Abstract: A noise suppressor selects, for individual frequency components, maximums by comparing a plurality of noise suppressed spectra 105 and 106 a plurality of noise suppressing units 4 and 5 output, thereby obtaining an output spectrum 107 having the frequency components selected as its components. A first noise suppressing unit 4 generates a noise suppressed spectrum 105 by multiplying an input spectrum 102 by amplitude suppression gains, and makes the amplitude suppression gains greater than most of the amplitude suppression gains in a noise signal intervals of a second noise suppressing unit 5.

Abstract: A system for noise removal is coupled to a signal unit that provides a digital signal. The noise removal system includes a transformation module to transform the digital signal into an f-digital signal, a threshold filter to generate a noiseless signal from the f-digital signal based on a threshold profile, and a signal synthesizer to provide a gain to the noiseless signal and to transform the noiseless signal into an output signal.

Abstract: A system, a method, logic embodied in a computer-readable medium, and a computer-readable medium comprising instructions that when executed carry out a method. The method processes: (a) a plurality of input signals, e.g., signals from a plurality of spatially separated microphones; and, for echo suppression, (b) one or more reference signals, e.g., signals from or to be rendered by one or more loudspeakers and that can cause echoes. The method processes the input signals and one or more reference signals to carry out in an integrated manner simultaneous noise suppression and out-of-location signal suppression, and in some versions, echo suppression.

Abstract: In an embodiment, a method of receiving a digital audio signal, using a processor, includes correcting the digital audio signal from lost data. Correcting includes copying frequency domain coefficients of the digital audio signal from a previous frame, adaptively adding random noise coefficients to the copied frequency domain coefficients, and scaling the random noise coefficients and the copied frequency domain coefficients to form recovered frequency domain coefficients. Scaling is controlled with a parameter representing a periodicity or harmonicity of the digital audio signal. A corrected audio signal is produced from the recovered frequency domain coefficients.

Abstract: The present invention provides a spatially pre-processed target-to-jammer ratio weighted filter and a method thereof, which uses two microphones to receive audio signals. The audio signals are divided into a plurality of sinusoidal waves by a fast Fourier transform (FFT) module, and a beamformer uses the sinusoidal waves to generate beamformed signals. A reference generator generates at least one reference signal. The beamformed signals and reference signals are used to work out power spectral densities (PSD), and a target-to-jammer ratio (TJR) is worked out with the power spectral densities. TJR is used to determine whether a sound source exists. According to the determination result, a noise estimator is switched to eliminate noise from the beamformed signals and generate output signals. An inverse fast Fourier transform (IFFT) module recombines the output signals and then outputs the recombined signals.

Abstract: A method estimates noise power spectral density (PSD) in an input sound signal to generate an output for noise reduction of the input sound signal. The method includes storing frames of a digitized version of the input signal, each frame having a predefined number N2 of samples corresponding to a frame length in time of L2=N2/sampling frequency. It further includes performing a time to frequency transformation, deriving a periodogram comprising an energy content |Y|2 from the corresponding spectrum Y, applying a gain function G(k,m)=f(?s2(km),?w2l (k,m?1), |Y(k,m)|2), to estimate a noise energy level |?|2 in each frequency sample, where ?s2 is the speech PSD and ?w2 the noise PSD. It further includes dividing spectra into a number of sub-bands, and providing a first estimate |{circumflex over (N)}|2 of the noise PSD level in a sub-band and a second, improved estimate |{circumflex over (N)}|2 of the noise PSD level in a subband by applying a bias compensation factor B to the first estimate.

Abstract: A method of post-processing raw banded gains for applying to an audio signal, an apparatus to generate banded post-processed gains, and a tangible computer-readable storage medium comprising instructions that when executed carry out the method. The raw banded gains are determined by input processing one or more input audio signals. The method includes applying post-processing to the raw banded gains to generate banded post-processed gains, generating a particular post-processed gain for a particular frequency band, including median filtering using raw gain values for frequency bands adjacent to the particular frequency band. One or more properties of the post-processing depend on classification of the one or more input audio signals.

Abstract: Using frequency characteristics determined for individual ones of a plurality of analyzed bands of a predetermined audio frequency range with frequency resolution that becomes finer in order of lowering frequencies of the analyzed bands, a synthesized band is set for each one or for each plurality of the analyzed bands, and then a time-axial response waveform is determined for each of the synthesized bands. The response waveforms of the synthesized bands are then added together to thereby provide a response waveform for the whole of the audio frequency range.

Abstract: A method of combining at least two audio signals for generating an enhanced system output signal is described.

Abstract: A method for dynamically adjusting the spectral content of an audio signal, which increases the harmonic content of said audio signal, said method comprising translating an encoded digital signal into data bands, creating a psychoacoustic model to identify sections of said data bands that are deficient in harmonic quality, analyzing the fundamental frequency and amplitude of said harmonically deficient data bands, creating additional higher order harmonics for said harmonically deficient data bands, adding said higher order harmonics back to said encoded digital signal to form a newly enhanced signal, inverse filtering said newly enhanced signal, and converting said inverse filtered signal to an analog waveform for consumption by the listener.

Abstract: A method and apparatus for providing uplink signal-to-noise ratio (SNR) estimation in a wireless communication system. A first signal is received over a first channel and a second signal is received over a second channel, where the second signal is received at a higher signal power level than said first signal. A signal-to-noise ratio (SNR) of the second signal is measured, and the SNR of the first signal is determined based at least in part upon the measured SNR of the second signal.

Abstract: The present invention relates to a method for signal processing comprising the steps of providing a set of prototype spectral envelopes, providing a set of reference noise prototypes, wherein the reference noise prototypes are obtained from at least a sub-set of the provided set of prototype spectral envelopes, detecting a verbal utterance by at least one microphone to obtain a microphone signal, processing the microphone signal for noise reduction based on the provided reference noise prototypes to obtain an enhanced signal and encoding the enhanced signal based on the provided prototype spectral envelopes to obtain an encoded enhanced signal.

Abstract: A method and apparatus are provided for estimating the noise spectrum of an audio signal occupying a certain frequency range. The amplitude of the audio signal is measured at only a subset of the frequency range having a low expected signal to noise ratio at the time of measuring such that the measured amplitude is indicative of noise amplitude at that subset of the frequency range. A predefined noise spectral shape is then adapted by fitting to the noise amplitude so as to form the estimated noise spectrum. The noise spectrum so estimated is used to remove interference noise from the audio signal.

Abstract: A method and system is provided for eliminating acoustical feedback in a system. The method determines a parameter for at least one notch filter, adjusting the notch filter based on the parameter, processing the digital signals through the notch filter, testing at the effect of the notch filter in the system, and removing the notch filter if the notch filter is not effective. Also disclosed is a method and system of selecting candidate frequencies which might be feedback, as opposed to other wanted sound frequencies. The selection method sampling the digital signals, converting the time domain digital signal samples by a fast Fourier transform algorithm into the frequency domain, using a ballistics approach to find prominences in the frequency spectrum, and testing the sizes of the prominences.

Abstract: There is provided a sound processing apparatus for processing received sounds. A plurality of sound receiving units which are included in the apparatus output individually a sound signal corresponding to a received sound, then the sound signals in a time domain are converted into respective converted signal in a frequency domain, and a spectral ratio between the two converted signals is calculated for driving a phase correction value which corrects a phase of the sound signal.

Abstract: A system and method for use in filtering of an acoustic signal are provided for producing an output signal of attenuated amount of diffuse sound in accordance with predetermined parameters of desired output directional response and required attenuation of diffuse sound. The system includes a filtration module and a filter generation module including a directional analysis module and filter construction module.

Abstract: System and method to improve intelligibility of coded speech, the method including: receiving an encoded speech signal from a network; extracting an encoded media data stream and one or more control data packets from the encoded speech signal; decoding the encoded media data stream to produce a decoded speech signal; boosting an upper spectral portion of the decoded speech signal to produce a boosted speech signal; and outputting the boosted speech signal. In another embodiment, the method may include: receiving an uncoded speech signal; processing the uncoded speech signal, wherein the processing comprises generating an unencoded data stream from the uncoded speech signal; boosting an upper spectral portion of the unencoded data stream to produce a boosted speech signal; encoding the boosted speech signal to produce an encoded speech signal; and outputting the boosted speech signal.

Abstract: A multiband dynamics compressor implements a solution for minimizing unwanted changes to the long-term frequency response. The solution essentially proposes undoing the multiband compression in a controlled manner using much slower smoothing times. In this regard, the compensation provided acts more like an equalizer than a compressor. What is applied is a very slowly time-varying, frequency-dependent post-gain (make-up gain) that attempts to restore the smoothed long-term level of each compressor band.

Abstract: A method and system is provided for eliminating acoustical feedback in a system. The method determines a parameter for at least one notch filter, adjusting the notch filter based on the parameter, processing the digital signals through the notch filter, testing at the effect of the notch filter in the system, and removing the notch filter if the notch filter is not effective. Also disclosed is a method and system of selecting candidate frequencies which might be feedback, as opposed to other wanted sound frequencies. The selection method sampling the digital signals, converting the time domain digital signal samples by a fast Fourier transform algorithm into the frequency domain, using a ballistics approach to find prominences in the frequency spectrum, and testing the sizes of the prominences.

Abstract: A sound processing device includes: a plurality of sound input units; a detecting unit for detecting a frequency component of each sound input to the plurality of sound signal unit, the each sound arriving from a direction approximately perpendicular to a line determined by arrangement positions of two sound input units among the plurality of sound input units; a correction coefficient unit for obtaining a correction coefficient for correcting a level of at least one of the sound signals generated from the input sounds by the two sound input units so as to match the levels of the sound signals with each other based on the sound of the detected frequency component; a correcting unit for correcting the level of at least one of the sound signals using the obtained correction coefficient; and a processing unit for performing a sound process based on the sound signal with the corrected level.

Abstract: The specification and drawings present a new method, apparatus and software product for calibrating multiple microphones (e.g., a microphone array) to match their sensitivity using an ambient noise by creating and updating one or more calibration signal level difference histograms.

Abstract: VoIP phones according to the present invention include a microphone, which may be internal or external, and allow the user to communicate unobtrusively, check voice mail and conduct other activities in an environment which can be noisy in general and extremely noisy sometimes. Speech recognition functionally may also be used to generate and send touch tone or DTMF tones such as in response to call trees or voice recognition functionality used by airlines, credit card companies, voice mail systems, and other applications. A system and method of audio processing which provides enhanced speech recognition is provided. Audio input is received at the microphone which is processed by adaptive noise cancellation to generate an enhanced audio signal. The operation of the speech recognition engine and the adaptive noise canceller may be advantageously controlled based on Voice Activity Detection (VAD).

Abstract: The present technology provides noise reduction of an acoustic signal using a configurable classification threshold which provides a sophisticated level of control to balance the tradeoff between positional robustness and noise reduction robustness. The configurable classification threshold corresponds to a configurable spatial region, such that signals arising from sources within the configurable spatial region are preserved, and signals arising from sources outside it are rejected. In embodiments, the configurable classification threshold can be automatically and dynamically adjusted in real-time based on evaluated environmental conditions surrounding an audio device implementing the noise reduction techniques described herein.

Abstract: The invention concerns a method and a system for sound spatialization of a first set of not less than one of the audio channels encoded on of a number of frequency subbands (SBk) and decoded in a transformed domain (Fl, C, Fr, Sr, SI, Ife) into a second set of not less than two (Bl, Br) sound channels in the time domain, from modelling filters converted into a gain and a delay applicable in the transformed domain involving: filtering (A) through equalization, subband delay of the signal by applying at least one gain and one delay to generate from each of said encoded channels an equalized and delayed component; adding (B) a subset of equalized and delayed signals to create a number of filtered signals corresponding to not less than two; synthesizing (C) each of said filtered signals to obtain the second set of not less than two reproduction sound channels (Bl, Br) in the time domain.

Abstract: A method of suppressing wind noise in a voice signal determines an upper frequency limit that lies within the frequency spectrum of the voice signal, and for each of a plurality of frequency bands below the upper frequency limit, compares the average power of signal components in a first portion of the signal to the average power of signal components in a second portion of the signal, where the second portion is successive to the first portion. Signal components are identified in at least one of the plurality of frequency bands as containing impulsive wind noise in dependence on the comparison, and the identified signal components are attenuated.

Abstract: It is described a method for controlling an adaptation of a behavior of an audio device (100) to a current acoustic environmental condition. The method comprises (a) monitoring an audio output signal (x(t), x?(t)) being provided to an acoustic output device (110) of the audio device (100) for outputting an acoustic output signal, (b) measuring an audio input signal (z(t)) being provided by an acoustic input device (120) of the audio device (100), wherein the audio input signal (z(t)) is indicative for a feedback portion of the acoustic output signal and for the current acoustic environmental condition, (c) determining a relation between the audio output signal (x?(t)) and the audio input signal (z(t)) and (d) adapting the behavior of the audio device (100) based on the determined relation.

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: A signal processing apparatus includes a source separation module for producing respective separation signals corresponding to a plurality of sound sources by applying an ICA (Independent Component Analysis) to observation signals produced based on mixture signals from the sound sources, which are taken by source separation microphones, to thereby execute a separation process of the mixture signals, and a signal projection-back module for receiving observation signals of projection-back target microphones and the separation signals produced by the source separation module, and for producing projection-back signals as respective separation signals corresponding to the sound sources, which are taken by the projection-back target microphones. The signal projection-back module produces the projection-back signals by receiving the observation signals of the projection-back target microphones which differ from the source separation microphones.

Abstract: In one aspect thereof the invention provides a method for noise suppression of a speech signal that includes, for a speech signal having a frequency domain representation dividable into a plurality of frequency bins, determining a value of a scaling gain for at least some of said frequency bins and calculating smoothed scaling gain values. Calculating smoothed scaling gain values includes, for the at least some of the frequency bins, combining a currently determined value of the scaling gain and a previously determined value of the smoothed scaling gain. In another aspect a method partitions the plurality of frequency bins into a first set of contiguous frequency bins and a second set of contiguous frequency bins having a boundary frequency there between, where the boundary frequency differentiates between noise suppression techniques, and changes a value of the boundary frequency as a function of the spectral content of the speech signal.

Abstract: An electronic device for suppressing noise in an audio signal is described. The electronic device includes a processor and instructions stored in memory. The electronic device receives an input audio signal and computes an overall noise estimate based on a stationary noise estimate, a non-stationary noise estimate and an excess noise estimate. The electronic device also computes an adaptive factor based on an input Signal-to-Noise Ratio (SNR) and one or more SNR limits. A set of gains is also computed using a spectral expansion gain function. The spectral expansion gain function is based on the overall noise estimate and the adaptive factor. The electronic device also applies the set of gains to the input audio signal to produce a noise-suppressed audio signal and provides the noise-suppressed audio signal.

Abstract: Feedback whistle in hearing devices is intended to be able to be suppressed without loss of output of the useful signal. To this end, it is provided to establish or predetermine a frequency range which is susceptible to feedback. From an input signal which has a spectral component in the frequency range susceptible to feedback, a predeterminable component is substituted with a synthetic signal. Mixing-in a synthetic signal is also possibly used to widen the spectrum of an input signal, which is limited.

Abstract: A device and method processing microphone signals from at least two microphones is presented. A first beamformer processes the signals from the microphones and provides a first beamformed signal. A power estimator processes the signals from the microphones and the first beamformed signal from the first beamformer in order to generate, in frequency bands, a first statistical estimate of the energy of a first part of an incident sound field. A gain controller processes said first statistical estimate in order to generate in frequency bands a first gain signal, and an audio processor for processing an input to the signal processing device in dependence of said generated first gain signal. The invention provides a new and improved noise reduction device and noise reduction method for use in the signal processing in devices processing acoustic signals, e.g. microphone devices.

Abstract: A method and an apparatus for removing white noise in a portable terminal are provided. The method for removing the white noise in the portable terminal includes measuring a volume variation of a voice signal output from a power amplifier; detecting a frequency band including white noise using the measured volume variation; and removing signals of the detected frequency band in the voice signal before output to speaker.

Abstract: A method is provided for allocating a frequency for wireless audio communications. A request is received for a frequency for wireless audio communications. The request contains the location, the time period, the quantity of required wireless communications paths and/or the available equipment for the wireless audio communications. This request is compared with information about frequency use which is stored in databases. Possible frequency ranges are outputted, and the possible frequency ranges are allocated to transmitting and/or receiving units of the requested audio communications.

Abstract: A method and an apparatus for obtaining a precoding matrix indicator is disclosed, which relates to the field of communication technologies. The method includes: obtaining a first rotation matrix according to first channel information; obtaining a first differential matrix according to the first rotation matrix and a currently-obtained instantaneous beam forming matrix/precoding matrix; and quantizing, according to a first differential codebook, a pre-acquired first rank indicator, and preset quantization criteria, the first differential matrix to obtain a differential precoding matrix indicator. The apparatus includes: a first rotation matrix obtaining module, a first differential matrix obtaining module, and a differential precoding matrix indicator obtaining module. The differential PMI is obtained according to the channel information, and the differential PMI is used for feedback.

Abstract: An audio signal decorrelator for deriving an output audio signal from an input audio signal has a frequency analyzer for extracting from the input audio signal a first partial signal descriptive of an audio content in a first audio frequency range and a second partial signal descriptive of an audio content in a second audio frequency range having higher frequencies compared to the second audio frequency range. A partial signal modifier modifies the first and second partial signals, to obtain first and second processed partial signals, so that a modulation amplitude of a time variant phase shift or time variant delay applied to the first partial signal is higher than that applied to the second partial signal, or for modifying only the first partial signal. A signal combiner combines the first and second processed partial signals, or combines the first processed partial signal and the second partial signal, to obtain an output audio signal.

Abstract: A robust noise reduction system may concurrently reduce noise and echo components in an acoustic signal while limiting the level of speech distortion. The system may receive acoustic signals from two or more microphones in a close-talk, hand-held or other configuration. The received acoustic signals are transformed to frequency domain sub-band signals and echo and noise components may be subtracted from the sub-band signals. Features in the acoustic sub-band signals are identified and used to generate a multiplicative mask. The multiplicative mask is applied to the noise subtracted sub-band signals and the sub-band signals are reconstructed in the time domain.

Abstract: A system and method for dynamic residual noise shaping configured to reduce hiss noise in an audio signal. The system and method may detect an amount and type of hiss noise. The system and method may limit calculated noise suppression gains responsive to the detected amount and type of hiss noise. The limited noise suppression gains may be applied to the audio signal and may reduce the hiss noise.

Abstract: A signal processing apparatus is configured to change volume level or frequency characteristics of an input signal with a limited bandwidth in a first frequency range. The apparatus includes: an information extracting unit configured to extract second frequency characteristic information from a collection signal with a limited bandwidth in a second frequency range different from the first frequency range; a frequency characteristic information extending unit configured to estimate first frequency characteristic information from the second frequency characteristic information extracted by the information extracting unit, the first frequency characteristic information including the first frequency range; and a signal correcting unit configured to change volume level or frequency characteristics of the input signal according to the first frequency characteristic information obtained by the frequency characteristic information extending unit.

Abstract: A noise suppressing device includes: a suppressing unit for suppressing noise by subtracting a noise component derived on the basis of a plurality of sound signals output by a sound receiving unit from the sound signals; a variation deriving unit for deriving a degree of variation by numerically quantizing the variation on the basis of a spectrum obtained by converting the plurality of sound signals to components on a frequency axis; a determining unit for determining that wind noise is occurring due to wind blowing against the sound receiving units when the degree of variation is equal to or greater than a value; and an adjusting unit for adjusting an amount of suppression according to the decision of the determining unit; wherein the suppressing unit suppresses noise by subtracting the noise component from the sound signals based on the amount of suppression adjusted by the adjusting unit.

Abstract: A method of enhancing an audio signal includes the steps of: a) receiving a primary audio input signal, b) receiving a detected audio signal which comprises: A) an echo component derived from play-out of the primary audio input signal and B) a noise component, and c) estimating from the primary audio input signal and the detected audio signal: 1) a set of frequency-specific lower bound gains, such that each frequency-specific lower bound gain, when applied to a respective frequency of the primary audio input signal, would cause the noise component to just mask the echo component at that respective frequency and 2) a set of frequency-specific upper bound gains, such that each frequency-specific upper bound gain, when applied to a respective frequency of the primary audio input signal, would cause the echo component to just mask the noise component at that respective frequency; d) estimating a set of frequency-specific gains in such a way that each frequency-specific gain falls between the respective frequency-

Abstract: A circuit configured to adjust the level of an audio signal includes filters each of which is configured to receive an audio signal, and to pass a band set for the filter. Variable gain amplifiers are severally provided to the respective filters, and each variable gain amplifier amplifies the output signal of the corresponding filter. An adder sums the input audio signal and the output signals of the variable gain amplifiers. A control unit controls the level of the output signal of each of the variable gain amplifiers, and controls the gain of each variable gain amplifier according to the level thus monitored.

Abstract: Embodiments relate to apparatuses for, and methods of, compensating for distance 206 and velocity present between a microphone and user's 202 mouth. Such devices and methods allow compensation for varying amplitudes of sound pressure received at a the due to a varying distance 206 between a microphone and an originating user's 202 mouth. Additionally, the devices and methods may also compensate for pitch shifts due to a Doppler effect caused by a non-zero velocity of a microphone relative to a user's 202 mouth.

Abstract: Embodiments of the present invention exploit redundancy of succeeding FFT spectra and use this redundancy for computing interpolated temporal supporting points. An analysis filter bank converts overlapped sequences of an audio (ex. loudspeaker) signal from a time domain to a frequency domain to obtain a time series of short-time loudspeaker spectra. An interpolator temporally interpolates this time series. The interpolation is fed to an echo canceller, which computes an estimated echo spectrum. A microphone analysis filter bank converts overlapped sequences of an audio microphone signal from the time domain to the frequency domain to obtain a time series of short-time microphone spectra. The estimated echo spectrum is subtracted from the microphone spectrum. Further signal enhancement (filtration) may be applied. A synthesis filter bank converts the filtered microphone spectra to the time domain to generate an echo compensated audio microphone signal.

Abstract: In one aspect, the invention divides an audio signal into auditory events, each of which tends to be perceived as separate and distinct, by calculating the spectral content of successive time blocks of the audio signal, calculating the difference in spectral content between successive time blocks of the audio signal, and identifying an auditory event boundary as the boundary between successive time blocks when the difference in the spectral content between such successive time blocks exceeds a threshold. In another aspect, the invention generates a reduced-information representation of an audio signal by dividing an audio signal into auditory events, each of which tends to be perceived as separate and distinct, and formatting and storing information relating to the auditory events. Optionally, the invention may also assign a characteristic to one or more of the auditory events. Auditory events may be determined according to the first aspect of the invention or by another method.

Abstract: A separation signal generation unit generates a plurality of separation signals which are independent from one another from the mixed signals for one frame which are converted into those in a frequency region. A mask processing unit judges a noise condition of a first separation signal for each frequency bin on the basis of the first separation signal and second separation signals. The mask processing unit further removes a first noise component obtained on the basis of a judgment result on the noise condition from the first separation signal. A noise amount measuring unit measures the amount of noise in the first separation signal. A noise signal selection unit selects a noise signal for each frequency bin on the basis of the amount of noise measured by the noise amount measuring unit. A noise removing unit removes a second noise component from a noise removal signal inputted from the mask processing unit.

Abstract: Methods, apparatus, and articles of manufacture for media monitoring are disclosed. In particular, the example methods, apparatus, and articles of manufacture generate signatures. Initially, a first plurality of filtered values are identified by performing a wavelet transform on a first frame of media samples. A first energy value is determined based on a first portion of the first plurality of filtered values, and a second energy value is determined based on a second portion of the first plurality of filtered values. A first descriptor of the first frame of media samples is determined based on a comparison of the first energy value and the second energy value. A first signature is generated based on the first descriptor.

Abstract: A system for detecting noise in a signal received by a microphone array and a method for detecting noise in a signal received by a microphone array is disclosed. The system also provides for the reduction of noise in a signal received by a microphone array and a method for reducing noise in a signal received by a microphone array. The signal to noise ratio in handsfree systems may be improved, particularly in handsfree systems present in a vehicular environment.

Abstract: Systems and methods are described herein for detecting and suppressing breathing noise in an audio signal. First, systems and methods are described that analyze audio signals generated by two or more microphones to detect breathing noise in one of the audio signals and that leverage the multiple microphones to suppress detected breathing noise in a manner that minimizes signal distortion. Then, systems and methods are described that are capable of analyzing the audio signal generated by a single microphone to detect breathing noise in the audio signal and thereafter suppress it.

Abstract: A sound source model storage section stores a sound source model that represents an audio signal emitted from a sound source in the form of a probability density function. An observation signal, which is obtained by collecting the audio signal, is converted into a plurality of frequency-specific observation signals each corresponding to one of a plurality of frequency bands. Then, a dereverberation filter corresponding to each frequency band is estimated by using the frequency-specific observation signal for the frequency band on the basis of the sound source model and a reverberation model that represents a relationship for each frequency band among the audio signal, the observation signal and the dereverberation filter. A frequency-specific target signal corresponding to each frequency band is determined by applying the dereverberation filter for the frequency band to the frequency-specific observation signal for the frequency band, and the resulting frequency-specific target signals are integrated.