Abstract: Various embodiments reduce noise within a particular environment, while isolating and capturing speech in a manner that allows operation within an otherwise noisy environment. In one embodiment, an array of one or more microphones is used to selectively eliminate noise emanating from known, generally fixed locations, and pass signals from a pre-specified region or regions with reduced distortion.

Abstract: The present invention discloses a circuit with three-stage of power-on sequence used for suppressing the pop noise in audio system. It mainly comprises a first resistor (R1); a capacitor (Cout); a first switch (SW1); a second switch (SW2); a soft start device; a first feedback amplifier; and a second feedback amplifier. By using the three-stage of power-on sequence, the present invention can effectively suppress the pop noise when the audio driver is power on.

Abstract: An information processing apparatus includes a jack, a noise-canceling adjustment section, a signal superposition section, an inter-device universal communication bus, and a control section. The jack is connectable with a plug of a headphone including a microphone, outputs a first signal to the headphone, and inputs a second signal from the microphone. The noise-canceling adjustment section generates and outputs, based on the second signal input from the jack, a third signal that cancels a noise component around the headphone. The signal superposition section superposes the third signal on the first signal output from the jack. The control section controls the noise-canceling adjustment section via the inter-device universal communication bus.

Abstract: A device in a headset having a throat microphone and ear speakers soundproofing against the auditory meatus. The device has a separate microphone connected to the ear speaker for picking up the sound of the surroundings that is transmitted to the car speaker. Furthermore, noise-suppressing means are included, which limit the sound level of the sound of the surroundings. The throat microphone communicates via a communication unit. The soundproofing ear speaker and the microphone are arranged so that the headset can be housed in a safety helmet.

Abstract: The invention provides a noise removal device and method capable of more proper interpolation on an input signal. The noise removal device is for removing noise in an input signal and includes: a noise detector detecting noise in an IF signal and outputting a noise detection signal; an interpolation controller determining a period and amount of interpolation for noise correction processing, based on the IF signal and the noise detection signal; and a noise gate processor performing the noise correction processing on the IF signal, based on the interpolation period and amount supplied from the interpolation controller. The interpolation controller sets a predetermined first interpolation period, based on a first noise detection signal inputted from the noise detector, and redefines a second interpolation period longer than the first interpolation period when a second noise detection signal is detected within the first interpolation period.

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: Techniques for introducing background noise segments into signal data are provided. The background noise segments are constructed from a background noise print extracted from the signal data. The background noise print may be user specified, or automatically identified by the signal editing tool. The background noise print may be stored with, and subsequently loaded as part of, the project associated with a signal. The background noise segments that are generated based on the background noise print may have different durations than the background noise print itself.

Abstract: This present invention discloses an audio-separating apparatus and operation method thereof. The audio-separating apparatus applies both blind source separation and noise reduction mechanisms. The audio-separating apparatus only uses one microphone to record mixed sound signals. After applying the noise reduction mechanism, noise reduced signals and the mixed sound signals are used as the inputs of the blind source separation. The method may avoid the spatial aliasing effect caused by using a microphone array to record the mixed sound signals. Besides, speech segment losses caused by processing the noise reduction will be effectively recovered, which may help the hearing impaired recognize target speech signals.

Abstract: The object is to improve the effect of a noise reduction algorithm for hearing apparatuses and in particular hearing aids. This is achieved by a method wherein the input signal is modeled by a wanted signal model and a noise signal model. In addition, a signal statistic of the input signal is recorded in a data logging unit. The wanted signal model and/or the noise signal model can now be changed as a function of said signal statistic. Finally the noise component of the input signal is reduced using the noise signal model and/or the wanted signal model. This means that the models used can be continuously adapted to the hearing apparatus user's current situation.

Abstract: An adaptive audio system can be implemented in a communication device. The adaptive audio system can enhance voice in an audio signal received by the communication device to increase intelligibility of the voice. The audio system can adapt the audio enhancement based at least in part on levels of environmental content, such as noise, that are received by the communication device. For higher levels of environmental content, for example, the audio system might apply the audio enhancement more aggressively. Additionally, the adaptive audio system can detect substantially periodic content in the environmental content. The adaptive audio system can further adapt the audio enhancement responsive to the environmental content.

Abstract: An active noise reduction system is provided for receiving an audio input signal and a noise interference signal and calculating an audio broadcasting signal according to a Feedback Filtered-X Least-Mean-Square (FFXLMS) algorithm, wherein the FFXLMS algorithm optimizes a (convergence factor) ? so as to decrease the numbers of divisions operated by the active noise reduction system and increase the operation speed of the active noise reduction system.

Abstract: A talk-through system for hearing protectors such as headphones, helmets, earplugs and the like, in which separate left and right microphones are controlled by separate left and right switches. The microphones allow ambient sound to be heard by the wearer of the hearing protector, and normally the wearer hears audio from both microphones in the appropriate ears. Pushing a switch causes the audio from the ear on which the switch is mounted to be enhanced and, preferably, switched to both ears. Various arrangements of control logic are provided such that activation of a switch can cause changes in audio processing.

Abstract: An audio signal processing method for processing input audio signals of plural channels includes calculating at least one feature quantity representing a difference between channels of input audio signals, selecting at least one weighting factor according to the feature quantity from at least one weighting factor dictionary prepared by learning beforehand, and subjecting the input audio signals of plural channels to signal processing including noise suppression and weighting addition using the selected weighting factor to generate output an output audio signal.

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 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: 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: An audio system is provided that employs time-frequency analysis and/or synthesis techniques for processing audio obtained from a microphone array. These time-frequency analysis/synthesis techniques can be more robust, provide better spatial resolution, and have less computational complexity than existing adaptive filter implementations. The time-frequency techniques can be implemented for dual microphone arrays or for microphone arrays having more than two microphones. Many different time-frequency techniques may be used in the audio system. As one example, the Gabor transform may be used to analyze time and frequency components of audio signals obtained from the microphone array.

Abstract: A dereverberation and noise reduction method adapted for a microphone array and an apparatus using the same are proposed. The microphone array receives a plurality of audio signals from an audio source. The dereverberation and noise reduction method includes the following steps. The received audio signals are processed by a beamforming processing, and a first audio signal is generated. Besides, the received audio signals are processed by a suppression processing, and a suppression audio vector is generated. Further, suppression audio vector is processed by an adaptive filtering processing, and a second audio signal is generated. In addition, the second audio signal is subtracted from the first audio signal to acquire an audio output signal, where parameters of the adaptive filtering processing are adjusted according to a feedback of the audio output signal.

Abstract: A microphone system has a base coupled with first and second microphone apparatuses. The first microphone apparatus is capable of producing a first output signal having a noise component, while the second microphone apparatus is capable of producing a second output signal. The first microphone apparatus may have a first back-side cavity and the second microphone may have a second back-side cavity. The first and second back-side cavities may be fluidly unconnected. The system also has combining logic operatively coupled with the first microphone apparatus and the second microphone apparatus. The combining logic uses the second output signal to remove at least a portion of the noise component from the first output signal.

Abstract: A first matrix (W(k)) indicating frequency characteristics of a separation filter is calculated from input signals of a plurality of channels. A second matrix (Ws(k)) is calculated by using the restriction coefficients (Ci(k)) for restricting the separation filter and the first matrix, and separation filter coefficients (wsij(s)) are calculated by using the second matrix. With use of the separation filter coefficients, separation signals (ysi(t)) are then calculated from the input signals. A third matrix (Ws?1(k)) is then calculated by transforming the second matrix into an inverse matrix at each frequency, and reproduction filter coefficients (a?I1(s), a?I2(s)) are calculated by using the third matrix. With use of the reproduction filter coefficients, the synthesized signal of each channel is calculated by using the separation signals.

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 new type of headset that employs adaptive noise suppression, multiple microphones, a voice activity detection (VAD) device, and unique mechanisms to position it correctly on either ear for use with phones, computers, and wired or wireless connections of any kind is described. In various embodiments, the headset employs combinations of new technologies and mechanisms to provide the user a unique communications experience.

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: 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: 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: Enhancement of audio quality (e.g., speech intelligibility) in a noisy environment, based on subband gain control using information from a noise reference, is described.

Abstract: Microphone arrays (MAs) are described that position and vent microphones so that performance of a noise suppression system coupled to the microphone array is enhanced. The MA includes at least two physical microphones to receive acoustic signals. The physical microphones make use of a common rear vent (actual or virtual) that samples a common pressure source. The MA includes a physical directional microphone configuration and a virtual directional microphone configuration. By making the input to the rear vents of the microphones (actual or virtual) as similar as possible, the real-world filter to be modeled becomes much simpler to model using an adaptive filter.

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.

Abstract: A microphone system has a primary microphone for producing a primary signal, a secondary microphone for producing a secondary signal, and a selector operatively coupled with both the primary microphone and the secondary microphone. The system also has an output for delivering an output audible signal principally produced by one of the to microphones. The selector selectively permits either 1) at least a portion of the primary signal and/or 2) at least a portion of the secondary signal to be forwarded to the output as a function of the noise in the primary signal.

Abstract: A sound reproducing device is provided including a communication unit that transmits/receives signals; at least one sound output unit that outputs sound based upon a signal having been received, a sound pickup unit that picks up sound and generates audio data, an echo canceller unit that stores any echo signal contained in the signal having been received at the communication unit and generates a dummy echo signal by using the stored echo signal, and a noise reducing unit that generates a cancel signal to be used to cancel noise by using the audio data if the sound picked up at the sound pickup unit contains noise originating from a noise source and outputs a composite signal generated by combining the output signal from the echo canceller unit and the cancel signal.

Abstract: Disclosed is an audio equipment, comprising: an audio output amplifier and a speaker, with a sound output from the speaker based on an audio signal that is input in the audio output amplifier, and with a high signal output as an error signal to protect the audio output amplifier when an abnormal operation occurs. The audio equipment further comprising a rectifier circuit to stabilize the error signal.

Abstract: A method, medium, and apparatus for extracting a target sound from mixed sound. The method includes receiving a mixed signal through a microphone array, generating a first signal whose directivity is emphasized toward a target sound source and a second signal whose directivity toward the target sound source is suppressed based on the mixed signal, and extracting a target sound signal from the first signal by masking an interference sound signal, which is contained in the first signal, based on a ratio of the first signal to the second signal. Therefore, a target sound signal can be clearly separated from a mixed sound signal which contains a plurality of sound signals and is input to a microphone array.

Abstract: This specification describes technologies relating to editing digital audio data. In general, one aspect of the subject matter described in this specification can be embodied in methods that include the actions of receiving an audio signal including audio data in multiple channels; identifying noise in the audio signal including identifying panning information for the audio data in the signal at each of multiple frequency bands; and attenuating the audio data at one or more frequency bands to generate an edited audio signal when the panning exceeds a specified threshold for each of the one or more frequency bands. Other embodiments of this aspect include corresponding systems, apparatus, and computer program products.

Abstract: A power spectrum estimation unit (200) obtains an estimated sound power spectrum Ps(?), based on a power spectrum P1(?) and on a first calculated value obtained by at least multiplying a power spectrum P2(?) by a weight coefficient A2(?). A coefficient update unit (300) updates the weight coefficient A2(?) and a weight coefficient A1(?) so that a second calculated value approximates to the power spectrum P1(?). The second calculated value is obtained by adding at least two values obtained by multiplying the power spectrum P2(?) and the estimated target sound power spectrum Ps(?) by the weight coefficient A2(?) and the weight coefficient A1(?), respectively.

Abstract: An audio device performs self calibration with respect to an audio source location when processing an audio signal frame determined likely to be dominated by the audio source. One or more conditions for sub-bands within the audio frame are evaluated to help identify whether the frame is dominated by the audio source. If the conditions meet a threshold value for a number of sub-bands within the frame, the audio signal may be identified as one dominated by the desired audio source and an audio source location coefficient may be adapted. Additionally, when the audio source location coefficient falls below a threshold value, (e.g., suggesting that one of two or more microphones is blocked), noise suppression is reduced or eliminated for the frame or frame sub-bands to prevent suppression of a desired audio source component along with the noise component.

Abstract: Systems and methods for adaptive processing of a close microphone array in a noise suppression system are provided. A primary acoustic signal and a secondary acoustic signal are received. In exemplary embodiments, a frequency analysis is performed on the acoustic signals to obtain frequency sub-band signals. An adaptive equalization coefficient may then be applied to a sub-band signal of the secondary acoustic signal. A forward-facing cardioid pattern and a backward-facing cardioid pattern are then generated based on the sub-band signals. Utilizing cardioid signals of the forward-facing cardioid pattern and backward-facing cardioid pattern, noise suppression may be performed. A resulting noise suppressed signal is output.

Abstract: A voice processing apparatus includes a band dividing portion dividing a first voice signal generated by a first microphone and a second voice signal generated by a second microphone into predetermined frequency bands, a sound source segregating portion segregating an echo component of a voice emitted by a first sound source included in a voice emitted by a second sound source in each of the predetermined frequency bands based on the power of the first and second microphones, and a band synthesis portion synthesizing the first and second voice signals from which the echo component of the first sound source has been segregated by the sound source segregating portion into a voice signal including the voice emitted by the first sound source and a voice signal including the echo component of the first sound source.

Abstract: Systems and methods for utilizing inter-microphone level differences (ILD) to attenuate noise and enhance speech are provided. In exemplary embodiments, primary and secondary acoustic signals are received by omni-directional microphones, and converted into primary and secondary electric signals. A differential microphone array module processes the electric signals to determine a cardioid primary signal and a cardioid secondary signal. The cardioid signals are filtered through a frequency analysis module which takes the signals and mimics a cochlea implementation (i.e., cochlear domain). Energy levels of the signals are then computed, and the results are processed by an ILD module using a non-linear combination to obtain the ILD. In exemplary embodiments, the non-linear combination comprises dividing the energy level associated with the primary microphone by the energy level associated with the secondary microphone.

Abstract: A sound collection apparatus includes a target sound collection unit that collects a sound including a target sound and outputs a collected-sound signal, a non-target sound collection unit, provided at positions different from each other, forms dead zones of sensitivity in a direction of the target sound source so as to collect a sound outside the dead zones and outputs a collected-sound signal. A sensitivity suppression unit generates a sensitivity suppression signal for suppressing a sound collection sensitivity in an overlap region in which dead zones overlap, as compared to a region surrounding the overlap region, by subjecting, to a predetermined signal processing, the collected-sound signal outputted by the non-target sound collection unit. An extraction unit removes, from the collected-sound signal, the sensitivity suppression signal generated, so as to extract a signal of a sound generated in the overlap region in which the dead zones overlap.

Abstract: A system reduces noise or other external signals that may affect communication. A device converts sound from two or more microphones into an operational signal. Based on one or both signals, a beamformer generates an intermediate signal. Reflected or other undesired signals may be estimated or measured by an echo canceller. Interference may be measured or estimated by processing the echo-reduced signal or estimate by a blocking matrix. An interference canceller may reduce the interference that may modify or disrupt a signal based on the output of the blocking matrix and the intermediate signal.

Abstract: Speakers are identified based on sound origination detection through use of infrared detection of satellite microphones, estimation of distance between satellite microphones and base unit utilizing captured audio, and/or estimation of satellite microphone orientation utilizing captured audio. Multiple sound source localization results are combined to enhance sound source localization and/or active speaker detection accuracy.

Abstract: Systems, methods, and apparatus for processing an M-channel input signal are described that include outputting a signal produced by a selected one among a plurality of spatial separation filters. Applications to separating an acoustic signal from a noisy environment are described, and configurations that may be implemented on a multi-microphone handheld device are also described.

Abstract: Disclosed are an apparatus and a method for separating sound sources capable of learning distributions of corresponding sound sources based on the assumption that specific sound sources have specific distributions based on interchannel correlation parameter in audio signals providing space perception through a plurality of channels to separate an amount corresponding to energy contribution of the corresponding sound sources from mixture signals. Exemplary embodiments of the present invention can more precisely predict the channel distributions of the specific sound sources included in the input mixture signals and more accurately separate sound sources than a method for separating a sound source based on the channel according to the related art, under conditions that general channel distribution information of the specific sound sources are approximately modeled.

Abstract: In a method for suppressing a noise by the spectrum subtraction method, it is possible to improve the noise suppression capability by simultaneously obtaining a frequency resolution required for the noise estimation spectrum and a temporal resolution required for the noise suppression spectrum. The signal length of an observation signal cut out for analyzing the spectrum of the observation signal used for estimation calculation of the noise spectrum is set longer than the signal length of an observation signal cut out for analyzing the spectrum of the observation signal as a value to be subtracted for performing subtraction with the noise spectrum.

Abstract: Methods, apparatus, and systems for source separation include a converged plurality of coefficient values that is based on each of a plurality of M-channel signals. Each of the plurality of M-channel signals is based on signals produced by M transducers in response to at least one information source and at least one interference source. In some examples, the converged plurality of coefficient values is used to filter an M-channel signal to produce an information output signal and an interference output signal.

Abstract: An audio signal separation apparatus for separating observation signals in the time domain of a mixture of a plurality of signals including audio signals into individual signals by means of independent component analysis to produce isolated signals adapted to produce isolated signals in the time-frequency domain from the observation signals in the time-frequency domain and a separation matrix substituted by initial values, compute the modified value of the separation matrix by using a score function using the isolated signals in the time-frequency domain and a multidimensional probability density function and the separation matrix, modify the separation matrix until the separation matrix substantially converges by using the modified value and produce isolated signals in the time-frequency domain by using the substantially converging separation matrix.

Abstract: A speaker array and microphone arrays positioned on both sides of the speaker array are provided. A plurality of focal points each serving as a position of a talker are set in front of the microphone arrays respectively symmetrically with respect to a centerline of the speaker array, and a bundle of sound collecting beams is output toward the focal points. Difference values between sound collecting beams directed toward the focal points that are symmetrical with respect to the centerline are calculated to cancel sound components that detour from the speaker array to microphones. Then, it is estimated based on totals of squares of peak values of the difference values for a particular time period that the position of the talker is close to which one of the focal points, and the position of the talker is decided by comparing the totals of the squares of the peak values of the sound collecting beams directed to the focal points that are symmetrical mutually.

Abstract: A signal processing apparatus includes a first audio output unit configured to output audio of a first audio signal input from a first signal input line, a first pickup unit connected to the first signal input line, a second audio output unit configured to output audio of a second audio signal input from a second signal input line, a second pickup unit connected to the second signal input line, a connecting line that connects the above units to ground, and a first reducing unit configured to at least reduce a first sound leakage signal, being the first audio signal leaking into the second signal input line from the first audio output unit, by using the first audio signal, or reducing a second sound leakage signal, being the second audio signal leaking into the second signal input line from the second audio output unit, by using the second audio signal.

Abstract: A microphone system has a base coupled with first and second microphone apparatuses. The first microphone apparatus is capable of producing a first output signal having a noise component, while the second microphone apparatus is capable of producing a second output signal. The system also has combining logic operatively coupled with the first microphone apparatus and the second microphone apparatus. The combining logic uses the second output signal to remove at least a portion of the noise component from the first output signal.