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Небесная энциклопедия

Космические корабли и станции, автоматические КА и методы их проектирования, бортовые комплексы управления, системы и средства жизнеобеспечения, особенности технологии производства ракетно-космических систем

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Мониторинг СМИ

Мониторинг СМИ и социальных сетей. Сканирование интернета, новостных сайтов, специализированных контентных площадок на базе мессенджеров. Гибкие настройки фильтров и первоначальных источников.

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Форма поиска

Поддерживает ввод нескольких поисковых фраз (по одной на строку). При поиске обеспечивает поддержку морфологии русского и английского языка
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Применить Всего найдено 13486. Отображено 100.
05-01-2012 дата публикации

Speech audio processing

Номер: US20120004909A1
Автор: Anand P. Rangarajan, Arijit Raychowdhury, Matias Zanartu, Michael E. Deisher, Willem M. Beltman
Принадлежит: Intel Corp

A speech processing engine is provided that in some embodiments, employs Kalman filtering with a particular speaker's glottal information to clean up an audio speech signal for more efficient automatic speech recognition.

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Номер записи: 1
12-01-2012 дата публикации

Sound processing device and sound processing method

Номер: US20120008797A1
Автор: Takeo Kanamori, Yutaka Banba
Принадлежит: Panasonic Corp

A sound processing apparatus ( 100 ), which can improve precision of analyses on ambient sounds, carries out analysis on the ambient sounds based upon collected sound signals acquired by two sound collectors (first sound collector 110 - 1 and second sound collector 110 - 2 ), and the sound processing apparatus ( 100 ) is provided with a level signal conversion section (first level signal conversion section 130 - 1, second level signal conversion section 130 - 2 ) that converts the collected sound signal into a level signal, from which phase information is removed, a level signal synthesizing section ( 140 ) that generates a synthesized level signal in which the level signals acquired from the collected sound signals of the two sound collectors (first sound collector 110 - 1 and second sound collector 110 - 2 ) are synthesized, and a detecting and identifying section ( 160 ) that carries out analysis on the ambient sounds based upon the synthesized level signal.

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Номер записи: 2
26-01-2012 дата публикации

Noise canceller and noise cancellation program

Номер: US20120020489A1
Автор: Tomohiro Narita
Принадлежит: Mitsubishi Electric Corp

A directivity control unit 10 calculates a main beam signal with its directivity turned toward an object sound direction and a sub-beam signal with its blind spot turned toward the object sound direction from output signals of a plurality of microphones 2 and 3 through signal processing, and a frequency analyzing unit 20 converts them to spectra. A sound source decision unit 30 decides on whether a sound source is voice, stationary noise or unstationary noise from the spectra of the main beam signal and sub-beam signal and outputs as a sound source decision result, and calculates the average spectrum which is a statistic of noise for the main beam signal. An interfering sound removing unit 50 subtracts the average spectrum from the spectrum of the main beam signal to remove interfering sounds.

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Номер записи: 3
23-02-2012 дата публикации

Sound source separation apparatus and sound source separation method

Номер: US20120045066A1
Автор: Hirofumi Nakajima, Kazuhiro Nakadai
Принадлежит: Honda Motor Co Ltd

A sound source separation apparatus includes a transfer function storage unit that stores a transfer function from a sound source, a sound change detection unit that generates change state information indicating a change of the sound source on the basis of an input signal input from a sound input unit, a parameter selection unit that calculates an initial separation matrix on the basis of the change state information generated by the sound change detection unit, and a sound source separation unit that separates the sound source from the input signal input from the sound input unit using the initial separation matrix calculated by the parameter selection unit.

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Номер записи: 4
03-05-2012 дата публикации

Handheld electronic device with microphone array

Номер: US20120106755A1
Автор: Ming Zhang
Принадлежит: Fortemedia Inc

A handheld electronic device includes a body and a microphone array. The body includes a side and a recess formed on the side. The microphone array includes a first microphone and a second microphone. Either the first and second microphones are disposed in the recess, or the first microphone is disposed in the recess while the second microphone is disposed outside the recess.

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Номер записи: 5
17-05-2012 дата публикации

Post-noise suppression processing to improve voice quality

Номер: US20120123775A1
Автор: Carlo Murgia, Scott Isabelle
Принадлежит: Individual

Provided are methods and systems for improving quality of speech communications. The method may be for improving quality of speech communications in a system having a speech encoder configured to encode a first audio signal using a first set of encoding parameters associated with a first noise suppressor. A method may involve receiving a second audio signal at a second noise suppressor which provides much higher quality noise suppression than the first noise suppressor. The second audio signal may be generated by a single microphone or a combination of multiple microphones. The second noise suppressor may suppress the noise in the second audio signal to generate a processed signal which may be sent to a speech encoder. A second set of encoding parameters may be provided by the second noise suppressor for use by the speech encoder when encoding the processed signal into corresponding data.

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Номер записи: 6
31-05-2012 дата публикации

Noise suppression apparatus, method, and a storage medium storing a noise suppression program

Номер: US20120134509A1
Автор: Chikako Matsumoto
Принадлежит: Fujitsu Ltd

A noise suppression apparatus includes: a conversion unit to convert a recorded sound signal in a time domain into a spectrum in a frequency domain; a setting unit to set a suppression gain indicating a degree of suppression on each spectrum for each frequency spectrum on the basis of a nonstationarity-value variation in time of the respective spectrum; a suppression unit to suppress each of the spectrum on the basis of the suppression gain set by the setting unit for each frequency spectrum; and an inverse conversion unit to perform an inverse conversion to the conversion by the conversion unit on the spectrum having been subjected to the suppression processing by the suppression unit.

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Номер записи: 7
21-06-2012 дата публикации

Sound processing apparatus and recording medium storing a sound processing program

Номер: US20120155674A1
Автор: Naoshi Matsuo
Принадлежит: Fujitsu Ltd

A sound processing apparatus includes a first calculator that calculates first power based on a first signal received by a first microphone that is among the first microphone and a second microphone; a second calculator that calculates second power based on a second signal received by the second microphone; a gain calculator that calculates a gain on the basis of the ratio of the first power to the second power; and a multiplier that processes the second signal using the gain calculated by the gain calculator.

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Номер записи: 8
04-10-2012 дата публикации

Noise removal device and noise removal program

Номер: US20120250883A1
Автор: Tomohiro Narita
Принадлежит: Mitsubishi Electric Corp

A noise removal unit 102 executes noise removal and flooring processing of an input signal, and a density calculating unit 104 calculates, as to a point of interest on a time-frequency plane of the input signal passing through the noise removal, a density of non-flooring processing points from the presence or absence of the flooring processing of individual points around the point of interest. A partial suppression unit 105 replaces, when the density is less than a threshold, the power of the point of interest with its flooring value by considering it as a musical noise component, thereby suppressing the musical noise component.

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Номер записи: 9
25-10-2012 дата публикации

Signal demultiplexing device, signal demultiplexing method and non-transitory computer readable medium storing a signal demultiplexing program

Номер: US20120269203A1
Автор: Kyota Higa, Toshiyuki Nomura
Принадлежит: NEC Corp

Provided is a signal demultiplexing system that can minimize losses in demultiplexing performance even if signals unsuited to demultiplexing are inputted. The provided signal demultiplexing device contains: an input signal analysis means for determining whether or not a plurality of input signals are suited to demultiplexing; a data memory means for storing data from frequency-domain input signals which result from transformation of the aforementioned input signals into frequency-domain signals; a selection control means for storing the frequency-domain input signals in the data memory means if the input signal analysis means has determined that the input signals are suited to the generation of a demultiplexing matrix for demultiplexing; and a demultiplexing matrix generation means for generating a demultiplexing matrix using frequency-domain input signals including the most recent and older frequency-domain input signals stored in the data memory means.

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Номер записи: 10
15-11-2012 дата публикации

Method and apparatus for processing multi-channel de-correlation for cancelling multi-channel acoustic echo

Номер: US20120288100A1
Автор: Nam-gook CHO
Принадлежит: SAMSUNG ELECTRONICS CO LTD

Provided are a method and apparatus for multi-channel de-correlation processing for cancelling a multi-channel acoustic echo. The method includes: dividing an input multi-channel audio signal into units of frames to form multi-channel audio signals in units of frames; analyzing eigen values and eigen vectors related to the multi-channel audio signals by using the multi-channel audio signals in units of frames every time contents are modified; and separating the multi-channel audio signals in units of frames into a plurality of signal component spaces by using the analyzed eigen values and eigen vectors.

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Номер записи: 11
31-01-2013 дата публикации

Methods and apparatuses for convolutive blind source separation

Номер: US20130031152A1
Автор: Xuejing Sun
Принадлежит: Dolby Laboratories Licensing Corp

Methods and apparatuses for convolutive blind source separation are described. Each of a plurality of input signals is transformed into frequency domain. Values of coefficients of unmixing filter corresponding to frequency bins are calculated by performing a gradient descent process on a cost function at least dependent on the coefficients of the unmixing filters. In each iteration of the gradient descent process, gradient terms for calculating the values of the same coefficient of the unmixing filters are adjusted to improve smoothness of gradient terms across the frequency bins. With respect to each of the frequency bins, source signals are estimated by filtering the transformed input signals through the respective unmixing filter configured with the calculated values of the coefficients. The estimated source signals on the respective frequency bins are transformed into time domain. The cost function is adapted to evaluate decorrelation between the estimated source signals.

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Номер записи: 12
14-02-2013 дата публикации

Methods and apparatuses for echo cancelation with beamforming microphone arrays

Номер: US20130039504A1
Автор: Ashutosh Pandey, Darrin T. Thurston, David K. Lambert, Tracy A. Bathurst
Принадлежит: ClearOne Communications Inc

Embodiments include methods and apparatuses for sensing acoustic waves for a conferencing application. A conferencing apparatus includes a plurality of microphones oriented to cover a corresponding plurality of direction vectors and to develop a corresponding plurality of microphone signals. A processor is operably coupled to the plurality of microphones. The processor is configured to perform a beamforming operation to combine the plurality of microphone signals to a plurality of combined signals that is greater in number than one and less in number than the plurality of microphone signals. The processor is also configured perform an acoustic echo cancelation operation on the plurality of combined signals to generate a plurality of combined echo-canceled signals and select one of the plurality of combined echo-canceled signals for transmission.

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Номер записи: 13
14-02-2013 дата публикации

Spatio-temporal speech enhancement technique based on generalized eigenvalue decomposition

Номер: US20130041659A1
Автор: Malay Gupta, Scott C. DOUGLAS
Принадлежит: Individual

Described herein is a speech enhancement method using microphone arrays and a new iterative technique for enhancing noisy speech signals under low signal-to-noise-ratio (SNR) environments. Included is the processing of observed noisy speech both in the spatial- and the temporal-domains to enhance the desired signal component speech and an iterative technique to compute the generalized eigenvectors of the multichannel data derived from the microphone array. The entire processing is done on the spatio-temporal correlation coefficient sequence of the observed data in order to avoid large matrix-vector multiplications. Also described is a speech enhancement system having two stages. In the first stage, the noise component of the observed signal is whitened, and in the second stage a spatio-temporal power method is used to extract the most dominant speech component. In both the stages, the filters are adapted using the multichannel spatio-temporal correlation coefficients of the data.

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Номер записи: 14
21-02-2013 дата публикации

Method, System and Computer Program Product for Suppressing Noise Using Multiple Signals

Номер: US20130046535A1
Автор: Devangi Nikunj Parikh, Muhammad Zubair Ikram, Takahiro Unno
Принадлежит: Texas Instruments Inc

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.

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Номер записи: 15
28-02-2013 дата публикации

Array microphone apparatus for generating a beam forming signal and beam forming method thereof

Номер: US20130051577A1
Автор: Alessandro Morcelli, Marco Veneri
Принадлежит: STMICROELECTRONICS SRL

Embodiments described in the present disclosure relate to an array microphone apparatus for generating a beam forming signal. The apparatus includes first, second, and third omni-directional microphones, each converting an audible signal into a corresponding electrical signal. The second omni-directional microphone is disposed between the other two omni-directional microphones. The apparatus includes a first directional microphone forming device to jointly output a first directional microphone signal with a first bi-directional pattern, and a magnitude and phase response handler device to output a second directional microphone signal with an omni-directional pattern shifted by a prefixed value with respect to first directional microphone signal. The apparatus further includes a combining device receiving the first and second directional microphone signals and outputting a combined directional microphone signal with a combined beam pattern correlated to the first bi-directional and second omni-directional patterns, the combined directional microphone signal being in a broadside configuration.

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Номер записи: 16
14-03-2013 дата публикации

Speech enhancement method

Номер: US20130066626A1
Автор: Hsien Cheng Liao
Принадлежит: Industrial Technology Research Institute ITRI

A speech enhancement method is disclosed. The method includes the steps of: receiving a plurality of frames of sound signals by a microphone array; calculating an inter-aural time difference for each frequency band of each frame of the sound signals corresponding to at least one two-microphone set of the microphone array; calculating a plurality of values of cumulative histograms according to the calculated inter-aural time differences; determining a first inter-aural time difference threshold according to the calculated value of the cumulative histograms; and filtering the plurality of frames of sound signals according to the first inter-aural time difference threshold.

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Номер записи: 17
14-03-2013 дата публикации

Apparatus and method for suppressing noise from voice signal by adaptively updating wiener filter coefficient by means of coherence

Номер: US20130066628A1
Автор: Katsuyuki Takahashi
Принадлежит: Oki Electric Industry Co Ltd

A voice signal processor detects background noise sections to reflect characteristics of the background noise on the Wiener filter coefficient to be used for suppressing noise components of input voice signals. In the voice signal processor, directivity signal generators form directivity signals having a directivity pattern. The directivity signals are used by a coherence calculator to obtain coherence, which is in turn used by a targeted voice section detector to detect a targeted voice section. A background noise section detector detects background noise sections containing no voice signal. When a background noise section is detected, a WF adapter uses characteristics of background noise in the detected temporal section to calculate a new WF coefficient.

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Номер записи: 18
28-03-2013 дата публикации

Information Terminal, Server Device, Searching System, and Searching Method Thereof

Номер: US20130080056A1
Автор: Fukunaga Kouichirou, HOMMA Takeshi, Motohashi Masataka, Obuchi Yasunari, Watarai Norio
Принадлежит:

An object of the present invention is to provide a technique of an information terminal that allows more efficient utilization of high-level searching functions. The information terminal is provided with an audio input accepting unit to accept an input of speech information, a communication unit to establish communication with a predetermined server device via a network, an output unit, a POI specifying unit to transmit the speech information accepted by the audio input accepting unit to the server device and receive information specifying a candidate of a POI (Point Of Interest) associated with the speech information, a POI candidate output unit to output to the output unit, the information specifying the candidate of the POI received by the POI specifying unit, and a route searching unit to accept a selective input of the information specifying the candidate of the POI, and search for a route directed to the POI. 1. An information terminal comprising ,an audio input accepting unit adapted to accept an input of speech information,a communication unit adapted to establish communication with a predetermined server device via a network,an output unit,a POI specifying unit adapted to transmit the speech information accepted by the audio input accepting unit to the server device, and receive information specifying a candidate of a POI (Point Of Interest) associated with the speech information,a POI candidate output unit adapted to output to the output unit, the information specifying the candidate of the POI received by the POI specifying unit, anda route searching unit adapted to accept a selective input of the information specifying the candidate of the POI, and search for a route directed to the POI, wherein,the POI specifying unit compresses the speech information accepted by the audio input accepting unit with a predetermined compression rate and transmits the speech information being compressed to the server device, andwhen a quality of the speech information ...

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Номер записи: 19
28-03-2013 дата публикации

Speech Enhancement with Minimum Gating

Номер: US20130080158A1
Автор: Hetherington Phillip A., Li Xueman, Paranjpe Shreyas
Принадлежит: QNX Software Systems Limited

A speech enhancement system enhances transitions between speech and non-speech segments. The system includes a background noise estimator that approximates the magnitude of a background noise of an input signal that includes a speech and a non-speech segment. A slave processor is programmed to perform the specialized task of modifying a spectral tilt of the input signal to match a plurality of expected spectral shapes selected by a Codec. 1. A system , comprising:a speech enhancement processor configured to receive an input signal and output a processed signal; andan encoder device coupled with the speech enhancement processor and configured to receive the processed signal from the speech enhancement processor, where the encoder device supports one or more spectral shapes to encode the processed signal for transmission over a communication channel;where the speech enhancement processor is configured to modify a spectral tilt of the input signal, based on a spectral tilt associated with at least one of the one or more spectral shapes supported by the encoder device, to generate the processed signal.2. The system of claim 1 , where the speech enhancement processor is configured to modify the spectral tilt of the input signal in response to a determination that an input noise tilt of the input signal surpasses a maximum tilt limitation that is based on one or more spectral shapes available at the encoder device.3. The system of claim 1 , where the encoder device is configured to perform a comparison between the processed signal that has a modified spectral tilt and a plurality of spectral shapes that represent comfort noise; andwhere the encoder device is configured to select, based on the comparison, a spectral shape of the plurality of spectral shapes that represent comfort noise for transmission over the communication channel.4. The system of claim 1 , where the speech enhancement processor is configured to modify the spectral tilt of the input signal by maintaining ...

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Номер записи: 20
18-04-2013 дата публикации

Satellite Microphones for Improved Speaker Detection and Zoom

Номер: US20130093831A1
Автор: Cutler Ross G.
Принадлежит: MICROSOFT CORPORATION

Architecture for exploiting satellite microphones and employing other techniques of conference room camera/microphone systems to significantly improve the true positive rate (reduce false positives) in sound source localization (SSL). Techniques for realizing the improvement include using an LED emitter to determine the precise location of the satellite microphones on a table, using the base SSL and external sounds to determine the approximate location of the satellite microphone on the table, using the satellite microphone phase to improve the SSL performance, using the satellite microphone amplitude to improve the active speaker detector (ASD) performance, and using the satellite microphones to estimate camera zoom. 1. A method of detecting a speaker comprising:extending audio range of a local microphone array of a conferencing base station using satellite microphones;computing location information for location of the satellite microphones relative to the conferencing base station; andprocessing the location information and audio signals of the local microphone array and the satellite microphones for automatic detection of a speaker relative to the conferencing base station.2. The method of claim 1 , further comprising locating the satellite microphones on opposing regions of the conferencing base station according to corresponding nominal locations and estimating location of a satellite microphone relative to the regions based on audio power.3. The method of claim 1 , further comprising uniquely identifying each of the satellite microphones using emitters that transmit unique wireless signals for capture by a camera system of the conferencing base station and image processing to identify the satellite microphones.4. The method of claim 1 , further comprising estimating the location distances of the satellite microphones from the conferencing base station based on camera images of the satellite microphones claim 1 , vertical field of view of a camera claim 1 , and ...

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Номер записи: 21
18-04-2013 дата публикации

Method and device for phase-sensitive processing of sound signals

Номер: US20130094664A1
Автор: Dietmar Ruwisch
Принадлежит: Individual

A method and device for phase-sensitive processing of sound signals of at least one sound source may include arranging two microphones at a distance d from each other, capturing sound signals with both microphones, generating associated microphone signals, and processing the sound signals of the microphones. During a calibration mode, a calibration-position-specific, frequency-dependent phase difference vector φ0(f) between the associated calibration microphone signals may be calculated from their frequency spectra for the calibration position. Then, during an operating mode, a signal spectrum S of a signal to be output is calculated by multiplication of at least one of the two frequency spectra of the current microphone signals with a spectral filter function F.

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Номер записи: 22
18-04-2013 дата публикации

System And Method For Utilizing Inter-Microphone Level Differences For Speech Enhancement

Номер: US20130096914A1
Автор: Avendano Carlos, Santos Peter, Watts Lloyd
Принадлежит:

Systems and methods for utilizing inter-microphone level differences to attenuate noise and enhance speech are provided. In exemplary embodiments, energy estimates of acoustic signals received by a primary microphone and a secondary microphone are determined in order to determine an inter-microphone level difference (ILD). This ILD in combination with a noise estimate based only on a primary microphone acoustic signal allow a filter estimate to be derived. In some embodiments, the derived filter estimate may be smoothed. The filter estimate is then applied to the acoustic signal from the primary microphone to generate a speech estimate. 1. A method for enhancing speech , comprising:receiving a primary acoustic signal at a primary microphone and a secondary acoustic signal at a secondary microphone;determining a filter estimate during a frame, the filter estimate based on a noise estimate of the primary acoustic signal, an energy estimate of the primary acoustic signal, and an inter-microphone level difference based on the primary and secondary acoustic signals; andapplying the filter estimate to the primary acoustic signal to produce a speech estimate.2. The method of further comprising determining an energy estimate for each of the acoustic signals during the frame.3E. The method of wherein the energy estimate of the primary acoustic signal is approximated as E(t claim 2 ,ω)=|X(t claim 2 ,ω)|+(1−λ)(t−1 claim 2 ,ω).4|XE. The method of wherein the energy estimate of the secondary acoustic signal is approximated as E(t claim 2 ,ω)=λ(t claim 2 ,ω)|+(1−λ)(t−1 claim 2 ,ω).5. The method of further comprising using the energy estimates to determine the inter-microphone level difference for the frame.8. The method of wherein the noise estimate is based on an energy estimate of the primary acoustic signal and the inter-microphone level difference.9. The method of wherein the noise estimate is approximated as{'br': None, 'i': N', 't', 't', 'E', 't', 't', 'N', 't−', 'E', 't, ' ...

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Номер записи: 23
02-05-2013 дата публикации

MICROPHONE

Номер: US20130108074A1
Автор: Reining Friedrich
Принадлежит: KNOWLES ELECTRONICS ASIA PTE. LTD.

A microphone comprises a sound inlet, a first MEMS microphone structure comprising a first membrane and associated backplate in the path of the sound inlet and a second MEMS microphone structure comprising a second membrane and associated backplate stacked with respect to the first microphone membrane, with an enclosed space between the first and second membranes. 1. A microphone comprising:a sound inlet;a first microphone structure comprising a first membrane and associated backplate in the path of the sound inlet;a second microphone structure comprising a second membrane and associated backplate stacked with respect to the first microphone structure, with an enclosed space between the first and second membranes,wherein the microphone structures are back to back, such that movement of one membrane towards its associated backplate is accompanied by movement of the other membrane away from its associated backplate; anda signal processor for combining microphone signals of the first and second microphone structures to generate the microphone output; a non-linear microphone signal response of one of the microphone structures; and', 'pull-in of the membranes of one or both of the microphone structures., 'characterized in that the signal processor is adapted to combine the microphone signals of the first and second microphone structures to detect at least one of2. (canceled)3. (canceled)4. (canceled)5. (canceled)6. A microphone as claimed in claim 1 , wherein the signal processor is adapted to determine the presence of a non-linear microphone signal response based on a threshold applied to the combined microphone signals.7. A microphone as claimed in claim 1 , further comprising means for:adjusting the microphone electrical load; orselecting different parts of the microphone signals of the first and second microphone structures to derive the microphone output signal,based on the detection of the non-linear microphone signal response.8. (canceled)9. (canceled)10. A method ...

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Номер записи: 24
09-05-2013 дата публикации

PIEZOELECTRIC MICROPHONES

Номер: US20130114822A1
Автор: Fazzio R. Shane, Goel Atul
Принадлежит: Avago Technologies General IP (Singapore) Pte. Ltd.

Electronic devices and microphone devices are described. 1. An electronic device , comprising:a first microphone operative to receive audio signals from a first, direction;a second microphone operative to receive audio signals from a second direction; anda controller operative to engage selectively the second microphone to receive ambient audio noise or to receive an audio input.2. An electronic device as claimed in claim 1 , further comprising a microprocessor adapted to receive an output signal from the second microphone and operative to provide noise cancellation to an output signal of the first microphone.3. An electronic device as claimed in claim 1 , wherein the audio input is an audio portion of an audio/video signal.4. An electronic device as claimed in claim 1 , wherein the first microphone is adapted to receive a voice input.5. An electronic device as claimed in claim 1 , further comprising at least one additional microphone claim 1 , which is adapted to receive ambient audio noise or to receive an audio input claim 1 , or both.6. An electronic device as claimed in claim 1 , wherein at least one of the microphones is a piezoelectric microphone.7. An electronic device as claimed in claim 1 , wherein the first microphone comprises a first film bulk acoustic (FBA) structure and the second microphone comprises a second FBA structure.8. An electronic device claim 1 , comprising:a first microphone operative to receive audio signals from a first, direction;a second microphone operative to receive audio signals from a second direction; anda controller operative to engage selectively the second microphone to receive ambient audio noise or to receive an audio input, wherein the controller is adapted to receive an output signal from the second microphone and operative to provide noise cancellation to an output signal of the first microphone.9. An electronic device claim 1 , comprising:a first microphone operative to receive audio signals from a first direction;a ...

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Номер записи: 25
16-05-2013 дата публикации

MICROPHONE ARRAY CONFIGURATION AND METHOD FOR OPERATING THE SAME

Номер: US20130121505A1
Автор: "ODonovan Adam", Duraiswami Ramani, Zotkin Dmitry
Принадлежит:

An apparatus comprises a plurality of microphone units including at least a first microphone unit and a second microphone unit, each of the first and second microphone units comprising a microphone, an analog-to-digital converter, and a local memory. The microphone is configured to capture an analog audio signal. The analog-to-digital converter is configured to convert the analog audio signal created by the microphone into a digital audio signal. The local memory is configured to store the digital audio signal. The apparatus further comprises, a controller unit comprising a processor configured to process the digital audio signals. The first microphone unit and the second microphone unit are operatively connected to the controller unit in a series configuration, the second microphone unit being configured to output the digital audio signal to the first microphone unit, and the first microphone unit being configured to output the digital audio signal to the controller unit. 1. An apparatus comprising:a plurality of microphone units including at least a first microphone unit and a second microphone unit, each of the first and second microphone units comprising a microphone, an analog-to-digital converter, and a local memory, wherein the microphone is configured to capture an analog audio signal, the analog-to-digital converter is configured to convert the analog audio signal created by the microphone into a digital audio signal, and the local memory is configured to store the digital audio signal; anda controller unit comprising a processor configured to process the digital audio signals,wherein the first microphone unit and the second microphone unit are operatively connected to the controller unit in a series configuration, the second microphone unit being configured to output the digital audio signal to the first microphone unit, and the first microphone unit being configured to output the digital audio signal to the controller unit.2. The apparatus of claim 1 , ...

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Номер записи: 26
23-05-2013 дата публикации

MULTICHANNEL ACOUSTIC ECHO REDUCTION

Номер: US20130129101A1
Автор: Acero Alejandro, Madhu Nilesh, TASHEV Ivan Jelev
Принадлежит: MICROSOFT CORPORATION

A multichannel acoustic echo reduction system is described herein. The system includes an acoustic echo canceller (AEC) component having a fixed filter for each respective combination of loudspeaker and microphone signals and having an adaptive filter for each microphone signal. For each microphone signal, the AEC component modifies the microphone signal to reduce contributions from the outputs of the loudspeakers based at least in part on the respective adaptive filter associated with the microphone signal and the set of fixed filters associated with the respective microphone signal. 1. A method executed by a processor of a computing device , the method comprising:transmitting a first calibration signal to a first speaker, the first calibration signal causing the first speaker to generate first output over a first time period;transmitting a second calibration signal to a second speaker, the second signal causing the second speaker to generate second output over a second time period that is non-overlapping with the first time period;receiving a first microphone signal from a first microphone, the first microphone signal corresponding to the first output from the first speaker over the first time period;receiving a second microphone signal from the first microphone, the second microphone signal corresponding to the second output from the second speaker over the second time period;computing a first coefficient of a first fixed filter for the first microphone based upon the first calibration signal and the first microphone signal;computing a second coefficient of a second fixed filter for the first microphone based upon the second calibration signal and the second microphone signal; andsubsequent to computing the first coefficient of the first fixed filter and the second coefficient of the second fixed filter, applying the first fixed filter and the second fixed filter to an acoustic signal captured by the first microphone.2. The method of claim 1 , wherein applying ...

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Номер записи: 27
23-05-2013 дата публикации

SOUND SOURCE SIGNAL FILTERING APPARATUS BASED ON CALCULATED DISTANCE BETWEEN MICROPHONE AND SOUND SOURCE

Номер: US20130129113A1
Автор: Jeong Jae-Hoon, Jeong So-Young, KIM Kyu-hong, Oh Kwang-Cheol
Принадлежит: SAMSUNG ELECTRONICS CO., LTD.

Provided is a sound source signal filtering method and apparatus. The sound source signal filtering method includes: generating two or more microphone output signals by combining sound source signals input through a plurality of microphones; calculating distances between the microphones and a sound source from which the sound source signals are emitted by using distance relationships according to frequencies of the sound source signals extracted from the generated microphone output signals; and filtering the sound source signals to obtain one or more sound source signals corresponding to a predetermined distance by using the calculated distances. Accordingly, it is possible to obtain only sound source signals emitted from a sound source at a particular distance from the microphone array among a plurality of sound source signals input through the microphone array. 1. A sound source signal filtering apparatus comprising:a microphone output signal generator generating two or more microphone output sound signals by combining sound source signals input through a plurality of microphones;a distance calculator calculating distances from the microphones to a sound source from which the sound source signals are emitted, by using distance relationships based on frequencies of the sound source signals of the generated microphone output sound signals; anda signal filter filtering the sound source signals to obtain one or more sound source signals corresponding to a predetermined distance by using the calculated distances.2. The apparatus of claim 1 , further comprising an operator generating result values independent from magnitudes of the generated microphone output sound signals by removing the magnitudes claim 1 ,wherein the distance calculator calculates a distance between the computation result value to the sound source.3. The apparatus of claim 1 , wherein the microphone output sound signals have different sound attenuation rates from each other based on the calculated ...

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Номер записи: 28
23-05-2013 дата публикации

SMART REJECTER FOR KEYBOARD CLICK NOISE

Номер: US20130132076A1
Автор: MINETT Ian Kenneth, RIDDER Robert Jan, VERITY Steven Burritt, VOGELSANG Klaas Carlo, YANG Jun
Принадлежит: CREATIVE TECHNOLOGY LTD

According to various embodiments of the invention, a new and effective keyboard click noise reduction scheme is presented. The keyboard click noise reduction scheme may have various processing units including: Dynamic Signal Modeler, Smart Model Selector, Adaptive Filtering Module, Keyboard/Impulse Noise and Voice Activity Detectors, and a Post-Processing Unit. By adaptively changing the coefficients of the proposed adaptive filter through minimizing the output energy, the scheme can provide the target signal/voice with nearly zero keyboard click noise. The scheme could be used in real-time to minimize keyboard click noise or any kind of unwanted noise, especially noise having transient impulse characteristics. 1. A method for an impulse noise filter to minimize impulse noise in a communication session , comprising:receiving an audio input from an audio source;determining whether the audio input includes impulse noise;determining whether the audio input includes voice; andgenerating an audio output by adaptively filtering the audio input based on the determination of impulse noise being included in the audio input and based on the determination of voice being included in the audio input, wherein the adaptive filtering minimizes the impulse noise and maximizes the voice in the audio input.2. The method as recited in claim 1 , wherein determining whether the audio input includes impulse noise comprises:applying an impulse noise detection to the audio input in identifying the impulse noise in the audio input, the impulse noise detection being selected from the group consisting of noisy excitation analysis and power estimation analysis.3. The method as recited in claim 2 , wherein determining whether the audio input includes impulse noise comprises:applying dynamic signal modeling to the audio input in modeling the audio input for impulse noise, the dynamic signal modeling being selected from the group consisting of linear prediction analysis and spectral whitening ...

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Номер записи: 29
13-06-2013 дата публикации

METHOD AND DEVICE FOR SUPPRESSING RESIDUAL ECHOES

Номер: US20130151247A1
Автор: Li Bo, Liu Song, Lou Shasha
Принадлежит: GOERTEK INC.

The present invention discloses a method and a device for suppressing residual echoes. The method comprises: performing adaptive filtering on M transmitter signals respectively to obtain M adaptive filtered signals; performing array-filtering on the M−1 adaptive filtered signals other than the first adaptive filtered signal to obtain M−1 array-filter output signals; subtracting each of the M−1 array-filter output signals from the first adaptive filtered signal respectively to obtain M−1 difference signals, performing time-domain/frequency-domain conversion on the M−1 difference signals respectively and selecting one of the frequency-domain signals that has the least energy; performing time-domain/frequency-domain conversion on the first adaptive filtered signal and the Madaptive filtered signal and then performing speech probability filtering on the converted first adaptive filtered signal and the converted Madaptive filtered signal to obtain one frequency-domain speech probability signal; and multiplying the frequency-domain speech probability signal with the selected signal that has the least energy, and performing frequency-domain/time-domain conversion on the multiplication result to obtain a signal as a transmitter output signal. The technical solutions of the present invention can suppress the residual echoes effectively without impairing near end speech. 1. A method for suppressing residual echoes , the method being suitable for use in a communication apparatus comprising M transmitters and one receiver , wherein M is a natural number greater than 1 , and the M transmitters are arranged in line to form an array , the method comprising:performing adaptive filtering on M transmitter signals respectively with a receiver signal to obtain M adaptive filtered signals;processing the M−1 adaptive filtered signals except the first adaptive filtered signal by respective array-filters to obtain M−1 array-filter output signals, wherein for each of the adaptive filtered ...

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Номер записи: 30
20-06-2013 дата публикации

SIGNAL PROCESSING APPARATUS AND SIGNAL PROCESSING METHOD

Номер: US20130156221A1
Автор: MATSUO Naoshi
Принадлежит:

A signal processing apparatus includes an adder that acquires a plurality of input signals from a plurality of microphones and calculates an added value obtained by adding the input signals together, a subtracter that acquires a plurality of input signals from the plurality of microphones and calculates a subtracted value obtained by subtracting one input signal from the other input signal, and a determination unit that determines whether noise is included in the input signals based on the added value and the subtracted value. 1. A signal processing apparatus comprising:an adder that acquires a plurality of input signals from a plurality of microphones and calculates an added value obtained by adding the input signals together;a subtracter that acquires a plurality of input signals from the plurality of microphones and calculates a subtracted value obtained by subtracting one input signal from the other input signal; anda determination unit that determines whether noise is included in the input signals based on the added value and the subtracted value.2. The signal processing apparatus according to claim 1 , further comprising:a gain calculator that calculates a gain based on the added value and the subtracted value; anda noise suppression unit that suppresses noise of the input signals based on the gain calculated by the gain calculator.3. The signal processing apparatus according to claim 1 , wherein the determination unit determines that noise is included in the input signals when a difference between the added value and the subtracted value is smaller than a threshold value.4. The signal processing apparatus according to claim 1 , further comprisinga ratio calculator that calculates a ratio of the subtracted value to the added value,wherein the determination unit determines that noise is included in the input signals when a value of the ratio calculated by the ratio calculator is greater than or equal to a threshold value.5. The signal processing apparatus ...

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Номер записи: 31
20-06-2013 дата публикации

INFORMATION PROCESSING APPARATUS AND PROGRAM

Номер: US20130158990A1
Автор: Miseki Kimio, Sudo Takashi
Принадлежит: KABUSHIKI KAISHA TOSHIBA

According to one embodiment, an information processing apparatus includes a first signal input unit configure to receive a first signal, a second signal input unit configure to receive a signal, a first control unit configure to acquire system resources, a second control unit configure to select, in accordance with information of the system resources acquired by the first control unit, a processing method for suppressing at least one of echo and noise of the second signal input from the second signal input unit containing the echo due to the first signal input from the first signal input unit, a third control unit configure to generate an output signal by suppressing at least one of the echo and the noise from the second signal by the processing method selected by the second control unit, and a signal output unit configure to output the output signal generated by the third control unit. 1. An information processing apparatus comprising:a first signal input unit configured to receive a first input signal;a second signal input unit configured to receive a second input signal;a first control unit configured to acquire system resources;a second control unit configured to select, in accordance with information of the system resources acquired by the first control unit, a processing method for suppressing at least one of echo and noise of the second input signal input from the second signal input unit containing the echo due to the first input signal input from the first signal input unit;a third control unit configured to generate an output signal by suppressing at least one of the echo and the noise from the second input signal by the processing method selected by the second control unit;a signal output unit configured to output the output signal generated by the third control unit;a buffer configured to store the first input signal; anda fourth control unit configured to control synchronism between the first input signal and the second input signal,wherein the fourth ...

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Номер записи: 32
27-06-2013 дата публикации

VOICE EMPHASIS DEVICE

Номер: US20130166289A1
Автор: Suzuki Ryoji
Принадлежит: Panasonic Corporation

There is provided a voice emphasis device with which voice clarity can be improved. This voice emphasis device comprises a correlation component removal filter circuit that removes a correlation component from a voice signal produced at a specific sampling frequency, a multiplication circuit that produces an extracted signal by multiplying a specific gain coefficient by the output of the correlation component removal filter circuit, and an arithmetic circuit that adds or subtracts the extracted signal to or from the voice signal. The correlation component removal filter circuit is a lattice-type filter circuit that combines a feedforward filter and a feedback filter. The feedforward filter and the feedback filter update the filter coefficient at the specific sampling frequency based on the formula k=k+α×f/b. 1. A voice emphasis device comprising:a correlation component removal filter circuit configured to remove a correlation component from a voice signal produced at a specific sampling frequency; anda voice signal processor configured to execute signal processing on the voice signal based on an output of the correlation component removal filter circuit,the correlation component removal filter circuit being a lattice-type filter circuit in which a feedforward filter and a feedback filter are combined, and {'br': None, 'i': k', '=k', '+α×f', '/b, 'sub': i,j+1', 'i,j', 'i', 'i−1}, 'the feedforward filter and the feedback filter configured to update a filter coefficient on the basis of the specific sampling frequency according to the following formula{'sub': i,j', 'i,j+1', 'i−1, '(where kis a filter coefficient at a i-th level of the lattice-type filter circuit at a time j, kis a filter coefficient at the i-th level of the lattice-type filter circuit at a time j+1, i is a natural number from 1 to n, n is a number of levels of the lattice-type filter circuit, α is a constant (0.0≦α≦2.0), fi is a feedforward predicted error signal at the i-th level of the lattice-type ...

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Номер записи: 33
04-07-2013 дата публикации

VOICE CLARIFICATION APPARATUS

Номер: US20130173262A1
Автор: AOKI Ryotaro, SHIDOJI Hiroomi
Принадлежит: YAMAHA CORPORATION

The voice clarification apparatus includes a plurality of band-pass filters that respectively extract a plurality of band components, which are included in a voice band, from an input audio signal; a gain determination unit that determines a gain according to the level of a signal of a band component which is extracted by at least one band-pass filter of the plurality of band-pass filters; a level adjustment unit that adjusts the levels of signals of the plurality of band components which are extracted by the plurality of band-pass filters using the gain; and a first addition unit that adds a signal which is based on the audio signal to a signal in which the gain is adjusted by the level adjustment unit, and outputs a signal obtained through the addition. 1. A voice clarification apparatus comprising:a plurality of band-pass filters that respectively extract a plurality of band components, which are included in a voice band, from an input audio signal;a gain determination unit that determines a gain according to a level of a signal of a band component which is extracted by at least one band-pass filter of the plurality of band-pass filters;a level adjustment unit that adjusts levels of signals of the plurality of band components which are extracted by the plurality of band-pass filters using the gain; anda first addition unit that adds a signal which is based on the audio signal to a signal in which the gain is adjusted by the level adjustment unit, and outputs a signal obtained through the addition.2. The voice clarification apparatus according to claim 1 ,wherein the gain determination unit includes a conversion unit which converts input levels based on a signal indicative of voice components into a gain which has predetermined input and output characteristics, andwherein the conversion unit outputs the gain which is greater than “1” when an absolute value of a level of the signal indicative of the voice components is equal to or less than a threshold, and outputs ...

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Номер записи: 34
11-07-2013 дата публикации

System and Method for Improved Use of Voice Activity

Номер: US20130179160A1
Автор: James James H., Rosenbluh Joshua Hal
Принадлежит: AT&T INTELLECTUAL PROPERTY II, L.P.

The present invention is a system and method for packetizing actual noise signals, typically background noise, received by an access gateway from a speaking party and transmitting these packetized noise signals via a network to an egress gateway. The egress gateway converts the packetized noise signal into noise signals suitable for output and transmits the output noise signals to a listening party. When the access gateway detects that no voice signal is being received and only a noise signal is being received for a predetermined period of time, the access gateway instructs the egress network to continually transmit output noise signals to the listening party and ceases to transmit packetized noise signals to the egress gateway. 1. A method comprising:receiving an input signal comprising a packetized voice communication and packetized noise;converting the packetized voice communication into an output voice signal;converting the packetized noise into an output noise signal;outputting the output voice signal and the output noise signal; andwhen the input signal no longer contains the packetized voice communication, continuing to output the output noise signal while ceasing to output the output voice signal.2. The method of claim 1 , wherein when the input signal no longer contains the packetized voice communication claim 1 , the input signal also no longer contains the packetized noise.3. The method of claim 1 , wherein the input signal is received from a first party and the output voice signal is output to a second party.4. The method of claim 3 , wherein the first party and the second party are distinct individuals.5. The method of claim 1 , wherein the input signal is received using one of voice over internet protocol claim 1 , voice over frame claim 1 , and voice over asynchronous transfer mode.6. The method of claim 1 , wherein the output noise signal is output continuously until the input signal contains another packetized voice communication.7. The method of ...

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Номер записи: 35
18-07-2013 дата публикации

Echo removing apparatus, echo removing method, program and recording medium

Номер: US20130185064A1
Автор: Mitsuhiro Suzuki
Принадлежит: Sony Corp

To be provided is an echo removing apparatus including a transmission path estimate update processing unit, and an output selection unit. A fixed section of the transmission path estimate is updated based on an error from an echo estimate determined using all of the fixed section, the holding section, and the update section. These sections are updated depending on whether an estimate obtained by adding the fixed section and the holding section is better than an estimate of the fixed section alone in every fixed period. Only when the estimate is better, the holding section is added to the fixed section cumulatively, and the update section is substituted into the holding section. Depending on whether an estimate is better, an error from the eco estimate determined using all these sections or the fixed section alone is selected as an output.

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Номер записи: 36
25-07-2013 дата публикации

NOISE SUPPRESSING DEVICE, NOISE SUPPRESSING METHOD, AND PROGRAM

Номер: US20130191118A1
Автор: Makino Kenichi
Принадлежит: SONY CORPORATION

Provided is a noise suppressing device including a framing unit that frames an input signal, a band division unit that obtains a band division signal, a band power computation unit that obtains a band power from each band division signal, a noise determination unit that determines whether each band is stationary noise or non-stationary noise, a noise band power estimation unit that estimates a band power of noise of each band, a noise suppression gain decision unit that decides a noise suppression gain of each band, a noise suppression unit that obtains a band division signal whose noise is suppressed, a band synthesis unit that obtains a framed signal whose noise is suppressed, and a frame synthesis unit that obtains an output signal whose noise is suppressed. 1. A noise suppressing device comprising:a framing unit that frames an input signal by dividing the input signal into frames having a predetermined frame length;a band division unit that obtains a band division signal by dividing a framed signal obtained in the framing unit into a plurality of bands;a band power computation unit that obtains a band power from each band division signal obtained in the band division unit;a noise determination unit that determines whether each band is stationary noise or non-stationary noise based on a characteristic of the framed signal;a noise band power estimation unit that estimates a band power of noise of each band from the band power of each band division signal obtained in the band power computation unit and a determination result of the noise determination unit;a noise suppression gain decision unit that decides a noise suppression gain of each band based on the band power of each band division signal obtained in the band power computation unit and the band power of noise of each band estimated in the noise band power estimation unit;a noise suppression unit that obtains a band division signal whose noise is suppressed by applying the noise suppression gain of each band ...

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Номер записи: 37
25-07-2013 дата публикации

SIGNAL PROCESSING DEVICE, SIGNAL PROCESSING METHOD AND SIGNAL PROCESSING PROGRAM

Номер: US20130191119A1
Автор: Sugiyama Akihiko
Принадлежит: NEC Corporation

From a mixed signal in which a first signal and a second signal are mixed, the second signal is removed at low processing cost and without delay. As a result, an estimated first signal which has low residue of the second signal and low distortion is obtained. 1. A signal processing device comprising:a first input unit which inputs a first mixed signal in which a first signal and a second signal are mixed;a second input unit which inputs a second mixed signal in which said first signal and said second signal are mixed in a different proportion from said first mixed signal;a first subtraction unit which outputs an estimated first signal by subtracting a pseudo second signal from said first mixed signal;a first adaptive filter which generates said pseudo second signal by filtering said second mixed signal using a first coefficient updated by a first control signal;a first compare unit which compares a value of said estimated first signal and a value of said first mixed signal; anda first control unit which outputs said first control signal to make an update amount of said first coefficient smaller in case a value of said estimated first signal is larger than a value of said first mixed signal as a result of comparison by said first compare unit as compared with an opposite case.2. The signal processing device according to claim 1 , wherein said first adaptive filter inputs said second mixed signal.3. The signal processing device according to claim 1 , whereinsaid first compare unit obtains a first mixed signal average value by averaging values of said first mixed signal, obtains an estimated first signal average value by averaging values of said estimated first signal, and compares said first mixed signal average value and said estimated first signal average value, andsaid first control unit outputs said first control signal to make said update amount of said first coefficient smaller in case said estimated first signal average value is larger than said first mixed ...

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Номер записи: 38
01-08-2013 дата публикации

MULTI-DIRECTIONAL AND OMNIDIRECTIONAL HYBRID MICROPHONE FOR HEARING ASSISTANCE DEVICES

Номер: US20130195297A1
Автор: Burns Thomas Howard
Принадлежит: Starkey Laboratories, Inc.

Disclosed herein, among other things, are methods and apparatus for an directional microphone arrays for hearing assistance devices. In various embodiments, the present subject matter provides a microphone array system for receiving sounds including a first directional microphone, a second directional microphone and an omnidirectional microphone. The first directional microphone has a first directional axis in a first direction, and the second directional microphone has a second directional axis that is collinear with the first direction and pointing in the same direction as the first direction. The omnidirectional microphone has a sound sampling position that is a disposed between the first directional microphone and the second directional microphone, and the omnidirectional microphone sound sampling position is on or about the first directional axis. Weighted outputs of the first directional microphone, second directional microphone, and omnidirectional microphone are processed to provide a second order directional microphone system, according to various embodiments. 1. A microphone array system for receiving sounds , comprising:a first directional microphone having a first directional axis in a first direction;a second directional microphone having a second directional axis that is collinear with the first direction and pointing in the same direction as the first direction;an omnidirectional microphone having a sound sampling position that is a disposed between the first directional microphone and the second directional microphone;wherein the omnidirectional microphone sound sampling position is on or about the first directional axis and wherein weighted outputs of the first directional microphone, second directional microphone, and omnidirectional microphone are processed to provide a second order directional microphone system.2. The system of claim 1 , wherein the first directional microphone claim 1 , the second directional microphone and the omnidirectional ...

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Номер записи: 39
01-08-2013 дата публикации

SIGNAL PROCESSING DEVICE, SIGNAL PROCESSING METHOD AND SIGNAL PROCESSING PROGRAM

Номер: US20130197905A1
Автор: Sugiyama Akihiko
Принадлежит: NEC Corporation

To achieve sufficient noise cancellation when a reference signal cannot be captured in the proximity of a noise source. 1. A signal processing device comprising:a first input unit which obtains a first mixed signal in which a first signal and a second signal are mixed;a second input unit which obtains a second mixed signal in which the first signal and the second signal are mixed in a different ratio from the first mixed signal;a delay unit which generates a delayed first mixed signal by delaying the first mixed signal with a delay amount based on transmission distance from a generation source of the second signal to the second input unit;a subtracting unit which outputs an estimated first signal in which a pseudo second signal is subtracted from the delayed first mixed signal; andan adaptive filtering unit which generates the pseudo second signal applying coefficients which are updated based on the estimated first signal to the second mixed signal.2. The signal processing device according to claim 1 , whereinthe delay unit controls the delay amount of the first mixed signal using the estimated first signal.3. The signal processing device according to claim 1 , whereinthe delay unit controls the delay amount of the first mixed signal using the coefficients of the adaptive filtering unit.4. The signal processing device according to claim 2 , whereinthe delay unit controls the delay amount in a range that does not exceed a predetermined upper limit value.5. The signal processing device according to claim 1 , whereinthe adaptive filtering means changes the coefficients according to a change of the delay amount.6. A signal processing method comprising:obtaining a first mixed signal in which a first signal and a second signal are mixed;obtaining a second mixed signal in which the first signal and the second signal are mixed in a different ratio from the first mixed signal;generating a delayed first mixed signal by delaying the first mixed signal with a delay amount based ...

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Номер записи: 40
08-08-2013 дата публикации

Computer-Implemented System and Method for Enhancing Audio to Individuals Participating in a Conversation

Номер: US20130204616A1
Автор: Aoki Paul M., Szymanski Margaret H., Thornton James D., Wilson Daniel H., Woodruff Allison G.
Принадлежит: PALO ALTO RESEARCH CENTER INCORPORATED

A computer-implemented system and method for enhancing audio to individuals participating in a conversation is provided. Audio data for individuals participating in one or more conversations is analyzed. Possible conversational configurations of the individuals are generated based on the audio data, and each possible conversational configuration includes one or more subconfigurations of at least two of the individuals. A probability weight is assigned to each of the subconfigurations and includes a likelihood that the individuals of that subconfiguration are participating in one of the conversations. A probability of each possible conversational configuration is determined by combining the probability weights for the subconfigurations of that possible conversational configuration. The possible conversational configuration with the highest probability is selected as a most probable configuration. The individuals participating in the conversations are determined based on the most probable configuration. Audio for each individual participating in the determined conversations is enhanced. 1. A computer-implemented system for enhancing audio to individuals participating in a conversation , comprising:an audio data analysis module to analyze audio data for a plurality of individuals participating in one or more conversations;a conversational configuration module to generate based on the audio data, a plurality of possible conversational configurations of the individuals, each possible conversational configuration comprising one or more subconfigurations of at least two of the individuals;a weight module to assign a probability weight to each of the subconfigurations comprising a likelihood that the individuals of that subconfiguration are participating in one of the conversations;a probability module to determine a probability of each possible conversational configuration by combining the probability weights for the subconfigurations of that possible conversational ...

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Номер записи: 41
15-08-2013 дата публикации

Temporal Interpolation Of Adjacent Spectra

Номер: US20130208905A1
Автор: Iser Bernd, Krini Mohamed, Schmidt Gerhard, Wolf Arthur
Принадлежит: NUANCE COMMUNICATIONS, INC.

Embodiments of the present invention exploit redundancy of succeeding FFT spectra and use this redundancy for computing interpolated temporal supporting points. An analysis filter bank converts overlapped sequences of an audio (ex. loudspeaker) signal from a time domain to a frequency domain to obtain a time series of short-time loudspeaker spectra. An interpolator temporally interpolates this time series. The interpolation is fed to an echo canceller, which computes an estimated echo spectrum. A microphone analysis filter bank converts overlapped sequences of an audio microphone signal from the time domain to the frequency domain to obtain a time series of short-time microphone spectra. The estimated echo spectrum is subtracted from the microphone spectrum. Further signal enhancement (filtration) may be applied. A synthesis filter bank converts the filtered microphone spectra to the time domain to generate an echo compensated audio microphone signal. Computational complexity of signal processing systems can, therefore, be reduced. 1. A method for echo compensation of at least one audio microphone signal that includes an echo signal contribution due to an audio loudspeaker signal in a loudspeaker-microphone system , the method comprising:converting overlapped sequences of the audio loudspeaker signal from a time domain to a frequency domain and obtaining a time series of short-time loudspeaker spectra with a predetermined number of sub-bands, wherein the sequences have a predetermined sequence length and an amount of overlapping of the overlapped sequences predetermined by a loudspeaker sub-sampling rate;temporally interpolating the time series of short-time loudspeaker spectra, including, for each pair of temporally adjacent short-time loudspeaker spectra, calculating an interpolated short-time loudspeaker spectrum by weighted addition of the temporally adjacent short-time loudspeaker spectra; first filter coefficients are used for weighting the current loudspeaker ...

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Номер записи: 42
15-08-2013 дата публикации

Microphone apparatus and method for removing unwanted sounds

Номер: US20130208923A1
Автор: Mikko Veli Aimo Suvanto
Принадлежит: Nokia Oyj

An apparatus comprises a first transducer configured to detect sound and generate a first signal based on the detected sound. The apparatus also comprises a second transducer configured to detect vibration and/or sound and generate a second signal based on the detected vibrations and/or sound. The second transducer is less acoustically responsive than the first transducer. The apparatus comprises an interface configured to send the first and second signals to a processor configured to modify the first signal on the basis of the second signal.

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Номер записи: 43
15-08-2013 дата публикации

ADAPTIVE SYSTEMS USING CORRENTROPY

Номер: US20130211829A1
Автор: Liu Weifeng `, Principe Jose Carlos, Singh Abhishek
Принадлежит:

Various methods and systems are provided for related to adaptive systems using correntropy. In one embodiment, a signal processing device includes a processing unit and a memory storing an adaptive system executable in the at least one processing unit. The adaptive system includes modules that, when executed by the processing unit, cause the signal processing device to adaptively filter a desired signal using a correntropy cost function. In another embodiment, a method includes adjusting a coefficient of an adaptive filter based at least in part on a correntropy cost function signal, providing an adaptive filter output signal based at least in part on the adjusted coefficient and a reference signal, and determining an error signal based at least in part on a received signal and the adaptive filter output signal. 1. A signal processing device , comprising:at least one processing unit; anda memory storing an adaptive system executable in the at least one processing unit, the adaptive system comprising modules that when executed by the at least one processing unit cause the signal processing device to adaptively filter a desired signal using a correntropy cost function.2. The signal processing device of claim 1 , wherein the adaptive system comprises:an adaptive filter module including a set of filter weights, the adaptive filter configured to provide an adaptive filter output signal based at least in part upon the set of filter weights and a reference signal; anda parameter learning module configured to adjust the set of filter weights based at least in part upon the correntropy cost function and an error signal based at least in part upon the difference between the desired signal and the adaptive filter output signal.3. The signal processing device of claim 2 , wherein the adaptive system adjusts the set of filter weights based upon the error signal associated with the previous filter weight adjustment.4. The signal processing device of claim 2 , wherein the adaptive ...

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Номер записи: 44
22-08-2013 дата публикации

Adaptive Approach to Improve G.711 Perceptual Quality

Номер: US20130218557A1
Автор: GAO Yang
Принадлежит: Huawei Technologies Co., Ltd.

In order to achieve the best improvement of ITU G.711 related codec perceptual quality, perceptual weighting controlling parameter(s) should be at least adaptive to relative quantization error statistics or adaptive to signal level. When the relative quantization error statistics are larger or the signal level is lower, the perceptual weighting should be “stronger”; when the relative quantization error statistics are smaller or the signal level is larger, the perceptual weighting should be “weaker”. 2. The method of claim 1 , wherein the G.711 codec performs as a core layer of a scalable encoder.3. The method of claim 1 , wherein the G.711 codec is compatible with International Telecommunication Union (ITU) G.711 A-law or μ-law codec standard.8. The method of claim 4 , wherein the G.711 encoder performs as a core layer of a scalable encoder.9. The method of claim 4 , wherein the G.711 encoder is compatible with International Telecommunication Union (ITU) G.711 A-law or μ-law codec standard.11. The codec according to claim 10 , wherein the codec performs as a core layer of a scalable encoder.12. The codec according to claim 10 , wherein the codec is compatible with International Telecommunication Union (ITU) G.711 A-law or β-law codec standard.17. The codec according to claim 13 , wherein the codec performs as a core layer of a scalable encoder.18. The codec according to claim 13 , wherein the codec is compatible with International Telecommunication Union (ITU) G.711 A-law or μ-law codec standard. This application is a continuation of U.S. application Ser. No. 12/203,052, filed on Sep. 2, 2008, which claims priority to U.S. Provisional Application No. 60/997,663, filed on Oct. 4, 2007. Both of the aforementioned patent applications are hereby incorporated by reference in their entireties.1. Field of the Invention The present invention is generally in the field of signal coding. In particular, the present invention is in the field of speech/signal coding and ...

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Номер записи: 45
22-08-2013 дата публикации

NOISE REDUCTION APPARATUS, AUDIO INPUT APPARATUS, WIRELESS COMMUNICATION APPARATUS, AND NOISE REDUCTION METHOD

Номер: US20130218559A1
Автор: Yamabe Takaaki
Принадлежит: JVC KENWOOD CORPORATION

A speech segment of a voice sound is detected based on a first sound pick-up signal obtained based on the voice sound. A voice incoming direction of the voice sound is determined using the first sound pick-up signal and a second sound pick-up signal obtained based on a picked-up sound. A noise reduction process is performed to reduce a noise component carried by the first sound pick-up signal by using the second sound pick-up signal, wherein a noise reduction amount adjusted in accordance with the voice incoming direction is used in the noise reduction process. 1. A noise reduction apparatus comprising:a speech segment determiner configured to detect a speech segment of a voice sound based on a first sound pick-up signal obtained based on the voice sound;a voice direction detector configured to determine a voice incoming direction of the voice sound using the first sound pick-up signal and a second sound pick-up signal obtained based on a picked-up sound; anda noise reduction processor configured to perform a noise reduction process to reduce a noise component carried by the first sound pick-up signal by using the second sound pick-up signal,wherein a noise reduction amount adjusted in accordance with the voice incoming direction is used in the noise reduction process.2. The noise reduction apparatus according to claim 1 , wherein the noise reduction processor includes:an adaptive filter configured to generate a noise-presumed signal corresponding to the noise component carried by the first sound pick-up signal by using the second sound pick-up signal;an adaptive coefficient adjuster configured to adjust adaptive coefficients of the adaptive filter based on a result of an arithmetic operation between the first and second sound pick-up signals;a noise reduction-amount adjuster configured to adjust the noise-presumed signal in accordance with the voice incoming direction; andan arithmetic unit configured to reduce the noise component carried by the first sound pick-up ...

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Номер записи: 46
29-08-2013 дата публикации

System and method for noise estimation with music detection

Номер: US20130226572A1
Автор: Phillip Alan Hetherington, Shreyas Paranjpe, Steven Mason
Принадлежит: QNX Software Systems Ltd

In a system and method for noise estimation with music detection described herein provides for generating a music classification for music content in an audio signal. The music detector may classify the audio signal as music or non-music. The non-music signal may be considered to be signal and noise. An adaption rate may be adjusted responsive to the generated music classification. A noise estimate is calculated applying the adjusted adaption rate. The system and method may mitigate the noise modeling algorithms being misled by the music components.

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Номер записи: 47
29-08-2013 дата публикации

NOISE REMOVING SYSTEM IN VOICE COMMUNICATION, APPARATUS AND METHOD THEREOF

Номер: US20130226573A1
Автор: HA Dong Gyung, JEONG Seong Il, PARK Seong-Soo, SONG Jae Hoon
Принадлежит:

Disclosed is the system and method to remove noises in voice signals in a voice communication. The at least one embodiment of the present disclosure performs a spectral subtraction (SS) for voice signals based on a gain function by a spectral subtraction apparatus, performs clustering of voice signals consecutive on a frequency axis of a spectrogram for the voice signals in which the spectral subtraction has been already performed to designate one or more clusters, and extracts musical noises by determining continuity of each of the designated clusters on the frequency axis and a time axis of the spectrogram to extract to musical noises. 1. A noise removing system in a voice communication , comprising:a spectral subtraction apparatus configured to perform a spectral subtraction (SS) for voice signals; and perform clustering of the voice signals, for which the spectral subtraction has been performed and which are consecutive on a frequency axis of a spectrogram, to designate one or more clusters, and', 'determine continuity of each of the designated clusters on the frequency axis and a time axis of the spectrogram to extract musical noises., 'a noise removing apparatus configured to'}2. The system of claim 1 , wherein the noise removing apparatus is configured tocompare a continuous length of each of the designated clusters on the frequency axis with a threshold to extract the clusters corresponding to the musical noises, andextract the clusters corresponding to the musical noises based on similarities among clusters for each of residual clusters.3. A noise removing apparatus claim 1 , comprising:a clustering unit configured to perform clustering of voice signals on a frequency axis of a spectrogram to designate one or more clusters;a first extractor configured to determine continuity of each of the designated clusters on the frequency axis to extract clusters corresponding to musical noises; anda second extractor configured to extract clusters corresponding to the ...

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Номер записи: 48
05-09-2013 дата публикации

APPARATUS AND METHOD FOR DERIVING A DIRECTIONAL INFORMATION AND COMPUTER PROGRAM PRODUCT

Номер: US20130230187A1
Автор: AHONEN Jukka, DEL GALDO Giovanni, KUECH Fabian, PULKKI Ville, THIERGART Oliver
Принадлежит: Fraunhofer-Gesellschaft zur Foerderung der angewandten Forschung e.V.

An apparatus for deriving a directional information from a plurality of microphone signals or from a plurality of components of a microphone signal, wherein different effective microphone look directions are associated with the microphone signals or components, has a combiner configured to obtain a magnitude value from a microphone signal or a component of the microphone signal. The combiner is further configured to combine direction information items describing the effective microphone look directions, such that a direction information item describing a given effective microphone look direction is weighted in dependence on the magnitude value of the microphone signal, or of the component of the microphone signal, associated with the given effective microphone look direction, to derive the directional information. 1. Apparatus for deriving a directional information from a plurality of microphone signals or from a plurality of components of a microphone signal , wherein different effective microphone look directions are associated with the microphone signals or components , the apparatus comprising:a combiner configured to acquire a magnitude value from a microphone signal or a component of the microphone signal, and to combine direction information items describing the effective microphone look directions, such that a direction information item describing a given effective microphone look direction is weighted in dependence on the magnitude value of the microphone signal, or of the component of the microphone signal, associated with the given effective microphone look direction, to derive the directional information;wherein a direction information item describing a given effective microphone look direction is a vector pointing in the given effective microphone look direction;wherein the combiner is configured to derive the directional information d(k, n) for a given time frequency tile corresponding to a linear combination of the direction information items weighted ...

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Номер записи: 49
05-09-2013 дата публикации

USER INTERFACE TONE ECHO CANCELLATION

Номер: US20130231158A1
Автор: Schell Stephan Vincent
Принадлежит: Apple Inc.

A multi-function communications device has a processor that generates a user interface audible tone signal. The device also has a downlink digital signal processor, and an uplink digital signal processor. A mixer has an input to receive the downlink signal and another input to receive the user interface tone signal. The uplink processor has an acoustic echo canceller having an input to receive the uplink signal and another input to receive an output from the mixer. The echo canceller may reduce the amount of both the far-end user's speech and the user interface tone that may be present in the uplink signal. The mixer may be positioned within the chain of audio signal processors, or it may be positioned outside the chain. Other embodiments are also described and claimed. 1. A multi-function communications device comprising a multi-function communications device housing having integrated therein:a microphone;a speaker;a display;memory having stored therein an operating system program and a plurality of application programs to perform functions of the multi-function communications device including visual voicemail, web browser, email, digital camera, and telephony;a processor to execute the operating system and application programs to perform said functions and to generate a user interface audible tone signal that causes a sound to be heard by a near-end user of the device from the speaker and that is a result of an event occurring in one of the plurality of application programs being executed;a downlink digital signal processor to process a downlink audio signal received from a far-end user's communications device, the downlink signal causes speech of the far-end user to be heard by the near-end user from the speaker; andan uplink digital signal processor to process an uplink audio signal picked up by the microphone and to be transmitted to the far-end user's device wherein the uplink processor has an acoustic echo canceller to receive the uplink audio signal, the ...

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Номер записи: 50
05-09-2013 дата публикации

Monaural Noise Suppression Based on Computational Auditory Scene Analysis

Номер: US20130231925A1
Автор: Avendano Carlos, Goodwin Michael M., Laroche Jean, Solbach Ludger
Принадлежит:

The present technology provides a robust noise suppression system that may concurrently reduce noise and echo components in an acoustic signal while limiting the level of speech distortion. A time-domain acoustic signal may be received and be transformed to frequency-domain sub-band signals. Features, such as pitch, may be identified and tracked within the sub-band signals. Initial speech and noise models may be then be estimated at least in part from a probability analysis based on the tracked pitch sources. Speech and noise models may be resolved from the initial speech and noise models and noise reduction may be performed on the sub-band signals. An acoustic signal may be reconstructed from the noise-reduced sub-band signals. 1. A method for performing noise reduction , the method comprising:executing a program stored in a memory to transform a time-domain acoustic signal into a plurality of frequency-domain sub-band signals;tracking multiple pitched sources within a sub-band signal in the plurality of sub-band signals;generating a speech model and one or more noise models based on the tracked pitch sources; andperforming noise reduction on the sub-band signal based on the speech model and the one or more noise models.2. The method of claim 1 , wherein tracking includes tracking the multiple pitched sources across successive frames of a sub-band signal.3. The method of claim 1 , wherein tracking includes:calculating at least one feature for each pitched source in the multiple pitched sources; anddetermining a probability for each pitched source that the pitched source is a speech source.4. The method of claim 3 , wherein the probability is based at least in part on pitch energy level claim 3 , pitch salience claim 3 , and pitch stationarity.5. The method of claim 1 , further comprising generating a speech model and a noise model from the multiple pitch tracks.6. The method of claim 1 , wherein generating a speech model and one or more noise models includes ...

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Номер записи: 51
12-09-2013 дата публикации

LOCAL PEAK WEIGHTED-MINIMUM MEAN SQUARE ERROR (LPW-MMSE) ESTIMATION FOR ROBUST SPEECH

Номер: US20130238324A1
Автор: Ichikawa Osamu
Принадлежит: INTERNATIONAL BUSINESS MACHINES CORPORATION

A system and method for noise reduction applied to a speech recognition front-end. An output of a front-end is optimized by giving, as a weight to the output for each band, a confidence index representing the remarkableness of the harmonic structure of observation speech. In a first method, when clean speech is estimated by executing MMSE estimation on a model that gives a probability distribution of noise-removed speech generated from observation speech, the posterior probability of the MMSE estimation is weighted using the confidence index as a weight. In a second method, linear interpolation is executed, for each band, between an observed value of observation speech and an estimated value of clean speech, with the confidence index serving as a weight. The first method and the second method can be combined. 1. A noise reduction method comprising:a step of generating a confidence index for each band based on a spectrum of observation speech; anda step of estimating a clean speech estimated value by executing MMSE estimation on a probability model of clean speech generated on the basis of the observation speech, and, for each band, weighting posterior probability of the MMSE estimation using the confidence index as a weight.2. The method according to claim 1 , wherein the confidence index is an index representing remarkableness of a harmonic structure of the observation speech.3. The method according to claim 1 , wherein the step of generating a confidence index includes:a step of extracting a harmonic structure from the spectrum of the observation speech and normalizing the harmonic structure;a step of smoothing normalized values on a mel scale; anda step of normalizing the smoothed values so that a mean of the smoothed values becomes 1.4. The method according to claim 1 ,wherein the step of generating a confidence index is a step of generating a first confidence index and a second confidence index,wherein the step of estimating a clean speech estimated value is a ...

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Номер записи: 52
12-09-2013 дата публикации

GEOTAGGED ENVIRONMENTAL AUDIO FOR ENHANCED SPEECH RECOGNITION ACCURACY

Номер: US20130238325A1
Автор: Kristjansson Trausti T., Lloyd Matthew I.
Принадлежит:

Methods, systems, and apparatus, including computer programs encoded on a computer storage medium, for enhancing speech recognition accuracy. In one aspect, a method includes receiving an audio signal that corresponds to an utterance recorded by a mobile device, determining a geographic location associated with the mobile device, identifying a set of geotagged audio signals that correspond to environmental audio associated with the geographic location, weighting each geotagged audio signal of the set of geotagged audio signals based on metadata associated with the respective geotagged audio signal, and using the set of weighted geotagged audio signals to perform noise compensation on the audio signal that corresponds to the utterance. 1. A system comprising:one or more computers; and receiving geotagged audio signals that correspond to environmental audio recorded by multiple mobile devices in multiple geographic locations,', 'receiving an audio signal that corresponds to an utterance recorded by a particular mobile device,', 'determining a particular geographic location associated with the particular mobile device,', 'generating a noise model for the particular geographic location using a subset of the geotagged audio signals, and', 'performing noise compensation on the audio signal that corresponds to the utterance using the noise model that has been generated for the particular geographic location., 'a computer-readable medium coupled to the one or more computers having instructions stored thereon which, when executed by the one or more computers, cause the one or more computers to perform operations comprising2. The system of claim 1 , wherein the operations further comprise performing speech recognition on the utterance using the noise-compensated audio signal.3. The system of claim 1 , wherein generating the noise model further comprises generating the noise model before receiving the audio signal that corresponds to the utterance.4. The system of claim 1 , ...

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Номер записи: 53
12-09-2013 дата публикации

SPEECH RECOGNITION PROCESSING DEVICE AND SPEECH RECOGNITION PROCESSING METHOD

Номер: US20130238327A1
Автор: NONAKA Tsutomu
Принадлежит: SEIKO EPSON CORPORATION

A speech recognition processing device includes a speech synthesis part, a speech output part, a speech input part, and a speech recognition part. A first synthesized sound and a second synthesized sound synthesized by the speech synthesis part are output from the speech output part. Noise information is obtained from a sound signal input from the speech input part between an output period of the first synthesized sound and an output period of the second synthesized sound, and the noise information is used for noise removal processing in the speech recognition part. 1. A speech recognition processing device comprising:a speech synthesis part;a speech output part that outputs speech synthesized in the speech synthesis part;a speech input part; anda speech recognition part that renders speech recognition on sound input from the speech input part,when a first sentence synthesized in the speech synthesis part contains a first word and a second word, the first word synthesized in the speech synthesis part defines a first synthesized sound, and the second word synthesized in the speech synthesis part defines a second synthesized sound,correction information used for removing noise from a speech signal to be used for the speech recognition being generated based on sound input from the speech input part in a third period when speech is not output from the speech output part, between a first period when the first synthesized sound is output and a second period when the second synthesized sound is output.2. The speech recognition processing device according to claim 1 , wherein the second word is a word next to the first word.3. The speech recognition processing device according to claim 1 , wherein the correction information is generated based on sound input in a plurality of the third periods.4. A speech recognition processing method for a speech recognition processing device claim 1 , the speech recognition processing device including a speech synthesis part claim 1 , a ...

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Номер записи: 54
19-09-2013 дата публикации

SYSTEM AND METHOD FOR PRODUCING AN AUDIO SIGNAL

Номер: US20130246059A1
Автор: Kechichian Patrick, Van Den Dungen Wilhelmus Andreas Martinus Arnoldus Maria
Принадлежит: KONINKLIJKE PHILIPS ELECTRONICS N.V.

There is provided a method of generating a signal representing the speech of a user, the method comprising obtaining a first audio signal representing the speech of the user using a sensor in contact with the user; obtaining a second audio signal using an air conduction sensor, the second audio signal representing the speech of the user and including noise from the environment around the user; detecting periods of speech in the first audio signal; applying a speech enhancement algorithm to the second audio signal to reduce the noise in the second audio signal, the speech enhancement algorithm using the detected periods of speech in the first audio signal; equalizing the first audio signal using the noise-reduced second audio signal to produce an output audio signal representing the speech of the user. 1. A method of generating a signal representing the speech of a user , the method comprising:{'b': '101', 'obtaining a first audio signal representing the speech of the user using a sensor in contact with the user ();'}{'b': '101', 'obtaining a second audio signal using an air conduction sensor, the second audio signal representing the speech of the user and including noise from the environment around the user ();'}{'b': '103', 'detecting periods of speech in the first audio signal ();'}{'b': '105', 'applying a speech enhancement algorithm to the second audio signal to reduce the noise in the second audio signal, the speech enhancement algorithm using the detected periods of speech in the first audio signal ();'}{'b': '107', 'equalizing the first audio signal using the noise-reduced second audio signal to produce an output audio signal representing the speech of the user ().'}2103. A method as claimed in claim 1 , wherein the step of detecting periods of speech in the first audio signal () comprises detecting parts of the first audio signal where the amplitude of the audio signal is above a threshold value.3105. A method as claimed in claim 1 , wherein the step of ...

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Номер записи: 55
19-09-2013 дата публикации

SIGNAL PROCESSING DEVICE, SIGNAL PROCESSING METHOD AND SIGNAL PROCESSING PROGRAM

Номер: US20130246060A1
Автор: Sugiyama Akihiko
Принадлежит: NEC Corporation

The purpose of the present invention is to obtain a higher-quality output signal by performing noise suppression in view of a background sound. The signal processing device disclosed in the present application is provided with suppression means for performing suppression of a second signal by processing a mixed signal in which a first signal and said second signal are contained. Moreover the signal processing device is provided with background sound estimation means for estimating a background sound signal in said mixed signal. Additionally, the signal processing device is provided with restriction means for restricting said suppression of said second signal such that a suppression result outputted by said suppression means does not become smaller than said estimated background sound signal. 19-. (canceled)10. A signal processing device comprising:a suppression unit which performs suppression of a second signal by processing a mixed signal in which a first signal and said second signal are contained;a background sound estimation unit which estimates a background sound signal in said mixed signal; anda restriction unit which restricts said suppression of said second signal such that a suppression result outputted by said suppression means does not become smaller than said estimated background sound signal.11. The signal processing device according to claim 10 , further comprising:an estimation unit which estimates said second signal contained in said mixed signal,wherein said restriction unit corrects said estimated second signal outputted from said estimation means in accordance with said background sound signal, andsaid suppression unit subtracts said corrected estimated second signal from said mixed signal to restrict said suppression.12. The signal processing device according to claim 10 , further comprising:a storage unit which stores therein an estimated second signal which is estimated to be contained in said mixed signal,wherein said restriction unit corrects ...

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Номер записи: 56
26-09-2013 дата публикации

PSYCHOACOUSTIC FILTER DESIGN FOR RATIONAL RESAMPLERS

Номер: US20130253917A1
Автор: Schildbach Wolfgang
Принадлежит: DOLBY INTERNATIONAL AB

The present document relates to the design of anti-aliasing and/or anti-imaging filters for resamplers using rational resampling factors. In particular, the present document relates to a method for designing such filters having a reduced number of filter coefficients or an increased perceptual performance, as well as to the filters designed using such method. A method for designing a filter () configured to reduce imaging and/or aliasing of an output audio signal () at an output sampling rate (fs) is described. The output audio signal () is a resampled version of an input audio signal () at an input sampling rate (fs). The ratio of the output sampling rate (fs) and the input sampling rate (fs) is a rational number N/M. The filter () operates at an upsampled sampling rate which equals N times the input sampling rate (fs). The method comprises the steps of selecting an allowed deviation of the frequency response () of the filter () within a stop band of the filter () based on a perceptual frequency response indicative of an auditory spectral sensitivity; wherein the allowed deviation indicates a deviation of the frequency response () of the filter () from a predetermined attenuation within the stop band; and of determining coefficients of the filter () such that the frequency response () of the filter () is fitted to the allowed deviation of the frequency response (). 134-. (canceled)35. A method for designing a filter configured to reduce imaging and/or aliasing of an output audio signal at an output sampling rate; wherein the output audio signal is a resampled version of an input audio signal at an input sampling rate;wherein the ratio of the output sampling rate and the input sampling rate is a rational number N/M;wherein the filter operates at an upsampled sampling rate which equals N times the input sampling rate; selecting an allowed deviation of the frequency response of the filter within a stop band of the filter based on a perceptual frequency response ...

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Номер записи: 57
03-10-2013 дата публикации

SOUND RECORDING DEVICE

Номер: US20130259262A1
Автор: OHTSUKA Yoshio
Принадлежит: Panasonic Corporation

A sound recording device of the present disclosure is can be connected with an external sound pickup device. The sound recording device includes a connector having a plurality of terminals to which the external sound pickup device can be connected, and a determiner that determines a type of the external sound pickup device when the external sound pickup device is connected to the connector, based on a correlation between signals of specific terminals out of the plurality of terminals. 1. A sound recording device to which an external sound pickup device can be connected , the sound recording device comprising:a connector having a plurality of terminals to which the external sound pickup device can be connected; anda determiner that determines a type of the external sound pickup device when the external sound pickup device is connected to the connector, based on a correlation between signals of specific terminals of the plurality of terminals.2. The sound recording device according to claim 1 , whereinthe plurality of terminals include a first terminal, a second terminal, and a third terminal, andthe determiner determines that the connected external sound pickup device is a specific external sound pickup device, when a value concerning a correlation between an output signal of the first terminal and an output signal of the second terminal is higher than a first threshold value and when a value concerning a correlation between an output signal of the first terminal and an output signal of the third terminal is higher than a second threshold value.3. The sound recording device according to claim 2 , whereinthe first terminal is a terminal to which a first external sound pickup device that picks up sound from a front direction is connected,the second terminal is a terminal to which a second external sound pickup device that picks up sound from a left direction is connected, andthe third terminal is a terminal to which a third external sound pickup device that picks up ...

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Номер записи: 58
03-10-2013 дата публикации

NOISE REDUCTION SYSTEM WITH REMOTE NOISE DETECTOR

Номер: US20130262101A1
Автор: Srinivasan Sriram
Принадлежит: KONINKLIJKE PHILIPS N.V.

Noise reduction system with remote noise detector The present invention relates to a noise reduction system with at least one remote noise detector placed close to at least one noise source, which transmits relevant information to a primary device where it is used for noise reduction. Thereby, acoustic signal enhancement can be achieved via the at least one remote noise detector in that a noise estimate is transmitted to controller for noise reduction in the signal obtained from a primary source. 1. A noise reduction apparatus for reducing at least one of background noise and interference during reception of an audio signal , said noise reduction apparatus comprising:a wireless receiver for receiving a noise estimate from at least one remote noise detector,an acoustic receiver for receiving an acoustic signal from a primary acoustic source,a noise reduction processor for reducing or cancelling a noise component in said received acoustic signal based on said received noise estimate, wherein said received noise estimate is power spectral density of a noise or interference received at said remote noise detector.2. The noise reduction apparatus according to claim 1 , wherein said acoustic receiver comprises a first microphone adapted to receive said acoustic signal from said primary acoustic source.3. The noise reduction apparatus according to claim 1 , wherein said noise reduction processor comprises a level adjustment unit for compensating a level difference between said received noise estimates and said noise component in said received acoustic signal based on a speech model on a frame-by-frame basis.4. (canceled)5. The noise reduction apparatus according to claim 1 , wherein said noise reduction processor comprises a path estimation unit for estimating an acoustic path between said remote noise detector and said acoustic receiver.6. The noise reduction apparatus according to claim 1 , wherein said noise reduction processor comprises a speech enhancement unit for ...

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Номер записи: 59
10-10-2013 дата публикации

APPARATUS AND METHOD FOR CANCELLING WIDEBAND ACOUSTIC ECHO

Номер: US20130268267A1
Автор: Choo Chang, HWANG In Ki, Lee Byung Sun
Принадлежит: Electronics and Telecommunications Research Institute

Disclosed is an apparatus for cancelling a wideband acoustic echo, the apparatus including a determining unit to determine whether a monitor coefficient value obtained by dividing a Near-End Talker (NET) energy value by an ERROR energy value converges to an acoustic echo cancelling reference value. 1. An apparatus for cancelling a wideband acoustic echo , the apparatus comprising:a determining unit to determine whether a monitor coefficient value obtained by dividing a Near-End Talker (NET) energy value by an ERROR energy value converges to an acoustic echo cancelling reference value;an internal memory to store a coefficient value of a filter when the monitor coefficient value converges to the acoustic echo cancelling reference value;a calculating unit to calculate a threshold value of the coefficient value of the filter and a performance index;a register to store data location information when the coefficient value of the filter is less than or equal to the threshold value;an index counter to increase a feedback counter value when the performance index is determined to be lower than a predetermined convergence index based on a result of the comparing the performance index and the predetermined convergence index, using the determining unit; anda restoring unit to restore the feedback counter value to the coefficient value of the filter when the feedback counter value corresponds to a predetermined restoring reference value.2. The apparatus of claim 1 , wherein the filter applies a Normalized Least Mean Square (NLMS) adaptive filter.3. The apparatus of claim 1 , wherein when a filtering operation with respect to a subsequent data sample is performed claim 1 , performing a filtering and a coefficient updating with respect to a location stored in the register is omitted.4. The apparatus of claim 1 , further comprising:a detecting unit to detect an occurrence of a double-talk with respect to noise of data.5. The apparatus of claim 4 , wherein the detecting unit ...

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Номер записи: 60
24-10-2013 дата публикации

Mixer with adaptive post-filtering

Номер: US20130279718A1
Автор: Phillip A. Hetherington
Принадлежит: QNX Software Systems Ltd

A noise reduction system includes multiple transducers that generate time domain signals. A transforming device transforms the time domain signals into frequency domain signals. A signal mixing device mixes the frequency domain signals according to a mixing ratio. Frequency domain signals are rotated in phase to generate phase rotated signals. A post-processing device attenuates portions of the output based on coherence levels of the signals.

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Номер записи: 61
24-10-2013 дата публикации

SYSTEMS AND METHODS FOR AUDIO SIGNAL PROCESSING

Номер: US20130282369A1
Автор: Guo Yinyi, Kim Lae-Hoon, Ryu Sang-Ut, SCHEVCIW Andre Gustavo P., Shin Jongwon, Visser Erik
Принадлежит: QUALCOMM INCORPORATED

A method for signal level matching by an electronic device is described. The method includes capturing a plurality of audio signals from a plurality of microphones. The method also includes determining a difference signal based on an inter-microphone subtraction. The difference signal includes multiple harmonics. The method also includes determining whether a harmonicity of the difference signal exceeds a harmonicity threshold. The method also includes preserving the harmonics to determine an envelope. The method further applies the envelope to a noise-suppressed signal. 1. A method for signal level matching by an electronic device , comprising:capturing a plurality of audio signals from a plurality of microphones;determining a difference signal based on an inter-microphone subtraction, wherein the difference signal comprises multiple harmonics;determining whether a harmonicity of the difference signal exceeds a harmonicity threshold;preserving the harmonics to determine an envelope; andapplying the envelope to a noise-suppressed signal.2. The method of claim 1 , further comprising:segmenting an input spectrum into one or more bands;measuring a signal-to-noise ratio for each band;determining if the signal-to-noise ratios are less than a first threshold;assembling a target spectrum; andadjusting a gain of one or more bands in the noise-suppressed signal based on the target spectrum.3. The method of claim 2 , wherein assembling a target spectrum comprises replacing a portion of a speech reference spectrum with a portion of a speech template spectrum.4. The method of claim 3 , wherein the portion of the speech reference spectrum that is replaced comprises one or more bands where the signal-to-noise ratio is less than the first threshold.5. The method of claim 3 , wherein the speech template spectrum is based on a codebook.6. The method of claim 3 , wherein the speech template spectrum is based on an interpolation of the bands of the input spectrum where the signal-to- ...

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Номер записи: 62
24-10-2013 дата публикации

SPEECH PROCESSING APPARATUS, CONTROL METHOD THEREOF, STORAGE MEDIUM STORING CONTROL PROGRAM THEREOF, AND VEHICLE, INFORMATION PROCESSING APPARATUS, AND INFORMATION PROCESSING SYSTEM INCLUDING THE SPEECH PROCESSING APPARATUS

Номер: US20130282370A1
Автор: Arakawa Takayuki, Sugiyama Akihiko
Принадлежит: NEC Corporation

An apparatus of this invention is a speech processing apparatus that acquires pseudo speech from a mixture sound including desired speech and noise. The speech processing apparatus includes a first microphone that inputs a first mixture sound including desired speech and noise and outputs a first mixture signal, a second microphone that is opened to the same sound space as that of the first microphone, inputs a second mixture sound including the desired speech and the noise at a ratio different from the first mixture sound, and outputs a second mixture signal, a first sound collector including a concave surface that collects the first mixture sound to the first microphone, a second sound collector including a concave surface that collects the second mixture sound to the second microphone and disposed in a direction different from the first sound collector, and a noise suppression circuit that suppresses an estimated noise signal based on the first mixture signal and the second mixture signal and outputs a pseudo speech signal. With this arrangement, it is possible to, in a single sound space where desired speech and noise mix, collect the desired speech and the noise, correctly estimate the noise, and reconstruct pseudo speech close to the desired speech. 1. A speech processing apparatus comprising:a first microphone that inputs a first mixture sound including desired speech and noise and outputs a first mixture signal;a second microphone that is opened to the same sound space as that of said first microphone, inputs a second mixture sound including the desired speech and the noise at a ratio different from the first mixture sound, and outputs a second mixture signal;a first sound collector including a concave surface that collects the first mixture sound to said first microphone;a second sound collector including a concave surface that collects the second mixture sound to said second microphone and disposed in a direction different from said first sound collector; ...

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Номер записи: 63
31-10-2013 дата публикации

NOISE SUPPRESSION BASED ON CORRELATION OF SOUND IN A MICROPHONE ARRAY

Номер: US20130287224A1
Автор: Nilsson Jesper, Nystrom Martin, Smailagic Sead
Принадлежит: SONY ERICSSON MOBILE COMMUNICATIONS AB

A microphone array includes a left microphone, a right microphone and a processor to receive a right microphone signal from the right microphone and a left microphone signal from the left microphone. The processor determines a timing difference between the left microphone signal and the right microphone signal. The processor determines whether the timing difference is within a time threshold. The processor time shifts one of the left microphone signal and the right microphone signal based on the timing difference. The processor also sums the shifted microphone signal and the other microphone signal to form an output signal. 1. A computer-implemented method in a microphone array , wherein the microphone array includes a left microphone and a right microphone , comprising:receiving a right microphone signal from the right microphone;receiving a left microphone signal from the left microphone;determining a timing difference between the left microphone signal and the right microphone signal;determining whether the timing difference is within a time threshold;time shifting one of the left microphone signal and the right microphone signal based on the timing difference when the timing difference is within the time threshold; andsumming the shifted microphone signal and the other microphone signal to form an output signal.2. The computer-implemented method of claim 1 , further comprising:identifying an average sound pressure level for a predetermined time slot for each of the left microphone signal and the right microphone signal;selecting one of the left microphone signal and the right microphone signal that has a lowest average sound pressure level as the output signal for the predetermined time slot.3. The computer-implemented method of claim 2 , further comprising:determining whether an output signal for a preceding time slot is from a same microphone signal as the output signal for the predetermined time slot;identifying a zero crossing point near a border of the ...

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Номер записи: 64
07-11-2013 дата публикации

IMAGING DEVICE, PROGRAM, MEMORY MEDIUM, AND NOISE REDUCTION METHOD

Номер: US20130293747A1
Автор: OKANO Kosuke, OKAZAKI Mitsuhiro, YOSHIZUKA Yoko
Принадлежит: NIKON CORPORATION

Provided are an imaging device, program, memory medium, and noise reduction method capable of appropriately reducing noise without causing degradation in a target sound such as voice. The imaging device of the present invention has: a video imaging unit for capturing video; a signal converter for converting a sound generated during video capture to a sound signal; subject determination units that predict or recognize a specific subject; a noise detector for detecting noise included in the sound generated during video capture; a noise reduction unit for reducing the noise signal from the sound signal; a voice detector for detecting non-noise signals in the sound signal; and a noise reduction performance change unit that lowers the noise signal reduction performance of the noise reduction unit when the subject determination units predict or recognize the specific subject. 1. An imaging device comprising:a movie shooting unit for shooting a movie;a signal conversion unit for converting sound generated during the shooting of the movie into a sound signal;a subject determination unit for predicting or recognizing a specific subject;a noise detection unit for detecting noise included in the sound generated during the shooting of the movie;a noise reduction unit for reducing a noise signal from the sound signal;a voice detection unit for detecting a non-noise signal from the sound signal; anda noise reduction performance change unit for decreasing performance of the noise reduction unit for reducing the noise signal when the subject determination unit predicts or recognizes the specific subject.2. The imaging device according to claim 1 , wherein:the noise reduction unit reduces the noise signal from the sound signal on the basis of a noise reduction coefficient; andwhen the subject determination unit predicts or recognizes the specific subject,the noise reduction performance change unit sets the noise reduction coefficient to a relatively low-level reduction coefficient ...

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Номер записи: 65
07-11-2013 дата публикации

AUTOMATIC MICROPHONE MUTING OF UNDESIRED NOISES BY MICROPHONE ARRAYS

Номер: US20130294612A1
Автор: Chu Peter L., Feng Jinwei
Принадлежит:

Methods and systems for cancelation of table noise in a speaker system used for video or audio conferencing are disclosed. Table noise is cancelled by using a vertical microphone array to distinguish the tilt angle of sound received by a microphone. If the sound is close to horizontal, the audio is muted. If the sound is above a given angle from horizontal, it is not muted, as this indicates a person speaking. This eliminates paper rustling, keyboard clicks and the like. 1. A microphone system comprising:a vertical microphone array having at least two microphones, with a first microphone near a horizontal surface and a second microphone located above said first microphone; anda processor coupled to said microphone array to receive sound from said first and second microphones and configured to determine an angle from horizontal of a received sound and mute said sound if said angle from horizontal is below a predetermined amount.2. The microphone system of claim 1 , further comprising at least one loudspeaker.3. The microphone system of claim 2 , wherein at least one of the microphones is a vertical distance away from the loudspeaker.4. The microphone system of claim 1 , further comprising an echo cancelation module for each microphone in said vertical microphone array.5. The microphone system of claim 4 , wherein said echo cancelation modules provide echo cancelation of sounds from a loudspeaker signal.6. A noise cancelation method comprising:providing a vertical microphone array having at least two microphones, with a first microphone near a horizontal surface and a second microphone located above said first microphone; andreceiving sounds from said first and second microphones; anddetermining an angle from horizontal of a received sound and muting said sound if said angle from horizontal is below a predetermined amount.7. The noise cancelation method of claim 6 , further providing at least one loudspeaker.8. The noise cancelation method of claim 7 , wherein at ...

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Номер записи: 66
07-11-2013 дата публикации

METHOD AND SYSTEM FOR SPEECH ENHANCEMENT IN A ROOM

Номер: US20130294616A1
Автор: Mülder Hans
Принадлежит: PHONAK AG

A system for speech enhancement in a room, having: a microphone arrangement with at least two spaced apart microphones for capturing audio signals from a speaker's voice, an acoustic beamforming unit for processing the captured audio signals in a manner so as to impart one of a plurality of different directional patterns to the microphone arrangement; a feedback cancellation unit for applying a feedback cancellation algorithm to the processed audio signals and for providing a feedback status signal indicating how close an acoustic feedback loop of the system is to feedback; an amplifier for amplifying the processed audio signals; a loudspeaker arrangement to be located in the room for generating sound according to the amplified audio signals; and a control for selecting the directional pattern imparted to the microphone arrangement as a function of the feedback status signal. 123-. (canceled)24. A system for speech enhancement in a room , comprising:a microphone arrangement comprising at least two spaced apart microphones for capturing audio signals from a speaker's voice,an acoustic beamforming unit for processing the captured audio signals in a manner so as to impart one of a plurality of different directional patterns to the microphone arrangement;a feedback cancellation unit for applying a feedback cancellation algorithm to the processed audio signals and for providing a feedback status signal indicating how close an acoustic feedback loop of the system is to feedback;means for amplifying the processed audio signals;a loudspeaker arrangement to be located in the room for generating sound according to the amplified audio signals; andmeans for selecting the directional pattern imparted to the microphone arrangement as a function of the feedback status signal.25. The system of claim 24 , wherein the microphone arrangement is a lapel microphone arrangement.26. The system of claim 24 , wherein the microphones of the microphone arrangement are of an omnidirectional ...

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Номер записи: 67
07-11-2013 дата публикации

Source separation by independent component analysis in conjunction with source direction information

Номер: US20130297296A1
Автор: Jaekwon Yoo, Ruxin Chen
Принадлежит: Sony Computer Entertainment Inc

Methods and apparatus for signal processing are disclosed. Source separation can be performed to extract source signals from mixtures of source signals by way of independent component analysis. Source direction information is utilized in the separation process, and independent component analysis techniques described herein use multivariate probability density functions to preserve the alignment of frequency bins in the source separation process. It is emphasized that this abstract is provided to comply with the rules requiring an abstract that will allow a searcher or other reader to quickly ascertain the subject matter of the technical disclosure. It is submitted with the understanding that it will not be used to interpret or limit the scope or meaning of the claims.

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Номер записи: 68
07-11-2013 дата публикации

Speech processing apparatus, control method thereof, storage medium storing control program thereof, and vehicle, information processing apparatus, and information processing system including the speech processing apparatus

Номер: US20130297303A1
Автор: Akihiko Sugiyama, Takayuki Arakawa
Принадлежит: NEC Corp

An apparatus of this invention is a speech processing apparatus that acquires pseudo speech from a mixture sound including desired speech and noise. The speech processing apparatus includes a first microphone that inputs a first mixture sound including desired speech and noise and outputs a first mixture signal, a second microphone that is opened to the same sound space as that of said first microphone and disposed at a focus position of an interface that is part of a boundary of the sound space and has one of a quadratic surface shape and a pseudo surface shape approximating a quadratic surface, inputs a second mixture sound including the desired speech reflected by the interface and the noise reflected by the interface at a ratio different from the first mixture sound, and outputs a second mixture signal, and a noise suppression circuit that suppresses an estimated noise signal based on the first mixture signal and the second mixture signal and outputs a pseudo speech signal.

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Номер записи: 69
14-11-2013 дата публикации

Voice Signals Improvements in Compressed Wireless Communications Systems

Номер: US20130301844A1
Автор: ALON KONCHITSKY
Принадлежит: Individual

Improvements in voice signals transmitted within communication systems are obtained by use of adaptive filters, front and rear microphones, noise cancelling systems and other means and methods. Disclosed embodiments include the use of directional microphones, primary inputs, secondary inputs, adaptive weight generators, canceller outputs to improve signal to noise ratios and other communication attributes.

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Номер записи: 70
14-11-2013 дата публикации

METHOD AND AN APPARATUS FOR VOICE QUALITY ENHANCEMENT

Номер: US20130304461A1
Автор: Taleb Anisse, Virette David, Xu Jianfeng
Принадлежит:

A voice quality enhancement (VQE) detector for a network element receiving an audio signal from a previous network element of a network, wherein said voice quality enhancement detector is adapted: to perform a voice quality enhancement detection based on the received audio signal, wherein said voice quality enhancement detection comprises detecting that at least one voice quality enhancement function, VQEF, was applied to the received audio signal by at least one previous network element of the network; and to control a voice quality enhancement processing of the received audio signal depending on the detection result. 1. A voice quality enhancement (VQE) detector for a network element receiving an audio signal from a previous network element of a network ,wherein the VQE detector is configured to:perform a VQE detection based on the received audio signal, wherein the voice quality enhancement detection comprises detecting that at least one voice quality enhancement function (VQEF) has been applied to the received audio signal by at least one previous network element of the network; andcontrol a VQE processing of the received audio signal according to the detection result.2. The VQE detector according to claim 1 ,wherein the at least one VQEF is at least one of the following: a noise reduction function, an echo cancellation function, a dynamic range compression function, and an automatic gain control function.3. The VQE detector according to claim 1 ,wherein received audio signal is a coded audio signal; andwherein the VQE detector is connected to one of a signal input and an output of a bit stream decoder;when the VQE detector is connected to the signal input, the VQE detector processes a bit stream of the coded audio signal received at the signal input to perform the voice quality enhancement detection; andwhen the VQE detector is connected to the output of the bit stream decoder, and the VQE detector processes a decoded signal generated by the bit stream de-coder ...

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Номер записи: 71
21-11-2013 дата публикации

SPEECH PROCESSING APPARATUS, CONTROL METHOD THEREOF, STORAGE MEDIUM STORING CONTROL PROGRAM THEREOF, AND VEHICLE, INFORMATION PROCESSING APPARATUS, AND INFORMATION PROCESSING SYSTEM INCLUDING THE SPEECH PROCESSING APPARATUS

Номер: US20130311175A1
Автор: Arakawa Takayuki, Sugiyama Akihiko
Принадлежит: NEC Corporation

An apparatus of this invention is a speech processing apparatus that acquires pseudo speech from a mixture sound including desired speech and noise. The speech processing apparatus includes a first microphone that inputs a first mixture sound including desired speech and noise and outputs a first mixture signal, a second microphone that is opened to the same sound space as that of the first microphone, inputs a second mixture sound including the desired speech and the noise at a ratio different from the first mixture sound, and outputs a second mixture signal, a sound insulator that is disposed between the first microphone and the second microphone, and a noise suppression circuit that suppresses an estimated noise signal based on the first mixture signal and the second mixture signal and outputs a pseudo speech signal. With this arrangement, it is possible to, in a single sound space where desired speech and noise mix, correctly estimate the noise and reconstruct pseudo speech close to the desired speech. 1. A speech processing apparatus comprising:a first microphone that inputs a first mixture sound including desired speech and noise, and outputs a first mixture signal;a second microphone that is opened to the same sound space as that of said first microphone, inputs a second mixture sound including the desired speech and the noise at a ratio different from the first mixture sound, and outputs a second mixture signal;a sound insulator that is disposed between said first microphone and said second microphone; anda noise suppression circuit that suppresses an estimated noise signal based on the first mixture signal and the second mixture signal and outputs a pseudo speech signal.2. The speech processing apparatus according to claim 1 , wherein said sound insulator includes a sound insulating portion that crosses a line segment connecting said first microphone and a source of the noise claim 1 , and shields an airborne sound of the noise.3. The speech processing ...

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Номер записи: 72
28-11-2013 дата публикации

SPATIAL ADAPTATION IN MULTI-MICROPHONE SOUND CAPTURE

Номер: US20130315403A1
Автор: Samuelsson Leif Jonas
Принадлежит: DOLBY INTERNATIONAL AB

A spatial adaptation system for multiple-microphone sound capture systems and methods thereof are described. A spatial adaptation system includes an inference and weight module configured to receive a inputs. The inputs based on two or more input signals captured by at least two microphones. The inference and weight module to determine one or more weight values base on at least one of the inputs. The spatial adaptation system also including a noise magnitude ratio update module coupled with the inference and weight module. The noise magnitude ratio update module to determine an updated noise target based on the one or more weight values from the inference and weight module. 1. A spatial adaptation system comprising:an inference and weight module configured to receive a plurality of inputs based on two or more input signals captured by at least two microphones, said inference and weight module to determine one or more weight values based on at least one of said plurality of inputs; anda noise magnitude ratio update module coupled with said inference and weight module, said noise magnitude ratio update module to determine an updated noise target based on said one or more weight values from said inference and weight module.2. The system of further comprising a spatial feature module coupled with said inference and weight module claim 1 , said spatial feature module to determine one or more spatial features based on said two or more input signals.3. The system of claim 2 , wherein said inference and weight module determines one or more weight values based on said one or more spatial features determined by said spatial feature module.4. The system of claim 3 , wherein said one or more spatial features include magnitude ratio based on said two or more input signals claim 3 , phase difference based on said two or more input signals claim 3 , and coherence based on said two or more input signals.5. The system of claim 1 , wherein at least two of said plurality of inputs are ...

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Номер записи: 73
28-11-2013 дата публикации

VARIABLE BEAMFORMING WITH A MOBILE PLATFORM

Номер: US20130316691A1
Автор: Forutanpour Babak, Momeyer Brian, SCHEVCIW Andre Gustavo P., Visser Erik
Принадлежит: QUALCOMM INCORPORATED

A mobile platform includes a microphone array and is capable of implementing beamforming to amplify or suppress audio information from a sound source. The sound source is indicated through a user input, such as pointing the mobile platform in the direction of the sound source or through a touch screen display interface. The mobile platform further includes orientation sensors capable of detecting movement of the mobile platform. When the mobile platform moves with respect to the sound source, the beamforming is adjusted based on the data from the orientation sensors so that beamforming is continuously implemented in the direction of the sound source. The audio information from the sound source may be included or suppressed from a telephone or video-telephony conversation. Images or video from a camera may be likewise controlled based on the data from the orientation sensors. 1. A method comprising:pointing a mobile platform towards a sound source to select a direction of the sound source with respect to the mobile platform for amplification or suppression of audio information;implementing beamforming with the mobile platform in the direction of the sound source to amplify or suppress audio information from the sound source;determining movement of the mobile platform with respect to the sound source; andusing the determined movement to adjust the beamforming to continue to implement beamforming in the direction of the sound source after the mobile platform has moved with respect to the sound source.2. The method of claim 1 , wherein the sound source is a first sound source in a first direction claim 1 , the method further comprising:indicating a second direction of a second sound source with respect to the mobile platform;implementing beamforming with the mobile platform in the second direction of the second sound source to amplify or suppress audio information from the sound source; andusing the determined movement to adjust the beamforming to continue to implement ...

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Номер записи: 74
05-12-2013 дата публикации

METHOD AND DEVICE FOR MICROPHONE SELECTION

Номер: US20130322655A1
Автор: Lindström Fredric, Schuldt Christian
Принадлежит: Limes Audio AB

The present invention relates to a device, such as an audio communication device, for combining a plurality of microphone signals x(k) into a single output signal y(k). The device comprises processing means configured to calculate control signals f(k), and control means configured to select which microphone signal x(k) or which combination of microphone signals x(k) to use as output signal y(k) based on said control signals f(k). To improve the selection, the device comprises linear prediction filters for calculating linear prediction residual signals e(k) from the plurality of microphone signals x(k), and the processing means is configured to calculate the control signals f(k) based on said linear prediction residual signals e(k). 1. A device for combining a plurality of microphone signals x(k) into a single output signal y(k) , comprising:{'sub': 'n', 'processing means configured to calculate control signals f(k);'}{'sub': n', 'n', 'n, 'control means configured to select which microphone signal x(k) or which combination of microphone signals x(k) to use as output signal y(k) based on said control signals f(k),'}{'sub': n', 'n', 'n', 'n, 'characterised in that said device comprises linear prediction filters for calculating linear prediction residual signals e(k) from said plurality of microphone signals x(k), and in that said processing means is configured to calculate said control signals f(k) based on said linear prediction residual signals e(k).'}2. Device according to claim 1 , further comprising delay processing means and a subtraction unit claim 1 , wherein the delay processing means is configured to delay said plurality of microphone signals x(k) claim 1 , the linear prediction filters are configured to filter the delayed microphone signals claim 1 , and the subtraction unit is configured to subtract said microphone signals x(k) from the delayed and filtered signals in order to obtain said linear prediction residual signals e(k).3. Device according to claim ...

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Номер записи: 75
05-12-2013 дата публикации

DYNAMIC MICROPHONE SIGNAL MIXER

Номер: US20130325458A1
Автор: Buck Markus, Eichentopf Achim, Matheja Timo
Принадлежит:

A system and method of signal combining that supports different speakers in a noisy environment is provided. Particularly for deviations in the noise characteristics among the channels, various embodiments ensure a smooth transition of the background noise at speaker changes. A modified noise reduction (NR) may achieve equivalent background noise characteristics for all channels by applying a dynamic, channel specific, and frequency dependent maximum attenuation. The reference characteristics for adjusting the background noise may be specified by the dominant speaker channel. In various embodiments, an automatic gain control (AGC) with a dynamic target level may ensure similar speech signal levels in all channels. 129-. (canceled)30. A method , comprising:receiving a plurality of signals containing sound information;performing, using at least in part a computer processor, dynamic noise reduction filtering on the plurality of signals to generate respective preprocessed signals having substantially equivalent noise characteristics; andcombining at least two of the preprocessed signals to provide an output signal.31. The method according to claim 30 , further comprising providing claim 30 , by respective microphones claim 30 , the plurality of signals claim 30 , wherein at least two of the microphones are positioned in different locations in a vehicle.32. The method according to claim 30 , wherein performing the dynamic noise reduction filtering of the plurality of signals further includes driving the preprocessed signals such that background noise is substantially equivalent as to at least one of spectral shape and power.33. The method according to claim 30 , wherein the noise characteristics include a substantially equivalent signal to noise ratio.34. The method according to claim 30 , wherein signals in the plurality of signals are each associated with a respective channel claim 30 , and wherein the dynamic noise reduction filtering includes determining a dynamic ...

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Номер записи: 76
12-12-2013 дата публикации

Sensor Fusion to Improve Speech/Audio Processing in a Mobile Device

Номер: US20130332156A1
Автор: Karahan Sinan, MATHIAS Arun G., Shao Tiange, Tackin Onur Ergin, Theverapperuma Lalin S., Zhang Haining
Принадлежит: Apple Inc.

The disclosed system and method for a mobile device combines information derived from onboard sensors with conventional signal processing information derived from a speech or audio signal to assist in noise and echo cancellation. In some implementations, an Angle and Distance Processing (ADP) module is employed on a mobile device and configured to provide runtime angle and distance information to an adaptive beamformer for canceling noise signals, provides a means for building a table of filter coefficients for adaptive filters used in echo cancellation, provides faster and more accurate Automatic Gain Control (AGC), provides delay information for a classifier in a Voice Activity Detector (VAD), provides a means for automatic switching between a speakerphone and handset mode of the mobile device, or primary microphone and reference microphones and assists in separating echo path changes from double talk. 1. A computer-implemented method performed by one or more processors of a mobile device , comprising:receiving data from one or more sensors of a mobile device;calculating an orientation and distance of a signal source relative to a first microphone of the mobile device based on the data;receiving a signal from the source through the first microphone; andprocessing the signal based on the calculated orientation and distance.2. The method of claim 1 , where processing comprises:calculating a gain based on the distance; andautomatically applying the gain to the signal received through the first microphone.3. The method of claim 2 , where automatically applying the gain claim 2 , comprises:comparing the calculated gain with an estimated gain; anddetermining whether to apply the calculated gain to the signal based on results of the comparison.4. The method of claim 3 , where processing comprises:determining a gain error based on the calculated gain and the estimated gain; andapplying either the calculated gain or the estimated gain to the signal received through the ...

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Номер записи: 77
26-12-2013 дата публикации

MICROPHONE MOUNTING STRUCTURE OF MOBILE TERMINAL AND USING METHOD THEREOF

Номер: US20130343572A1
Автор: KIM Joontae, LEE Keansub, Lee Yonghee, SON Youngjoo, SONG Hosung
Принадлежит:

Disclosed are a microphone mounting structure of a mobile terminal, capable of capturing a sound generated from a subject in an optimum manner while capturing an image and reproducing the captured image, and a using method thereof. In the present invention, a first microphone and a second microphone are arranged on one side surface of a terminal body, in a spaced manner from each other on different axes, so that they can be used in a sound capturing mode. Three or more microphones are arranged on a plurality of surfaces of the terminal body, at various intervals, by interworking with various situation changes. Then, the number of microphones to be used for capturing a sound, and a microphone combination are selected according to a user's behavior scenario. Under such configuration, an optimum audio zooming function can be implemented. 1. A microphone mounting structure of a mobile terminal , comprising:a first microphone mounted to one side surface of a body; anda second microphone arranged on a different axis from the first microphone, and spaced from the first microphone in directions of a horizontal axis and a vertical axis by a prescribed distance.2. The microphone mounting structure of claim 1 , wherein both of the first microphone and the second microphone are arranged on a rear surface of the body.3. The microphone mounting structure of claim 1 , wherein the prescribed distance comprises the same distance or a different distance claim 1 , andwherein the same distance is 1.45 cm based on a sampling rate of 22.05 KHz.4. The microphone mounting structure of claim 1 , wherein the first microphone and the second microphone are arranged at a right side or a left side of a camera.5. The microphone mounting structure of claim 1 , wherein if the first microphone is positioned on a horizontal axis based on a user claim 1 , the second microphone is positioned on a vertical axis claim 1 , at an upper or lower right side based on the first microphone claim 1 , at an angle ...

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Номер записи: 78
26-12-2013 дата публикации

DEVICE AND METHOD FOR ACOUSTIC COMMUNICATION

Номер: US20130346070A1
Автор: JUNG Hee-Won, KOH Jun-Ho, LEE Gi-Sang, Lee Sang-Mook, Zhidkov Sergey
Принадлежит: SAMSUNG ELECTRONICS CO., LTD.

Disclosed is an acoustic communication method that includes filtering an audio signal to attenuate a high frequency section of the audio signal; generating a residual signal which corresponds to a difference between the audio signal and the filtered signal; generating a psychoacoustic mask for the audio signal based on a predetermined psychoacoustic model; generating a psychoacoustic spectrum mask by combining the residual signal with the psychoacoustic mask; generating an acoustic communication signal by modulating digital data according to the acoustic signal spectrum mask; and combining the acoustic communication signal with the filtered signal. 1. An acoustic communication method comprising:filtering an audio signal to attenuate a high frequency section of the audio signal;generating a residual signal which corresponds to a difference between the audio signal and the filtered signal;generating a psychoacoustic spectrum mask for the audio signal based on a predetermined psychoacoustic model and the residual signal;generating an acoustic communication signal by modulating digital data according to the psychoacoustic spectrum mask; andcombining the acoustic communication signal with the filtered signal.2. The acoustic communication method of claim 1 , wherein generating the psychoacoustic spectrum mask includes:generating a psychoacoustic mask for the audio signal based on the predetermined psychoacoustic model; andgenerating the psychoacoustic spectrum mask by combining the residual signal with the psychoacoustic mask.3. The acoustic communication method of claim 1 , wherein filtering of the audio signal is performed by a frequency selection attenuation filter which has a frequency response that reduces from a low frequency to a high frequency.4. The acoustic communication method of claim 1 , further comprising:detecting a spectrum envelope of the residual signal.5. The acoustic communication method of claim 4 , wherein detecting of the spectrum envelope comprises ...

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Номер записи: 79
02-01-2014 дата публикации

SYSTEMS AND METHODS FOR SURROUND SOUND ECHO REDUCTION

Номер: US20140003611A1
Автор: Kim Lae-Hoon, Mohammad Asif I., Visser Erik
Принадлежит:

A method for echo reduction by an electronic device is described. The method includes nulling at least one speaker. The method also includes mixing a set of runtime audio signals based on a set of acoustic paths to determine a reference signal. The method also includes receiving at least one composite audio signal that is based on the set of runtime audio signals. The method further includes reducing echo in the at least one composite audio signal based on the reference signal. 1. A method for echo reduction by an electronic device , comprising:nulling at least one speaker;mixing a set of runtime audio signals based on a set of acoustic paths to determine a reference signal;receiving at least one composite audio signal that is based on the set of runtime audio signals; andreducing echo in the at least one composite audio signal based on the reference signal.2. The method of claim 1 , further comprising:outputting a set of output calibration audio signals;receiving a set of input calibration audio signals based on the set of output calibration audio signals; anddetermining the set of acoustic paths based on the set of input calibration audio signals.3. The method of claim 2 , wherein each of the set of output calibration audio signals is output individually in an output sequence claim 2 , and wherein each of the set of input calibration audio signals is received individually in an input sequence.4. The method of claim 1 , wherein the at least one composite audio signal is received by two or more microphones in a wireless communication device claim 1 , and wherein mixing the set of runtime audio signals and reducing the echo is performed by an audio processing device.5. The method of claim 1 , further comprising applying a first acoustic path to multiple runtime audio signals.6. The method of claim 1 , wherein the at least one composite audio signal is received by two or more microphones in a wireless communication device claim 1 , wherein a mixed-down source per each ...

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Номер записи: 80
02-01-2014 дата публикации

APPARATUS FOR AUDIO SIGNAL PROCESSING

Номер: US20140006019A1
Автор: Niemistö Riitta Elina, Paajanen Erkki Juhani
Принадлежит: Nokia Corporation

A method for estimating background noise of an audio signal comprises detecting voice activity in one or more frames of the audio signal based on one or more first conditions. The method also comprises estimating a first background noise estimation if voice activity is not detected based on the one or more first conditions. Voice activity in the one or more frames of the audio signal based on one or more second conditions is detected. A second background noise estimation is estimated if voice activity is not detected based on the one or more second conditions. The voice activity is detected in the one or more frames less often based on the one or more first conditions than based on the one or more second conditions. 161-. (canceled)62. An apparatus comprising:a first voice activity detection module configured to detect a first voice activity in one or more frames of an audio signal;a first background noise estimation module configured to estimate a first background noise estimation if the first voice activity is not detected;a second voice activity detection module configured to detect a second voice activity in the one or more frames of the audio signal;a second background noise estimation module configured to estimate a second background noise estimation if the second voice activity is not detected; andwherein the first voice activity is detected in the one or more frames less often-than the second voice activity.63. The apparatus as claimed in claim 62 , wherein the second background noise estimation module is configured to update the second background noise estimation based on the first background noise estimation.64. The apparatus as claimed in claim 62 , wherein the second background noise estimation module is configured to update the second background noise estimation with at least one of:a combination of the first and second background noise estimates; anda weighted mean of the first and second background noise estimates.65. The apparatus as claimed in claim ...

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Номер записи: 81
23-01-2014 дата публикации

NOISE REDUCTION USING DIRECTION-OF-ARRIVAL INFORMATION

Номер: US20140023199A1
Автор: Giesbrecht David
Принадлежит: QSound Labs, Inc.

Systems and methods of improved noise reduction using direction of arrival information include: receiving an audio signal from two or more acoustic sensors; applying a beamformer module to employ a first noise cancellation algorithm to the audio signal; applying a noise reduction post-filter module to the audio signal, the application of which includes: estimating a current noise spectrum of the received audio signal after the application of the first noise cancellation algorithm; using spatial information derived from the audio signal received from the two or more acoustic sensors to determine a measured direction-of-arrival by estimating the current time-delay between the acoustic sensor inputs; comparing the measured direction-of-arrival to a target direction-of-arrival; applying a second noise reduction algorithm to the audio signal in proportion to the difference between the measured direction-of-arrival and the target direction-of-arrival; and outputting a single audio stream with reduced background noise. 1. An audio device comprising:an audio processor and memory coupled to the audio processor, wherein the memory stores program instructions executable by the audio processor, wherein, in response to executing the program instructions, the audio processor is configured to:receive an audio signal from two or more acoustic sensors;apply a beamformer module to employ a first noise cancellation algorithm to the audio signal; estimating a current noise spectrum of the received audio signal after the application of the first noise cancellation algorithm;', 'using spatial information derived from the audio signal received from the two or more acoustic sensors to determine a measured direction-of-arrival;', 'comparing the measured direction-of-arrival to a target direction-of-arrival;', 'applying a second noise reduction algorithm in proportion to the difference between the measured direction-of-arrival and the target direction-of-arrival; and, 'apply a noise ...

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Номер записи: 82
06-02-2014 дата публикации

Multi-microphone noise reduction using enhanced reference noise signal

Номер: US20140037100A1
Автор: David Giesbrecht
Принадлежит: QSound Labs Inc

Systems and methods of improved noise reduction include the steps of: receiving an audio signal from two or more acoustic sensors; applying a beamformer to employ a first noise cancellation algorithm; applying a noise reduction post-filter module to the audio signal including: estimating a current noise spectrum of the received audio signal after the application of the first noise cancellation algorithm, wherein the current noise spectrum is estimated using the audio signal received by the second acoustic sensor; determining a punished noise spectrum using the time-average level difference between the audio signal received by the first acoustic sensor and the current noise spectrum; determining a final noise estimate by subtracting the punished noise spectrum from the current noise spectrum; and applying a second noise reduction algorithm to the audio signal received by the first acoustic sensor using the final noise estimate; and outputting an audio stream with reduced background noise.

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Номер записи: 83
06-02-2014 дата публикации

Headphone device, wearing state detection device, and wearing state detection method

Номер: US20140037101A1
Автор: Kohei Asada, Yasunobu Murata
Принадлежит: Sony Corp

Provided is a headphone device including an outside microphone attached to a position at which an extraneous sound is picked up without passing through a shield in a state in which a user is wearing the headphone device, an inside microphone attached to a position at which the extraneous sound is picked up via the shield in the state in which the user is wearing the headphone device, a driver unit which performs an acoustic output, and a wearing state detection unit which detects a wearing or non-wearing state using a signal comparison result between sound signals obtained by the outside and inside microphones, respectively, a pre-stored non-wearing state reference value which is a signal comparison result when the extraneous sound arrives in the non-wearing state, and a pre-stored wearing state reference value which is a signal comparison result when the extraneous sound arrives in the wearing state.

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Номер записи: 84
06-02-2014 дата публикации

Method and device for forming a digital audio mixed signal, method and device for separating signals, and corresponding signal

Номер: US20140037110A1
Автор: Antoine Liuktus, Gael Richard, Laurent Girin, Roland Badeau
Принадлежит: Institut Polytechnique de Grenoble, Telecom ParisTech

The invention relates to a method of forming one or more digital audio mixed signals (S out ) on the basis of at least two digital audio source signals (S 1 , S 2 ), in which the digital audio mixed signal or signals are formed by mixing the digital audio source signals. A characteristic digital magnitude of at least one digital audio source signal is compressed into a series of bits and said series of bits is inserted into said digital audio source signal or into the digital audio mixed signals in an almost inaudible or inaudible manner, The characteristic digital magnitude is the temporal, spectral or spectro-temporal distribution of said digital audio source signal or the temporal, spectral or spectro-temporal contribution of said digital audio source signal in the mixed signal or signals, or said digital audio source signal. The invention also relates to a method of separation intended for separating, at least partially, at least one digital audio source signal contained in one or more digital audio mixed signals obtained previously. The invention also relates to the corresponding digital audio mixed signal (S out ), as well as to the corresponding devices.

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Номер записи: 85
06-03-2014 дата публикации

Binaural enhancement of tone language for hearing assistance devices

Номер: US20140064496A1
Автор: Ning Li
Принадлежит: Starkey Laboratories Inc

Disclosed herein, among other things, are methods and apparatus for binaural enhancement of tone language for hearing assistance devices. One aspect of the present subject matter includes a method for enhancing pitch in a hearing assistance system having a first and second hearing assistance device. A signal is received using a microphone of the first hearing assistance device. Pitch detection is performed on the signal to obtain a pitch value. The pitch value is wirelessly transmitted from the first hearing assistance device to the second hearing assistance device. In various embodiments, the pitch value of the first hearing assistance device is combined with a pitch value of the second hearing assistance device. The gain is adjusted based on the combined pitch value, in various embodiments.

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Номер записи: 86
06-03-2014 дата публикации

TARGET SOUND ENHANCEMENT DEVICE AND CAR NAVIGATION SYSTEM

Номер: US20140064514A1
Автор: Mikami Takashi, Yano Atsuyoshi
Принадлежит: Mitsubishi Electric Corporation

A target sound enhancement device has a first beamformer and a second beamformer which are of different types. A vehicle interior environment model in which this target sound enhancement device is mounted is stored in a vehicle interior environment model storage unit . A beamformer type determining unit selects a most suitable beamformer according to the vehicle interior environment model for each of predetermined frequency bands, and a BF selector outputs a signal in each frequency band to the beamformer selected. A signal combining unit 18 combines signals in the frequency bands in each of which the driver's voice outputted from the first beamformer or the second beamformer is enhanced. 1. A target sound enhancement device comprising:an operation unit for converting output signals from two or more microphones mounted in an interior of a vehicle into signals in a frequency domain;a beamformer group having two or more different types of beamformers each for generating a signal including an enhanced target sound for each predetermined frequency band from the plural signals in the frequency domain into which the output signals are converted by said operation unit;a vehicle interior environment model storage unit for holding information about noise characteristics for said each predetermined frequency band in said vehicle interior environment, and information about directional characteristics of each of said beamformers;a beamformer type determining unit for evaluating each of said beamformers for said each predetermined frequency band on a basis of the directional characteristic and the noise characteristics held by said vehicle interior environment model storage unit to select a beamformer having a highest level of evaluation from said beamformers;an output switching unit for outputting the signals in the frequency domain into which the output signals are converted by said operation unit in units of said each predetermined frequency band to the beamformer selected by ...

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Номер записи: 87
06-03-2014 дата публикации

SOUND PROCESSING DEVICE, SOUND PROCESSING METHOD, AND SOUND PROCESSING PROGRAM

Номер: US20140067385A1
Автор: GOUYON Fabien, INCE Gökhan, NAKADAI Kazuhiro, Nakamura Keisuke, OKUNO Hiroshi, OLIVEIRA João Labato, REIS Luís Paulo
Принадлежит:

A sound processing device includes a separation unit configured to separate at least a music signal and a speech signal from a recorded audio signal, a noise suppression unit, a music feature value estimation unit, a speech recognition unit, a noise-processing confidence calculation unit, a music feature value estimation confidence calculation unit, a speech recognition confidence calculation unit, and a control unit configured to calculate at least one behavioral decision function of a speech behavioral decision function associated with speech and a music behavioral decision function associated with music based on a noise-processing confidence value, a music feature value estimation confidence value, and a speech recognition confidence value and to determine behavior corresponding to the calculated behavioral decision function. 1. A sound processing device comprising:a separation unit configured to separate at least a music signal and a speech signal from a recorded audio signal;a noise suppression unit configured to perform a noise suppression process of suppressing noise from at least one of the music signal and the speech signal separated by the separation unit;a music feature value estimation unit configured to estimate a feature value of the music signal from the music signal;a speech recognition unit configured to recognize speech from the speech signal;a noise-processing confidence calculation unit configured to calculate a noise-processing confidence value which is a confidence value relevant to the noise suppression process;a music feature value estimation confidence calculation unit configured to calculate a music feature value estimation confidence value which is a confidence value relevant to the process of estimating the feature value of the music signal;a speech recognition confidence calculation unit configured to calculate a speech recognition confidence value which is a confidence value relevant to the speech recognition; anda control unit ...

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Номер записи: 88
06-03-2014 дата публикации

METHOD AND SYSTEM FOR NOISE REDUCTION

Номер: US20140067386A1
Автор: Feng Yuhong, Zhang Chen
Принадлежит: VIMICRO CORPORATION

A method for noise reduction is provided including: beamforming audio signals sampled by a microphone array to get a signal with an enhanced target voice and a signal with a weakened target voice; locating a target voice in the audio signal sampled by the microphone array; determining a credibility of the target voice when the target voice is located; updating an adaptive filter coefficient according to the credibility, and filtering the signal with the enhanced target voice and the signal with the weakened target voice according to the updated adaptive filter coefficient to get a signal with reduced noise; and weighing a voice presence probability by the credibility, and enhancing the signal with reduced noise according to the weighed voice presence probability. 1. A method for noise reduction , comprising: beamforming audio signals sampled by a microphone array to get a signal with an enhanced target voice and a signal with a weakened target voice; locating a target voice in the audio signal sampled by the microphone array; determining a credibility of the target voice when the target voice is located; updating an adaptive filter coefficient according to the credibility , and filtering the signal with the enhanced target voice and the signal with the weakened target voice according to the updated adaptive filter coefficient to get a signal with reduced noise; and weighing a voice presence probability by the credibility , and enhancing the signal with reduced noise according to the weighed voice presence probability;Wherein, the locating a target voice in the audio signal sampled by the microphone array comprises: computing a maximum cross-correlation value of the audio signals sampled by the microphone array; determining a time difference that the target voice arrives at different microphones of the microphone array based on the maximum cross-correlation value; and determining an incidence angle of the target voice relative to the microphone array based on the ...

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Номер записи: 89
20-03-2014 дата публикации

Wearable Communication System With Noise Cancellation

Номер: US20140081631A1
Автор: Kopit Danny, Li Qi, Luo Xuling, Zhu Manli
Принадлежит:

A method and a wearable communication system for personal face-to-face and wireless communications in high noise environments are provided. A noise cancellation device (NCD) operably coupled to a wireless coupling device (WCD) includes a speech acquisition unit, an audio signal processing unit, one or more loudspeakers, and a communication module. The NCD receives voice vibrations from user speech via a contact microphone and a second microphone and converts the voice vibrations into an audio signal. The NCD processes the audio signal to remove noise signals and enhance a speech signal contained in the audio signal. A loudspeaker emits the speech signal during face-to-face communication. The NCD transmits the speech signal to a communication device via the WCD and receives an external speech signal from the communication device during wireless communication. With the NCD, the signal intelligibility and signal-to-noise ratio can be improved, for example, from −10 dB to 20 dB. 1. A noise cancellation device for personal face-to-face communication and wireless communication in a high noise environment , comprising:a speech acquisition unit comprising a contact microphone operably positioned with respect to a wearable unit, said contact microphone configured to receive voice vibrations from user speech in said high noise environment via said wearable unit, and to convert said voice vibrations into an audio signal;an audio signal processing unit, in operative communication with said speech acquisition unit, configured to process said audio signal, remove noise signals from said audio signal, and enhance a speech signal contained in said audio signal;a communication interface configured to connect said noise cancellation device to a communication device, wherein said communication interface, in operative communication with said audio signal processing unit, is configured to transmit said speech signal to said communication device for facilitating said wireless ...

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Номер записи: 90
20-03-2014 дата публикации

Method and Device for Voice Operated Control

Номер: US20140081644A1
Автор: Boillot Marc, Goldstein Steve, Usher John
Принадлежит: PERSONICS HOLDINGS, INC.

At least one exemplary embodiment is directed to a method and device for voice operated control. The method can include measuring a first sound received from a first microphone, measuring a second sound received from a second microphone, detecting a spoken voice based on an analysis of measurements taken at the first and second microphone, mixing the first sound and the second sound to produce a mixed signal, and controlling the production of the mixed signal based on one or more aspects of the spoken voice. 1. An acoustic device , comprising:a first microphone configured to detect a first acoustic signal;a second microphone configured to detect a second acoustic signal;a processor operatively coupled to the first microphone and the second microphone, the processor being configured to detect a spoken voice based on an analysis of the first acoustic signal captured by the first microphone and the second acoustic signal captured by the second microphone; anda voice operated control configured to mix the first acoustic signal and the second acoustic signal to produce a mixed signal.2. The acoustic device of claim 1 , wherein the voice operated control is configured to control the production of the mixed signal based on one or more aspects of the spoken voice.3. The acoustic device of claim 1 , wherein the voice operated control is configured to increase a first gain of one of the first acoustic signal and the second acoustic signal claim 1 , wherein the mixed signal includes a combination of the first acoustic signal and the second acoustic signal.4. The acoustic device of claim 1 , wherein the analysis is at least one among a sound pressure level comparison claim 1 , a correlation claim 1 , a coherence claim 1 , a spectral difference or a ratio between the first acoustic signal and the second acoustic signal.5. The acoustic device of claim 1 , further comprising a speaker claim 1 , wherein the voice operated control mixes an audio content with the mixed signal and ...

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Номер записи: 91
03-04-2014 дата публикации

SYSTEM AND METHOD OF DETECTING A USER'S VOICE ACTIVITY USING AN ACCELEROMETER

Номер: US20140093093A1
Автор: Andersen Esge B., Bright Andrew P., Dusan Sorin V., Lindahl Aram
Принадлежит: Apple Inc.

A method of detecting a user's voice activity in a mobile device is described herein. The method starts with a voice activity detector (VAD) generating a VAD output based on (i) acoustic signals received from microphones included in the mobile device and (ii) data output by an inertial sensor that is included in an earphone portion of the mobile device. The inertial sensor may detect vibration of the user's vocal chords modulated by the user's vocal tract based on vibrations in bones and tissue of the user's head. A noise suppressor may then receive the acoustic signals from the microphones and the VAD output and suppress the noise included in the acoustic signals received from the microphones based on the VAD output. The method may also include steering one or more beamformers based on the VAD output. Other embodiments are also described. 1. A method of detecting a user's voice activity in a mobile device comprising:generating by a voice activity detector (VAD) a VAD output based on (i) acoustic signals received from microphones included in the mobile device and (ii) data output by an inertial sensor that is included in an earphone portion of the mobile device, the inertial sensor to detect vibration of the user's vocal chords modulated by the user's vocal tract based on vibrations in bones and tissue of the user's head.2. The method of claim 1 , wherein inertial sensor is an accelerometer.3. The method of claim 2 , wherein the accelerometer has a sampling rate greater than 2000 Hz.4. The method of claim 2 , wherein the accelerometer has a sampling rate between 2000 Hz and 6000 Hz.5. The method of claim 2 , wherein the microphones included in the mobile device are a microphone array.6. The method of claim 5 , wherein the vibrations in the bones and tissue of the user's head further comprises the vibrations detected from portions of the user's ear and head that are in contact with the earphone portion of the mobile device.7. The method of claim 6 , wherein the ...

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Номер записи: 92
03-04-2014 дата публикации

Single Channel Suppression Of Impulsive Interferences In Noisy Speech Signals

Номер: US20140095156A1
Автор: HOFMANN Christian, Wolff Tobias
Принадлежит:

Methods and apparatus for reducing impulsive interferences in a signal, without necessarily ascertaining a pitch frequency in the signal, detect onsets of the impulsive interferences by searching a spectrum of high-energy components for large temporal derivatives that are correlated along frequency and extend from a very low frequency up, possibly to about several kHz. The energies of the impulsive interferences are estimated, and these estimates are used to suppress the impulsive interferences. Optionally, techniques are employed to protect desired speech signals from being corrupted as a result of the suppression of the impulsive interferences. 1. A method for reducing impulsive interferences in a signal , the method comprising automatically:identifying a plurality of high-energy components of the signal, wherein energy of each of the plurality of identified high-energy components exceeds a predetermined threshold;identifying a plurality of temporal derivatives of the plurality of identified high-energy components;morphologically filtering the identified plurality of temporal derivatives, including detecting onsets of the impulsive interferences and estimating a plurality of interference energies in the signal, based at least in part on the plurality of identified temporal derivatives; andsuppressing portions of the signal, based on the plurality of estimated interference energies.2. A method according to claim 1 , wherein identifying the plurality of high-energy components comprises determining the threshold claim 1 , such that the threshold is below a spectral envelope of the signal.3. A method according to claim 1 , wherein identifying the plurality of high-energy components comprises determining the threshold claim 1 , based at least in part on a spectral envelope of the signal and at least in part on a power spectral density of stationary noise in the signal.4. A method according to claim 3 , wherein determining the threshold comprises determining the ...

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Номер записи: 93
10-04-2014 дата публикации

ARRAY MICROPHONE DEVICE AND GAIN CONTROL METHOD

Номер: US20140098972A1
Автор: YAMADA Katsushi
Принадлежит: Panasonic Corporation

The array microphone device has a microphone array composed of a plurality of microphone units, having a signal input section for inputting a signal from the microphone units to be corrected as a signal to be corrected; a reference signal input section; a gain variable section for making the levels of the signal to be corrected and the reference signal equal; and a gain control section. The gain control section includes a high-speed gain update section for changing the gain with a first amount of change per unit time upon time elapsed since the array microphone device is starting-up being below a predetermined period of time; and a low-speed gain update section for changing the gain with a second amount of change per unit time upon the elapsed time being above the predetermined period of time, the second amount of change being smaller than the first amount of change. 1. An array microphone device having a microphone array composed of a plurality of microphone units , comprising:a to-be-corrected signal input section configured to input a signal from a microphone unit to be corrected out of the plurality of microphone units as a signal to be corrected;a reference signal input section configured to input a reference signal;a gain variable section configured to amplify or attenuate the signal to be corrected so that a level of the signal to be corrected and a level of the reference signal is substantially equal to each other; anda gain control section configured to control a gain at the time of amplifying or attenuating the signal to be corrected, wherein:the gain control section includes:a first gain update section configured to change the gain with a first amount of change per unit time upon time elapsed since the array microphone device is starting-up being below a predetermined period of time; anda second gain update section configured to change the gain with a second amount of change per unit time upon the elapsed time being above or equal to the predetermined ...

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Номер записи: 94
06-01-2022 дата публикации

AUTOMATED TRANSCRIPT GENERATION FROM MULTI-CHANNEL AUDIO

Номер: US20220005492A1
Автор: DaSilva David Joseph, Donofrio Anthony, JR. James Andrew, Kaplan Jonathan Mordecai, Maraska
Принадлежит:

Systems and methods are described for generating a transcript of a legal proceeding or other multi-speaker conversation or performance in real time or near-real time using multi-channel audio capture. Different speakers or participants in a conversation may each be assigned a separate microphone that is placed in proximity to the given speaker, where each audio channel includes audio captured by a different microphone. Filters may be applied to isolate each channel to include speech utterances of a different speaker, and these filtered channels of audio data may then be processed in parallel to generate speech-to-text results that are interleaved to form a generated transcript. 1a plurality of microphones;audio mixer hardware configured to process a plurality of audio channels, wherein each of the plurality of microphones corresponds to a different channel of the plurality of audio channels; and receiving speaker identification information for each of the plurality of audio channels, wherein the speaker identification information for each individual audio channel identifies a person assigned to the individual audio channel and vocal characteristic information of the person, wherein the person assigned to the individual audio channel is physically located closer to a microphone assigned to the individual audio channel than to any other microphone of the plurality of microphones;', 'selecting a speech model to be used with respect to audio data for each of two or more of the plurality of audio channels, wherein a first speech model selected for a first audio channel is based at least in part on vocal characteristic information of a first person assigned to the first audio channel;', 'receiving at least a portion of multi-channel streaming audio from the audio mixer hardware, wherein the multi-channel streaming audio comprises audio signals captured from each of the plurality of microphones on a different channel of the plurality of audio channels;', 'applying one or ...

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Номер записи: 95
05-01-2017 дата публикации

Noise cancelation system and techniques

Номер: US20170004818A1
Автор: Chidananda Khatua
Принадлежит: Zpillow Inc

Techniques for noise cancelation include an automated method having the steps of: receiving signals from a plurality of microphones positioned within a microphone array outside a target area; identifying, from the received signals, a noise and position information for a source for the noise external to the target area before the noise reaches the target area; before the noise reaches the target area, determining a cancelation sound for the noise based on the noise and the position information; and playing the cancelation sound as the noise reaches the target area so as to significantly cancel the noise within the target area.

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Номер записи: 96
05-01-2017 дата публикации

Audio decoder for wind and microphone noise reduction in a microphone array system

Номер: US20170004836A1
Автор: Scott Patrick Campbell, Zhinian Jing
Принадлежит: GoPro Inc

An audio system encodes and decodes audio captured by a microphone array system in the presence of wind noise. The encoder encodes the audio signal in a way that includes beamformed audio signal and a “hidden” representation of a non-beamformed audio signal. The hidden signal is produced by modulating the low frequency signal to a high frequency above the audible range. A decoder can then either output the beamformed audio signal or can use the hidden signal to generate a reduced wind noise audio signal that includes the non-beamformed audio in the low frequency range.

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Номер записи: 97
05-01-2017 дата публикации

ENHANCEMENT OF NOISY SPEECH BASED ON STATISTICAL SPEECH AND NOISE MODELS

Номер: US20170004841A1
Автор: JENSEN Jesper
Принадлежит: OTICON A/S

A system for enhancement of noisy speech comprises an input unit is configured to subdivide the spectrum of the input signal into a plurality of frequency sub-bands and to provide time-frequency coefficients X(k,m) for a sequence [X(k, m′−D+1) . . . X(k,m′)] of observable noisy signal samples for each of said frequency sub-bands, where k and m are frequency and time indices, respectively, and D is larger than 1. The system further comprises enhancement processing unit configured to receive X(k,m) and to provide enhanced time-frequency coefficients Ŝ(k, m), a storage for statistical model(s) of speech and for statistical model(s) of noise, and an optimizing unit configured to provide said enhanced time-frequency coefficients Ŝ(k,m) using said statistical model of speech and said statistical model of noise, while considering said sequence [X(k, m′−D+1) . . . X(k, m′)] of observable noisy signal samples. Thereby the enhancement processing unit is able to determine the enhanced time-frequency coefficients based on the time-frequency coefficients for each of said frequency sub-bands. 1. A method for enhancement of speech in noise , the method comprising:providing a noisy input signal in a plurality of frequency sub-bands (k);for each of said frequency sub-bands providing time-frequency coefficients X(k,m) corresponding to a sequence [X(k,m′−D+1) . . . X(k,m′)] of observable noisy signal samples, where k and m are frequency and time indices, respectively, and D is larger than 1,enhancing said time-frequency coefficients X(k,m) thereby providing enhanced time-frequency coefficients Ŝ(k,m);providing a statistical model of speech;providing a statistical model of noise;providing said enhanced time-frequency coefficients Ŝ(k,m) using said statistical model of speech and said statistical model of noise, while considering said sequence [X(k, m′−D+1) . . . X(k,m′)] of observable noisy signal samples.2. The method according to wherein said statistical model of speech comprises a ...

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Номер записи: 98
05-01-2017 дата публикации

Accurate Forward SNR Estimation Based on MMSE Speech Probability Presence

Номер: US20170004842A1
Автор: Joshi Bijal, Lamy Guillaume
Принадлежит:

Acoustic noise in an audio signal is reduced by calculating a speech probability presence (SPP) factor using minimum mean square error (MMSE). The SPP factor, which has a value typically ranging between zero and one, is modified or warped responsive to a value obtained from the evaluation of a sigmoid function, the shape of which is determined by a signal-to-noise ratio (SNR), which is obtained by an evaluation of the signal energy and noise energy output from a microphone over time. The shape and aggressiveness of the sigmoid function is determined using an extrinsically-determined SNR, not determined by the MMSE determination. The extrinsically-determined SNR is obtained from a long term history of previously-determined speech presence probabilities and a long term history of previously-determined noise histories. 1. A method of reducing noise in an audio signal received at a microphone for a speech-processing device , the audio signal , that is received at the microphone being represented by a plurality of consecutive frames of data , each consecutive frame of data representing a plurality of consecutive samples of the received audio signal , the method comprising:converting the audio signal received at the microphone to a plurality of consecutive frames of data representing said audio signal;determining a signal to noise ratio (SNR) for a first frame responsive to energy generated by the microphone, and responsive to the determination of a softSNR and the determination of a realSNR for the first frame;determining a warped speech probability presence (SPP) factor for the first frame using a minimum mean square error (MMSE) determiner, which uses a SPP factor determined for the first frame, multiplied by a sigmoid function having a shape, the warped SPP factor for the first frame being determined by the determiner using the signal to noise ratio determined for the first frame;determining if the warped SPP factor is between pre-determined maximum and minimum values ...

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Номер записи: 99
05-01-2017 дата публикации

Externally Estimated SNR Based Modifiers for Internal MMSE Calculations

Номер: US20170004843A1
Автор: Lamy Guillaume
Принадлежит:

Acoustic noise in an audio signal is reduced by calculating a speech probability presence (SPP) factor using minimum mean square error (MMSE). The SPP factor, which has a value typically ranging between zero and one, is modified or warped responsive to a value obtained from the evaluation of a sigmoid function, the shape of which is determined by a signal-to-noise ratio (SNR), which is obtained by an evaluation of the signal energy and noise energy output from a microphone over time. The shape and aggressiveness of the sigmoid function is determined using an extrinsically-determined SNR, not determined by the MMSE determination. 1. A method of reducing noise in an audio signal received at a microphone for a speech-processing device , the audio signal , being represented by a plurality of consecutive frames of data , each frame representing a plurality of consecutive samples of the received signal , the method comprising:determining a speech probability presence (SPP) factor for a first frame using a minimum mean square error (MMSE) calculation, which uses a SPP factor determined for a previous frame, the SPP factor for the first frame also being determined using a first actual signal-to-noise ratio obtained from an actual audio signal and not from the MMSE calculation for the first frame of data;determining a warping factor for the SPP factor for the first frame of data, responsive to a determination of the actual signal-to-noise ratio;multiplying the SPP factor for the first frame of data by the determined warping factor to provide a gain factor, to be applied to a second frame of data that comes after the first frame of data;multiplying a second frame of data by the determined gain factor to provide a noise-suppressed second frame of data; andproviding the noise-suppressed second frame of data to the speech-processing device.2. (canceled)3. (canceled)4. The method of claim 1 , wherein the step of determining a warping factor for the SPP factor comprises:evaluating ...

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Номер записи: 100