference signals on each subband; and then respectively removing the noiseirrelative to the target voice form the fixed beam forming device on corresponding subband through an adaptive filtering processing algorithm, and then merging the subbands, thereby forming an initial gain signal; and meanwhile, making use of the previously compensated any two array signals to obtain a filter for inhibiting the noise signal related to the target voice through a recursive mutual power spectral density, thereby obtaining the final target voice signal through combining the initial gain signal.
Owner:ZHEJIANG UNIV

Dual-microphone-based speech enhancement device and speech enhancement method

The invention relates to a dual-microphone-based speech enhancement device. The device comprises a digital microphone array module, a first pulse density modulation decoder module, a directivity formation module, a second pulse density modulation decoder module, a speech enhancement processing module and an output module, wherein the digital microphone array module is used for receiving a time domain noisy speech digital signal; the first pulse density modulation decoder module is used for performing frame separation, down-sampling and filtering on the time domain noisy speech digital signal to form an initial down-sampling rate digital signal; the directivity formation module is used for performing directive formation on the initial down-sampling rate digital signal to form an initially noise-reduced directive output signal; the second pulse density modulation decoder module is used for performing the frame separation, the down-sampling, the filtering and accentuation on the initially noise-reduced directive output signal to form a second down-sampling rate digital signal; the speech enhancement processing module is used for performing speech enhancement processing on the second down-sampling rate digital signal; and the output module is used for outputting a background noise-removed speech signal. The invention also provides a dual-microphone speech enhancement device-based speech enhancement method. The device and the method have a good denoising effect.
Owner:AAC TECH PTE LTD +1

Deep learning speech extraction and noise reduction method fusing bone vibration sensor and microphone signals

The invention relates to a deep learning noise reduction method fusing bone vibration sensor and microphone signals. The deep learning noise reduction method comprises the following steps: enabling abone vibration sensor and a microphone to collect audio signals, and respectively obtaining a bone vibration sensor audio signal and a microphone audio signal; inputting an audio signal of the bone vibration sensor into a high-pass filtering module, and carrying out high-pass filtering; inputting the bone vibration sensor audio signal subjected to high-pass filtering or the signal subjected to frequency band widening and a microphone audio signal into a deep neural network module; and obtaining the voice after noise reduction through prediction by the deep neural network module. According to the deep learning noise reduction method fusing bone vibration sensor and microphone signals, signals of a bone vibration sensor and a traditional microphone are combined, and the strong modeling capability of the deep neural network is utilized to realize high human voice restoration degree and extremely strong noise suppression capability, and the problem of human voice extraction in a complex noise scene can be solved, and the target human voice is extracted, and the interference noise is reduced, and a single-microphone structure can be adopted to reduce the cost. In addition, the signals obtained after the frequency band of the bone vibration sensor audio signals is broadened can be directly used as output.
Owner:ELEVOC TECH CO LTD

Restraining device and restraining method for burst impulse noise of power line communication channel

The invention discloses a restraining device and restraining method for burst impulse noise of a power line communication channel. The restraining device comprises a Fourier transformation module, a masking subcarrier selection module and a compressed sensing estimation module. The restraining method for the burst impulse noise of the power line communication channel is based on compressed sensing and comprises the steps of carrying out Fourier transformation on a received signal to obtain a frequency domain signal, carrying out masking subcarrier selection to obtain a subcarrier only comprising background noise and the burst impulse noise, estimating the burst impulse noise in the power line communication channel by using a signal recover method based on compressed sensing, and finally, removing the impulse noise from the received signal to obtain a signal with noise removed. Due to the facts that the burst impulse noise in the power line communication channel has the characteristic of sparsity, and the OFDM masking subcarrier selection and compressed sensing technologies are applied, the effect of restraining the noise can be remarkably improved, and signal information of positions interfered by burst impulse can be effectively kept.
Owner:DALIAN UNIV OF TECH

Adaptive voice separating method based on sound source positioning

The invention provides an adaptive voice separating method based on sound source positioning, and relates to the technical field of information processing. The method includes steps: acquiring an audio signal of an observed environment, and confirming the number of sound sources and the direction of arrival of each sound source; generating a dimension reduction matrix P; generating a voice transfer matrix and a delay superposed wave beam coefficient; determining an active sound source of a frequency point and separating voice components; obtaining the obtained voice components and setting non-activated sound source components as zero; and obtaining time domain voice signals of the sound sources. According to the method, the number and the orientation of the sound sources in a current environment can be obtained through a sound source positioning technology, dimension reduction of each frequency band of the voice signal is performed with the cooperation of a PCA whitening technology toobtain an initial separation matrix, frequency components of each sound source channel are separated through the number of the activated sound sources at the frequency point by adaptive usage of the beam forming technology and the FDICA technology to restore the voice components, the obtained signal-to-noise ratio improvement characteristic is higher, better noise suppression performance is achieved, and the method is applicable to any sound source situations in the real voice environment.
Owner:NORTHEASTERN UNIV
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