摘要:
The invention is directed to a multi-channel echo compensation system, comprising two loudspeaker input channels, each loudspeaker input channel being connected to a loud-speaker for providing a loudspeaker input signal to be emanated by the loudspeaker, a microphone output channel being connected to at least one microphone for receiving a microphone output signal from the at least one microphone, wherein each microphone is configured to acquire a signal emanating from the loudspeakers, a compensation channel for each loudspeaker input channel, each compensation channel connecting a respective loudspeaker input channel and the microphone output channel, an adaptive compensation filter for each compensation channel, wherein each adaptive compensation filter is configured to filter a signal on the respective compensation channel such that a compensation output signal is provided to compensate a microphone output signal for a signal emanating from the loudspeakers, a pre-processing means for pre-processing loudspeaker input signals on the compensation channels, the pre-processing means being configured to determine a correlation value of the loudspeaker input signals for the two loudspeakers according to a pre-determined criterion and to de-activate one of the adaptive compensation filters if the determined correlation value passes a pre-determined threshold.
摘要:
An adaptive signal processing system eliminates noise from input signals while retaining desired signal content, such as speech. The resulting low noise output signal delivers improved clarity and intelligibility. The low noise output signal also improves the performance of subsequent signal processing systems, including speech recognition systems. An adaptive beamformer in the signal processing system consistently updates beamforming signal weights in response to changing microphone signal conditions. The adaptive weights emphasize the contribution of high energy microphone signals to the beamformed output signal. In addition, adaptive noise cancellation logic removes residual noise from the beamformed output signal based on a noise estimate derived from the microphone input signals.
摘要:
A speech processing device includes an automotive device that filters data that is sent and received across an in-vehicle bus. The device selectively acquires vehicle data related to a user settings or adjustments. An interface acquires the selected vehicle data from in-vehicle sensors in response to a user's articulation of a first code phrase. A memory stores the selected vehicle data with unique identifying data associated with the user and establishes a connection between the selected vehicle data and the user when a second code phrase is articulated. A data interface provides access to the selected vehicle data and stored relationship data and enables the processing of the data to customize the in-vehicle system. The data interface is responsive to the user's articulation of a third code phrase to process the selected vehicle data that enables the setting or adjustment of the in-vehicle system.
摘要:
An approach for adjusting an adjustable element, such as a mirror, head rest, steering wheel, heating/air condition blower, associated with a vehicle by determining the position of a speaker in the vehicle.
摘要:
The invention provides a computer-implemented method for determining a time delay for time delay compensation of a microphone signal from a microphone array in a beamformer arrangement. For a given time, an instantaneous estimate of a position of a wanted sound source and/or of a direction of arrival of a signal originating from the wanted sound source is determined. The computer system then determines whether the instantaneous estimate deviates from a preset estimate of a position of the wanted sound source and/or of a direction of arrival of a signal originating from the wanted sound source according to a predetermined criterion. The predetermined criterion comprises a check whether the instantaneous estimate deviates from the preset estimate by at least a predetermined deviation threshold. If the predetermined criterion is fulfilled, the instantaneous estimate for the given time is set by the computer system as the preset estimate, and the computer system determines the time delay for time delay compensation of the microphone signal based on the instantaneous estimate.
摘要:
An echo compensation system may remove undesirable audio signals. The echo compensation system may utilize adaptive filters to remove echoes and undesirable signals received by a microphone. The echo compensation system may inhibit adaptation of an adaptive filter when left and right channel audio signals are highly correlated. In a two-channel system, inhibiting adaptation of one of two adaptive filters may reduce computational power requirements, while still removing undesirable signals. In a four-channel system, inhibiting adaptation of all but one of the four adaptive filters may reduce computational power requirements by a greater percentage.
摘要:
A microphone compensation system responds to changes in the characteristics of individual microphones in an array of microphones. The microphone compensation system provides a communication system with consistent performance despite microphone aging, widely varying environmental conditions, and other factors that alter the characteristics of the microphones. Furthermore, lengthy, complex, and costly measurement and analysis phases for determining initial settings for filters in the communication system are eliminated.
摘要:
An audio processing system controls an audio input signal. The audio processing system includes a signal analyzer that detects content information and source information corresponding to the audio input signal. The system generates an analyzed audio signal. A signal processor receives the analyzed audio signal and generates a processed audio signal based on the content information and/or source information.
摘要:
An audio processing system controls an audio input signal. The audio processing system includes a signal analyzer that detects content information and source information corresponding to the audio input signal. The system generates an analyzed audio signal. A signal processor receives the analyzed audio signal and generates a processed audio signal based on the content information and/or source information.
摘要:
The present invention relates to a vehicle communication system comprising a plurality of microphones adapted to detect speech signals of different vehicle passengers, a mixer combining the audio signal components of the different microphones to a resulting speech output signal, a weighting unit determining the weighting of the audio signal components for the resulting speech output signal, where the weighting unit determines the weighting of the signal components based upon non-acoustical information about the presence of a vehicle passenger.