Spectral Adjustment Patents (Class 381/94.2)
  • Publication number: 20100272288
    Abstract: A method and an apparatus for removing white noise in a portable terminal are provided. The method for removing the white noise in the portable terminal includes measuring a volume variation of a voice signal output from a power amplifier; detecting a frequency band including white noise using the measured volume variation; and removing signals of the detected frequency band in the voice signal before output to speaker.
    Type: Application
    Filed: April 15, 2010
    Publication date: October 28, 2010
    Applicant: SAMSUNG ELECTRONICS CO., LTD.
    Inventor: Jung-Eun HWANG
  • Publication number: 20100260352
    Abstract: A system identifying device for identifying an unknown system interposed between first and second input terminals.
    Type: Application
    Filed: September 19, 2008
    Publication date: October 14, 2010
    Inventor: Osamu Hoshuyama
  • Patent number: 7813499
    Abstract: A regression-based residual echo suppression (RES) system and process for suppressing the portion of the microphone signal corresponding to a playback of a speaker audio signal that was not suppressed by an acoustic echo canceller (AEC). In general, a prescribed regression technique is used between a prescribed spectral attribute of multiple past and present, fixed-length, periods (e.g., frames) of the speaker signal and the same spectral attribute of a current period (e.g., frame) of the echo residual in the output of the AEC. This automatically takes into consideration the correlation between the time periods of the speaker signal. The parameters of the regression can be easily tracked using adaptive methods. Multiple applications of RES can be used to produce better results and this system and process can be applied to stereo-RES as well.
    Type: Grant
    Filed: March 31, 2005
    Date of Patent: October 12, 2010
    Assignee: Microsoft Corporation
    Inventors: Amit Chhetri, Arungunram Surendran, Jack Stokes, John Platt
  • Patent number: 7809560
    Abstract: In a method and system for identifying speech sound and non-speech sound in an environment, a speech signal and other non-speech signals are identified from a mixed sound source having a plurality of channels. The method includes the following steps: (a) using a blind source separation (BSS) unit to separate the mixed sound source into a plurality of sound signals; (b) storing spectrum of each of the sound signals; (c) calculating spectrum fluctuation of each of the sound signals in accordance with stored past spectrum information and current spectrum information sent from the blind source separation unit; and (d) identifying one of the sound signals that has a largest spectrum fluctuation as the speech signal.
    Type: Grant
    Filed: January 26, 2006
    Date of Patent: October 5, 2010
    Assignee: Panasonic Corporation
    Inventors: Chia-Shin Yen, Chien-Ming Wu, Che-Ming Lin
  • Publication number: 20100239104
    Abstract: A noise attenuation system attenuates noise in an input signal. The system may estimate a power of the input signal, and determine a noise power value based on the input power estimate. The noise power value corresponds to an estimate of a noise power within the input signal. The system may determine an attenuation factor based on the noise power value, and attenuate the input signal by using the attenuation factor.
    Type: Application
    Filed: March 18, 2010
    Publication date: September 23, 2010
    Applicant: Harman Becker Automotive Systems GmbH
    Inventors: Bernd Iser, Gerhard Schmidt, Mathias Roder
  • Publication number: 20100232622
    Abstract: A novel system prevents surrounding sound to enter through a hearing apparatus, for instance through a ventilation opening, and reach an eardrum of the wearer in the form of interference sound. Contrary to auditory accessories designed especially to protect against noise, it is not possible for many hearing apparatus to compensate for such an interference sound by means of active noise cancellation. The hearing apparatuses do not have the special components needed. No compensation sound signal can therefore form with a correct phase. In accordance with the invention, a compensation sound is only generated for a relatively narrow spectral band. This spectral band is determined as a function of a hearing ability of the wearer of the hearing apparatus and/or as a function of a spectral distribution of the energy of the interference sound or a sound producing the interference sound. The improvement is particularly suited to compensating for an interference sound in a hearing device.
    Type: Application
    Filed: March 9, 2010
    Publication date: September 16, 2010
    Applicant: SIEMENS MEDICAL INSTRUMENTS PTE. LTD.
    Inventors: Robert Kasanmascheff, Ulrich Kornagel
  • Patent number: 7797154
    Abstract: Provision to reduce production of musical noise. A noise reduction device includes: means for calculating a rank for each element included in a first region having predetermined sizes in the time axis direction and in the frequency axis direction, depending on a value of the element, in a noise section of an observed signal indicating variation of a frequency spectrum with time; means for calculating a rank for each element included in a second region, depending on a value of the element, the second region having predetermined sizes in the time axis direction and in the frequency axis direction in the observed signal; and means for subtracting, from the values of the respective elements in the second region, values based on the values of the respective elements in the first region whose ranks correspond to ranks of respective elements in the second region.
    Type: Grant
    Filed: May 27, 2008
    Date of Patent: September 14, 2010
    Assignee: International Business Machines Corporation
    Inventor: Osamu Ichikawa
  • Patent number: 7787636
    Abstract: A sound recording device includes a disk drive, a microphone, and a controller. The disk drive writes data on a rotatable data storage disk therein. The microphone generates a microphone signal which includes a desired sound component and a noise component from noise that is generated by the disk drive. The controller filters the microphone signal to reduce the noise component from the disk drive relative to the desired sound component, and writes the filtered microphone signal on the disk. Accordingly, noise from the disk drive can be attenuated during recording by the sound recording device.
    Type: Grant
    Filed: February 6, 2006
    Date of Patent: August 31, 2010
    Assignee: Seagate Technology, LLC
    Inventors: Liu Yanning, Ben Chang, Babu Rahman, Timothy Glassburn, Erhard Schreck
  • Publication number: 20100215190
    Abstract: A noise suppressing device includes a plurality of sound input units inputting sounds from a given sound source and converting the sounds to sound signals on a time axis, a transfer characteristic obtaining unit performing frequency transform of the sound signals after dividing the sound signals into frames and calculating respective transfer characteristics of the sounds for each given frequency band, a storage unit storing the calculated transfer characteristics of the sounds, a frequency obtaining unit obtaining a frequency for updating the transfer characteristics stored in the storage unit for the frequency band, an updating unit updating the transfer characteristics every given number of frames corresponding to the obtained frequency based on the transfer characteristics for each frequency band, a generating unit generating suppression information for suppressing the noise component based on the updated transfer characteristics, and a suppression unit suppressing the noise component based on the suppres
    Type: Application
    Filed: February 23, 2010
    Publication date: August 26, 2010
    Applicant: FUJITSU LIMITED
    Inventor: Taisuke ITOU
  • Publication number: 20100215191
    Abstract: A noise removal device includes: an FFT analysis unit which receives a mixed sound including to-be-extracted sounds and noises, and determines frequency signals at time points in a time width; and a to-be-extracted sound determination unit which determines, for each to-be-extracted sound, frequency signals at the time points, satisfying conditions of (i) being equal to or greater than a first threshold value in number and (ii) having a phase distance between the frequency signals that is equal to or smaller than a second threshold value, wherein the phase distance is a distance between phases ??(t) of the condition-satisfying frequency signals when a phase of a frequency signal at a current time point t is ?(t) (radian) and the phase ??(t) is mod 2?(?(t)?2?ft), f denoting a reference frequency, and the predetermined time width is within 2 to 4 times the time window widths of the window functions.
    Type: Application
    Filed: May 4, 2010
    Publication date: August 26, 2010
    Inventors: Shinichi YOSHIZAWA, Yoshihisa Nakatoh
  • Patent number: 7778828
    Abstract: A method and system for automatic gain control of a speech signal in a communication system are disclosed. The gain of the speech signal can be controlled, based on a calculated gain value. This gain value is calculated on the basis of energy calculation and speech activity identification in the speech signal which is done by means of the encoder. Encoding the gain controlled speech signal for transmission follows the step of gain control.
    Type: Grant
    Filed: August 4, 2006
    Date of Patent: August 17, 2010
    Assignee: Sasken Communication Technologies Ltd.
    Inventors: Sachin Ghanekar, Anoop Deoras
  • Publication number: 20100202621
    Abstract: There is provided a signal processing device including: an audio signal acquisition portion that acquires audio signals; an external signal acquisition portion that acquires external signals; an output signal generation portion that generates output signals from the audio signals and the external signals; a mode setting portion that sets an external mode as an operation mode; and a fade control portion that controls the output signal generation portion in accordance with the operation mode. When the external mode is set, the fade control portion causes the output signal generation portion to generate the output signal for one of the right ear and the left ear of the user from at least the external signal, and also to generate the output signal for the other ear from at least the audio signal.
    Type: Application
    Filed: January 27, 2010
    Publication date: August 12, 2010
    Applicant: Sony Corporation
    Inventors: Yasunobu Murata, Kohei Asada
  • Patent number: 7756498
    Abstract: Disclosed is a channel estimator and a method for changing a coefficient of an IIR filter depending on a moving speed of a mobile communication terminal. In the channel estimator, a coefficient changing unit receives I and Q signals from a current base station, and selects a coefficient of the IIR filter optimized depending on the moving speed of the current mobile communication terminal. The coefficient changing unit sets the selected coefficient of the IIR filter to the IIR filter of the channel estimator. Accordingly, it is possible to prevent the performance degradation of the channel estimator caused by the speed of the mobile communication terminal.
    Type: Grant
    Filed: August 6, 2007
    Date of Patent: July 13, 2010
    Assignee: Samsung Electronics Co., Ltd
    Inventor: Soo-Jin Park
  • Publication number: 20100158269
    Abstract: Techniques pertaining to techniques to reduce wind noises effectively in recorded signals are disclosed. According to one aspect of the present invention, there is a strong correlation between two voice signals from target voices in the same frequency band sampled simultaneously by a pair of microphones in a common scene while there is a weak correlation between wind noises in the same frequency band of the two voice signals sampled simultaneously by the pair of microphones in the common scene. Taking advantage of this feature to provide a larger gain to the frequency band having a strong correlation and a smaller gain to the frequency band having a weak correlation, thereby the wind noise is reduced efficiently with minimum impact on the target voices.
    Type: Application
    Filed: May 31, 2009
    Publication date: June 24, 2010
    Inventor: Chen Zhang
  • Patent number: 7742608
    Abstract: A method and apparatus for detecting a singing frequency in a signal processing system using two neural-networks is disclosed. The first one (a hit neural network) monitors the maximum spectral peak FFT bin as it changes with time. The second one (change neural network) monitors the monotonic increasing behavior. The inputs to the neural-networks are the maximum spectral magnitude bin and its rate of change in time. The output is an indication whether howling is likely to occur and the corresponding singing frequency. Once the singing frequency is identified, it can be suppressed using any one of many available techniques such as notch filters. Several improvements of the base method or apparatus are also disclosed, where additional neural networks are used to detect more than one singing frequency.
    Type: Grant
    Filed: March 31, 2005
    Date of Patent: June 22, 2010
    Assignee: Polycom, Inc.
    Inventors: Kwan Kin Truong, James Steven Joiner
  • Patent number: 7725314
    Abstract: A method and apparatus identify a clean speech signal from a noisy speech signal. To do this, a clean speech value and a noise value are estimated from the noisy speech signal. The clean speech value and the noise value are then used to define a gain on a filter. The noisy speech signal is applied to the filter to produce the clean speech signal. Under some embodiments, the noise value and the clean speech value are used in both the numerator and the denominator of the filter gain, with the numerator being guaranteed to be positive.
    Type: Grant
    Filed: February 16, 2004
    Date of Patent: May 25, 2010
    Assignee: Microsoft Corporation
    Inventors: Jian Wu, James G. Droppo, Li Deng, Alejandro Acero
  • Patent number: 7720233
    Abstract: A signal processor includes: a first adaptive filter that takes a first signal as input and generates a first pseudo signal; a first subtractor that subtracts the first pseudo signal from a second signal to supply a first differential signal as output; a second adaptive filter that takes the first signal as input to generate a second pseudo signal; a second subtractor that subtracts the second pseudo signal from the second signal to supply a second differential signal as output; a first step size control circuit that generates a first step size used in updating the first adaptive filter in accordance with the relation between the second pseudo signal and the second differential signal; and a second step size control circuit that generates a second step size used in updating the second adaptive filter in accordance with the relation between the first signal and the second signal.
    Type: Grant
    Filed: August 31, 2004
    Date of Patent: May 18, 2010
    Assignee: NEC Corporation
    Inventors: Miki Sato, Akihiko Sugiyama
  • Patent number: 7715567
    Abstract: A method of denoise a stereo signal comprising a stereo sum signal and a stereo difference signal, performs a frequency selective stereo to mono blending based on the masking effect of the human auditory system. Therefore, a stereo signal noise reducer, comprising a first filter bank (1) to split the stereo difference signal (l?r) into a plurality of subbands, respective first multipliers (CO, . . . , CN) to weight each of the subbands of the stereo difference signal with a respective corresponding control signal (CO, . . . , CN), and a first adder (3) to sum all weighted subbands of the stereo difference signal (l?r) to build a frequency selective weighted stereo difference signal (diff), within which a number and width of the subbands obtained via the first filter bank (1) are choosen according to the properties of the human auditory system, further comprises a weighting factor determination unit which determines a respective control signal (CO, . . .
    Type: Grant
    Filed: August 18, 2006
    Date of Patent: May 11, 2010
    Assignee: Sony Deutschland GmbH
    Inventor: Jens Wildhagen
  • Publication number: 20100104113
    Abstract: In a noise suppression device, an audio detector detects presence or absence of audio in an input signal. A first noise spectrum estimator estimates a noise spectrum contained in the input signal based on the input signal and detection result of the audio detector. A second noise spectrum estimator estimates the noise spectrum based on the input signal regardless of the detection result of the audio detector. A noise spectrum calculator calculates a final noise spectrum estimation value according to a length of detecting time during which the audio detector continuously detects the audio and based on first and second noise spectrum estimation values that are obtained as estimation results by the first and second noise spectrum estimators. A gain calculator calculates a noise suppression gain based on the final noise spectrum estimation value. A noise suppressor suppresses noise contained in the input signal by applying the noise suppression gain to the input signal.
    Type: Application
    Filed: October 23, 2009
    Publication date: April 29, 2010
    Applicant: YAMAHA CORPORATION
    Inventor: Encai LIU
  • Publication number: 20100092000
    Abstract: Provided are an apparatus and method for estimating noise and a noise reduction apparatus employing the same. The noise estimation apparatus estimates noise by blocking audio signals from a direction of a target sound source from received audio signals, and compensating for distortions from directivity gains of a target sound blocker blocking the audio signals from the target sound source.
    Type: Application
    Filed: September 10, 2009
    Publication date: April 15, 2010
    Inventors: Kyu-hong KIM, Kwang-cheol Oh
  • Patent number: 7693293
    Abstract: Provided is a sound processing device including: a sound input unit for dividing an input sound into predetermined time units; a sound processing unit for encoding the input sound thus divided; a noise detecting unit; and an output control unit for replacing encoded data on the input sound with silent data according to detection results of the noise detecting unit. Also provided is an input sound processing method including: encoding an input sound; judging whether or not the input sound contains a noise; and replacing a noise portion contained in the encoded input sound with silent data.
    Type: Grant
    Filed: August 26, 2005
    Date of Patent: April 6, 2010
    Assignee: NEC Corporation
    Inventors: Miyako Nemoto, Satoshi Hosokawa
  • Patent number: 7689275
    Abstract: A method and apparatus for filtering an electromyogram (EMG) signal from a raw signal which includes a contribution from an electrocardiogram (EKG) signal is disclosed. The method includes the steps of estimating an attribute (such as a Fourier transform) of both the EMG contribution to the raw signal and the EKG contribution to the raw signal and, dependent on both frequency spectrums, determining an EMG window in a frequency range and obtaining the EMG signal by passing it through a filter defined by the frequency range. The method is particularly used when monitoring a multi-channel electrical recording from a plurality of electrodes attached to a patient's diaphragm.
    Type: Grant
    Filed: November 18, 2004
    Date of Patent: March 30, 2010
    Assignee: Maquet Critical Care AB
    Inventors: Urban Blomberg, Fredrik Jalde
  • Patent number: 7676046
    Abstract: A method of removing noise and interference from a signal by receiving the signal, calculating a joint time-frequency domain of the signal, estimating instantaneous frequencies of the joint time-frequency domain, modifying each estimated instantaneous frequency, if necessary, to correspond to a frequency of the joint time-frequency domain to which it most closely compares, redistributing the elements within the joint time-frequency domain according to the estimated instantaneous frequencies as modified, computing a magnitude for each element in the joint time-frequency domain as redistributed, plotting the results as the time-frequency representation of the signal, identifying in the plot any noise and interference components in the received signal, eliminating from the redistributed joint time-frequency domain elements that correspond to noise and interference, and recovering a signal devoid of noise and interference from the modified redistributed joint time-frequency domain.
    Type: Grant
    Filed: June 9, 2005
    Date of Patent: March 9, 2010
    Assignee: The United States of America as represented by the Director of the National Security Agency
    Inventors: Douglas J. Nelson, David C. Smith
  • Patent number: 7672466
    Abstract: An audio signal processing apparatus includes a splitting unit for splitting an audio signal of a first system and another audio signal of a second system into pluralities of frequency band components, a level comparing unit for calculating a level ratio or a level difference between each of the frequency bands of the first system and each of the frequency bands of the second systems, and an output control unit for removing frequency band components whose level ratio or level difference calculated by the level comparing unit is equal and substantially equal to a predetermined value from at least one of the first and second systems.
    Type: Grant
    Filed: September 19, 2005
    Date of Patent: March 2, 2010
    Assignee: Sony Corporation
    Inventors: Yuji Yamada, Koyuru Okimoto
  • Patent number: 7672842
    Abstract: A method and system processes a speech signal. A fast Fourier transform is performed on a speech signal to produce a speech signal having a plurality of frequency bands in a frequency domain.
    Type: Grant
    Filed: July 26, 2006
    Date of Patent: March 2, 2010
    Assignee: Mitsubishi Electric Research Laboratories, Inc.
    Inventors: Bhiksha Ramakrishnan, Bent Schmidt-Nielsen, Lorenzo Turicchia, Rahul Sarpeshkar
  • Publication number: 20100036659
    Abstract: The present invention relates to a method for signal processing comprising the steps of providing a set of prototype spectral envelopes, providing a set of reference noise prototypes, wherein the reference noise prototypes are obtained from at least a sub-set of the provided set of prototype spectral envelopes, detecting a verbal utterance by at least one microphone to obtain a microphone signal, processing the microphone signal for noise reduction based on the provided reference noise prototypes to obtain an enhanced signal and encoding the enhanced signal based on the provided prototype spectral envelopes to obtain an encoded enhanced signal.
    Type: Application
    Filed: August 7, 2009
    Publication date: February 11, 2010
    Applicant: Nuance Communications, Inc.
    Inventors: Tim Haulick, Mohamed Krini, Shreyas Paranjpe, Gerhard Schmidt
  • Publication number: 20100008520
    Abstract: A noise suppression estimation device calculates a noise index value which varies according to kurtosis of a frequence distribution of magnitude of a sound signal before or after suppression of the noise component, the noise index value indicating a degree of occurrence of musical noise after suppression of the noise component in a frequency domain. For example, the noise suppression estimation device calculates first kurtosis of a frequence distribution of magnitude of the sound signal before suppression of the noise component, calculates second kurtosis of a frequence distribution of magnitude of the sound signal after suppression of the noise component, and calculates the noise index value from the first kurtosis and the second kurtosis.
    Type: Application
    Filed: July 8, 2009
    Publication date: January 14, 2010
    Applicants: Yamaha Corporation, Nara Institute of Science and Technology National University Corporation
    Inventors: Hiroshi SARUWATARI, Yoshihisa Uemura, Kazunobu Kondo
  • Publication number: 20090296958
    Abstract: It is possible to provide a noise suppression method, device, and program capable of realizing a sound image positioning of an output side corresponding to an input side with a small calculation amount. The device includes a common suppression coefficient calculation unit for receiving conversion outputs from a plurality of channels and calculating a suppression coefficient common to the channels.
    Type: Application
    Filed: June 29, 2007
    Publication date: December 3, 2009
    Applicant: NEC CORPORATION
    Inventor: Akihiko Sugiyama
  • Patent number: 7613310
    Abstract: A method for reducing noise associated with an audio signal received through a microphone sensor array is provided. The method initiates with enhancing a target signal component of the audio signal through a first filter. Simultaneously, the target signal component is blocked by a second filter. Then, the output of the first filter and the output of the second filter are combined in a manner to reduce noise without distorting the target signal. Next, an acoustic set-up associated with the audio signal is periodically monitored. Then, a value of the first filter and a value of the second filter are both calibrated based upon the acoustic set-up. A system capable of isolating a target audio signal from multiple noise sources, a video game controller, and an integrated circuit configured to isolate a target audio signal are included.
    Type: Grant
    Filed: August 27, 2003
    Date of Patent: November 3, 2009
    Assignee: Sony Computer Entertainment Inc.
    Inventor: Xiadong Mao
  • Patent number: 7609841
    Abstract: A decorrelation method for improving feedback cancellation utilizes a small frequency shifting ratio, on the order of 0.3 percent. Frequency shifting is applied only to the high frequency portion of the signal, which is shifted alternately upward and downward.
    Type: Grant
    Filed: August 4, 2004
    Date of Patent: October 27, 2009
    Assignee: House Ear Institute
    Inventors: Daniel J. Freed, Sigfrid D. Soli
  • Patent number: 7602926
    Abstract: An audio enhancement system (1) for speech recognition or voice control is described, comprising a signal input for carrying a distorted desired signal (z), a reference signal input, and a spectral processor (SP) coupled to both signal inputs for processing the distorted desired signal (z) by means of a reference signal (x) acting as an estimate for the distortion of the desired signal. The spectral processor (SP) is equipped for said processing such that a factor C? is determined, whereby said estimate is a function of the factor C? times the spectral power of the reference signal (x), and the factor C is determined as the spectral ratio between those components of the signals z and x, which are essentially stationary with time. Such a factor determined by stationary parts of those signals makes application of a critical speech detector in the audio enhancement system superfluous.
    Type: Grant
    Filed: June 19, 2003
    Date of Patent: October 13, 2009
    Assignee: Koninklijke Philips Electronics N.V.
    Inventor: David Antoine Christian Marie Roovers
  • Publication number: 20090252379
    Abstract: An information processing apparatus includes a feature value detecting section, an image processing section, and an audio processing section. The feature value detecting section determines, when a first image and a second image that are captured at different positions include a specific subject, a feature value of the subject included in the supplied first and second images. The image processing section detects motion of the subject on the basis of the feature value determined by the feature value detecting section. The audio processing section localizes a sound image of the subject in accordance with the motion of the subject detected by the image processing section.
    Type: Application
    Filed: March 27, 2009
    Publication date: October 8, 2009
    Applicant: Sony Corporation
    Inventors: Tetsujiro Kondo, Tetsushi Kokubo, Kenji Tanaka, Hitoshi Mukai, Hirofumi Hibi, Kazumasa Tanaka, Takuro Ema, Hiroyuki Morisaki
  • Patent number: 7596231
    Abstract: Methods, machines, systems and machine-readable instructions for processing input audio signals are described. In one aspect, an input audio signal has a noise period that includes a targeted noise signal and a noise-free period free of the targeted noise signal. The input audio signal in the noise-free period is divided into spectral time slices each having a respective spectrum. Ones of the spectral time slices of the input audio signal are selected based on the respective spectra of the spectral time slices. An output audio signal is composed for the noise period based at least in part on the selected ones of the spectral time slices of the input audio signal in the noise-free period.
    Type: Grant
    Filed: May 23, 2005
    Date of Patent: September 29, 2009
    Assignee: Hewlett-Packard Development Company, L.P.
    Inventor: Ramin Samadani
  • Publication number: 20090196434
    Abstract: A method, an apparatus, and a computer program, which can suppress a low frequency range component with a small amount of calculation, and can achieve a noise suppression of high quality, are provided. The noise superposed in a desired signal of an input signal is suppressed by converting the input signal to a frequency domain signal; correcting an amplitude of the frequency domain signal to obtain an amplitude corrected signal; obtaining an estimated noise by using the amplitude corrected signal; determining a suppression coefficient by using the estimated noise and the amplitude corrected signal; and weighting the amplitude corrected signal with the suppression coefficient.
    Type: Application
    Filed: August 28, 2006
    Publication date: August 6, 2009
    Applicant: NEC Corporation
    Inventors: Akihiko Sugiyama, Masanori Katou
  • Patent number: 7567548
    Abstract: The perceptual quality of voice signals used for Voice over IP (VoIP) systems is assessed. The content of the voice data packets may be altered in order to increase the perceived quality of the VoIP system.
    Type: Grant
    Filed: June 26, 2001
    Date of Patent: July 28, 2009
    Assignee: British Telecommunications plc
    Inventors: Richard J B Reynolds, Philip Gray, Michael P Hollier, Antony W Rix
  • Patent number: 7562013
    Abstract: The present invention provides a method for recovering target speech based on shapes of amplitude distributions of split spectra obtained by use of blind signal separation.
    Type: Grant
    Filed: August 31, 2004
    Date of Patent: July 14, 2009
    Assignee: Kitakyushu Foundation For The Advancement of Industry, Science and Technology
    Inventors: Hiromu Gotanda, Keiichi Kaneda, Takeshi Koya
  • Patent number: 7561702
    Abstract: Method and system adapted to modifying an audio signal or speech signal comprising a step in which the frequency spectrum S(k) of the signal is converted by the application of a non-linear function. The method comprises at least the following steps: firstly, determining the signal level A(k), B(k) associated with a frequency k by taking account of different levels a(k), b(k) of the signal for the frequency k concerned and/or the neighboring frequencies (step 2a, step 8a); secondly, applying the non-linear function to said level A(k), B(k).
    Type: Grant
    Filed: June 21, 2002
    Date of Patent: July 14, 2009
    Assignee: Thales
    Inventor: Pierre André Laurent
  • Publication number: 20090175466
    Abstract: In one embodiment, a directional microphone array having (at least) two microphones generates forward and backward cardioid signals from two (e.g., omnidirectional) microphone signals. An adaptation factor is applied to the backward cardioid signal, and the resulting adjusted backward cardioid signal is subtracted from the forward cardioid signal to generate a (first-order) output audio signal corresponding to a beampattern having no nulls for negative values of the adaptation factor. After low-pass filtering, spatial noise suppression can be applied to the output audio signal. Microphone arrays having one (or more) additional microphones can be designed to generate second- (or higher-) order output audio signals.
    Type: Application
    Filed: March 9, 2007
    Publication date: July 9, 2009
    Applicant: MH ACOUSTICS, LLC
    Inventors: Gary W. Elko, Jens M. Meyer, Tomas Fritz Gaensler
  • Patent number: 7542577
    Abstract: An input sound processor compares power at each frequency component of an input sound with a reference value, and sets multiplication points indicating frequency components at which the total power of the input sound is to be determined. A product-sum operation is performed at the multiplication points on the power at each frequency component and the square amplitude of each filter coefficient indicating the transfer characteristic from a loudspeaker to a microphone to estimate the total power of the input sound at the position of the microphone.
    Type: Grant
    Filed: March 1, 2005
    Date of Patent: June 2, 2009
    Inventor: Shingo Kiuchi
  • Publication number: 20090116662
    Abstract: An audio processing method used in a microphone is provided. Firstly, a sound signal is received. Next, the sound signal is transduced to a first voltage signal. The first voltage signal is interfered with by a second voltage signal resulting from electromagnetic wave penetrating into the microphone. Next, the second voltage signal is filtered out from the interfered first voltage signal. Finally, the filtered first voltage signal is amplified.
    Type: Application
    Filed: November 6, 2007
    Publication date: May 7, 2009
    Applicant: FORTEMEDIA, INC.
    Inventor: Li-Te Wu
  • Publication number: 20090112579
    Abstract: A system improves speech intelligibility by reconstructing speech segments. The system includes a low-frequency reconstruction controller programmed to select a predetermined portion of a time domain signal. The low-frequency reconstruction controller substantially blocks signals above and below the selected predetermined portion. A harmonic generator generates low-frequency harmonics in the time domain that lie within a frequency range controlled by a background noise modeler. A gain controller adjusts the low-frequency harmonics to substantially match the signal strength to the time domain original input signal.
    Type: Application
    Filed: May 23, 2008
    Publication date: April 30, 2009
    Applicant: QNX SOFTWARE SYSTEMS (WAVEMAKERS), INC.
    Inventors: Xueman Li, Rajeev Nongpiur, Frank Linseisen, Phillip A. Hetherington
  • Patent number: 7515703
    Abstract: A method for providing an embellishment representation of a noise information is discloses.
    Type: Grant
    Filed: May 19, 2008
    Date of Patent: April 7, 2009
    Assignee: International Business Machines Corporation
    Inventors: Travis M. Grigsby, Steven Michael Miller, Lisa Anne Seacat
  • Patent number: 7508948
    Abstract: A method of removing reverberation from audio signals is disclosed. The method comprises spectro-temporally analyzing the first audio signal and the second audio signal to derive an energy function of time for a plurality of frequency bands. The method further comprises determining a delay stability between the energy function of time for the first audio signal and the second audio signal in each band, determining a gain function in each band based on the delay stability, adjusting the energy of the first audio signal and the second audio signal using the gain function within each band, and resynthesizing audio signals from the energy in each band of the first audio signal and the second audio signal.
    Type: Grant
    Filed: October 5, 2004
    Date of Patent: March 24, 2009
    Assignee: Audience, Inc.
    Inventors: David Justin Klein, Lloyd Watts
  • Publication number: 20090074203
    Abstract: A portable assistive listening system for enhancing sound for hearing impaired individuals includes a functional hearing aid and a separate handheld digital signal processing (DSP) device. The focus of the embodiments is directed to the handheld DSP device and a method of processing audio signals. The DSP device includes a programmable digital signal processor, a UWB transceiver for communicating with the hearing aid and/or other wireless audio sources, an LCD display, and a user input device (keypad). The handheld device is user programmable to apply different sound processing algorithms for processing sound signals received from the hearing aid and/or other audio source. The handheld device is capable of receiving audio signals from multiple sources, and gives the user control over selection of incoming sound sources and selective processing of audio.
    Type: Application
    Filed: September 13, 2007
    Publication date: March 19, 2009
    Applicant: BIONICA CORPORATION
    Inventors: KIPP BRADFORD, RALPH A. BECKMAN, JOHN F. MURPHY, III
  • Publication number: 20090073950
    Abstract: A wireless, multi-function audio gateway device provides communication between the headset and at least one audio gateway. The headset includes a housing having at least one multifunction button, and a first and a second microphone. The first microphone is located closer to a user's mouth than the second microphone. The headset also includes a flexible ear bud, a speaker, a volume control button, a rechargeable battery, a USB port, a detachable ear wrap, and a programmable baseband IC. The programmable baseband IC is configured for wireless communications to allow interfacing between the at least one audio gateway and the wireless headset. Methods for improved noise suppression, pairing and communicating with multiple audio gateways simultaneously, and allowing headset-to-headset communications between two wireless, multiple audio gateway headsets are also disclosed.
    Type: Application
    Filed: September 19, 2007
    Publication date: March 19, 2009
    Applicant: CALLPOD INC.
    Inventors: Darren S. Guccione, Craig B. Lurey
  • Publication number: 20090046865
    Abstract: The present invention is to provide a sound image localization apparatus which can prevent the lowering of the amplitude of the sound image localizing signal, the occurrence of clipping, and deterioration of the sound image localization component of the sound image localizing signal.
    Type: Application
    Filed: March 12, 2007
    Publication date: February 19, 2009
    Applicant: MATSUSHITA ELECTRIC INDUSTRIAL CO., LTD.
    Inventor: Gempo Ito
  • Patent number: 7489789
    Abstract: The invention regards a method for noise reduction in an audio device whereby an electrical and/or digital signal which represents sound is routed simultaneously through:—a signal analysis path, and—a signal processing path wherein the signal amplification is individually controllable in specific frequency bands by attenuation values derived from the signal analysis path, whereby the signal in the signal analysis path is routed simultaneously through:—a first detector which identifies the presence of speech indicators in the overall signal, and—a second detector which in a predefined number of frequency bands detects the modulation amplitude, and—where attenuation values in each of the predefined frequency bands are calculated based on the combined results of the first detector and the modulation amplitude in the specific frequency band detected by the second detector,—where the attenuation values in the predefined number of frequency bands are routed to the signal processing path in order to attenuate the si
    Type: Grant
    Filed: February 28, 2005
    Date of Patent: February 10, 2009
    Assignee: Oticon A/S
    Inventor: Thomas Kaulberg
  • Patent number: 7480614
    Abstract: The present invention provides an energy feature extraction method for noisy speech recognition. At first, noisy speech energy of an input noisy speech is computed. Next, the noise energy in the input noisy speech is estimated. Then, the estimated noise energy is subtracted from the noisy speech energy to obtain estimated clean speech energy. Finally, delta operations are performed on the log of the estimated clean speech energy to determine the energy derivative features for the noisy speech.
    Type: Grant
    Filed: December 30, 2003
    Date of Patent: January 20, 2009
    Assignee: Industrial Technology Research Institute
    Inventor: Tai-Huei Huang
  • Publication number: 20080304679
    Abstract: An apparatus processes an acoustic input signal to provide an output signal with reduced noise. The apparatus weights the input signal based on a frequency-dependent weighting function. A frequency-dependent threshold function bounds the weighting function from below.
    Type: Application
    Filed: May 9, 2008
    Publication date: December 11, 2008
    Inventors: Gerhard Uwe Schmidt, Raymond Bruckner, Markus Buck, Ange Tchinda-Pockem, Mohamed Krini
  • Publication number: 20080294432
    Abstract: Provides speech enhancement techniques which are effective even for extemporaneous noise without a noise interval and unknown extemporaneous noise. An example of a signal enhancement device includes: spectral subtraction means for subtracting a given reference signal from an input signal containing a target signal and a noise signal by spectral subtraction; an adaptive filter applied to the reference signal; and coefficient control means for controlling a filter coefficient of the adaptive filter in order to reduce components of the noise signal in the input signal. In the signal enhancement device, a database of a signal model concerning the target signal expressing a given feature by means of a given statistical model is provided, and the filter coefficient is controlled based on the likelihood of the signal model with respect to an output signal from the spectral subtraction means.
    Type: Application
    Filed: May 26, 2008
    Publication date: November 27, 2008
    Inventors: Tetsuya Takiguchi, Masafumi Nishimura