Abstract: Provided are methods and systems for noise suppression within multiple time-frequency points of spectral representations. A multi-feature cluster tracker is used to track signal and noise sources and to predict signal versus noise dominance at each time-frequency point. Multiple features, such as binaural and monaural features, may be used for these purposes. A Gaussian mixture model (GMM) is developed and, in some embodiments, dynamically updated for distinguishing signal from noise and performing mask-based noise reduction. Each frequency band may use a different GMM or share a GMM with other frequency bands. A GMM may be combined from two models, with one trained to model time-frequency points in which the target dominates and another trained to model time-frequency points in which the noise dominates. Dynamic updates of a GMM may be performed using an expectation-maximization algorithm in an unsupervised fashion.

Abstract: A Voice Activity Detection (VAD) algorithm provides a simple binary signal indicating the presence or absence of speech in a microphone signal. The VAD algorithm includes a first step of noise suppression which both estimates and removes (i.e., filters) ambient noise from the microphone signal to create a filtered signal. The magnitude of the filtered signal is then compared to a threshold in order to produce a VAD output signal. The threshold is dynamic and may be derived either from the filtered signal itself, or from a noise spectrum estimate calculated by the noise suppression step.

Abstract: A band separating unit 5 carries out a band division of a plurality of power spectra into which an input signal is converted by a time-to-frequency converting unit 2 to combine power spectra into each subband, and a band representative component generating unit 6 defines a power spectrum having a maximum among the plurality of power spectra within each subband as a representative power spectrum. A noise suppression amount generating unit 7 calculates an amount of noise suppression for each subband by using the representative power spectrum and a noise spectrum, and a noise suppressing unit 9 suppresses the amplitudes of the power spectra according to the amount of noise suppression.

Abstract: A device and method for estimating a tonal stability of a sound signal include: calculating a current residual spectrum of the sound signal; detecting peaks in the current residual spectrum; calculating a correlation map between the current residual spectrum and a previous residual spectrum for each detected peak; and calculating a long-term correlation map based on the calculated correlation map, the long-term correlation map being indicative of a tonal stability in the sound signal.

Abstract: Embodiments of methods and devices for classifying background noise contained in an audio signal are disclosed. In one embodiment, the device includes a module for extracting from the audio signal a background noise signal, termed the noise signal. Also included is a second that calculates a first parameter, termed the temporal indicator. The temporal indicator relates to the temporal evolution of the noise signal. The second module also calculates a second parameter, termed the frequency indicator. The frequency indicator relates to the frequency spectrum of the noise signal. Finally, the device includes a third module that classifies the background noise by selecting, as a function of the calculated values of the temporal indicator and of the frequency indicator, a class of background noise from among a predefined set of classes of background noise.

Abstract: The present disclosure provides a audio signal processing apparatus including, an amplitude detector configured to detect a noise start point of an audio signal including a noise signal by comparing an amplitude value of the audio signal with a threshold value, a frequency feature calculator configured to calculate a frequency feature representing at least a frequency characteristic of the audio signal after the noise start point, and a noise determiner configured to determine a leg continuously including high-frequency components equal to or higher than a reference frequency in the audio signal after the noise start point as a noise leg based on the frequency feature.

Abstract: A method and system for improving a perceived duplexity of handsfree telephone applications is disclosed. An echo suppression circuit for a device comprising loudspeaker and microphone is described. A circuit attenuates a subband of a transmit signal, wherein the transmit signal is captured by the microphone and wherein the transmit signal comprises an echo of a far-end signal rendered by the loudspeaker and a near-end signal. The attenuation circuit further determines a subband far-end indicator of a voice activity in the far-end signal; determines a subband near-end indicator of a voice activity of the near-end signal; determines a subband masking weight; determines a subband attenuation for the transmit signal in the subband; and attenuates the subband of the transmit signal using the determined subband attenuation.

Abstract: Multi-channel noise suppression systems and methods are described that omit the traditional delay-and-sum fixed beamformer in devices that include a primary speech microphone and at least one noise reference microphone with the desired speech being in the near-field of the device. The multi-channel noise suppression systems and methods use a blocking matrix (BM) to remove desired speech in the input speech signal received by the noise reference microphone to get a “cleaner” background noise component. Then, an adaptive noise canceler (ANC) is used to remove the background noise in the input speech signal received by the primary speech microphone based on the “cleaner” background noise component to achieve noise suppression. The filters implemented by the BM and ANC are derived using closed-form solutions that require calculation of time-varying statistics of complex frequency domain signals in the noise suppression system.

Abstract: A personal audio device, such as a wireless telephone, includes noise canceling circuit that adaptively generates an anti-noise signal from a reference microphone signal and injects the anti-noise signal into the speaker or other transducer output to cause cancellation of ambient audio sounds. An error microphone may also be provided proximate the speaker to estimate an electro-acoustical path from the noise canceling circuit through the transducer. A processing circuit uses the reference and/or error microphone, optionally along with a microphone provided for capturing near-end speech, to determine whether one of the reference or error microphones is obstructed by comparing their received signal content and takes action to avoid generation of erroneous anti-noise.

Abstract: A microphone array apparatus includes: an acquisition unit configured to acquire samples from a sound signal inputted from each of a plurality of microphones, at predetermined time intervals; an operation unit configured to calculate a value based on volumes of the sound signal possessed by a plurality of the samples for each of the sound signals inputted from the plurality of microphones; a correlation coefficient calculator configured to calculate a coefficient of correlation between the sound signals, on the basis of the values calculated for the respective sound signals; and a gain calculator configured to calculate reduction gain for the sound signals inputted from the plurality of microphones, on the basis of the coefficient of correlation.

Abstract: A listening device for processing an input sound to an output sound, includes an input transducer for converting an input sound to an electric input signal, an output transducer for converting a processed electric output signal to an output sound, a forward path being defined between the input transducer and the output transducer and including a signal processing unit for processing an input signal in a number of frequency bands and an SBS unit for performing spectral band substitution from one frequency band to another and providing an SBS-processed output signal, and an LG-estimator unit for estimating loop gain in each frequency band thereby identifying plus-bands having an estimated loop gain according to a plus-criterion and minus-bands having an estimated loop gain according to a minus-criterion.

Abstract: The present invention is directed to a wireless telephone having a first microphone and a second microphone and a method for processing audio signal in a wireless telephone having a first microphone and a second microphone. The wireless telephone includes a first microphone, a second microphone, and a signal processor. The first microphone outputs a first audio signal, the first audio signal comprising a voice component and a background noise component. The second microphone outputs a second audio signal. The signal processor increases a ratio of the voice component to the noise component of the first audio signal based on the content of at least one of the first audio signal and the second audio signal to produce a third audio signal.

Abstract: Systems and methods for controlling adaptivity of noise cancellation are presented. One or more audio signals are received by one or more corresponding microphones. The one or more signals may be decomposed into frequency sub-bands. Noise cancellation consistent with identified adaptation constraints is performed on the one or more audio signals. The one or more audio signals may then be reconstructed from the frequency sub-bands and outputted via an output device.

Abstract: A device includes a HPF 702 that modifies frequency characteristics of a target signal; a phase correcting unit 701 that corrects the phase characteristics of the target signal to make the phase characteristics nearly equal to phase characteristics of the HPF 702; a first multiplier 705 that adjusts the gain of the signal output from the phase correcting unit 701; a second multiplier 706 that adjusts the gain of the signal output from the HPF 702; a coefficient determining unit that determines the gain coefficients of the first and second multipliers 705 and 706 in such a manner that the sum of the gain coefficient of the first multiplier 705 and the gain coefficient of the second multiplier 706 becomes a fixed value; and an adder 713 that adds the two signals output from the first multiplier 705 and second multiplier 706.

Abstract: Systems and methods for reconstructing decomposed audio signals are presented. In exemplary embodiments, a decomposed audio signal is received. The decomposed audio signal may include a plurality of frequency sub-band signals having successively shifted group delays as a function of frequency from a filter bank. The plurality of frequency sub-band signals may then be grouped into two or more groups. A delay function may be applied to at least one of the two or more groups. Subsequently, the groups may be combined to reconstruct the audio signal, which may be outputted accordingly.

Abstract: In a multiband compressor 100, a level calculation unit 121 calculates a signal level inputted for each of bands, a gain calculation unit 122 calculates a gain value from the calculated signal level, and a gain limitation unit 130 limits a gain value by comparison with a gain value of the other band in a compressor for each band. With this configuration, provided is a multiband compressor capable of achieving a balance between the quality of sound and the effect of enhancing the sound level at a high level.

Abstract: A method is disclosed for acoustic feedback attenuation at a telecommunications terminal. A speakerphone equipped with a loudspeaker and two microphones is featured. Signals from the two microphones are subjected to a calibration stage and then to a runtime stage. The purpose of the calibration stage is to match the microphones to each other by advantageously using both magnitude and phase equalization across the frequency spectrum of the microphones. During the runtime stage, the microphones monitor the ambient sounds received from sound sources, such as the speakerphone's users and the loudspeaker itself, during a conference call. The speakerphone applies the generated set of filter coefficients to the optimized microphone's signals. By combining the signal from the reference microphone with the filtered signal from the optimized microphone, the speakerphone is able to attenuate the sounds from the loudspeaker that would otherwise be transmitted back to other conference call participants.

Abstract: A pseudo noise superimposing unit superimposes a pseudo noise (M-sequence) to an audio signal picked up by a microphone and outputs the superimposed signal to an amplifying system. An calculating unit calculates a correlation value between the audio signal picked up by the microphone and the pseudo noise. The calculating unit estimates a gain of a closed loop based on the correlation value. A gain control unit suppresses a gain of the audio signal based on the estimated gain of the closed loop.

Abstract: Corrupted portions of an audio signal are detected and repaired. An audio signal, including numerous sequential frames, may be received from an audio input device. One or more corrupted frames included in the audio signal may be identified. A frame approximating an uncorrupted frame and corresponding to each corrupted frame may be constructed. Each corrupted frame may be replaced with a corresponding constructed frame to generate a repaired audio signal. The repaired audio signal may be outputted via an audio output device.

Abstract: A sound signal processing device that is capable of suitably extracting main sound from mixed sound in which unnecessary sound (for example, leakage sound and reverberant sound) is mixed with the main sound. More specifically, a mixed sound signal in the time domain including first sound and second sound, and a target sound signal in the time domain including sound corresponding to at least the second sound, which have temporal relation in their entirety or in part, are each divided into a plurality of frequency bands. A level ratio between the two signals is calculated at each frequency. Based on the level ratio, a signal of the first sound that is included in the mixed sound signal is extracted.

Abstract: The present invention discloses a microphone array structure able to reduce noise and improve speech quality and a method thereof. The method of the present invention comprises steps: using at least two microphone to receive at least two microphone signals each containing a noise signal and a speech signal; using FFT modules to transform the microphone signals into frequency-domain signals; calculating an included angle between a speech signal and a noise signal of the microphone signal, and selecting a phase difference estimation algorithm, a noise reduction algorithm or both to reduce noise according to the included angle; if the phase difference estimation algorithm is used, calculating phase difference of the microphone signals to obtain a time-space domain mask signal; and multiplying the mask signal and the average of the microphone signals to obtain the speech signals of the microphone signals. Thereby is eliminated noise and improve speech quality.

Abstract: Improved methods of presenting speech prompts to a user as part of an automated system that employs speech recognition or other voice input are described. The invention improves the user interface by providing in combination with at least one user prompt seeking a voice response, an enhanced user keyword prompt intended to facilitate the user selecting a keyword to speak in response to the user prompt. The enhanced keyword prompts may be the same words as those a user can speak as a reply to the user prompt but presented using a different audio presentation method, e.g., speech rate, audio level, or speaker voice, than used for the user prompt. In some cases, the user keyword prompts are different words from the expected user response keywords, or portions of words, e.g., truncated versions of keywords.

Abstract: The present invention relates to a signal processing apparatus and method, a program, and a data recording medium configured such that the playback level of an audio signal can be easily and effectively enhanced without requiring prior analysis. An analyzer 21 generates mapping control information in the form of the root mean square of samples in a given segment of a supplied audio signal. A mapping processor 22 takes a nonlinear function determined by the mapping control information taken as a mapping function, and conducts amplitude conversion on a supplied audio signal using the mapping function. In this way, by conducting amplitude conversion of an audio signal using a nonlinear function that changes according to the characteristics in respective segments of an audio signal, the playback level of an audio signal can be easily and effectively enhanced without requiring prior analysis. The present invention may be applied to portable playback apparatus.

Abstract: Systems and methods improve audio signals and include means and methods of reducing stochastic noise in wideband audio signals. Multiple microphones may acquire near and far end audio signals, the audio signals may undergo transformations via a general or specialized digital signal processor.

Abstract: An information processing device includes: an output device configured to perform notification to a user by outputting ringing sound; a sound pickup device configured to pick up surrounding sound as ambient sound; an adaptive filtering process device configured to perform an adaptive filtering process using the picked-up ambient sound and the ringing sound output from the output device, to thereby extract, from the ambient sound, estimated environmental sound from which the ringing sound picked up by the sound pickup device has been removed; and a control device configured to control, on the basis of the feature quantity of a predetermined feature extracted from the estimated environmental sound, the adjustment of at least one of the sound volume and the sound quality of the ringing sound.

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 system, method, and computer program product are provided for cleaning an audio segment. For a given audio segment, an offset amount is calculated where the audio segment is maximally correlated to the audio segment as offset by the offset amount. The audio segment and the audio segment as offset by the offset amount are averaged to produce a cleaned audio segment, which has had noise features reduced while having signal features (such as voiced audio) enhanced.

Abstract: A method and device are provided for the objective evaluation of voice quality of a speech signal. The device includes: a module for extracting a background noise signal, referred to as a noise signal, from the speech signal; a module for calculating the audio parameters of the noise signal; a module for classifying the background noise contained in the noise signal on the basis of the calculated audio parameters, according to a predefined set of background noise classes; and a module for evaluating the voice quality of the speech signal on the basis of at least the resulting classification relative to the background noise in the speech signal.

Abstract: An audio processing pipeline, for an auditory prosthesis, includes: a common stage, including a common frequency analysis filter bank, configured to generate a common set of processed signals based on an input audio signal; and first and second stimulator-specific stages, responsive to the common set of signals and including first and second frequency-analysis filter banks, configured to generate first and second sets of processed signals adapted for the first and second hearing stimulators, respectively.

Abstract: The present invention relates to a multi-microphone system and method adapted to determine phase angle differences between a first microphone and a second microphone signal to detect presence of wind noise.

Abstract: To provide a noise reduction transmitter which can secure clarity of sounds collected in very noisy environments and maintain a quality of sounds without devising a noise insulation cover particularly. A transmission microphone 7 is arranged inside a noise insulation cover 2 worn on and covering at least a user's 1 mouth. A noise detection microphone 9 which detects external noises is arranged outside the noise insulation cover, and a noise component cancellation circuit 11 is provided which generates a noise component cancellation signal based on an output signal from the noise detection microphone. An electroacoustic transducer 8 is arranged in the noise insulation cover to reproduce a noise component cancellation sound based on an output signal from the noise component cancellation circuit 11.

Abstract: Systems and methods for utilizing inter-microphone level differences to attenuate noise and enhance speech are provided. In exemplary embodiments, energy estimates of acoustic signals received by a primary microphone and a secondary microphone are determined in order to determine an inter-microphone level difference (ILD). This ILD in combination with a noise estimate based only on a primary microphone acoustic signal allow a filter estimate to be derived. In some embodiments, the derived filter estimate may be smoothed. The filter estimate is then applied to the acoustic signal from the primary microphone to generate a speech estimate.

Abstract: A method of compensating for noise in a receiver having a first receiver unit and a second receiver unit, the method includes receiving a first transmission at the first receiver unit, the first transmission having a first signal component and a first noise component; receiving a second transmission at the second receive unit, the second transmission having a second signal component and a second noise component; determining whether the first noise component and the second noise component are incoherent and; only if it is determined that the first and second noise components are incoherent, processing the first and second transmissions in a first processing path, wherein the first processing path is configured to compensate for incoherent noise.

Abstract: An exemplary method includes a sound processing subsystem 1) dividing an audio signal presented to an auditory prosthesis patient into a plurality of signals each representative of a distinct frequency portion of the audio signal and each contained within a distinct analysis channel included in a plurality of analysis channels, 2) determining a sound level of each signal included in the plurality of signals, and 3) setting an amount of noise reduction applied to each signal included in the plurality of signals in accordance with the determined sound level of each signal included in the plurality of signals. Corresponding methods and systems are also disclosed.

Abstract: To provide a signal analysis apparatus and a signal analysis method which can expand a dynamic range. A spectrum analyzer 1 includes a frequency conversion unit 10 that includes an S-ATT 11 which adjusts the level of an analog input signal and converts the input signal into a predetermined intermediate frequency signal, a V-ATT 21 that adjusts the level of an output signal from the frequency conversion unit 10, an ADC 23 that converts an output signal from the V-ATT 21 into a digital signal, an f-response correction filter 24 that corrects the frequency response of an output signal from the ADC 23, and a noise floor level subtraction unit 25 that subtracts the noise level of the ADC 23 from a noise floor level indicating an overall noise level of the S-ATT 11 to the f-response correction filter 24 in a predetermined frequency band.

Abstract: The initial values of parameter estimates are set, including reverberation parameter estimates, which includes a regression coefficient used in a linear convolutional operation for calculating an estimated value of reverberation included in an observed signal, source parameter estimates, which includes estimated values of a linear prediction coefficient and a prediction residual power that identify the power spectrum of a source signal, and noise parameter estimates, which include noise power spectrum estimates. Then, the maximum likelihood estimation is used to alternately repeat processing for updating at least one of the reverberation parameter estimates and the noise parameter estimates and processing for updating the source parameter estimates until a predetermined termination condition is satisfied.

Abstract: A system and method may be provided to segment and/or classify an audio signal from transformed audio information. Transformed audio information representing a sound may be obtained. The transformed audio information may specify magnitude of a coefficient related to energy amplitude as a function of frequency for the audio signal and time. Features associated with the audio signal may be obtained from the transformed audio information. Individual ones of the features may be associated with a feature score relative to a predetermined speaker model. An aggregate score may be obtained based on the feature scores according to a weighting scheme. The weighting scheme may be associated with a noise and/or SNR estimation. The aggregate score may be used for segmentation to identify portions of the audio signal containing speech of one or more different speakers. For classification, the aggregate score may be used to determine a likely speaker model to identify a source of the sound in the audio signal.

Abstract: A method for the automatic removal of speech contamination from an acoustic noise signal. The method includes the steps of: (a) receiving an input acoustic noise signal; (b) automatically detecting speech contamination in the received acoustic noise signal using a VAD; (c) automatically identifying uncontaminated segments of the received acoustic noise signal based upon a decision value output by the VAD; (d) automatically assembling a congruous uncontaminated acoustic noise signal from the identified uncontaminated segments of the received acoustic noise signal; and (e) outputting the congruous uncontaminated acoustic noise signal. Also, systems implementing such a method.

Abstract: A novel adaptive beamforming technique with enhanced noise suppression capability. The technique incorporates the sound-source presence probability into an adaptive blocking matrix. In one embodiment the sound-source presence probability is estimated based on the instantaneous direction of arrival of the input signals and voice activity detection. The technique guarantees robustness to steering vector errors without imposing ad hoc constraints on the adaptive filter coefficients. It can provide good suppression performance for both directional interference signals as well as isotropic ambient noise.

Abstract: Frequency domain signal extraction methods and apparatuses include receiving a reference signal, which contains mostly undesired audio and is substantially void of desired audio. The reference signal is decomposed into at least two reference frequency components. Filtering the at least two reference frequency components with at least two adaptive filters to form at least two filtered reference frequency components. The filtered reference frequency components are recombined in an IFFT component, to produce a filtered reference signal. A delayed signal is input to an adder. The delayed signal contains desired audio and undesired audio. The filtered reference signal is subtracted from the delayed signal to form an output signal containing desired audio. The output signal is decomposed into at least two frequency components. The filtering is adapted with the at least two frequency components.

Abstract: An echo suppression system and method, and a computer-readable storage medium that is configured with instructions that when executed carry out echo suppression. Each of the system and the method includes the elements of a linear echo suppressor having a reference signal path, with a nonlinearity introduced in the reference signal path to introduce energy in spectral bands. Unlike an echo canceller, the echo suppression system and method are relatively robust to errors in the introduced nonlinearity.

Abstract: According to an embodiment, a method of reducing noise in a signal received at a processing stage of an acoustic system includes, at the processing stage identifying at least one frequency which causes a system gain of the acoustic system to be above an average system gain of the acoustic system; providing a noise attenuation factor for reducing noise in the signal for the at least one frequency, the noise attenuation factor for the at least one frequency based on the system gain for that frequency; and applying the noise attenuation factor to a component of the signal at that frequency.

Abstract: A signal processing method for converting a signal received via a transmission path or read from a storage medium into a first audible signal, and suppressing a noise other than a desired signal contained in the first audible signal based on predetermined audio quality adjustment information, comprising steps of: in suppressing a noise other than a desired signal contained in the first audible signal to generate an enhanced signal, receiving audio quality adjustment information for adjusting audio quality; and adjusting audio quality of the enhanced signal using the audio quality adjustment information.

Abstract: A noise estimation control system may limit increases of a stored background noise estimate in response to a detected noise feedback situation. The system receives an input audio signal detected within a space, and a reference audio signal that is transmitted by a speaker as an aural signal into the space. A signal processor processes the input audio signal and the reference audio signal to determine a coherence value based on an amount of the aural signal that is included in the input audio signal. The signal processor also calculates an amount to adjust the stored background noise estimate based on the coherence value and a determined background noise level of the input audio signal.

Abstract: A method and apparatus for generating a noise reduced output signal from sound received by a first and second microphone arranged as a microphone array. The method includes transforming sound received by the first microphone into a first input signal and sound received by a second microphone into a second input signal and calculating, for each of the frequency components, a weighted sum of at least two intermediate signals calculated from the input signals by means of complex valued transfer functions and real valued Equalizer functions. The method includes a weighing function with range between zero and one, with quotients of signal energies of the intermediate functions as arguments of the weighing function, and generating the noise reduced output signal based on the weighted sum of the intermediate functions and based on the weighted sum of the first and second intermediate function at each of the frequency components.

Abstract: A method and computing system for suppressing noise in an audio signal, comprising: receiving the audio signal at signal processing means; determining that another signal is input to the signal processing means, the input signal resulting from an activity which generates noise in the audio signal; and selectively suppressing noise in the audio signal in dependence on the determination that the input signal is input to the signal processing means to thereby suppress the generated noise in the audio signal.

Abstract: A signal processing apparatus includes: one or more detection means for detecting movement of a diaphragm of a speaker in correspondence with feedback methods that are different feedback methods; analog-to-digital conversion means for converting one or more detection signals acquired by the detection means into a digital form; feedback signal generating means for generating feedback signals corresponding to the feedback methods using the digital detection signals; synthesis means for combining an audio signal to be output as a driving signal of the speaker with the feedback signals; correction equalizer means for setting an equalizing characteristic to allow a sound reproduced by the speaker to have a target frequency characteristic by changing the digital audio signal; feedback operation setting means for setting feedback methods in which a feedback operation up to combining the audio signal with the feedback signal is performed and the feedback operation is not performed equalizing characteristic changing a

Abstract: The present invention provides a speech enhancing method for communication earphone including two parts: sending end noise reduction processing and receiving end noise reduction processing, wherein the sending end noise reduction processing part includes: determining a wearing condition of the earphone by comparing energy difference of sound signals picked up by microphones of the communication earphone; if the earphone is normally worn, subjecting the sound signal first to multi-microphone noise reduction and then to single channel noise reduction to further suppress residuary stationary noise; otherwise suppressing stationary noise in the sound signal by single channel noise reduction directly.

Abstract: One aspect of the invention provides a method for enhancing speech output by an electro-acoustical transducer in a noisy listening environment. In some embodiments, this method includes: filtering an input audio signal x(t) using a filter H(z) to produce a filtered audio signal x(t) formula (I), wherein x(t) formula (I)—H(z)x(t); providing to an electro-acoustical transducer a signal corresponding to the filtered audio signal x(t) formula (I) to produce a sound wave corresponding to the filtered audio signal; and prior to filtering the audio signal using the filter, configuring the filter such that, with respect to one or more frequencies, the filtered audio signal has a higher signal level than the input audio signal, and such that the overall signal level of the filtered audio signal (slƒ) is substantially related to the overall signal level of the input signal (slr) such that si/=sl/×c.

Abstract: Zoom motor noise in a camera audio recording is reduced by detecting activity of the zoom motor, transforming a audio signal into the frequency domain during zoom motor activity, and scaling the frequency domain signal during zoom motor activity in each of a series of frequency bins by a scaling factor derived from a pre-stored zoom motor noise spectrum to produce a processed audio signal in the frequency domain. The processed audio signal is then transformed back to the time domain.