Abstract: 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.

Abstract: A method for determining coefficients of a family of cascaded second order Infinite Impulse Response (IIR) parametric filters used for equalizing a room response. The method includes determining parameters of each IIR parametric filter from poles or roots of a reasonably high-order Linear Predictive Coding (LPC) model. The LPC model is able to accurately model the low-frequency room response modes providing better equalization of loudspeaker and room acoustics, particularly at the low frequencies. Advantages of the method include fast and efficient computation of the LPC model using a Levinson-Durbin recursion to solve the normal equations that arise from the least squares formulation. Due to possible band interactions between the cascaded IIR parametric filters, the method further includes optimizing the Q value of each filter to better equalize the room response.

Abstract: Disclosed are methods and devices for communicating between a user device, such as a mobile terminal, and a three wire headset assembly having two microphones coupled thereto. The methods provide two microphone uplink communication and stereo audio playback on a three wire headset assembly without requiring a fourth wire or the addition of a separate voice processor. The headset assembly includes a primary microphone and secondary microphone spaced some distance apart for applying techniques of noise reduction and suppression. The headset assembly includes two speakers. Controlled switches are also included in both the user device and headset assembly to achieve dual use for a single wire of the three wires. The three wire headset may advantageously support both stereo playback and two microphone voice processing. In some embodiments, two microphone noise suppression for a wired headset mode of the user device is provided.

Abstract: A method for separating a sound source from a mixed signal, includes Transforming a mixed signal to channel signals in frequency domain; and grouping several frequency bands for each channel signal to form frequency clusters. Further, the method for separating the sound source from the mixed signal includes separating the frequency clusters by applying a blind source separation to signals in frequency domain for each frequency cluster; and integrating the spectrums of the separated signal to restore the sound source in a time domain wherein each of the separated signals expresses one sound source.

Abstract: In one embodiment, an automated interferometric noise measurement system includes: a signal source adapted to provide a carrier signal; a delay line adapted to delay a first version of the carrier signal to provide a delayed signal to a device-under-test (DUT); a variable attenuator adapted to attenuate a second version of the carrier signal to provide an attenuated signal; a first variable phase-shifter adapted to phase-shift the attenuated signal to provide a first phase-shifted signal; a hybrid coupler adapted to receive an output signal from the DUT and the first phase-shifted signal to provide a carrier-suppressed signal and a carrier-enhanced signal; a low-noise amplifier adapted to amplify the carrier-suppressed signal to provide an amplified signal; a second variable phase-shifter adapted to phase-shift a version of the carrier-enhanced signal to provide a second phase-shifted signal; a first mixer adapted to mix a first version of the amplified signal and the second phase-shifted signal to provide a

Abstract: A system, method, and program product are provided for filtering sound from a selected application on a computer without interrupting voice communications on the computer. The method comprises: monitoring a selected program for an outgoing digital audio signal from a selected application; detecting said digital audio signal; and filtering an analog microphone input with the digital audio signal.

Abstract: A method and an acoustic signal processing system for noise reduction of a binaural microphone signal are proposed. A source signal and two interfering signals input to a left and a right microphone of a binaural microphone system respectively. A left and a right microphone signal is filtered by a Wiener filter to obtain binaural output signals of the source signal. The Wiener filter is calculated as H W ? ( ? ) = 1 - S y ? ? 1 , y ? ? 1 ? ( ? ) S v ? ? 1 , v ? ? 1 ? ( ? ) + S v ? ? 2 , v ? ? 2 ? ( ? ) , wherein Sy1,y1(?) is the auto power spectral density of the sum of the interfering signals contained in the left and right microphone signal, Sv1,v1(?) is the auto power spectral density of the filtered left microphone signal and Sv2,v2 is the auto power spectral density of the filtered right microphone signal.

Abstract: A microphone system and an associated method are proposed. The microphone system comprises at least two omnidirectional, microphone signal-emitting directional microphones connected electrically to one another to establish directivity, at least one filter unit with at least one adaptation parameter for the adaptive filtering of the at least two microphone signals and a control unit to change the at least one adaptation parameter such that the sum of interference power is reduced. The value range of the at least one adaptation parameter is limited. The control unit determines limits from a comparison of the noise floor of the ambient noise with a microphone noise number. The adaptation range of an adaptive differential directional microphone is a function of stationary component of background noise, so the directivity can be selected such that the non-stationary microphone noise resulting due to directivity is masked by the stationary component of the background noise.

Abstract: The present invention provides an audio apparatus capable of noise suppression, including a first signal processing circuit which is set with two input ends and one output end, where one input end is configured to receive noise and the other input end is configured to receive a digital music signal; and a second signal processing circuit which is connected to the output end of the first signal processing circuit. The first signal processing circuit is configured to generate a noise inversion signal, and mix and superpose the noise inversion signal with the digital music signal to obtain a mixed signal. The second signal processing circuit is configured to perform digital/analog conversion and amplification on the mixed signal, and output the signal to an earphone. The present invention further provides a noise-suppressed mobile phone where the earphone does not need to carry a power supply and a signal processing module.

Abstract: A mobile terminal including an output unit configured to output sound, an equalizer configured to adjust parameters of the sound output by the output unit, a display unit including a touch screen and configured to display a Graphic User Interface (GUI) including a graphical guide that can be touched and moved to adjust the parameters of the sound output by the output unit, and a controller configured to control the equalizer to adjust the parameters of the sound output by the output unit in accordance with a shape of the graphical guide that is touched and moved.

Abstract: A microphone capable of canceling vibration noise caused by mechanical vibration is provided with, in capsules, a pair of diaphragms and a pair of back plates opposite to the respective diaphragms. A printed circuit board is disposed at the middle of capsules. A pair of diaphragms is disposed close and opposite to the surfaces of the printed circuit board with the printed circuit board disposed therebetween. The difference in distance from a vibration source to the two diaphragms is made small. The microphone has a high canceling effect for canceling vibration noise caused by mechanical vibration.

Abstract: The present system proposes a technique called the spectro-temporal varying technique, to compute the suppression gain. This method is motivated by the perceptual properties of human auditory system; specifically, that the human ear has higher frequency resolution in the lower frequencies band and less frequency resolution in the higher frequencies, and also that the important speech information in the high frequencies are consonants which usually have random noise spectral shape. A second property of the human auditory system is that the human ear has lower temporal resolution in the lower frequencies and higher temporal resolution in the higher frequencies. Based on that, the system uses a spectro-temporal varying method which introduces the concept of frequency-smoothing by modifying the estimation of the a posteriori SNR. In addition, the system also makes the a priori SNR time-smoothing factor depend on frequency.

Abstract: An audio input signal is filtered using an adaptive filter to generate a prediction output signal with reduced noise, wherein the filter is implemented using a plurality of coefficients to generate a plurality of prediction errors and to generate an error from the plurality of prediction errors, wherein the absolute values of the coefficients are continuously reduced by a plurality of reduction parameters.

Abstract: A method for automatic speech recognition includes determining for an input signal a plurality scores representative of certainties that the input signal is associated with corresponding states of a speech recognition model, using the speech recognition model and the determined scores to compute an average signal, computing a difference value representative of a difference between the input signal and the average signal, and processing the input signal in accordance with the difference value.

Abstract: A method for measuring performance of a noise cancellation system that is operable to cancel noise is provided. The method includes generating a first model of a target noise. The first model represents the target noise in a form that is received at a location remote from a noise source of the target noise and within a defined environment. The method also includes generating a second model of a cancellation noise. The cancellation noise is configured to at least partially cancel the target noise when combined with the target noise. The second model represents the cancellation noise in a form that is received at the location. The method also includes determining, using the first model and the second model, a cancellation error value indicative of only a portion of the target noise that remains when the target noise and the cancellation noise are combined.

Abstract: A playback device includes: a playback portion that plays back content and outputs at least an audio signal; an acquisition portion that acquires an external audio signal; a generating portion that, based on noise collected by a sound collecting device, generates a noise cancellation signal to reduce the noise; a switching portion that, if the acquisition portion has acquired the external audio signal when the playback portion is playing back content, switches an output signal from the audio signal to the external audio signal; and a synthesizing portion that synthesizes the output signal from the switching portion with the noise cancellation signal.

Abstract: Disclosed is a wireless MIC system for automatically paring, which comprises a wireless MIC which includes a paring button at its one side and facilitates a connection mode as it is automatically paired with an amplifier by transmitting a paring request signal when the paring button is on, and initializes a connection related information stored in a storing part when the connection mode is cancelled and converts an acoustic signal into a digital audio signal and transmits to the paired amplifier; and an amplifier which facilitates a connection mode as the wireless MIC and the amplifier are paired by setting a transmission and receiving channel of the amplifier and the wireless MIC, and amplifies a digital audio signal from the wireless MIC and outputs in an acoustic form.

Abstract: A method and apparatus for attenuating applause from a satellite radio signal, including, a spectrum analyzer and comparator having an internal memory for obtaining a spectral signature of different sounds in the radio signal, a detector automatically detecting whether the sounds contain applause, an attenuator attenuating the sounds containing applause; and a digital to analog converter converting the attenuated sounds for audio listening. A comparator compares applause in the radio signal to a threshold level and disables the attenuator when the applause abates below the threshold level.

Abstract: An automatic external defibrillator produces audible prompts for a user by providing the audible information with default audio characteristics in response to a first ambient noise condition and providing the audible information having different audio characteristics than the default audio characteristics in response to a second ambient noise condition. An automatic external defibrillator provides voice prompts by selecting one of a plurality of different levels of sound quality at which to playback the voice prompts and playing back the voice prompts at the selected level of sound quality.

Abstract: In summary, this application describes a psycho-acoustically motivated, parametric description of the spatial attributes of multichannel audio signals. This parametric description allows strong bitrate reductions in audio coders, since only one monaural signal has to be transmitted, combined with (quantized) parameters which describe the spatial properties of the signal. The decoder can form the original amount of audio channels by applying the spatial parameters. For near-CD-quality stereo audio, a bitrate associated with these spatial parameters of 10 kbit/s or less seems sufficient to reproduce the correct spatial impression at the receiving end.

Abstract: A circuit assembly for processing an electrical signal of a microphone is provided. The microphone has an inherent impedance. The circuit assembly comprises an impedance circuit, a signal processing unit, a first electrical signal path, and a second electrical signal path. The first electrical signal path is coupleable to a first electrical output of the microphone and is furthermore coupled to a first input of the signal processing unit. The second electrical signal path is coupleable to a second electrical output of the microphone and is furthermore coupled to a second input of the signal processing unit via the impedance circuit. An impedance value of the impedance circuit is selected based on an impedance value of the inherent impedance of the microphone.

Abstract: This specification describes technologies relating to editing audio data. In general, one aspect of the subject matter described in this specification can be embodied in methods that include receiving an audio signal including digital audio data; receiving an input identifying particular audio data of the audio signal corresponding to a noise pulse; and replacing the audio data corresponding to the detected noise pulse using interpolation of adjacent audio data to generate an edited audio signal. Other embodiments of this aspect include corresponding systems, apparatus, and computer program products.

Abstract: Method for processing an input signal in a hearing aid, with the input signal being broken down into a discrete signal for each source relative to an acoustic signal, with the discrete signals being assigned to a spatial position of the source and with the discrete signals being output, or output attenuated, relative to the spatial position.

Abstract: An interference signal removing apparatus of a radio frequency (RF) receiver includes a low noise amplification unit which performs low noise amplification, a feedback processing unit which removes a necessary signal in a desired band from a signal output from the low noise amplification unit, and performs feedback of the signal from which the necessary signal is removed, and a signal processing unit which transmits a processed RF signal by synthesizing an input RF signal and the feedback signal to the noise amplification unit.

Abstract: The invention provides a method for directing operation of a microphone system. In one embodiment, the microphone system comprises a plurality of component modules. First, a diagnostic test is performed to determine a diagnostic result indicating whether the component modules have failed the diagnostic test. Whether a plurality of required component modules corresponding to a current application mode for operating the microphone system have failed the diagnostic test is then determined according to the diagnostic result, wherein the application mode requires cooperation of the required component modules selected from the component modules of the microphone system.

Abstract: The present invention provides a system and method for introducing white noises into a digital audio signal so that there is progressive and cumulative degradation in audio quality after each successive reproduction of the audio sound signal in a fashion analogous to analog audio reproduction. The invention provides a white noise generator, and a digital entroping unit. In a preferred embodiment, the white noise generator is implemented by a hardware random number generator. The digital entroping unit controls the magnitude of white noise desired based on a random number generated by the random number generator, and adds the white noise to the input audio sound signal to produce a degraded audio sound signal. The magnitude of white noise can be controlled by using various masking and formatting of random number data.

Abstract: An electronic audio device with a digital audio output channel in which an amplifier output voltage is gradually ramped up and down to avoid causing a popping sound when the device is turned on and off. This is accomplished without employing any additional hardware, by incrementally changing a digital input word applied to a digital audio source, such as a DSP, so as to gradually change the amplifier output voltage between a minimum, such as zero volts, and a DC working voltage. On powering up, the amplifier is only turned on after the digital word is applied, but while it still results in a minimum amplifier output, and on powering down the amplifier is turned off after it's output has been ramped down, but before removing the digital input word. Sources and output channels can also be switched over by powering down, and then powering up, following the same method.

Abstract: A method for compensating for audio signal distortion in a mobile terminal includes generating a digital audio signal at a gain, amplifying the digital audio signal into an amplified audio signal, reproducing the amplified audio signal, determining whether the amplified audio signal is distorted on the basis of at least one reference voltage, and adjusting the gain if the amplified audio signal is distorted.

Abstract: An earpiece (100) and a method (640) for acoustic management of multiple microphones is provided. The method can include capturing an ambient acoustic signal from an Ambient Sound Microphone (ASM) to produce an electronic ambient signal, capturing in an ear canal an internal sound from an Ear Canal Microphone (ECM) to produce an electronic internal signal, measuring a background noise signal, and mixing the electronic ambient signal with the electronic internal signal in a ratio dependent on the background noise signal to produce a mixed signal. The mixing can adjust an internal gain of the electronic internal signal and an external gain of the electronic ambient signal based on the background noise characteristics. The mixing can account for an acoustic attenuation level and an audio content level of the earpiece.

Abstract: A method, an apparatus, and a computer program, which can suppress a low frequency range component with a small amount of calculation, and can achieve a noise suppression of high quality, are provided. The noise superposed in a desired signal of an input signal is suppressed by converting the input signal to a frequency domain signal; correcting an amplitude of the frequency domain signal to obtain an amplitude corrected signal; obtaining an estimated noise by using the amplitude corrected signal; determining a suppression coefficient by using the estimated noise and the amplitude corrected signal; and weighting the amplitude corrected signal with the suppression coefficient.

Abstract: A method, an apparatus, and a computer program, which can suppress a low frequency range component with a small amount of calculation, and can achieve a noise suppression of high quality, are provided. The noise superposed in a desired signal of an input signal is suppressed by converting the input signal to a frequency domain signal; correcting an amplitude of the frequency domain signal to obtain an amplitude corrected signal; obtaining an estimated noise by using the amplitude corrected signal; determining a suppression coefficient by using the estimated noise and the amplitude corrected signal; and weighting the amplitude corrected signal with the suppression coefficient.

Abstract: A howling suppression apparatus suppresses a howling caused in an acoustic system including a sound collection device and a sound emission device. An estimation part generates an estimated signal by estimating a feedback sound reaching the sound collection device from the sound emission device. An adjustment part generates an estimated signal by adjusting the estimated signal. A spectrum subtraction part generates an acoustic signal using a result of subtracting a frequency spectrum of the estimated signal from a frequency spectrum of an acoustic signal. A filter part generates an acoustic signal by suppressing a component of a frequency band including a howling frequency F among the acoustic signal. An acoustic signal in which the acoustic signal is amplified by an amplifier is supplied to the sound emission device.

Abstract: An audio signal processing apparatus includes: a dividing section dividing each of audio signals of a plurality of channels into a plurality of frequency bands; a phase difference calculating section calculating a phase difference between the audio signals of the plurality of channels, for each of the plurality of frequency bands divided by the dividing section; a level ratio calculating section calculating a level ratio between the audio signals of the plurality of channels, for each of the plurality of frequency bands divided by the dividing section; and an audio signal processing section performing output gain setting with respect to divided signals obtained by the dividing section, on the basis of the phase difference and the level ratio for each of the plurality of frequency bands calculated by the phase difference calculating section and the level ratio calculating section.

Abstract: A noise extraction device of the present invention includes: first and second microphone units (11 and 12) each picking up a sound; a directivity synthesis unit which performs a directivity synthesis on output signals respectively received from the first and second microphone units (11 and 12) and generates two directionally synthesized signals which have: different sensitivities to noise; the same directional pattern with respect to sound pressure; and the same effective acoustic center position; and an acoustic cancellation unit which cancels an acoustic component of one of the two directionally synthesized signals by subtracting the one of the two directionally synthesized signals from the other of the two directionally synthesized signal, so as to extract a noise component.

Abstract: Disclosed is a method of reducing clicking sounds in an audio data stream. The samples of the audio data are delayed by a predetermined amount in a sliding window containing a predetermined number of samples. The presence of a clicking sound in the delayed data within said sliding window is detected and the corresponding audio data replaced by substitute data derived from the audio data stream.

Abstract: An automatic gain controller and a method using the same are provided. The automatic gain controller and method analyze background noise by operating a microphone mounted in a mobile communication terminal and automatically control the gain of the signal part which is non-audible due to the background noise. Thus, a user may listen to music by using an earphone or a headphone connected to the mobile communication terminal in an environment with background noise. The method includes receiving an audio signal to be reproduced, receiving a background noise signal introduced through a microphone, controlling a gain of the audio signal by comparing the background noise signal and the audio signal and outputting the gain-controlled audio signal so that the user can listen to the gain-controlled audio signal.

Abstract: A method for determining leakage factors or adaptation rates, or both, for adaptive filters in an active noise reduction system. The leakage factor or adaptation rate, or both, may vary depending on a parameter of an input reference signal. The parameter may include one or more of reference signal input frequency, rate of change of reference input signal frequency, if a predetermined triggering condition exits, or if a predetermined event has occurred.

Abstract: A mobile wireless communications device includes a housing and circuit board carried by the housing and having radio frequency (RF) circuitry and a processor operative with each other. Audio circuitry is carried by the circuit board and an audio transducer assembly, such as a speaker, has electrical contacts that electrically engage the audio circuitry for carrying audio signals between the audio circuitry and audio transducer assembly. A filter is mounted at the audio transducer assembly and reduces radio frequency (RF) electromagnetic interference to the audio transducer assembly during device operation.

Abstract: Provided are an apparatus and method for preventing noise. The apparatus estimates a noise signal from a signal transformed into a frequency domain, uses the estimated noise signal to estimate the amplitude of the frequency-domain signal according to a frequency band, and then calculates a phase difference according to a frequency band and eliminates or prevents noise from the amplitude-estimated frequency-domain signal based on the calculated phase difference according frequency band.

Abstract: According to embodiments, systems, devices, and methods for ridge selection in scalograms are disclosed. Ridges or ridge components are features within a scalogram which may be computed from a signal such as a physiological (e.g., photoplethysmographic) signal. Ridges may be identified from one or more scalograms of the signal. Parameters characterizing these ridges may be determined. Based at least in part on these parameters, a ridge density distribution function is determined. A ridge is selected from analyzing this ridge density distribution function. In some embodiments, the selected ridge is used to determine a physiological parameter such as respiration rate.

Abstract: The present invention discloses a wireless telephone system using microphone arrays together with additional signal processing to suppress the background noise in the surrounding environment. The signal processing resources of a wireless telephone and multi-channel transmission capabilities of the Bluetooth transmission are used to suppress the background noise. The wireless telephone system includes a Bluetooth transceiver communicating to a wireless telephone through a multi-channel Bluetooth transmission, and an array of microphones coupled to the Bluetooth transceiver. The array of microphones includes a first microphone producing a first audio signal output and a second microphone producing a second audio signal output. The first audio signal output and second audio signal output are transmitted to the wireless telephone through the first channel and second channel of multi-channel Bluetooth transmission respectively.

Abstract: Audio cables used in deploying audio equipment are typically selected to minimize signal transmission losses and may be chosen based on a distance between, and terminations of, an amplifier system and a loudspeaker system. Since this is usually unknown prior to deployment, multiple audio cables with different sizes and terminations and a variety of electrical extension cords to connect to a power supply line are transported. A loudspeaker adapter for the loudspeaker system and an amplifier adapter for the amplifier system can be used to interconnect the two systems via electrical extension cords. This precludes the need for transporting a large number of audio cables, and allows for interconnecting the loudspeaker and the amplifier systems without prior knowledge of the distance between the systems. Safety hazards and component damage can also be minimized by not connecting a hot terminal in the AC connectors of the loudspeaker and amplifier adapters.

Abstract: A method for reducing keyboard noise in conferencing equipment which includes a microphone and a keyboard, the method including: providing, at a noise reduction apparatus, an audio signal originating from the microphone; detecting, at the noise reduction apparatus, an operation of the keyboard; providing, at the noise reduction apparatus, a pre-stored estimate of a keyboard audio noise resulting from the operation of the keyboard upon detection of the operation of the keyboard; and calculating, at the noise reduction apparatus, a noise reduced output signal based on the estimate of the keyboard audio noise and the audio signal.

Abstract: A method and an acoustic signal processing system for noise reduction of a binaural microphone signal (x1, x2) with one target point source and M interfering point sources (n1, n2, . . . , nM) as input sources to a left and a right microphone of a binaural microphone system, include: filtering a left and a right microphone signal by a Wiener filter to obtain binaural output signals of a target point source, where the Wiener filter is calculated as: H W = 1 - ? ( x 1 , n + x 2 , n ) ? ( x 1 , n + x 2 , n ) ? ( x 1 + x 2 ) ? ( x 1 + x 2 ) , where HW is the Wiener filter, ?(x1,n+x2,n)(x1,n+x2,n) is the auto power spectral density of the sum of all of the M interfering point sources components (x1,n, x2,n) contained in the left and right microphone signals and ?(x1+x2)(x1+x2) is the auto power spectral density of the sum of the left and right microphone signals.

Abstract: An audio broadcast processing method includes a plurality of steps. One step involves processing an audio broadcast having at least one occurrence of signal degradation that adversely affects audio quality of a corresponding segment of the audio performance. Another step involves modifying the segment of the audio performance with a plurality of differing signal degradation compensation factors to selectively compensate for the at least one occurrence of signal degradation to produce a plurality of compensated audio performances. Another step involves evaluating the compensated audio performances. Still another step involves determining at least one preferred signal degradation compensation factor based on data from the evaluating of the compensated audio performances. Yet another step includes configuring an audio device with the at least one preferred processing factor.

Abstract: A system that models a microphone may include capsules that receive individual signals. The signals may be combined and modified based on a weighting factor. Directivity patterns of a converted signal may be modified or controlled based on the weighting of the signals.

Abstract: A noise removal unit 102 executes noise removal and flooring processing of an input signal, and a density calculating unit 104 calculates, as to a point of interest on a time-frequency plane of the input signal passing through the noise removal, a density of non-flooring processing points from the presence or absence of the flooring processing of individual points around the point of interest. A partial suppression unit 105 replaces, when the density is less than a threshold, the power of the point of interest with its flooring value by considering it as a musical noise component, thereby suppressing the musical noise component.

Abstract: A method for echo cancellation and noise suppression is disclosed. Linear echo cancellation (LEC) is performed for a primary microphone channel on an entire frequency band or in a range of frequencies where echo is audible. LEC is performed on one or more secondary microphone channels only on a lower frequency range over which spatial processing is effective. The microphone channels are spatially processed over the lower frequency range after LEC. Non-linear noise suppression post-processing is performed on the entire frequency band. Echo post-processing is performed on the entire frequency band.

Abstract: Disclosed herein are apparatus, method, and computer program product whereby a device receives an acoustic signal. In response to the received acoustic signal, the device outputs electrical signals from a first input audio transducer and a second input audio transducer. The second input audio transducer is less sensitive than the first input audio transducer.

Abstract: A system for correcting erroneous microphone readings in a vehicle engine harmonic cancellation (EHC) system. A method for operating an engine harmonic cancelling system, includes receiving, from a first microphone at a first location in a vehicle cabin, a signal representative of noise in the vehicle cabin; receiving, from a second microphone at a second location in the vehicle cabin, a signal representative of noise in the vehicle cabin; and correlating the signal from the first microphone with the signal from the second microphone.