Abstract: A decorrelation method for improving feedback cancellation utilizes a small frequency shifting ratio, on the order of 0.3 percent. Frequency shifting is applied only to the high frequency portion of the signal, which is shifted alternately upward and downward.

Abstract: An audio enhancement system (1) for speech recognition or voice control is described, comprising a signal input for carrying a distorted desired signal (z), a reference signal input, and a spectral processor (SP) coupled to both signal inputs for processing the distorted desired signal (z) by means of a reference signal (x) acting as an estimate for the distortion of the desired signal. The spectral processor (SP) is equipped for said processing such that a factor C? is determined, whereby said estimate is a function of the factor C? times the spectral power of the reference signal (x), and the factor C is determined as the spectral ratio between those components of the signals z and x, which are essentially stationary with time. Such a factor determined by stationary parts of those signals makes application of a critical speech detector in the audio enhancement system superfluous.

Abstract: An information processing apparatus includes a feature value detecting section, an image processing section, and an audio processing section. The feature value detecting section determines, when a first image and a second image that are captured at different positions include a specific subject, a feature value of the subject included in the supplied first and second images. The image processing section detects motion of the subject on the basis of the feature value determined by the feature value detecting section. The audio processing section localizes a sound image of the subject in accordance with the motion of the subject detected by the image processing section.

Abstract: Methods, machines, systems and machine-readable instructions for processing input audio signals are described. In one aspect, an input audio signal has a noise period that includes a targeted noise signal and a noise-free period free of the targeted noise signal. The input audio signal in the noise-free period is divided into spectral time slices each having a respective spectrum. Ones of the spectral time slices of the input audio signal are selected based on the respective spectra of the spectral time slices. An output audio signal is composed for the noise period based at least in part on the selected ones of the spectral time slices of the input audio signal in the noise-free period.

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: The perceptual quality of voice signals used for Voice over IP (VoIP) systems is assessed. The content of the voice data packets may be altered in order to increase the perceived quality of the VoIP system.

Abstract: Method and system adapted to modifying an audio signal or speech signal comprising a step in which the frequency spectrum S(k) of the signal is converted by the application of a non-linear function. The method comprises at least the following steps: firstly, determining the signal level A(k), B(k) associated with a frequency k by taking account of different levels a(k), b(k) of the signal for the frequency k concerned and/or the neighboring frequencies (step 2a, step 8a); secondly, applying the non-linear function to said level A(k), B(k).

Abstract: The present invention provides a method for recovering target speech based on shapes of amplitude distributions of split spectra obtained by use of blind signal separation.

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: An input sound processor compares power at each frequency component of an input sound with a reference value, and sets multiplication points indicating frequency components at which the total power of the input sound is to be determined. A product-sum operation is performed at the multiplication points on the power at each frequency component and the square amplitude of each filter coefficient indicating the transfer characteristic from a loudspeaker to a microphone to estimate the total power of the input sound at the position of the microphone.

Abstract: An audio processing method used in a microphone is provided. Firstly, a sound signal is received. Next, the sound signal is transduced to a first voltage signal. The first voltage signal is interfered with by a second voltage signal resulting from electromagnetic wave penetrating into the microphone. Next, the second voltage signal is filtered out from the interfered first voltage signal. Finally, the filtered first voltage signal is amplified.

Abstract: A system improves speech intelligibility by reconstructing speech segments. The system includes a low-frequency reconstruction controller programmed to select a predetermined portion of a time domain signal. The low-frequency reconstruction controller substantially blocks signals above and below the selected predetermined portion. A harmonic generator generates low-frequency harmonics in the time domain that lie within a frequency range controlled by a background noise modeler. A gain controller adjusts the low-frequency harmonics to substantially match the signal strength to the time domain original input signal.

Abstract: A method for providing an embellishment representation of a noise information is discloses.

Abstract: A method of removing reverberation from audio signals is disclosed. The method comprises spectro-temporally analyzing the first audio signal and the second audio signal to derive an energy function of time for a plurality of frequency bands. The method further comprises determining a delay stability between the energy function of time for the first audio signal and the second audio signal in each band, determining a gain function in each band based on the delay stability, adjusting the energy of the first audio signal and the second audio signal using the gain function within each band, and resynthesizing audio signals from the energy in each band of the first audio signal and the second audio signal.

Abstract: A portable assistive listening system for enhancing sound for hearing impaired individuals includes a functional hearing aid and a separate handheld digital signal processing (DSP) device. The focus of the embodiments is directed to the handheld DSP device and a method of processing audio signals. The DSP device includes a programmable digital signal processor, a UWB transceiver for communicating with the hearing aid and/or other wireless audio sources, an LCD display, and a user input device (keypad). The handheld device is user programmable to apply different sound processing algorithms for processing sound signals received from the hearing aid and/or other audio source. The handheld device is capable of receiving audio signals from multiple sources, and gives the user control over selection of incoming sound sources and selective processing of audio.

Abstract: A wireless, multi-function audio gateway device provides communication between the headset and at least one audio gateway. The headset includes a housing having at least one multifunction button, and a first and a second microphone. The first microphone is located closer to a user's mouth than the second microphone. The headset also includes a flexible ear bud, a speaker, a volume control button, a rechargeable battery, a USB port, a detachable ear wrap, and a programmable baseband IC. The programmable baseband IC is configured for wireless communications to allow interfacing between the at least one audio gateway and the wireless headset. Methods for improved noise suppression, pairing and communicating with multiple audio gateways simultaneously, and allowing headset-to-headset communications between two wireless, multiple audio gateway headsets are also disclosed.

Abstract: The present invention is to provide a sound image localization apparatus which can prevent the lowering of the amplitude of the sound image localizing signal, the occurrence of clipping, and deterioration of the sound image localization component of the sound image localizing signal.

Abstract: The invention regards a method for noise reduction in an audio device whereby an electrical and/or digital signal which represents sound is routed simultaneously through:—a signal analysis path, and—a signal processing path wherein the signal amplification is individually controllable in specific frequency bands by attenuation values derived from the signal analysis path, whereby the signal in the signal analysis path is routed simultaneously through:—a first detector which identifies the presence of speech indicators in the overall signal, and—a second detector which in a predefined number of frequency bands detects the modulation amplitude, and—where attenuation values in each of the predefined frequency bands are calculated based on the combined results of the first detector and the modulation amplitude in the specific frequency band detected by the second detector,—where the attenuation values in the predefined number of frequency bands are routed to the signal processing path in order to attenuate the si

Abstract: The present invention provides an energy feature extraction method for noisy speech recognition. At first, noisy speech energy of an input noisy speech is computed. Next, the noise energy in the input noisy speech is estimated. Then, the estimated noise energy is subtracted from the noisy speech energy to obtain estimated clean speech energy. Finally, delta operations are performed on the log of the estimated clean speech energy to determine the energy derivative features for the noisy speech.

Abstract: An apparatus processes an acoustic input signal to provide an output signal with reduced noise. The apparatus weights the input signal based on a frequency-dependent weighting function. A frequency-dependent threshold function bounds the weighting function from below.

Abstract: Provides speech enhancement techniques which are effective even for extemporaneous noise without a noise interval and unknown extemporaneous noise. An example of a signal enhancement device includes: spectral subtraction means for subtracting a given reference signal from an input signal containing a target signal and a noise signal by spectral subtraction; an adaptive filter applied to the reference signal; and coefficient control means for controlling a filter coefficient of the adaptive filter in order to reduce components of the noise signal in the input signal. In the signal enhancement device, a database of a signal model concerning the target signal expressing a given feature by means of a given statistical model is provided, and the filter coefficient is controlled based on the likelihood of the signal model with respect to an output signal from the spectral subtraction means.

Abstract: A noise suppression system reduces low-frequency noise in a speech signal using linear predictive coefficients in an adaptive filter. A digital filter may update or adapt a limited set of linear predictive coefficients on a sample-by-sample basis. The linear predictive coefficients may be used to provide an error signal based on a difference between the speech signal and a delayed speech signal. The error signal represents an enhanced speech signal having attenuated and normalized low-frequency noise components.

Abstract: A mobile audio device (for example, a cellular telephone, personal digital audio player, or MP3 player) performs Audio Dynamic Range Control (ADRC) and Automatic Volume Control (AVC) to increase the volume of sound emitted from a speaker of the mobile audio device so that faint passages of the audio will be more audible. This amplification of faint passages occurs without overly amplifying other louder passages, and without substantial distortion due to clipping. Multi-Microphone Active Noise Cancellation (MMANC) functionality is, for example, used to remove background noise from audio information picked up on microphones of the mobile audio device. The noise-canceled audio may then be communicated from the device. The MMANC functionality generates a noise reference signal as an intermediate signal. The intermediate signal is conditioned and then used as a reference by the AVC process. The gain applied during the AVC process is a function of the noise reference signal.

Abstract: A communication end device of a two-way communication system is shown. The device includes an audio signal capture device for capturing local audio to be transmitted to another end device, an audio signal rendering device for playing remote audio received from the other end device, and buffers for buffering the captured and rendered audio signals. The device also includes an audio echo canceller operating to predict echo from the rendered audio signal at a calculated relative offset in the captured audio signal based on an adaptive filter, and subtract the predicted echo from the signal transmitted to the other end device The calculated relative offset that is used by the audio echo canceller for a current signal sample is adjusted if a difference between it and an adjusted relative offset of a preceding sample exceeds a threshold value.

Abstract: A noise suppressing apparatus suppresses a noise component of a sound signal which contains the noise component and a signal component. In the apparatus, a frequency analyzing section divides the sound signal into a plurality of frames such that adjacent frames overlap with each other along a time axis, and computes a first spectrum of each frame. A noise suppressing section suppresses a noise component of the first spectrum so as to provide a second spectrum of each frame in which the noise component is suppressed. A frequency specifying section specifies a frequency of a noise component of each frame. A phase controlling section varies a phase of the noise component corresponding to the specified frequency in the second spectrum by a different variation amount each frame.

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

Abstract: A system and method are disclosed for transient detection and modification in audio signals. Digital signal processing techniques are used to detect transients and modify an audio signal to enhance or suppress such transients, as desired. A transient audio event is detected in a first portion of the audio signal. A graded response to the detected transient audio event is determined. The first portion of the audio signal is modified in accordance with the graded response. The extent of enhancement or suppression (as applicable) may be determined at least in part by a measure of the significance or magnitude of the transient.

Abstract: The present invention relates to a wave-filtering device with an adjustable frequency bandwidth installed in an output unit of an integrated system. The wave-filtering device is used for receiving a first output signal comprising an alternating current signal and a direct current bias voltage, and comprises a direct current bias voltage detector for receiving the first output signal and obtaining a control voltage according to the direct current bias voltage. A low-pass wave filter is electrically connected to the direct current bias voltage detector and receives the control voltage so as to adjust the frequency bandwidth. The low-pass wave filter is used for receiving the first output signal and obtaining a second output signal to be the output of the wave-filtering device with adjustable frequency bandwidth. The low-pass wave filter comprises a variable resistance the value of which varies in accordance with the value of the control voltage.

Abstract: The present system proposes a technique called the spectro-temporal varying technique, to compute the suppression gain. This method is motivated by the perceptual properties of human auditory system; specifically, that the human ear has higher frequency resolution in the lower frequencies band and less frequency resolution in the higher frequencies, and also that the important speech information in the high frequencies are consonants which usually have random noise spectral shape. A second property of the human auditory system is that the human ear has lower temporal resolution in the lower frequencies and higher temporal resolution in the higher frequencies. Based on that, the system uses a spectro-temporal varying method which introduces the concept of frequency-smoothing by modifying the estimation of the a posteriori SNR. In addition, the system also makes the a priori SNR time-smoothing factor depend on frequency.

Abstract: Near-field sensing of wave signals, for example for application in headsets and earsets, is accomplished by placing two or more spaced-apart microphones along a line generally between the headset and the user's mouth. The signals produced at the output of the microphones will disagree in amplitude and time delay for the desired signal—the wearer's voice—but will disagree in a different manner for the ambient noises. Utilization of this difference enables recognizing, and subsequently ignoring, the noise portion of the signals and passing a clean voice signal. A first approach involves a complex vector difference equation applied in the frequency domain that creates a noise-reduced result. A second approach creates an attenuation value that is proportional to the complex vector difference, and applies this attenuation value to the original signal in order to effect a reduction of the noise. The two approaches can be applied separately or combined.

Abstract: A method for reducing audio noise in an audio signal acquisition is described herein. The method includes: receiving an input audio signal; separating the input audio signal into a high-frequency portion and a low-frequency portion based on a threshold frequency; synthesizing the low-frequency portion to at least reduce any audio noise therein to generate a new low-frequency portion; combining the high-frequency portion and the new low-frequency portion to form a new audio signal representing the input audio signal; and outputting the new audio signal for the audio signal acquisition.

Abstract: A method for effecting aliasing cancellation in an audio effects algorithm using a delay modulated signal, derived from interpolation of a delay modulator at an instantaneous sampling frequency, including: determining the instantaneous sampling frequency 1/Tisf and band limiting an input signal, to which the audio effects algorithm is to be applied to ½ Tisf prior to interpolation.

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: A device for detecting the presence of wind noise in an array of microphones including two or more separate sound inlet openings, and a sound to electrical converting element or microphone in relation to each sound inlet opening. The microphones each generate time dependent signals which are fed to a signal processing device that provides one or more output signals. The signal processing device has means for generating a time dependent cross correlation function between a first and a second microphone signal, and means for generating a signal corresponding to a time dependent auto correlation function of either the first or the second of the microphone signals.

Abstract: A method and apparatus for de-noising weak bio-signals having a relatively low signal to noise ratio utilizes an iterative process of wavelet de-noising a data set comprised of a new set of frames of wavelet coefficients partially generated through a cyclic shift algorithm. The method preferably operates on a data set having 2N frames, and the iteration is performed N?1 times. The resultant wavelet coefficients are then linearly averaged and an inverse discrete wavelet transform is performed to arrive at the de-noised original signal. The method is preferably carried out in a digital processor.

Abstract: A noise reduction system is used in a BTSC system to reduce noise of an audio signal. The noise reduction system has an audio spectral compressing unit that has a filter and a memory in the approach of the digital processing. The filter is arranged to filter an input signal according to a transfer function, a variable d, and several parameters b0/a0, a0/b0, b1/b0 and a1/a0. The memory is arranged to store the parameters.

Abstract: An electronic device can be operated to detect noise, such as wind noise. A microphone signal is generated by a microphone. Autocorrelation coefficients are determined based on the microphone signal. Gradient values are determined from the autocorrelation coefficients.

Abstract: An amplitude suppression quantity denoting a noise suppression level of a current frame is calculated in an amplitude suppression quantity calculating unit (20), a perceptual weight distributing pattern of both a spectral subtraction quantity and a spectral amplitude suppression quantity is determined in a perceptual weight pattern adjusting unit (21), the spectral subtraction quantity and the spectral amplitude suppression quantity given by the perceptual weight distributing pattern are corrected according to a frequency band SN ratio in a perceptual weight correcting unit (7), a noise subtracted spectrum is calculated from an amplitude spectrum, a noise spectrum and a corrected spectral subtraction quantity in a spectrum subtracting unit (8), and a noise suppressed spectrum is calculated from the noise subtracted spectrum and a corrected spectral amplitude suppression quantity in a spectrum suppressing unit (9).

Abstract: A method and apparatus for de-noising weak bio-signals having a relatively low signal to noise ratio utilizes an iterative process of de-noising a data set comprised of a new set of frames. The method separately performs a non-linear de-noising operation on each of the component frames and combines the resultant de-noised frames to form a combined resultant de-noised input signal. The method is preferably carried out in a digital processor.

Abstract: A method is presented for eliminating an unwanted signal (e.g., background music, interference, etc.) from a mixture of a desired signal and the unwanted signal via time-frequency masking. Given a mixture of the desired signal and the unwanted signal, the goal of the present invention is to eliminate or at least reduce the effects of the unwanted signal to obtain an estimate of the desired signal.

Abstract: A system and method are disclosed for enhancing audio signals by nonlinear spectral operations. Successive portions of the audio signal are processed using a subband filter bank. A nonlinear modification is applied to the output of the subband filter bank for each successive portion of the audio signal to generate a modified subband filter bank output for each successive portion. The modified subband filter bank output for each successive portion is processed using an appropriate synthesis subband filter bank to construct a modified time-domain audio signal. High modulation frequency portions of the audio signal may be emphasized or de-emphasized, as desired. The modification may be applied within one or more frequency bands.

Abstract: A first band analyzer divides an acoustic signal received from a sound playback system through an input unit into frequency bands, and generates a first band level. An acoustic signal estimator estimates the band level of the original acoustic signal at the input unit, and generates a second band level for each band. A processor extracts an external noise component which is contained in the acoustic signal using the first band level and the second band level. The external noise can be accurately estimated with less computation than in the related art.

Abstract: According to a disclosed embodiment, an auscultatory training apparatus includes a database of pre-recorded physiological sounds stored on a computer for playing on a playback system. A user-friendly, graphical user interface software program is stored on the computer for use with a conventional computer mouse. The program allows a user to select one of the pre-recorded sounds for playback. In addition, the program is operable to generate an inverse model of the playback system in the form of a digital filter. If employed by the user, the inverse model processes the selected sound to cancel the distortions of the playback system so that the sound is accurately reproduced in the playback system. The program also permits the extraction of a specific sound component from a pre-recorded sound so that only the extracted sound component is audible during playback.

Abstract: A signal encoding apparatus (10) limits an inputted time series signal to a low frequency band signal having a certain cut-off frequency or less to include the low frequency band signal into code train for outputting encoded low frequency band code train. In addition, the signal encoding apparatus (10) adaptively determines aliasing frequency fa, shift frequency fsh or tone•noise synthesis information r used for generation of high frequency band signal at the decoding side to include these information into code train outputted along with high frequency band spectrum envelope information as high frequency band generation information.

Abstract: In an apparatus which estimates characteristics of a surrounding noise only when an input signal is soundless and performs a noise reduction or suppression of the input signal based on the estimated result, a signal noise ratio is estimated from the input signal, and an automatic switch or an automatic adjustment is performed so as to execute a noise reduction only when the signal noise ratio is good, otherwise to avoid the noise reduction or make the noise reduction degree smaller.

Abstract: A howling control apparatus and howling control method for controlling time up to the cancellation of howling suppression according to a howling occurrence situation, thereby eliminating the repetition of howling suppression and cancellation. A howling detecting section (104) detects howling based on the band level and the band level average value and measures time for which no howling occurs, a waiting time setting section (105) decides waiting time to be set this time from time for which no howling occurs and a previous waiting time, and a gain control section (106) causes a gain that is set to a howling suppressing section (107) to be retuned to a normal value during the waiting time.

Abstract: Systems and methods that enable high quality audio teleconferencing are disclosed. In one embodiment of the present invention, a signal processor receives signals from a spatially dispersed set of directional microphones, processing the microphone signals and the far-end received audio into a signal for transmission to a far-end party. The processing may comprise the use of one or more algorithms that reduce conference room noise and may selectively increase participant audio levels by processing the microphone signals using beamforming techniques. An embodiment of the present invention may also comprise one or more omni-directional microphones that may be used in cooperation with the directional microphones to adjust for background noise, acoustic echo, and the existence of side conversations.

Abstract: Techniques are provided to suppress noise and interference using an array microphone and a combination of time-domain and frequency-domain signal processing. In one design, a noise suppression system includes an array microphone, at least one voice activity detector (VAD), a reference generator, a beam-former, and a multi-channel noise suppressor. The array microphone includes multiple microphones—at least one omni-directional microphone and at least one uni-directional microphone. Each microphone provides a respective received signal. The VAD provides at least one voice detection signal used to control the operation of the reference generator, beam-former, and noise suppressor. The reference generator provides a reference signal based on a first set of received signals and having desired voice signal suppressed. The beam-former provides a beam-formed signal based on a second set of received signals and having noise and interference suppressed.

Abstract: An object is to provide an automatic wind noise reducing circuit and an automatic wind noise reducing method, which are capable of coping with a multichanneling trend of audio signals, and improving performance as well as the degree of freedom in system design. Arithmetic units (26, 27, 28) obtain added signals of audio signals of audio channels excluding respective selected audio channels, which are selected so as to be different from each other. Arithmetic units (29, 30, 31) subtract respective added signals of the arithmetic units (26, 27, 28) from corresponding audio signals of the selected audio channels. The subtraction signals from the arithmetic units (29, 30, 31) are subjected to a band limit control and limited to a frequency band of a wind noise signal by LPFs (21, 23, 25).