Abstract: A lanyard device, comprising a main body portion, an elastic support, a colloid, a rotary pull head and a lanyard.

Abstract: The present application discloses an electronic equipment support frame, which comprises a protective cover and a supporting structure, wherein the protective cover comprises a main plate and a side plate bent along the edge of the main plate, an accommodating space is enclosed by the main plate and the side plate for accommodating the electronic equipment, and a groove is formed on the side of the main plate facing away from the accommodating space; the supporting structure is disposed in the groove and comprises a first plate and a second plate, one end of the first plate is pivotally connected with the main plate, and the other end thereof is pivotally connected with the second plate. The electronic equipment support frame provided according to the present application can provide multi-angle support, and it has a higher adjustable range of supporting height and occupies a small area.

Abstract: An audio signal processing method includes: acquiring audio signals from at least two sound sources respectively through at least two microphones (MICs) to obtain respective original noisy signals of the at least two MICs in a time domain; for each frame in the time domain, using a first asymmetric window to perform a windowing operation on the respective original noisy signals of the at least two MICs to acquire windowed noisy signals; performing time-frequency conversion on the windowed noisy signals to acquire respective frequency-domain noisy signals of the at least two sound sources; acquiring frequency-domain estimated signals of the at least two sound sources according to the frequency-domain noisy signals; and obtaining audio signals produced respectively by the at least two sound sources according to the frequency-domain estimated signals.

Abstract: An original noisy signal of each of at least two microphones is acquired by acquiring, using the at least two microphones, an audio signal emitted by each sound source. For each frame in time domain, an estimated frequency-domain signal of each sound source is acquired according to the original noisy signal of each of the at least two microphones. A frequency collection containing a plurality of predetermined static frequencies and dynamic frequencies is determined in a predetermined frequency band range. A weighting coefficient of each frequency contained in the frequency collection is determined according to the estimated frequency-domain signal of the each frequency in the frequency collection. A separation matrix of the each frequency is determined according to the weighting coefficient. The audio signal emitted by each of the at least two sound sources is acquired based on the separation matrix and the original noisy signal.

Abstract: A method for responding to a voice signal includes: receiving a voice signal by the plurality of electronic devices; determining, for each of the electronic devices, a target sound source point associated with the voice signal; determining, for each of the electronic devices, a direct sound energy associated with the voice signal according to the target sound source point; and selecting from the plurality of electronic devices at least one electronic device with a direct sound energy satisfying a predetermined condition, and responding to the voice signal by the at least one electronic device. A sound pickup decision system based on a plurality of electronic devices, can make more accurate, more reasonable, and more targeted response to a user's voice, improving voice interaction experience.

Abstract: A method for processing an audio signal is provided. In the method, audio signals sent by at least two sound sources are acquired by at least two microphones to obtain multiple frames of original noisy signals of each microphone on a time domain. For each frame, frequency-domain estimation signals of each sound source are acquired according to the original noisy signals of the at least two microphones. For each sound source, the frequency-domain estimation signals are divided into multiple frequency-domain estimation components on a frequency domain. For each sound source, feature decomposition is performed on a related matrix of each frequency-domain estimation component to obtain a target feature vector. A separation matrix of each frequency point is obtained based on target feature vectors and the frequency-domain estimation signals. The audio signals of sounds are obtained based on the separation matrixes and the original noisy signals.

Abstract: An original noisy signal of each of at least two microphones is acquired by acquiring, using the at least two microphones, an audio signal emitted by each sound source. For each frame in time domain, an estimated frequency-domain signal of each sound source is acquired according to the original noisy signal of each of the at least two microphones. A frequency collection containing a plurality of predetermined static frequencies and dynamic frequencies is determined in a predetermined frequency band range. A weighting coefficient of each frequency contained in the frequency collection is determined according to the estimated frequency-domain signal of the each frequency in the frequency collection. A separation matrix of the each frequency is determined according to the weighting coefficient. The audio signal emitted by each of the at least two sound sources is acquired based on the separation matrix and the original noisy signal.

Abstract: Provided are an audio signal processing method and device. The method can include acquiring audio signals from at least two sound sources through at least two microphones to obtain multiple frames of original noise signals of the at least two microphones in a time domain, and, for each frame in the time domain, acquiring respective frequency-domain estimated signals of the at least two sound sources according to the respective original noise signals. The method can further include, for each sound source, dividing the frequency-domain estimated signal into frequency-domain estimated components which each corresponds to a frequency-domain sub-band and includes multiple frequency point data in a frequency domain, determining a weighting coefficient of each frequency point in the frequency-domain sub-band, and updating a separation matrix of each frequency point according to the weighting coefficient and obtaining the audio signals based on the updated separation matrices and the original noise signals.

Abstract: A method for processing audio signal includes that: audio signals emitted respectively from at least two sound sources are acquired through at least two microphones to obtain respective original noisy signals of the at least two microphones; sound source separation is performed on the respective original noisy signals of the at least two microphones to obtain respective time-frequency estimated signals of the at least two sound sources; a mask value of the time-frequency estimated signal of each sound source in the original noisy signal of each microphone is determined based on the respective time-frequency estimated signals; the respective time-frequency estimated signals of the at least two sound sources are updated based on the respective original noisy signals of the at least two microphones and the mask values; and the audio signals emitted respectively from the at least two sound sources are determined.

Abstract: An audio signal processing method includes: acquiring audio signals from at least two sound sources respectively through at least two microphones (MICs) to obtain respective original noisy signals of the at least two MICs in a time domain; for each frame in the time domain, using a first asymmetric window to perform a windowing operation on the respective original noisy signals of the at least two MICs to acquire windowed noisy signals; performing time-frequency conversion on the windowed noisy signals to acquire respective frequency-domain noisy signals of the at least two sound sources; acquiring frequency-domain estimated signals of the at least two sound sources according to the frequency-domain noisy signals; and obtaining audio signals produced respectively by the at least two sound sources according to the frequency-domain estimated signals.

Abstract: A method for processing an audio signal is provided. In the method, audio signals sent by at least two sound sources are acquired by at least two microphones to obtain multiple frames of original noisy signals of each microphone on a time domain. For each frame, frequency-domain estimation signals of each sound source are acquired according to the original noisy signals of the at least two microphones. For each sound source, the frequency-domain estimation signals are divided into multiple frequency-domain estimation components on a frequency domain. For each sound source, feature decomposition is performed on a related matrix of each frequency-domain estimation component to obtain a target feature vector. A separation matrix of each frequency point is obtained based on target feature vectors and the frequency-domain estimation signals. The audio signals of sounds are obtained based on the separation matrixes and the original noisy signals.

Abstract: Provided are an audio signal processing method and device. The method can include acquiring audio signals from at least two sound sources through at least two microphones to obtain multiple frames of original noise signals of the at least two microphones in a time domain, and, for each frame in the time domain, acquiring respective frequency-domain estimated signals of the at least two sound sources according to the respective original noise signals. The method can further include, for each sound source, dividing the frequency-domain estimated signal into frequency-domain estimated components which each corresponds to a frequency-domain sub-band and includes multiple frequency point data in a frequency domain, determining a weighting coefficient of each frequency point in the frequency-domain sub-band, and updating a separation matrix of each frequency point according to the weighting coefficient and obtaining the audio signals based on the updated separation matrices and the original noise signals.

Abstract: A method for processing audio signal includes that: audio signals emitted respectively from at least two sound sources are acquired through at least two microphones to obtain respective original noisy signals of the at least two microphones; sound source separation is performed on the respective original noisy signals of the at least two microphones to obtain respective time-frequency estimated signals of the at least two sound sources; a mask value of the time-frequency estimated signal of each sound source in the original noisy signal of each microphone is determined based on the respective time-frequency estimated signals; the respective time-frequency estimated signals of the at least two sound sources are updated based on the respective original noisy signals of the at least two microphones and the mask values; and the audio signals emitted respectively from the at least two sound sources are determined.

Abstract: A method for sound collection includes: converting time domain signals with a number of M collected by devices for sound collecting with a number of M into original frequency domain signals with a number of M; performing beam-forming on the M original frequency domain signals at each of preset grid points, to obtain beam-forming frequency domain signals with a number of N in one-to-one correspondence with the preset grid points; determining an average amplitude of frequency components with a number of N corresponding to each of frequency points with a number of K based on the beam-forming frequency domain signals with a number of N, and synthesizing a synthesized frequency domain signal including the frequency points and having an average amplitude as an amplitude at the each of the frequency points with a number of K; and converting the synthesized frequency domain signal into a synthesized time domain signal.

Abstract: A method for responding to a voice signal includes: receiving a voice signal by the plurality of electronic devices; determining, for each of the electronic devices, a target sound source point associated with the voice signal; determining, for each of the electronic devices, a direct sound energy associated with the voice signal according to the target sound source point; and selecting from the plurality of electronic devices at least one electronic device with a direct sound energy satisfying a predetermined condition, and responding to the voice signal by the at least one electronic device. A sound pickup decision system based on a plurality of electronic devices, can make more accurate, more reasonable, and more targeted response to a user's voice, improving voice interaction experience.

Abstract: A method for sound collection includes: converting time domain signals with a number of M collected by devices for sound collecting with a number of M into original frequency domain signals with a number of M; performing beam-forming on the M original frequency domain signals at each of preset grid points, to obtain beam-forming frequency domain signals with a number of N in one-to-one correspondence with the preset grid points; determining an average amplitude of frequency components with a number of N corresponding to each of frequency points with a number of K based on the beam-forming frequency domain signals with a number of N, and synthesizing a synthesized frequency domain signal including the frequency points and having an average amplitude as an amplitude at the each of the frequency points with a number of K; and converting the synthesized frequency domain signal into a synthesized time domain signal.

Abstract: An audio signal noise estimation method includes: for multiple preset sampling points, a noise Steered Response Power (SRP) value of a Microphone (MIC) array at each preset sampling point within a preset noise sampling period is determined to obtain a noise SRP multidimensional vector including the multiple noise SRP values corresponding to the multiple preset sampling points; a present frame SRP value for a present frame of an audio signal acquired by the MIC array at each preset sampling point is determined to obtain a present frame SRP multidimensional vector including the multiple present frame SRP values corresponding to the multiple preset sampling points; and whether the audio signal acquired by the MIC array in the present frame is a noise signal is determined according to the present frame SRP multidimensional vector and the noise SRP multidimensional vector.