32results about How to "Suppress ambient noise" patented technology

The invention discloses an absolute detection method for a surface shape error of a spherical surface. The absolute detection method comprises the steps of: establishing an equation system about Zernike multinomial coefficients of a measured spherical surface and a reference surface by utilizing measurement data of multiple rotation and concentric translation of the measured spherical surface at a con-focal position and adopting a Zernike multinomial fitting based rotation translation algorithm; and obtaining the Zernike multinomial coefficient by applying a least square method so as to obtain absolute surface shape information of the measured spherical surface and the reference surface. The absolute detection method is also available for the detection of the surface shape error of a plane. As a global optimized idea is adopted to resolve multinomial coefficients of the measured spherical surface and the reference surface at the same time, the absolute detection method for the surface shape error of the spherical surface can suppress system error and random noise for the better, and achieves stronger interference resistance and important application values.

The invention relates to an onsite classroom pickup and loudspeaking system adapted to the position of a speaker. The system comprises a pickup unit, a loudspeaker array and a processing circuit, wherein the pickup unit converts obtained audio signals of the speaker into electrical signal based on the position of the speaker, the processing circuit removes noises of the electrical signals and amplifies the de-noised signals to generate driving signals, and the loudspeaker array plays sounds of the speaker according to the driving signals. The system of the invention needs only one pickup unit, the speaker can walk in the classroom to teach class needless of wearing extra devices, local environmental noises can be effectively inhibited, and the problem of howling is eliminated.

The invention discloses a wave arrival estimation method based on the high-order accumulated amount, and belongs to the technical field of signal processing. According to the method, observation signals received by an equal-interval linear sensor array are used to estimate the direction of arrival and arrival time of a signal source. The method comprises the following steps that 1) spatial-time domain smoothing is carried out on the observation signals after Fourier Transform; 2) a four-order accumulated amount matrix of the signals after spatial-time domain smoothing is constructed; 3) an iterative part SVD method is used to construct a signal subspace and a noise subspace of the observation signals according to the four-order accumulated amount matrix; and 4) according to the orthogonality between the signal subspace and the noise subspace of the observation signals, the direction of arrival and arrival time of the signal source are estimated. The invention also discloses an ocean acoustic tomography method based on sound ray spreading time tomography and a positioning method. According to the invention, the computing complexity of an algorithm can be greatly reduced, the instantaneity of the algorithm is improved, and consumption of hardware resource is reduced.

The invention discloses a high-order cumulant-based method for the multi-path propagation and separation of acoustic signals. According to the technical scheme of the invention, an original second-order cumulant-based active broadband signal separation method is extended to a fourth-order cumulant. According to the fourth-order cumulant, a corresponding duplicate vector and noise sub-space is constructed. Meanwhile, an additive noise is assumed to be a colored noise during the acoustic signal propagation process. Finally, an acoustic ray path is found out based on the orthogonality of the noise sub-space. The invention also discloses an acoustic ray propagation time tomography-based ocean acoustic tomography method and a sound source localization method. Compared with the prior art, the above method is higher in separation accuracy and can correctly separate acoustic ray paths of smaller intervals. Meanwhile, the method can be applied to conditions wherein the number of sensors is smaller than the number of ray paths. At the same time, the influence of both Gaussian noises and non-Gaussian noises can be inhibited.

The embodiment of the invention provides a voice signal enhancement processing method and an earphone. The method comprises the following steps of: acquiring a first voice signal and a second voice signal, wherein the first voice signal and the second voice signal are voice signals acquired based on a first position and a second position respectively, and the first position and the second positionare taken as an outside-ear voice signal acquisition point and an inside-ear voice signal acquisition point respectively; at least performing echo cancellation processing on the second voice signal to obtain a third voice signal; and fusing the third voice signal and the first voice signal to obtain a fourth voice signal, and outputting the fourth voice signal. The earphone comprises a first microphone, a second microphone and an audio processor. Through adoption of the voice signal enhancement processing method and the earphone, the voice of a wearer can be effectively picked up and other noises can be suppressed in complex acoustic scenes such as low signal-to-noise ratio and multi-speaker interference. Moreover, the earphone is small in size, is lighter, and is lower in production andmanufacturing costs.

The invention discloses a voice enhancement method and system and a communication device. The method comprises the following steps: acquiring voice input signals; performing time-frequency compensation on the voice input signals; calculating multiple pieces of voice characteristic information subjected to time-frequency compensation, and classifying the voice input signals to obtain a voice classification result; determining a voice masking value according to the multiple pieces of voice characteristic information and the voice classification result; and determining a voice output signal basedon the voice masking value. The method solves the technical problem of poor communication effect due to the fact that large noise is left or sound of a target object is damaged because a fixed masking value is adopted in a voice enhancement mode in related technologies.

Disclosed are a smart headphone device personalization system and a method for using the system. A user downloads a headphone personalization application program unit from the server device for installation in a mobile device. The Mobile device and the smart headphone device are paired. The user enters corresponding data through the register module of the mobile device. The user wears the smart headphone device to allow a hearing inspection module of the mobile device to inspect hearing capabilities of left and right ears of the user and generate an inspection result. The automatic compensation unit of the smart headphone device may automatically carry out compensation according to the inspection result. Then, the manual compensation module of the mobile device further carries out compensation. Finally, the user adjusts the sound effect mode by using the preference setting module of the mobile device to complete personalization of the smart headphone device.

The invention discloses a seismic data event detection method of an environment suppression and contour combination model based on visual cognition. The method comprises steps that an event is picked up based on pre-stack seismic data which contains abundant information such as amplitude and frequency, and can reflect a tiny structure of a stratum; firstly, seismic data are preprocessed, wavelet fusion of envelope peak instantaneous frequency and oblique superposition peak instantaneous amplitude of the seismic data is solved, and optimal mass data can be provided to the maximum extent. While the seismic data event contour is enhanced, anisotropy suppression is carried out on environment non-correlation noise, significant extraction of the event target contour is realized, and preparation is made for subsequent seismic interpretation, reservoir prediction and the like. Compared with an isotropic method, the anisotropic environment suppression method adopted by the invention can more accurately detect edge variation.

A method and a device for improving voice quality on transparent telecommunications-transmission paths. The audio-transmission channel is adapted to the respective audio-spectrum of the speaker at predetermined intervals. As a result, means for analyzing the voice signals in relation to the useful band width and amplitude thereof are provided in addition to means for influencing the band width and amplitude of the audio-transmission channel according to the determined useful band and amplitude. Such means can be installed in the communication terminal as well as in the communication network.

The present disclosure provides a method of processing an audio signal, a user terminal, an electronic device, a computer readable storage medium, and a computer program product. For example, intelligent noise reduction can be carried out on the recording (especially the recording containing singing), whether the current recording frame is used for recording human voice or not is intelligently determined, and different noise reduction schemes are adaptively applied to different segments in the recording based on the determined result; therefore, the human voice signal in the recording is reserved as much as possible while the environmental noise is suppressed.

The invention provides a music playing method. The method comprises the following steps: a voice receiving device is separately arranged with an audio host, the voice receiving device such as a microphone receives near field voice control information of a user, the near field voice control information is used for ensuring the clear recording of the voice control information of the user, the far-field recording of sound is avoided, and the near field voice control information is transmitted to the remote audio host to indicate the audio host to play music. The problem of poor voice control effect during the use of AI sound box in related art is solved, and the clear recording of the voice control information of the user is ensured to achieve precise control.

The invention relates to a self-adaptive amplifying and filtering system for laser energy measurement of a back calorimeter, which comprises a calorimetric detector, an impedance matching device, a front programmable gain amplifier, a programmable switch capacitance low-pass filter, a post programmable gain amplifier, a signal acquisition and control module and a gain and bandwidth control module, wherein the signal acquisition and control module analyzes and processes an acquired signal by adopting a self-adaptive frequency spectrum analysis method and transmits processing results to the gain and bandwidth control module; and the gain and bandwidth control module downloads the processing results to the front programmable gain amplifier, the post programmable gain amplifier and the programmable switch capacitance low-pass filter. The self-adaptive amplifying and filtering system has the benefits that the gain of all amplifiers and the bandwidth of the filter are realized through online adjustment, the use is convenient, and the signal to noise ratio of an output signal is improved while the environmental noise is effectively restrained.