摘要:
A system reduces noise or other external signals that may affect communication. A device converts sound from two or more microphones into an operational signal. Based on one or both signals, a beamformer generates an intermediate signal. Reflected or other undesired signals may be estimated or measured by an echo canceller. Interference may be measured or estimated by processing the echo-reduced signal or estimate by a blocking matrix. An interference canceller may reduce the interference that may modify or disrupt a signal based on the output of the blocking matrix and the intermediate signal.
摘要:
A system estimates the spectral noise power density of an audio signal includes a spectral noise power density estimation unit, a correction term processor, and a combination processor. The spectral noise power density estimation unit may provide a first estimate of the spectral noise power density of the audio signal. The correction term processor may provide a time dependent correction term based, at least in part, on a spectral noise power density estimation error of the actual spectral noise power density. The correction term may be determined so that the spectral noise power density estimation error is reduced. The combination processor may combine the first estimate with the correction term to obtain a second estimate of the spectral noise power density that may be used for subsequent signal processing to enhance a desired signal component of the audio signal.
摘要:
To accommodate a press tooling carrier without interference, a control arm straight guide having at least one drive, e.g., having motor angular gears (worm gears), cardan shaft or the like is provided. A control arm straight guide permits a very shallow position of the joints in the completely assembled state. A hydraulic cylinder may be used to support startup of the motor, i.e., to move the control arm straight guide out of its bottom dead center position.
摘要:
An arrangement is described for speech signal processing. An input microphone signal is received that includes a speech signal component and a noise component. The microphone signal is transformed into a frequency domain set of short-term spectra signals. Then speech formant components within the spectra signals are estimated based on detecting regions of high energy density in the spectra signals. One or more dynamically adjusted gain factors are applied to the spectra signals to enhance the speech formant components.
摘要:
An In-Car Communication (ICC) system supports the communication paths within a car by receiving the speech signals of a speaking passenger and playing it back for one or more listening passengers. Signal processing tasks are split into a microphone related part and into a loudspeaker related part. A sound processing system suitable for use in a vehicle having multiple acoustic zones includes a plurality of microphone In-Car Communication (Mic-ICC) instances coupled and a plurality of loudspeaker In-Car Communication (Ls-ICC) instances. The system further includes a dynamic audio routing matrix with a controller and coupled to the Mic-ICC instances, a mixer coupled to the plurality of Mic-ICC instances and a distributor coupled to the Ls-ICC instances.
摘要:
Embodiments disclosed herein may include determining a signal parameter of a first microphone and a second microphone associated with a computing device. Embodiments may include generating a reference parameter based upon at least one of the parameter of the first microphone and the parameter of the second microphone. Embodiments may include adjusting a tolerance of at least one of the first microphone and the second microphone, based upon the reference parameter. Embodiments may include receiving, at the first microphone, a first speech signal, the first speech signal having a first speech signal magnitude and receiving, at the second microphone, a second speech signal, the second speech signal having a second speech signal magnitude. Embodiments may include comparing at least one of the first speech signal magnitude and the second speech signal magnitude with a third speech signal magnitude and detecting an obstructed microphone based upon the comparison.
摘要:
A method of frequency-domain filtering is provided that includes a plurality of filters, the plurality of filters including at least one constrained filter(s) W=I, I and at least one unconstrained filter(s) W=1,K− The method includes cascading the W k=i,K unconstrained filter(s). A single constraint window C is applied to the cascaded W=i,K unconstrained filter(s). The W=1,I constrained filter(s) are cascaded with the constrained cascaded Wk=1,K unconstrained filter(s) to form a resulting filter Wll=C(W 1{circle around (x)} . . . {circle around (x)} W){circle around (x)} W . . . W. The frequency domain representation of the single constraint window C may be based, at least in part, on a time domain representation of a single constraint window C that has been circularly shifted such that the frequency domain representation of the constraint window matches a property of the frequency domain representation of the cascaded W=1,K unconstrained filters.
摘要:
A system estimates the spectral noise power density of an audio signal includes a spectral noise power density estimation unit, a correction term processor, and a combination processor. The spectral noise power density estimation unit may provide a first estimate of the spectral noise power density of the audio signal. The correction term processor may provide a time dependent correction term based, at least in part, on a spectral noise power density estimation error of the actual spectral noise power density. The correction term may be determined so that the spectral noise power density estimation error is reduced. The combination processor may combine the first estimate with the correction term to obtain a second estimate of the spectral noise power density that may be used for subsequent signal processing to enhance a desired signal component of the audio signal.
摘要:
Embodiments of the invention disclose computer-implemented methods, systems, and computer program products for estimating signal coherence. First, a sound generated by a sound source is detected by a first microphone to obtain a first microphone signal and by a second microphone to obtain a second microphone signal. The first microphone signal is filtered by a first adaptive finite impulse response filter to obtain a first filtered signal. The second microphone signal is filtered by a second adaptive finite impulse response filter, to obtain a second filtered signal. The coherence of the first filtered signal and the second filtered signal is determined based upon the filtered signals. The first and the second microphone signals are filtered such that the difference between the acoustic transfer function for the transfer of the sound from the sound source to the first microphone and the transfer of the sound from the sound source to the second microphone is compensated in the first and second filtered signals.
摘要:
The invention provides a method for determining a signal component for reducing noise in an input signal, which comprises a noise component, comprising the steps of: estimating the noise component in the input signal, estimating a reverberation component in the noise component, and removing the estimated reverberation component from the estimated noise component to obtain a modified estimate of the noise component.