Abstract:
A method and device for identifying a human movement state that includes: determining according to acceleration signals that are collected by a three-axis acceleration sensor that a human is in a walking state, calculating a walking step number of the human, and calculating a walking step frequency according to the number; calculating a corresponding physical sign frequency during the walking process according to a physical sign signal that is collected; and comparing the walking step frequency and the physical sign frequency that are obtained by calculating respectively with a step frequency threshold and a physical sign frequency threshold, and if the walking step frequency is greater than the step frequency threshold, and the physical sign frequency is greater than the physical sign frequency threshold, determining that the human movement state is a running state, and recording the calculated walking step number to be a running step number.
Abstract:
A method and circuit for acquiring an output quantity of a linear resonance actuator are disclosed. The method comprises the steps of: establishing a circuit that simulates the linear resonance actuator by using passive electrical devices according to an electrical parameter and a kinematic parameter of the linear resonance actuator, the passive electrical devices comprise at least a resistor, a capacitor and an inductor; selecting a measuring point in the circuit according to an output quantity that the linear resonance actuator needs; and inputting a driving signal of an input source to an input end of the circuit, and collecting an electrical signal that is outputted at the measuring point to obtain the output quantity of the simulated linear resonance actuator. According to the technical solution of the application, a circuit module or system model that simulates the linear resonance actuator is established and used in the process of developing and debugging of projects, to replace the technical solutions that directly use physical actuators, thereby improving the efficiency and avoiding the relying on physical actuators.
Abstract:
A method and a system for achieving a self-adaptive surround sound. The method comprises: recognizing specific positions of a room and a user in the room by using an object recognition technology, capturing focusing images of recognized objects by controlling a camera using a focusing control technology, and recording corresponding focusing parameters (S110); calculating position information of the room relative to the camera and position information of the user relative to the camera according to the images and the parameters (S120); calculating sound beams that can achieve the surround sound at the position of the user in said room according to aforesaid calculated position information of the room and the user (S130); obtaining parameters of a filter group according to the calculated sound beams, and adjusting the filter group of a loudspeaker array according to the parameters (S140); and playing an audio signal via the loudspeaker array after the audio signal is filtered by the filter group that has been adjusted according to the parameters to form surround sound at the position of the user in the room (S150).
Abstract:
A method and circuit for acquiring an output quantity of a linear resonance actuator are disclosed. The method comprises the steps of: establishing a circuit that simulates the linear resonance actuator by using passive electrical devices according to an electrical parameter and a kinematic parameter of the linear resonance actuator, the passive electrical devices comprise at least a resistor, a capacitor and an inductor; selecting a measuring point in the circuit according to an output quantity that the linear resonance actuator needs; and inputting a driving signal of an input source to an input end of the circuit, and collecting an electrical signal that is outputted at the measuring point to obtain the output quantity of the simulated linear resonance actuator. According to the technical solution of the application, a circuit module or system model that simulates the linear resonance actuator is established and used in the process of developing and debugging of projects, to replace the technical solutions that directly use physical actuators, thereby improving the efficiency and avoiding the relying on physical actuators.
Abstract:
A method and a system for achieving a self-adaptive surround sound. The method comprises: recognizing specific positions of a room and a user in the room by using an object recognition technology, capturing focusing images of recognized objects by controlling a camera using a focusing control technology, and recording corresponding focusing parameters (S110); calculating position information of the room relative to the camera and position information of the user relative to the camera according to the images and the parameters (S120); calculating sound beams that can achieve the surround sound at the position of the user in said room according to aforesaid calculated position information of the room and the user (S130); obtaining parameters of a filter group according to the calculated sound beams, and adjusting the filter group of a loudspeaker array according to the parameters (S140); and playing an audio signal via the loudspeaker array after the audio signal is filtered by the filter group that has been adjusted according to the parameters to form surround sound at the position of the user in the room (S150).
Abstract:
The present invention discloses a speech enhancement method and device for mobile phones. By the method and device provided by the present invention, the mobile phone holding state of a user is detected when the user is talking on the phone, so that different denoising solutions will be employed according to the state of the user in holding the mobile phone. When the user holds the mobile phone normally, a solution integrating multi-microphone denoising and single-microphone denoising will be employed to effectively suppress both the steady noise and the non-steady noise; and when the user holds the mobile phone abnormally, a solution of single-microphone denoising will be employed only to suppress the steady noise. The distortion of speech by multi-microphone denoising is avoided, and the speech quality is ensured.
Abstract:
The present invention discloses a heart rate detection method used in an earphone and an earphone capable of detecting heart rate. The method comprises: providing a cavity inside the earphone, and installing a microphone in the cavity; providing an acceleration sensor in the earphone; performing self-adaptive filtering process on signals collected by the acceleration sensor, and obtaining estimated signals of the signals generated by body movement of a wearer in the signals collected by the microphone; subtracting the estimated signals from the signals collected by the microphone to obtain signals related to heart rate; and detecting heart rate according to the signals related to heart rate. The technical scheme of the invention adopts an enclosed cavity to place the microphone to reduce interference of external noises and reinforce signal information collected by the microphone. By performing self-adaptive filtering on signals collected by the acceleration sensor to obtain estimated signals, subtracting the estimated signals from the signals collected by the microphone, and then detecting the heart rate, the influence of the body movement of the wearer on heart rate detection can be eliminated.
Abstract:
The present invention relates to a method and device for dereverberation of single-channel speech. The method includes the following steps of framing an input single channel speech signal, and processing the frame signals as follows according to a time sequence: performing short-time Fourier transform on a current frame to obtain a power spectrum and a phase spectrum of the current frame; selecting several frames previous to the current frame and having a distance from the current frame within a set duration range, and performing linear superposition on the power spectra of these frames to estimate the power spectrum of a late reflection sound of the current frame; removing the estimated power spectrum of the late reflection sound of the current frame from the power spectrum of the current frame by a spectral subtraction method to obtain the power spectra of a direct sound and an early reflection sound of the current frame; and performing inverse short-time Fourier transform on the power spectra of the direct sound and the early reflection sound of the current frame and the phase spectrum of the current frame together to obtain a signal of the current frame after dereverberation. The dereverberation method and device can solve the problem that the estimation of a transfer function of a reverberation environment or the estimation of reverberation time is difficult in the dereverberation of single-channel speech.
Abstract:
Disclosed in the invention is a method and system for sampling rate mismatch correction of transmitting and receiving terminals, which can obtain a high-precision sampling rate mismatch in real time, carry out sampling rate correction on transmitting and receiving terminal signals, and send the transmitting terminal signal and the receiving terminal signal that have the same sampling rate after corrected to an echo cancellation system to carry out echo cancellation. The present invention can improve the quality of echo cancellation, simplify the computation and reduce the cost. The method for sampling rate mismatch correction of transmitting and receiving terminals provided in the embodiments of the invention comprises: calculating a transfer function of a receiving terminal signal relative to a transmitting terminal signal at each sampling timing according to the transmitting and receiving terminal signals; obtaining a transmission time delay of the transmitting and receiving terminals at each sampling timing using the transfer function; obtaining a sampling rate mismatch of the transmitting and receiving terminals at each sampling timing by means of parameter fitting using the transmission time delay and the linear relationship between the transmission time delay and the sampling rate mismatch; and adjusting the sampling rate of the transmitting terminal signal or the receiving terminal signal at each sampling timing according to the sampling rate mismatch.
Abstract:
The present invention relates to a method and device for dereverberation of single-channel speech. The method includes the following steps of framing an input single channel speech signal, and processing the frame signals as follows according to a time sequence: performing short-time Fourier transform on a current frame to obtain a power spectrum and a phase spectrum of the current frame; selecting several frames previous to the current frame and having a distance from the current frame within a set duration range, and performing linear superposition on the power spectra of these frames to estimate the power spectrum of a late reflection sound of the current frame; removing the estimated power spectrum of the late reflection sound of the current frame from the power spectrum of the current frame by a spectral subtraction method to obtain the power spectra of a direct sound and an early reflection sound of the current frame; and performing inverse short-time Fourier transform on the power spectra of the direct sound and the early reflection sound of the current frame and the phase spectrum of the current frame together to obtain a signal of the current frame after dereverberation. The dereverberation method and device can solve the problem that the estimation of a transfer function of a reverberation environment or the estimation of reverberation time is difficult in the dereverberation of single-channel speech.