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 an echo elimination device and method for a miniature hands-free voice communication system. The system comprises a receiver, a primary transmitter and an auxiliary transmitter, a distance from the primary transmitter to the receiver being greater than that from the auxiliary transmitter to the receiver. The device comprises an array echo elimination unit, a self-adaptive echo elimination unit and a residual echo elimination unit, which are structurally cascaded in turn. The array echo elimination unit, with inputs being a signal of the primary transmitter and a signal of the auxiliary transmitter, performs array filtering to obtain one path of output signals; the self-adaptive echo elimination unit, with the input signals being a signal of the receiver, the output signal of the array echo elimination unit and a signal of the auxiliary transmitter, performs self-adaptive filtering to obtain two paths of output signals; the residual echo elimination unit, with the input signals being the two paths of output signals of the self-adaptive echo elimination unit, performs voice probability estimation and echo matching to obtain an echo-eliminated voice signal. Thus, the duplex performance can be enhanced, and the phase consistency of the transmitters is not strictly required.
Abstract:
The invention discloses a method and a device for reducing voice reverberation based on double microphones. The method comprises the steps of calculating a transfer function h(t) from a secondary microphone to a primary microphone according to an input signal x2(t) of the primary microphone and an input signal x1(t) of the secondary microphone; judging the strength of reverberation according to h(t) and calculating a regulatory factor β of a gain function by taking a tail section hr(t) of the h(t); obtaining a late reverberation estimation signal {circumflex over (r)}(t) of x2(t) with the convolution of x1(t) and hr(t); calculating the gain function according to the frequency spectrum of x2(t), β and frequency spectrum of {circumflex over (r)}(t); obtaining the reverberation removed frequency spectrum of x2(t) by multiplying the frequency spectrum of x2(t) by the gain function; and obtaining a late reverberation removed time-domain signal of x2(t) by frequency-time conversion. Thus, the late reverberation can be removed from the input signal of the primary microphone, early reverberation can be preserved, processed voice is not caused to be thin, and the voice quality is improved. Meanwhile, spectral subtraction intensity is adjusted according to the strength of the reverberation so as to ensure that the voice is not damaged on the condition that the reverberation is weak and the voice intelligibility is originally high. Accurate estimation of DOA of direct sound is not needed, and therefore the microphones are not required to have high consistency.
Abstract:
The present invention discloses an echo elimination device and method for a miniature hands-free voice communication system. The system comprises a receiver, a primary transmitter and an auxiliary transmitter, a distance from the primary transmitter to the receiver being greater than that from the auxiliary transmitter to the receiver. The device comprises an array echo elimination unit, a self-adaptive echo elimination unit and a residual echo elimination unit, which are structurally cascaded in turn. The array echo elimination unit, with inputs being a signal of the primary transmitter and a signal of the auxiliary transmitter, performs array filtering to obtain one path of output signals; the self-adaptive echo elimination unit, with the input signals being a signal of the receiver, the output signal of the array echo elimination unit and a signal of the auxiliary transmitter, performs self-adaptive filtering to obtain two paths of output signals; the residual echo elimination unit, with the input signals being the two paths of output signals of the self-adaptive echo elimination unit, performs voice probability estimation and echo matching to obtain an echo-eliminated voice signal. Thus, the duplex performance can be enhanced, and the phase consistency of the transmitters is not strictly required.
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:
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 provides a speech enhancing method for communication earphone including two parts: sending end noise reduction processing and receiving end noise reduction processing, wherein the sending end noise reduction processing part includes: determining a wearing condition of the earphone by comparing energy difference of sound signals picked up by microphones of the communication earphone; if the earphone is normally worn, subjecting the sound signal first to multi-microphone noise reduction and then to single channel noise reduction to further suppress residuary stationary noise; otherwise suppressing stationary noise in the sound signal by single channel noise reduction directly.
Abstract:
A tactile vibration control system and method for a smart terminal. The system includes: a command generator, a tactile driver, a linear resonant actuator, a sensing module, a feedback unit and a comparator; by arranging a plurality of sensors that monitor or sense the vibrating status of the linear resonant actuator, channels of the sensor signals are generated when the actuator vibrates; the feedback unit sends the sensing signals characterizing the physical quantities related to the vibration modes output by the plurality of sensors to the comparator as the feedback signal; and the comparator generates an error signal according to the feedback signal and a desired signal in the input signal and sends the error signal to the command generator so that the command generator adjusts the generated initial commanding signal according to the error signal and achieves the close-loop control of the linear resonant actuator.
Abstract:
A tactile vibration control system and method for a smart terminal. The system includes a command generator, a filter, a tactile driver and a linear resonant actuator; the command generator generates an initial commanding signal according to an input signal; the filter filters the initial commanding signal so that amplitudes of a predetermined number of initial pulses of the filtered commanding signal are larger than a set threshold and phases of a predetermined number of ending pulses reverse; and the tactile driver generates a driving signal according to the filtered commanding signal for driving the actuator to vibrate. The initial commanding signal generated by the command generator is filtered so that when the actuator is driven to vibrate by the driving signal generated subsequently, the actuator has a quick starting response and a quick braking response.
Abstract:
The present method comprises: providing a feedforward microphone outside of each earphone of the active noise-reduction earphones; detecting an amount of external noise by using the feedforward microphone; calculating a weighted energy of a noise signal; and determining whether it is needed to activate the active noise-reduction system based on the weighted energy. When the active noise-reduction control is needed, calculating energy values of two sub-bands, corresponding to the feedforward noise-reduction amount and the feedback noise-reduction amount respectively, in the noise signal, thereby determining the noise-reduction amounts of the feedforward noise reduction system and the feedback noise-reduction system, and controlling the earphone to perform corresponding feedforward noise reduction and feedback noise reduction. Compared with the existing active noise-reduction technologies with a fixed noise reduction, the present invention can optimize the noise-reduction effect.