摘要:
A method for altering an audio signal of interest in a multi-channel soundfield representation of an audio enviroment, the method including the steps of: (a) extracting the signal of interest from the soundfield representation; (b) determining a residual soundfield signal; (c) inputting a further associated audio signal, which is associated with the signal of interest; (d) transforming the associated audio signal into a corresponding associated soundfield signal compatable with the residual soundfield; and (e) combining the residual soundfield signal with the associated soundfield signal to produce an output soundfield signal.
摘要:
Embodiments of the present invention disclose a data transmission method and system, and a related device, which are applicable to the field of communications technologies. In the data transmission method of the embodiments, a host acquires parameter information of a wireless communication channel between a wireless microphone array and the host, that is, a signal-to-noise ratio and/or bandwidth; if the acquired parameter information satisfies a first preset condition, the host reduces sampling frequency of the wireless microphone array or decreases a quantity of data transmission paths between the wireless microphone array and the host, so that bandwidth occupied when the wireless microphone array transmits data is reduced. In this way, the host can dynamically adjust a data transmission parameter of the wireless microphone array according to an actual status of communication between the wireless microphone array and the host, which satisfies a demand of the wireless microphone array that communicates with the host as much as possible.
摘要:
Systems, processes, devices, apparatuses, algorithms and computer readable medium for suppressing spatial interference using a dual microphone array for receiving, from a first microphone and a second microphone that are separated by a predefined distance, and that are configured to receive source signals, respective first and second microphone signals based on received source signals. A phase difference between the first and the second microphone signals is calculated based on the predefined distance. An angular distance between directions of arrival of the source signals and a desired capture direction is calculated based on the phase difference. Directional-filter coefficients are calculated based on the angular distance. Undesired source signals are filtered from an output based on the directional-filter coefficients.
摘要:
In a signal processing system, a set of channel signals from an array of sensors of different operating characteristics are processed in calibration circuitry that calculates individual average values of the channel signals and calculates an average of the individual average values of channel signals as a reference value. Reciprocal calculators calculate reciprocal values of the individual average values of the channel signals. Scaling circuitry scales the reciprocal values by the reference value to produce a set of amplitude calibration signals and scales the channel signals by the calibration signals respectively. As a result, the channel signals are normalized by their own average values and scaled by the reference value to produce a set of calibrated channel signals.
摘要:
In a method and a system non-uniformities of a plurality of microphones are compensated for by adaptively filtering the microphone signals on the basis of a reference signal that is derived from the microphone signals. In this way, the filter coefficients are updated so as to respond to varying environmental conditions and/or changes in the microphone characteristics.