Abstract: An audio separating apparatus obtains a matrix by performing time-frequency conversion on an input audio signal. The audio separating apparatus divides the obtained matrix into at least a basis matrix and an activity matrix, and classifies base spectra that configure the respective columns of the basis matrix into first base spectra corresponding to a target sound and second base spectra corresponding to a non-target sound.

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

Abstract: A device, system and method of playing back a digital audio stream wherein large amounts of pre-emphasis of the high frequencies is applied before the digital to analog conversion and before an interpolation or digital filter, followed by de-emphasis in the analog domain in order to yield better audio fidelity.

Abstract: The disclosure provides a network signal receiving system and a network signal receiving method. The network signal receiving system comprises: a high pass filter, a canceller, and an adder. The high pass filter is utilized for performing a high pass filtering operation for an audio data signal to output at least a signal corresponding to transitions of the audio data signal, wherein the audio data signal is synchronized with a network data signal. The canceller is coupled to the high pass filter, and utilized for generating a noise cancelling signal according to the at least a signal output by the high pass filter. The adder is coupled to the canceller, utilized for receiving the network data signal and the noise cancelling signal, so as to use the noise cancelling signal to cancel at least a noise in the network data signal, which is corresponding to the at least a signal.

Abstract: The embodiments herein disclose a device and a method for controlling noise in a wideband communication system. In one embodiment herein, multiple microphones for receiving wideband audio signals are provided. A processor is configured to analyze each wideband audio signal received by each microphone. Further, unique signal patterns are generated based on each analyzed wideband signals for each microphone and the unique signal patterns are compared to detect any identical signal patterns. A controller is also provided for controlling gains of those microphones that are detected to be receiving wideband audio signal of identical signal patterns.

Abstract: Provided are methods and systems for enhancing the intelligibility of an audio (e.g., speech) signal rendered in a noisy environment, subject to a constraint on the power of the rendered signal. A quantitative measure of intelligibility is the mean probability of decoding of the message correctly. The methods and systems simplify the procedure by approximating the maximization of the decoding probability with the maximization of the similarity of the spectral dynamics of the noisy speech to the spectral dynamics of the corresponding noise-free speech. The intelligibility enhancement procedures provided are based on this principle, and all have low computational cost and require little delay, thus facilitating real-time implementation.

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

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

Abstract: Null processing noise subtraction is performed per sub-band and time frame for acoustic signals received from multiple microphones. The acoustic signals may include a primary acoustic signal and one or more additional acoustic signals. A noise component signal may be determined for each additional acoustic signal in each sub-band of signals received by N microphones by subtracting a desired signal component within every other acoustic signal weighted by a complex-valued coefficient ? from the secondary acoustic signal. The noise component signals, each weighted by a corresponding complex-valued coefficient ?, may then be subtracted from the primary acoustic signal resulting in an estimate of a target signal (i.e., a noise subtracted signal).

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

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

Abstract: A method, an apparatus, and logic to post-process raw gains determined by input processing to generate post-processed gains, comprising using one or both of delta gain smoothing and decision-directed gain smoothing. The delta gain smoothing comprises applying a smoothing filter to the raw gain with a smoothing factor that depends on the gain delta: the absolute value of the difference between the raw gain for the current frame and the post-processed gain for a previous frame. The decision-directed gain smoothing comprises converting the raw gain to a signal-to-noise ratio, applying a smoothing filter with a smoothing factor to the signal-to-noise ratio to calculate a smoothed signal-to-noise ratio, and converting the smoothed signal-to-noise ratio to determine the second smoothed gain, with smoothing factor possibly dependent on the gain delta.

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

Abstract: In one embodiment, a directional microphone array having (at least) two microphones generates forward and backward cardioid signals from two (e.g., omnidirectional) microphone signals. An adaptation factor is applied to the backward cardioid signal, and the resulting adjusted backward cardioid signal is subtracted from the forward cardioid signal to generate a (first-order) output audio signal corresponding to a beampattern having no nulls for negative values of the adaptation factor. After low-pass filtering, spatial noise suppression can be applied to the output audio signal. Microphone arrays having one (or more) additional microphones can be designed to generate second- (or higher-) order output audio signals.

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

Abstract: Unlike sound based pressure waves that go everywhere, air turbulence caused by wind is usually a fairly local event. Therefore, in a system that utilizes two or more spatially separated microphones to pick up sound signals (e.g., speech), wind noise picked up by one of the microphones often will not be picked up (or at least not to the same extent) by the other microphone(s). Embodiments of methods and apparatuses that utilize this fact and others to effectively detect and suppress wind noise using multiple microphones that are spatially separated are described.

Abstract: In a system and method for maintaining the spatial stability of a sound field a balance gain may be calculated for two or more microphone signals. The balance gain may be associated with a spatial image in the sound field. Signal values may be calculated for each of the microphone. The signal values may be signal estimates or signal gains calculated to improve a characteristic of the microphone signals. The differences between the signal values associated with each microphone signal may be limited although some difference between signal values may be allowable. One or more microphone signals are adjusted responsive to the two or more balance gains and the signal gains to maintain the spatial stability of the sound field. The adjustments of one or more microphone signals may include mixing of two or more microphone. The signal gains are applied to the two or more microphone signals.

Abstract: In a system and method for maintaining the spatial stability of a sound field a balance gain may be calculated for two or more microphone signals. The balance gain may be associated with a spatial image in the sound field. Signal values may be calculated for each of the microphone. The signal values may be signal estimates or signal gains calculated to improve a characteristic of the microphone signals. The differences between the signal values associated with each microphone signal may be limited although some difference between signal values may be allowable. One or more microphone signals are adjusted responsive to the two or more balance gains and the signal gains to maintain the spatial stability of the sound field. The adjustments of one or more microphone signals may include mixing of two or more microphone. The signal gains are applied to the two or more microphone signals.

Abstract: In a system and method for maintaining the spatial stability of a sound field a balance gain may be calculated for two or more microphone signals. The balance gain may be associated with a spatial image in the sound field. Signal values may be calculated for each of the microphone. The signal values may be signal estimates or signal gains calculated to improve a characteristic of the microphone signals. The differences between the signal values associated with each microphone signal may be limited although some difference between signal values may be allowable. One or more microphone signals are adjusted responsive to the two or more balance gains and the signal gains to maintain the spatial stability of the sound field. The adjustments of one or more microphone signals may include mixing of two or more microphone. The signal gains are applied to the two or more microphone signals.

Abstract: Disclosed herein are systems, methods, and non-transitory computer-readable storage media for suppressing spatial noise based on phase information. The method transforms audio signals to frequency-domain data and identifies time-frequency points that have a parameter (e.g., signal-to-noise ratio) above a threshold. Based on these points, unwanted signals can be attenuated the desired audio source can be isolated. The method can work on a microphone array that includes two microphones or more.

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

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

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

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

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

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

Abstract: Psychoacoustic Bass Enhancement (PBE) is integrated with one or more other audio processing techniques, such as active noise cancellation (ANC), and/or receive voice enhancement (RVE), leveraging each technique to achieve improved audio output. This approach can be advantageous for improving the performance of headset speakers, which often lack adequate low-frequency response to effectively support ANC.

Abstract: A concept is proposed for a MEMS microphone which may be operated at a relatively low voltage level and still have comparatively high sensitivity. The component according to the present invention includes a micromechanical microphone structure having an acoustically active diaphragm which functions as a deflectable electrode of a microphone capacitor (1), and a stationary acoustically permeable counterelement which functions as a counter electrode of the microphone capacitor (1).

Abstract: A voice processing apparatus includes: a phase difference calculation unit which calculates for each frequency band a phase difference between first and second frequency signals obtained by applying a time-frequency transform to sounds captured by two voice input units; a detection unit which detects a frequency band for which the percentage of the phase difference falling within a first range that the phase difference can take for a specific sound source direction, the percentage being taken over a predetermined number of frames, does not satisfy a condition corresponding to a sound coming from the direction; a range setting unit which sets, for the detected frequency band, a second range by expanding the first range; and a signal correction unit which makes the amplitude of the first and second frequency signals larger when the phase difference falls within the second range than when the phase difference falls outside the second range.

Abstract: Methods and systems for masking audio noise are disclosed. One apparatus includes a silence detector configured to detect a period of substantial silence in an audio signal; a masking noise source operably coupled to the silence detector, the masking noise source configured to generate a noise signal in response to the silence detector detecting the period of substantial silence; and at least one combining device operably coupled to the masking noise source, the at least one combining device configured to contribute to combining the audio signal and the noise signal. A method includes detecting a period of substantial silence in an audio signal; and combining masking noise with the audio signal during the period of substantial silence.

Abstract: In response to a first envelope within a kth frequency band of a first channel, a speech level within the kth frequency band of the first channel is estimated. In response to a second envelope within the kth frequency band of a second channel, a noise level within the kth frequency band of the second channel is estimated. A noise suppression gain for a time frame n is computed in response to the estimated speech level for a preceding time frame, the estimated noise level for the preceding time frame, the estimated speech level for the time frame n, and the estimated noise level for the time frame n. An output channel is generated in response to multiplying the noise suppression gain for the time frame n and the first channel.

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

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

Abstract: A sound processing apparatus includes a target sound emphasizing unit configured to acquire a sound frequency component by emphasizing target sound in input sound in which the target sound and noise are included, a target sound suppressing unit configured to acquire a noise frequency component by suppressing the target sound in the input sound, a gain computing unit configured to compute a gain value to be multiplied by the sound frequency component using a gain function that provides a gain value and has a slope that are less than predetermined values when an energy ratio of the sound frequency component to the noise frequency component is less than or equal to a predetermined value, and a gain multiplier unit configured to multiply the sound frequency component by the gain value computed by the gain computing unit.

Abstract: Systems and methods for adaptive power control are provided. In exemplary embodiments, a primary signal is received. A noise power level of the primary signal is then estimated. The noise power level may then be compared to at least one power threshold. Subsequently, a large power consuming system is controlled based on the comparison of the noise power level to the power threshold.

Abstract: A sound quality correcting apparatus includes: an input module; a feature quantity calculator; a score calculator; a modulation spectrum power calculator; a score corrector; and a signal corrector. The input module receives an input audio signal. The feature quantity calculator calculates feature quantities of the input audio signal for each of a plurality of first intervals having a certain time length. The score calculator calculates a score value for each the first interval based on the feature quantities. The modulation spectrum power calculator calculates a power value, at a certain modulation frequency, of a modulation spectrum of the input audio signal. The score corrector corrects score values in the plurality of first intervals that belong to a second interval if a power value calculated in the second interval is larger than or equal to a certain value. The signal corrector corrects the audio signal based on the score values.

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 audio circuit.

Abstract: A device and method for manipulating an audio signal includes a windower for generating a plurality of consecutive blocks of audio samples, the plurality of consecutive blocks including at least one padded block of audio samples, the padded block having padded values and audio signal values, a first converter for converting the padded block into a spectral representation having spectral values, a phase modifier for modifying phases of the spectral values to obtain a modified spectral representation and a second converter for converting the modified spectral representation into a modified time domain audio signal.

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

Abstract: In order to reduce interference in an audio signal during a call on a mobile communication device, a plurality of transforms of the audio signal is performed, each transform containing phase information and amplitude information of corresponding samples of the audio signal. The results of the transforms are then averaged in order to generating a compensation signal that can be subtracted from the audio signal.

Abstract: Systems, methods, and apparatus to filter audio are disclosed. An example device includes first and second audio speakers having first audio characteristics, a third audio speaker having second audio characteristics, wherein the third speaker is positioned between the first and second audio speakers, a first audio filter to process an audio input signal to have a first frequency response including a first cutoff frequency, the first audio filter to output a first audio output signal to the first audio speaker, and a second audio filter to process the audio input signal to have a second frequency response to compensate for interference between the first and second frequency responses caused by a position of the first audio speaker relative to the second audio speaker.

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

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

Abstract: A system and method for audio bandwidth dependent noise suppression may detect the audio bandwidth of an audio signal responsive to one or more audio indicators. The audio indicators may include the audio sampling rate and characteristics of an associated compression format. Noise suppression gains may be calculated responsive to the audio signal. Noise suppression gains may mitigate undesirable noise in the reproduced output signal. The noise suppression gains may be modified responsive to the detected audio bandwidth. Less noise reduction may be desirable when more audio bandwidth is available. The modified noise suppression gains may be applied to the audio signal.

Abstract: A noise suppressing device includes a plurality of sound input units inputting sounds from a given sound source and converting the sounds to sound signals on a time axis, a transfer characteristic obtaining unit performing frequency transform of the sound signals after dividing the sound signals into frames and calculating respective transfer characteristics of the sounds for each given frequency band, a storage unit storing the calculated transfer characteristics of the sounds, a frequency obtaining unit obtaining a frequency for updating the transfer characteristics stored in the storage unit for the frequency band, an updating unit updating the transfer characteristics every given number of frames corresponding to the obtained frequency based on the transfer characteristics for each frequency band, a generating unit generating suppression information for suppressing the noise component based on the updated transfer characteristics, and a suppression unit suppressing the noise component based on the suppres

Abstract: A perceptually motivated, frequency domain active matrix decoder and decoding method which decodes N audio input signals to generate M audio output signals, where M is greater than N, including by generating M streams of output frequency components which determine the audio output signals, in response to N streams of input frequency components indicative of the audio input signals, determining power ratios from the input frequency components without use of feedback, including at least one power ratio for each critical frequency band in a set of critical frequency bands, and determining gain control values for each of the critical frequency bands from the power ratios including by shaping the power ratios in nonlinear fashion without use of feedback. An active matrix element is steered using the gain control values.

Abstract: Noise reduction is provided to audio captured by a headset by employing spatially separated microphones provided by a headset and a mobile phone. Primary audio is captured by a headset microphone and includes both voice audio and ambient noise. Secondary audio is captured by a mobile phone microphone and includes ambient audio. Noise reduction is performed using the primary and secondary audio to generate a noise-reduce audio.

Abstract: In a signal processing apparatus, a high frequency level detecting unit detects a level of a high frequency component adjusted with a first adjusting unit, and a first control unit controls a first gain of the adjusting unit according to the level of the high frequency component thus detected. Further, a low frequency level detecting unit detects a level of a low frequency component adjusted with a second adjusting unit. A second control unit controls a second gain according to the level of the high frequency component and the level of the low frequency component thus adjusted, so that a difference between the level of the high frequency component adjusted with the first adjusting unit and the level of the low frequency component adjusted with the second adjusting unit becomes smaller than a specific level determined in advance.

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