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

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

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: In public audio presentation it is desirable that the playback loudness is continuously adapted to the background noise loudness. Such adaptation of the playback volume requires a continuous measurement of the loudness somewhere within the sound exposure area. A dipole loudspeaker panel has a strong attenuation of the emitted sound in the panel plane. When placing a microphone capsule in that plane but outside the panel, the sound emitted from the panel will be recorded by the microphone with a significant sound pressure level attenuation. Such specific microphone arrangement can capture the background noise without being unduly disturbed by the direct sound emitted from the loudspeakers.

Abstract: In a noise reduction apparatus for controlling noise up to a predetermined upper limited frequency, a distance from a noise source to control point X is made larger than a distance obtained by subtracting a one-half wavelength from a distance, obtained by adding up a distance from the noise source to a noise detecting microphone, a distance corresponding to time as a sum of respective delay time of the noise detecting microphone, a noise controller, and a control speaker, and a distance from the control speaker to control point X, where one wavelength is a period corresponding to the upper limited frequency.

Abstract: A system and method may be configured to reconstruct an audio signal from transformed audio information. The audio signal may be resynthesized based on individual harmonics and corresponding pitches determined from the transformed audio information. Noise may be subtracted from the transformed audio information by interpolating across peak points and across trough points of harmonic pitch paths through the transformed audio information, and subtracting values associated with the trough point interpolations from values associated with the peak point interpolations. Noise between harmonics of the sound may be suppressed in the transformed audio information by centering functions at individual harmonics in the transformed audio information, the functions serving to suppress noise between the harmonics.

Abstract: An apparatus for providing an upmix signal representation on the basis of a downmix signal representation and an object-related parametric information, which are included in a bitstream representation of an audio content, and in dependence on a rendering information, has a distortion limiter configured to adjust upmix parameters using a distortion control scheme to avoid or limit audible distortions which are caused by an inappropriate choice of rendering parameters. The distortion limiter is configured to obtain a distortion limitation control parameter, which is included in the bitstream representation of the audio content, and to adjust a distortion control scheme in dependence on the distortion limitation control parameter.

Abstract: The controller controls the audio processor to perform the first reduction process, in a case where the type of the connected removable lens unit is a first type lens unit, the controller controls the audio processor to perform the second reduction process, in a case where the type of the connected removable lens unit is a second type lens unit.

Abstract: An apparatus for processing an audio signal and method thereof are disclosed, by which a local dynamic range of an audio signal can be adaptively normalized as well as a maximum dynamic range of the audio signal. The present invention includes receiving a signal, by an audio processing apparatus; computing a long-term power and a short-term power by estimating power of the signal; generating a slow gain based on the long-term power; generating a fast gain based on the short-term power; obtaining a final gain by combining the slow gain and the fast gain; and, modifying the signal using the final gain.

Abstract: An apparatus for processing an audio signal and method thereof are disclosed, by which a local dynamic range of an audio signal can be adaptively normalized as well as a maximum dynamic range of the audio signal. The present invention includes receiving, by an audio processing apparatus, a signal, and feedback information estimated based on a normalizing gain; generating a noise estimation based on the signal; computing a gain filter for noise canceling, based on the noise estimation and the signal; and, obtaining a restricted gain filter by applying the feedback information to the gain filter.

Abstract: A system and methods for distortion reduction in a non-linear audio system are disclosed. A method includes the operation of defining a non-linear system transfer function which relates a system electronic input signal with a system acoustic output over a range in which the system acoustic output is in a transition between non-saturated and saturated conditions in a medium in which the system acoustic output of the non-linear audio system is broadcast. A preprocessing transfer function is created that is the inverse of the non-linear system transfer function. The preprocessing transfer function is applied to the system electronic input signal to reduce the level of harmonic distortion at the system acoustic output.

Abstract: An embodiment of an acoustic echo cancellation system is disclosed. The system comprises an echo cancellation unit, a second filter and a subtraction unit. The echo cancellation unit comprises a first attenuator, a first filter and a first subtractor. The first attenuator has a first down-scaling factor for attenuating a first signal. The first filter generates a first echo signal estimate based on the attenuated first signal. The first subtractor generates a third signal by subtracting the first echo signal estimate from a second signal. The second filter generates a second echo signal estimate based on the first signal. The subtraction unit subtracts the second echo signal estimate from the third signal.

Abstract: A method for processing sound that includes, generating one or more noise component estimates relating to an electrical representation of the sound and generating an associated confidence measure for the one or more noise component estimates. The method further comprises processing, based on the confidence measure, the sound.

Abstract: An audio testing system is configured for receiving an audio signal from the an audio emitting device. The system samples the audio signal and obtains sampling points from the audio signal for determining if the audio signal has been distorted. A related method is also disclosed.

Abstract: According to an aspect of the invention, there is provided a noise suppressing apparatus comprising: a fifth unit configured to calculate a gain for noise suppression, based on the first signal-to-noise ratio for each frequency band and the second signal-to-noise ratio for an entire frequency band; an eighth unit configured to calculate an upper limit value of a noise suppression amount for each frequency band, based on the second signal-to-noise ratio; a ninth unit configured to calculate the noise suppression amount for each frequency band, based on the first signal-to-noise ratio; and a tenth unit configured to limit, based on the upper limit value, the noise suppression amount so as to calculate the gain.

Abstract: Broadly speaking, the embodiments disclosed herein describe an apparatus, system, and method for managing the effects of TDMA noise emitted by a communication device on an audible circuit.

Abstract: A method and a hearing device for cancelling or preventing feedback are disclosed. The hearing device comprises a microphone (1), a transfer function (2) and a receiver (3), wherein the transfer function (2) defines relation between an input signal (12) of the hearing device and an output signal (13, 13?) of the hearing device. The method according to the present invention comprises the steps of estimating an external transfer function (21) of an external feedback path (11) defined by sound traveling from the receiver (3) to the microphone (1), estimating the input signal (12?) having no feedback components of the external feedback path (11) using an auxiliary signal (15), which does not comprise feedback components of the external feedback path (11), and using the estimated input signal (14) for estimating the external transfer function (21) of the external feedback path (11).

Abstract: Provided is a method and apparatus for encoding/decoding an audio signal. Sections which are not used to output noise components near important spectral components and sub-bands which are not used to output noise components, are determined to be encoded or decoded, so that the efficiency of encoding and decoding an audio signal increases, and sound quality can be improved using less bits.

Abstract: A system and method for analyzing a signal to monitor the dynamics of its magnitude and frequency characteristics over time. An electronic circuit for identifying feedback in an audio signal, formed in accordance with embodiments of the invention may comprise a feedback control block operable to determine a candidate frequency having potential feedback such that the feedback control block is further operable to perform an iterative analysis of the magnitude of the audio signal at the candidate frequency to determine the growth characteristics of the signal. The electronic circuit may further include a test filter block operable to deploy a test filter at a candidate frequency and a permanent filter block operable to deploy a permanent filter at the candidate frequency if the feedback control block determines that the growth characteristics of the signal at the candidate frequency comprises feedback characteristics after the test filter has been deployed.

Abstract: An audio cable capable of increasing gain and filtering noise includes a signal amplifier arranged between an input end and an output end of the audio cable. The signal amplifier includes at least a primary coil and a secondary coil. The number of turns of the secondary coil is greater than that of the primary coil. A mutual inductance between the primary coil and the secondary coil can transform the level of voltage, current and impedance of the signal amplifier to generate an output gain greater than 1, so as to amplify the audio signal. Moreover, the primary and secondary coils have a characteristic of blocking fast current variation; thereby the higher the gain magnification, the more stable is the sound quality and the smaller the noise becomes.

Abstract: To sufficiently suppress an impact sound in a noisy signal. The impact sound in the noisy signal is suppressed. For this, the impact sound is detected in the noisy signal. It is characterized in that phase information of the detected impact sound is processed by using the phase information of a signal other than the impact sound in the noisy signal so that an amount of change in the phase information is reduced.

Abstract: A method for reducing noise in a vehicle by adjusting the rotational speed of various rotational components.

Abstract: Methods and arrangements for providing soundproof audio communication from a mask assembly. A mask assembly includes a soundproof mask, the soundproof mask being adapted to cover a human nose and mouth. A microphone disposed in a portion of the mask and a venting arrangement provides fluid communication from an interior of the mask to an exterior of the mask. A communication connection which provides audio communication between the microphone and a processor.

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 method for processing an input signal to create an enhanced output signal includes obtaining an envelope of the input signal, determining a logarithm signal of the envelope, determining a rate of change of the logarithm signal to obtain a slope value, and applying a value derived from the slope value to the input signal to thereby generate an enhanced output signal.

Abstract: An oscillation-echo preventing circuit has a microphone/speaker unit and a voltage-canceling circuit for canceling voltages of audio receive signals. The microphone/speaker unit has a main body, at least two microphones, and a speaker or an earphone. The microphone seals a first inside space from an outside space. The microphone seals the first inside space from a second inside space. The speaker or the earphone seals the first inside space from the outside space. The voltage-canceling circuit cancels out the voltages of audio receive signals coming from the microphones, respectively, generating an output of minimum magnitude. Thus, the circuit can sufficiently suppress oscillation and echoing.

Abstract: A sound input device includes a differential microphone, configured to receive sound including noise, and generate a first signal in accordance with the sound; a detector, configured to detect the noise, and generate a second signal in accordance with the detected noise; and a controller, configured to control at least one of suppression of high-frequency components of the first signal and changing of a frequency band to be suppressed of the first signal based on the second signal.

Abstract: A system and method for actively changing the sound perceived by listeners in an audio environment. A single transducer is used as both a sensing microphone and as an output driver. In one embodiment, the invention is implemented as an active noise cancellation system. The sensed noise signals are phase shifted to provide a cancellation effect, combined with the desired audio program signals, and output to the transducer, thereby reducing the level of unwanted noise heard by they listener. In other embodiments, the system can be used to sense the frequency response of a listening room and make appropriate equalization adjustments to the output.

Abstract: An audio playback apparatus is disclosed. The audio playback apparatus processes an audio signal and provides at least one output signal to at least one speaker or an earphone connected to a phone-jack. The audio playback apparatus comprises a clock generating module, a processing module, an amplifier module, and a detecting module. The processing module is connected to the clock generating module and processes the audio signal to generate a processed signal according to a clock signal generated by the clock generating module. The amplifier module amplifies the processed signal to generate an output signal. The detecting module detects whether the earphone is plugged in the phone-jack. If the earphone is plugged in the phone-jack, the amplifier module provides the output signal to the earphone. If not, the amplifier module provides the output signal to the speaker.

Abstract: In a voice intelligibility enhancement system that controls a gain of a voice signal based on noise power and voice power of the voice signal generated by a voice signal generation unit, it is detected whether the voice power is equal to or greater than a predetermined level, noise power output when the voice power is less than the predetermined level is measured and stored, noise power to be output when the voice power exceeds the predetermined level is estimated to be the stored noise power, and gain of a voice signal is controlled on the basis of the voice power and the estimated noise power.

Abstract: A microphone-array-based speech recognition system using a blind source separation (BBS) and a target speech extraction method in the system are provided. The speech recognition system performs an independent component analysis (ICA) to separate mixed signals input through a plurality of microphone into sound-source signals, extracts one target speech spoken for speech recognition from the separated sound-source signals by using a Gaussian mixture model (GMM) or a hidden Markov Model (HMM), and automatically recognizes a desired speech from the extracted target speech. Accordingly, it is possible to obtain a high speech recognition rate even in a noise environment.

Abstract: Systems and methods are described which facilitate quick and accurate extraction of the true noise level from a noise signal that includes additional signals, such as speech, in a cost effective implementation. Aspects of the invention allow the use of one microphone to simultaneously detect background noise as well as speech, while avoiding problems associated with artificially high background noise indication due to inclusion of the speech component in the noise determination. Additionally, systems and methods are described for altering system gain based on accurate noise level determinations.

Abstract: A sound signal correcting apparatus converts an acquired sound signal into a phase spectrum and an amplitude spectrum by an FFT process, compares the amplitude spectrum of the obtained sound signal with a noise model so that a correction coefficient used for correcting the amplitude spectrum of the sound signal is derived, smoothes waveform of the amplitude spectrum of the sound signal using the derived correction coefficient, and converts the sound signal into a sound signal where the amplitude spectrum is corrected by performing an inverse FFT process on the phase spectrum and the smoothed amplitude spectrum.

Abstract: Provided is a noise suppressing technology capable of suppressing various noises including unknown noises without storing information relating to a large number of noises in advance. Noises in a degraded signal are suppressed and noise information is generated on the basis of a noise suppression result. The noises in the degraded signal are suppressed using the generated noise information.

Abstract: A system and method for suppressing noise in one or more of at least first and second channels include obtaining a magnitude difference of signals in the first and second channels, obtaining a magnitude sum of signals in the first and second channels, obtaining a ratio of the magnitude difference to the magnitude sum, generating an attenuation value based on the ratio, selecting an attenuator based on the magnitude difference, and attenuating a signal in a channel by the attenuation value using the selected attenuator.

Abstract: A system for regulating the volume and frequency content of audio producing devices, the system includes: one or more noise making objects (NMO) configured with individual sound control devices in electrical communication with a noise management server and one or more audio producing devices configured with individual sound control devices; wherein the sound control devices have electronic logic processing, storage, and communication capabilities; wherein the noise management server utilize the sound control devices to: determine whether the NMO are producing noise in the audible range of one or more audio producing devices; determine a noise characteristic of the one or more NMO; command the one or more NMO to send the noise characteristic to the one or more audio producing devices; and wherein the volume and frequency content of audio produced by the one or more audio producing devices is adjusted in response to the received noise characteristic.

Abstract: A system and method for analyzing a signal to identify one or more candidate frequencies that may have a potential feedback problem. An electronic circuit for identifying feedback in an audio signal, formed in accordance with embodiments of the invention, may comprise a feedback control block operable to: 1) determine a magnitude of a first dominant spectral component in a wide-band span of frequencies of a signal, 2) determine a magnitude of a second dominant spectral component in a mid-band span of frequencies of the signal, and 3) determine a magnitude of a third dominant spectral component in a narrow-band span of frequencies of the signal. The feedback control block may then further analyze the magnitude and frequency of the first, second, and third dominant spectral components to determine the presence of feedback characteristics in the signal.

Abstract: An audio control apparatus. The audio control apparatus comprises a processing module, an amplification module, and a control module. The processing module receives and processes an audio signal according to a control signal, and generates a processed signal. The amplification module coupled to the processing module, amplifies the processed signal to generate an output signal. The control module coupled to the processing module and the amplification module, receives the output signal to generate the control signal.

Abstract: A teleconferencing apparatus includes the functions of a transmitting unit and a receiving unit and the transmitting unit transmits a sound signal formed from sound pick-up signals of a microphone array made up of microphones Mi (i=1 to N) and position information. The position information is provided by forming a plurality of sound pick-up beams directed in a specific direction and selecting the sound pick-up beam with the largest volume. In the receiving unit, a parameter calculation section sets a virtual sound source based on data of a reception signal and sets a delay parameter. A virtual sound source generation signal processing section forms a sound emission beam based on the parameters and outputs the beam to a loudspeaker SPi.

Abstract: A system and method for providing microphonic isolation on a transmission line. The transmission line has a first part and a second part. The first part of transmission line carries a data signal and a microphonic signal. The microphonic signal has frequencies that include those in a range of substantially 20 Hz to substantially 20 kHz. The system includes an isolation apparatus. The isolation apparatus has an input in electrical communication with a first part of the transmission line, an output in electrical communication with the second part of the transmission line, and a filter in electrical communication with the input and the output. The filter is arranged to substantially remove the microphonic signal received at the input from first part of transmission line and pass the data signal to the output.

Abstract: Compensating for varying degrees of ambient noise in a mobile device. One possible noise compensation application has a plurality of selectable settings associated with a range of speeds of motion. This noise compensation application, when executed by the processor, configures the control circuit to: receive an input from the detector, detect an amplitude of the input, compare the amplitude with a selected one of the settings, and selectively adjust an amplification device when the detected amplitude is within a prescribed range. Another possible noise compensation application has plurality of selectable settings each associated with a range of audio frequencies. This noise compensation application, when executed by the processor, configures the control circuit to: receive an input from the detector, detect a frequency of the input, compare the frequency with a selected one of the settings, and selectively filter the ambient noise when the detected frequency is within a prescribed range.

Abstract: An architecture and framework for speech/noise classification of an audio signal using multiple features with multiple input channels (e.g., microphones) are provided. The architecture may be implemented with noise suppression in a multi-channel environment where noise suppression is based on an estimation of the noise spectrum. The noise spectrum is estimated using a model that classifies each time/frame and frequency component of a signal as speech or noise by applying a speech/noise probability function. The speech/noise probability function estimates a speech/noise probability for each frequency and time bin. A speech/noise classification estimate is obtained by fusing (e.g., combining) data across different input channels using a layered network model.

Abstract: Embodiments of the invention disclose computer-implemented methods, systems, and computer program products for estimating signal coherence. First, a sound generated by a sound source is detected by a first microphone to obtain a first microphone signal and by a second microphone to obtain a second microphone signal. The first microphone signal is filtered by a first adaptive finite impulse response filter to obtain a first filtered signal. The second microphone signal is filtered by a second adaptive finite impulse response filter, to obtain a second filtered signal. The coherence of the first filtered signal and the second filtered signal is determined based upon the filtered signals.

Abstract: An architecture and framework for speech/noise classification of an audio signal using multiple features with multiple input channels (e.g., microphones) are provided. The architecture may be implemented with noise suppression in a multi-channel environment where noise suppression is based on an estimation of the noise spectrum. The noise spectrum is estimated using a model that classifies each time/frame and frequency component of a signal as speech or noise by applying a speech/noise probability function. The speech/noise probability function estimates a speech/noise probability for each frequency and time bin. A speech/noise classification estimate is obtained by fusing (e.g., combining) data across different input channels using a layered network model.

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

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

Abstract: A noise control device includes a signal processor that detects a noise outputted from a noise source, and generates a control signal based on the noise and a control acoustic system that generates a control sound for canceling the noise, based on the control signal outputted from the signal processor. The noise control device also includes an output correction section that corrects the control signal outputted from the signal processor, in a frequency band for which a noise control process time ?, which is a time period from when the noise is outputted from the noise source to pass through the signal processor and the control acoustic system to when the control sound reaches the control point, is larger than a noise transfer time T, which is a time period from when the noise is outputted from the noise source to when the noise reaches the control point via the noise transfer system (?>T).

Abstract: A desired signal is extracted with a higher accuracy from a mixed signal wherein a plurality of signals are mixed. At the time of extracting a first signal from a first mixed signal and a second mixed signal, said first mixed signal and second mixed signal having the first signal and second signal mixed therein, an estimate value of the first signal in the past is obtained as a first estimate value, and an estimate value of the second signal in the past is obtained as a second estimate value. Then, a first isolation signal is generated by subtracting the second estimate value from the first mixed signal, and a second isolation signal is generated by subtracting the first estimate value from the second mixed signal. Then, the signal generated using the first isolation signal and the second isolation signal is outputted as the first signal.

Abstract: A conversation shielding system comprises sensors that detect the location of a confidential conversation and the presence and location of a potential eavesdropper, audio output devices that produce masking sounds to shield the conversation from the eavesdropper, and a controller that automatically controls the operation of the output devices in response to data from the sensors. An optional portable controller may manually engage the system. A method for shielding conversation comprises identifying a conversation to be shielded, detecting a potential eavesdropper, automatically determining masking sound types, locations and volume that will shield the conversation, directing emission of masking sounds from at least one audio output device in order to shield the conversation, including adjusting the masking sound type, location, and volume in response to movement of the conversation or the eavesdropper, and continuing to shield the conversation until it ends or the eavesdropper is no longer detected.

Abstract: An apparatus for sound enhancement has at least two microphones (9) that provide a directional microphone array which is arranged to be pointed in the direction of a sound source. The directional microphone array thereby receives sound emitted by the sound source and generates sound signals. A processor (20) processors the sound signals generated by the microphone array to enhance the sound received by the directional microphone array from the sound source relative to other sound received by the directional microphone array. The processor (20) generates a corresponding enhanced signal (ES). Loud speakers (22) reproduce the enhanced signal as audible sound. Furthermore, sound suppression devices (7, 7a) are provided to suppress ambient should from reaching the eardrums of the user. This sound suppression acts in conjunction with the directional microphone array and the processor (20) which enhance the SOI to provide a listening environment in which the SOI is enhanced.