摘要:
In one embodiment, the invention is a microphone system for adjusting the final output sensitivity of a microphone. The system includes transducers that output transducer signals. The system also includes bias circuits providing bias signals to the transducers, as well as amplifiers to receive the transducer signals and output amplified signals. The amplified signals are summed by a summer, which outputs a summed signal. A controller receives the summed signal, and is configured to obtain a desired microphone output characteristic and calculate adjustment amounts based on the characteristic. The controller modifies signals from the transducers based on the adjustment amounts. The controller then outputs a microphone signal based on the summed signal. In another embodiment, the invention provides a method for adjusting the final output sensitivity of a microphone.
摘要:
An electronic device or method for adjusting a gain on a voice operated control system can include one or more processors and a memory having computer instructions. The instructions, when executed by the one or more processors causes the one or more processors to perform the operations of receiving a first microphone signal, receiving a second microphone signal, updating a slow time weighted ratio of the filtered first and second signals, and updating a fast time weighted ratio of the filtered first and second signals. The one or more processors can further perform the operations of calculating an absolute difference between the fast time weighted ratio and the slow time weighted ratio, comparing the absolute difference with a threshold, and increasing the gain when the absolute difference is greater than the threshold. Other embodiments are disclosed.
摘要:
A method includes receiving, at a processor, a first data frame at a first time from a first microphone. The method also includes receiving a second data frame at the first time from a second microphone. The method further includes calculating a power ratio of the first microphone and the second microphone based on the first data frame and the second data frame in response to determining that the first data frame and the second data frame are noise data frames.
摘要:
A directional microphone apparatus and directivity control method that corrects a level difference and a phase difference generated in a low band in a plurality of non-directional microphone units, improve the directivity, and reduce the size are provided. Level difference calculation section (105) calculates the level difference between first signal x1(t) obtained by first non-directional microphone unit (101) and second signal x2(t) obtained by second non-directional microphone unit (102), and correction parameter calculation section (106) calculates coefficients of a linear IIR filter configuring correction process section (103) based on the level difference. Correction process section (103) simultaneously corrects the level difference and a phase difference in the low band between two non-directional microphone units by using the calculated coefficients.
摘要:
A condenser microphone includes multiple condenser microphone units. Each unit includes an impedance converter. The condenser microphone units are connected in series such that outputs of the impedance converter in one of the condenser microphone units drive another of the condenser microphone units. A polarization voltage is accumulated to a DC voltage supplied from a DC voltage supply through a voltage adjuster to be applied to one of a diaphragm and a fixed electrode, and a voltage applied to the one of the diaphragm and the fixed electrode is adjusted by the voltage adjuster.
摘要:
Methods, systems, and apparatuses are described for improved multi-microphone source tracking and noise suppression. In multi-microphone devices and systems, frequency domain acoustic echo cancellation is performed on each microphone input, and microphone levels and sensitivity are normalized. Methods, systems, and apparatuses are also described for improved acoustic scene analysis and source tracking using steered null error transforms, on-line adaptive acoustic scene modeling, and speaker-dependent information. Switched super-directive beamforming reinforces desired audio sources and closed-form blocking matrices suppress desired audio sources based on spatial information derived from microphone pairings. Underlying statistics are tracked and used to updated filters and models. Automatic detection of single-user and multi-user scenarios, and single-channel suppression using spatial information, non-spatial information, and residual echo are also described.
摘要:
A method for processing the signals from two or more microphones in a listening device, and a listening device for conducting the method which has a casing holding the microphones, a signal processing unit which provides an output signal corresponding to the microphone signals and suited to the user's hearing, and a receiver unit for delivering the output signal to the user whereby the signals from the microphones are analyzed in order to detect when the casing of the listening device is being touched, whereby further the signal processing of the signal processing unit changes whenever touching of the casing is detected.
摘要:
A microphone array processing system and method carried out in the system. In one embodiment, the system includes: (1) a beamformer configured to perform adaptive beamforming on gain-compensated signals received from a plurality of microphones, the adaptive beamforming including dynamic range compression and diagonal loading of a sample correlation matrix based on order statistics and (2) a postfilter configured to receive an output of the beamformer and reduce noise components remaining from the beamforming.
摘要:
An exemplary embodiment illustrates an equalization pre-processing method, adapted for characterizing a second sound receptor unit in a sound receiving system based on knowing the internal structure parameters of a first sound receptor unit. The method includes: measuring a first sensitivity response of the first sound receptor unit and a second sensitivity response of the second sound receptor unit; equalizing the second sensitivity response according to the first sensitivity response and obtaining the differences in the sensitivity response; conducting simulation to obtain the third sensitivity response associated with the first sound receptor unit in the given sound receiving system; compensating the third sensitivity response to generate the fourth sensitivity response associated with the second sound receptor unit in the sound receiving system according to the differences in the sensitivity response; analyzing the fourth sensitivity response to characterize the second sound receptor unit in the given sound receiving system.
摘要:
A system includes a plurality of inputs each configured to receive a filtered version of a source signal. The system extracts the energy information from each input signal and compares the energy information of a plurality of input signals. Alternatively, the system extracts energy information from a signal that is the difference of two input signals. Based on the energy information, the system determines at least one parameter that may be changed in at least one circuit in a plurality of circuits to minimize the differences in energy of the input signals or to minimize the energy of the difference signal. Parameters may include for example amplification, delay, and corner frequency values.The set of circuits may include microphone interface circuits. Merely by way of example, a system with microphone interface circuits may be included in a hearing enhancement device or in a hands-free earpiece.