Spectral Adjustment Patents (Class 381/94.2)
  • Patent number: 10448142
    Abstract: Disclosed is a signal processing apparatus including a surrounding sound signal acquisition unit, a NC (Noise Canceling) signal generation part, a cooped-up feeling elimination signal generation part, and an addition part. The surrounding sound signal acquisition unit is configured to collect a surrounding sound to generate a surrounding sound signal. The NC signal generation part is configured to generate a noise canceling signal from the surrounding sound signal. The cooped-up feeling elimination signal generation part is configured to generate a cooped-up feeling elimination signal from the surrounding sound signal. The addition part is configured to add together the generated noise canceling signal and the cooped-up feeling elimination signal at a prescribed ratio.
    Type: Grant
    Filed: November 28, 2017
    Date of Patent: October 15, 2019
    Assignee: SONY CORPORATION
    Inventors: Yasunobu Murata, Kohei Asada, Yushi Yamabe
  • Patent number: 10339953
    Abstract: A howling detection method is provided. A window separation processing is processed on an audio signal to obtain a plurality of analysis windows. A signal energy indicator value of each preset frequency in at least one analysis window is obtained by using a preset perceptual coefficient corresponding to each frequency, to obtain a perceptual energy indicator value of each frequency, the preset perceptual coefficient corresponding to each frequency indicating a sensitivity of a human ear to a sound of each frequency. It is determined whether howling occurs according to the perceptual energy indicator value of each frequency in the at least one analysis window.
    Type: Grant
    Filed: July 24, 2018
    Date of Patent: July 2, 2019
    Assignee: TENCENT TECHNOLOGY (SHENZHEN) COMPANY LIMITED
    Inventor: Junbin Liang
  • Patent number: 10241667
    Abstract: A method consistent with certain implementation involves presenting a graphical user interface (GUI) to a user on a display, where the GUI presents a visual representation of a room that is adapted to be adjusted in size and shape by user manipulation of a controller; the GUI has a drop and drag menu adapted to selection of an object from a plurality of objects for placement at any selected position within the room; at least one of the objects comprising a loudspeaker; and where the GUI provides for input of data characterizing the loudspeaker. This abstract is not to be considered limiting, since other embodiments may deviate from the features described in this abstract.
    Type: Grant
    Filed: January 30, 2015
    Date of Patent: March 26, 2019
    Assignee: SONY CORPORATION
    Inventors: Djung Nguyen, Ted Dunn, Andy Nguyen, Nobukazu Sugiyama, Lobrenzo Wingo
  • Patent number: 10140089
    Abstract: A system and method that enhances spoken utterances by capturing one or more microphone signals. The system and method estimates a plurality of echo paths from each of the one or more microphone signals and synthesizes a speech reinforcement signal in response to and corresponding to the one or more microphone signals. The system and method concatenates portions of the synthesized reinforcement signal with the captured microphone signals and processes the captured microphone signals in response to the estimated plurality of echo paths.
    Type: Grant
    Filed: August 9, 2017
    Date of Patent: November 27, 2018
    Assignees: 2236008 Ontario Inc., BlackBerry Limited
    Inventors: Shreyas Paranjpe, Phillip Alan Hetherington, Leonard Charles Layton
  • Patent number: 10128905
    Abstract: A system for classifying impulsive noise on a communications signal comprises an impulse signal generator, an integrator, a first comparator, and an impulse peak detector. The impulse signal generator receives a communications signal that includes impulsive noise and is configured to provide an impulse signal that includes just the impulsive noise. The integrator receives the impulse signal and integrates the impulse signal to determine the power of the impulse signal. The first comparator receives the impulse signal and is configured to compare the impulse signal to a first reference signal and indicate the time during which the value of the impulse signal is greater than the value of the first reference signal. The impulse peak detector receives the impulse signal and is configured to process the impulse signal, compare the processed signal to a second reference signal, and detect the peak value of the impulse signal.
    Type: Grant
    Filed: June 26, 2013
    Date of Patent: November 13, 2018
    Assignee: Alarm.com Incorporated
    Inventors: Alain Charles Briancon, John Berns Lancaster, Curtis Scott Crawford, Robert Leon Lutes, Eric Alexander Shumaker, Marc Anthony Epard
  • Patent number: 10083005
    Abstract: A user speech interface for interactive media guidance applications, such as television program guides, guides for audio services, guides for video-on-demand (VOD) services, guides for personal video recorders (PVRs), or other suitable guidance applications is provided. Voice commands may be received from a user and guidance activities may be performed in response to the voice commands.
    Type: Grant
    Filed: April 21, 2016
    Date of Patent: September 25, 2018
    Assignee: Rovi Guides, Inc.
    Inventors: M. Scott Reichardt, David M. Berezowski, Michael D. Ellis, Toby DeWeese
  • Patent number: 10043530
    Abstract: A noise suppressor has a band extractor to separate signal by frequency band; and per-band units for each of band including noise estimator and SNR computation units. The per-band unit has a histogrammer to give histograms of current and past SNRs, and a gain-curve updater computes gain curves from the histogram. Gain curves are used to determine raw gains from current SNRs, raw gain is filtered and controls a variable gain unit to provide band-specific gain-adjusted, signals that are recombined into a noise-reduced frequency-domain output. Raw gain filtering may include finite-impulse-response filtering and weighted averaging of intermediate gains of a current and adjacent-band per-band unit. The method includes separating an input into frequency bands, estimating in-band noise, and deriving a band SNR. Then, histogramming the SNR and updating a gain curve from the histogram, and finding a raw gain using the gain curve and current SNR.
    Type: Grant
    Filed: February 8, 2018
    Date of Patent: August 7, 2018
    Assignee: OmniVision Technologies, Inc.
    Inventors: Dong Shi, Chung-An Wang
  • Patent number: 9992444
    Abstract: In some embodiments, a method includes receiving, at a device, digital audio content to be converted by a digital-to-analog converter to produce analog audio content. The digital audio content has at least one audible frequency. The method also includes selecting, at the device, a first signal and a second signal to output with the analog audio content. The first signal has an inaudible carrier frequency and a bandwidth, and the second signal also has an inaudible carrier frequency and a bandwidth. A difference between the first signal and the second signal is an undesired audible signal. The method further includes outputting, from the device, the analog audio content, the first signal and the second signal, such that when the analog audio content is detected by a recording device the undesired audible signal is detected with the analog audio content.
    Type: Grant
    Filed: August 26, 2016
    Date of Patent: June 5, 2018
    Assignee: Pegasus Media Security, LLC
    Inventors: Paul A. Kline, Gil Kline, Allan Weinstein, David J. Weinstein
  • Patent number: 9980043
    Abstract: A method comprising determining an audio noise spectrum that represents external audio noise, determining an audio reference spectrum that represents audio content to be perceived by a listener, and determining a compensation spectrum based on the audio noise spectrum and based on the audio reference spectrum.
    Type: Grant
    Filed: March 21, 2016
    Date of Patent: May 22, 2018
    Assignee: SONY CORPORATION
    Inventor: Michael Enenkl
  • Patent number: 9691382
    Abstract: A voice control device has a speech command recognizer, a sensor data processor and a decision making circuit. The speech command recognizer is arranged for performing speech command recognition to output a recognized speech command. The sensor data processor is arranged for processing sensor data generated from at least one auxiliary sensor to generate a detection output. The decision making circuit is arranged for deciding a response of the voice control device according to the recognized speech command and the detection output. The same speech command is able to trigger difference responses according to the detection output (e.g., detected motion). Besides, an adaptive training process may be employed to improve the accuracy of the sensor data processor. Hence, the voice control device may have improved performance of the voice control feature due to a reduce occurrence probability of miss errors and false alarm errors.
    Type: Grant
    Filed: November 13, 2013
    Date of Patent: June 27, 2017
    Assignee: MEDIATEK INC.
    Inventors: Chao-Ling Hsu, Yiou-Wen Cheng, Xin-Wei Shih, Jyh-Horng Lin
  • Patent number: 9685921
    Abstract: Loudness control systems or methods may normalize audio signals to a predetermined loudness level. If the audio signal includes moderate background noise, then the background noise may also be normalized to the target loudness level. Noise signals may be detected using content-versus-noise classification, and a loudness control system or method may be adjusted based on the detection of noise. Noise signals may be detected by signal analysis in the frequency domain or in the time domain. Loudness control systems may also produce undesirable audio effects when content shifts from a high overall loudness level to a lower overall loudness level. Such loudness drops may be detected, and the loudness control system may be adjusted to minimize the undesirable effects during the transition between loudness levels.
    Type: Grant
    Filed: March 15, 2013
    Date of Patent: June 20, 2017
    Assignee: DTS, INC.
    Inventors: Brandon Smith, Aaron Warner, Jeff Thompson
  • Patent number: 9635473
    Abstract: The application relates to a hearing device comprising a beamformer of the generalized sidelobe canceler (GSC) type. The application further relates to a method of operating a hearing device. The disclosure addresses a problem which occurs when using a GSC structure in a hearing device application. The problem arises due to a non-ideal target-cancelling beamformer. As a consequence, a target signal impinging from the look direction can—unintentionally—be attenuated by as much as 30 dB. To resolve this problem, it is proposed to monitor the difference between the output signals from the all-pass beamformer and the target-cancelling beamformer to control a time-varying regularization parameter in the GSC update. This has the advantage of providing a computationally simple solution to the non-ideality of the GSC beamformer. The invention may e.g. be used in hearing aids, headsets, ear phones, active ear protection systems, or combinations thereof.
    Type: Grant
    Filed: September 16, 2015
    Date of Patent: April 25, 2017
    Assignee: OTICON A/S
    Inventors: Meng Guo, Jan Mark de Haan, Jesper Jensen
  • Patent number: 9589577
    Abstract: A speech recognition apparatus and a speech recognition method are provided. In the invention, whether an original voice sampling signal corresponding to a target voice frame is a consonant signal is determined according to at least one of a ratio of an energy of a low-pass sampling signal to an energy of the original voice sampling signal and a ratio value of an energy of a second consonant frequency band signal.
    Type: Grant
    Filed: March 17, 2015
    Date of Patent: March 7, 2017
    Assignee: Acer Incorporated
    Inventors: Po-Jen Tu, Jia-Ren Chang, Kai-Meng Tzeng
  • Patent number: 9495973
    Abstract: A speech recognition apparatus and a speech recognition method are provided. In the invention, whether an original voice sampling signal corresponding to a target voice frame is a noise signal is determined according to a ratio of an energy of a first consonant frequency band signal to an energy of a second consonant frequency band signal, a ratio of an energy of the first consonant frequency band signal to an energy of the original voice sampling signal and a ratio of an energy of the second consonant frequency band signal to an energy of the original voice sampling signal.
    Type: Grant
    Filed: March 16, 2015
    Date of Patent: November 15, 2016
    Assignee: Acer Incorporated
    Inventors: Po-Jen Tu, Jia-Ren Chang, Kai-Meng Tzeng
  • Patent number: 9443531
    Abstract: In accordance with an embodiment of the present invention, a noise reduction method for speech processing includes detecting if two signals from two microphones are so close to each other in non voice area that the two microphones are equivalent to Single-Microphone for noise/interference reduction processing. Single-Microphone noise/interference reduction processing algorithm is selected if the equivalent Single-Microphone is detected; Multiple-Microphone noise/interference reduction processing algorithm is selected if the equivalent Single-Microphone is not detected.
    Type: Grant
    Filed: May 2, 2015
    Date of Patent: September 13, 2016
    Inventor: Yang Gao
  • Patent number: 9440071
    Abstract: An exemplary system for facilitating binaural hearing by a cochlear implant patient includes 1) a spectral analysis facility configured to divide a first audio signal presented to a first ear of the patient and a second audio signal presented to a second ear of the patient into first and second sets of analysis channels, respectively, and 2) a processing facility configured to process acoustic content contained in a first analysis channel included in the first set of analysis channels and acoustic content contained in a second analysis channel included in the second set of analysis channels, mix the processed acoustic content contained in the first and second analysis channels, and direct a cochlear implant to apply electrical stimulation representative of the mixed acoustic content to the first ear by way of a stimulation channel that corresponds to the first analysis channel.
    Type: Grant
    Filed: December 29, 2011
    Date of Patent: September 13, 2016
    Assignee: Advanced Bionics AG
    Inventors: Lakshmi N. Mishra, Leonid M. Litvak, Abhijit Kulkarni, Lee F. Hartley
  • Patent number: 9438281
    Abstract: Disclosed is a distortion compensation apparatus that, by appropriately generating a distortion-compensation coefficient, makes it possible to obtain a desired transmission output, and substantially reduce the amount of power leakage to an adjacent channel. Reception section (103) of the distortion compensation apparatus acquires and demodulates a transmission signal to generate a demodulation signal. Delay adjustment section (104) computes the delay amount of the demodulation signal with respect to the baseband signal, eliminates the delay of the demodulation signal with respect to the baseband signal based on the delay amount, and outputs the baseband signal and the demodulation signal in which the delay is eliminated.
    Type: Grant
    Filed: November 29, 2013
    Date of Patent: September 6, 2016
    Assignee: PANASONIC CORPORATION
    Inventors: Yoshito Hirai, Kouji Okamoto
  • Patent number: 9392360
    Abstract: Disclosed is a video controlled beam steering mechanism for an adaptive filter in a sensor array system that receives input from a target and applies an averaging filter and appropriately steers the beam. An adaptive filter is then used if the SNR of the output of the averaging filter reaches a threshold.
    Type: Grant
    Filed: June 30, 2014
    Date of Patent: July 12, 2016
    Assignee: Andrea Electronics Corporation
    Inventor: Douglas Andrea
  • Patent number: 9319150
    Abstract: An information handling system has a haptic generation module to generate haptic effects including haptic noise and a haptic noise reduction module. The haptic noise reduction module receives characteristics of sound representative of haptic noise generated by a haptic generation module of a device and entering an audio input module of the device, the characteristics including frequencies and timing. It also detects the generation of haptic effects, the generations occurring after the receiving characteristics. It also reduces the effects of haptic noise on digital data representing audio input to the device based upon the received characteristics of the sound. It may reduce the effects by subtracting amplitudes of audio waves representing the haptic noise from amplitudes of audio waves representing the audio input.
    Type: Grant
    Filed: October 29, 2012
    Date of Patent: April 19, 2016
    Assignee: DELL PRODUCTS, LP
    Inventors: Douglas J. Peeler, Richard W. Schuckle
  • Patent number: 9305540
    Abstract: A system and method for processing close talking differential microphone array (CTDMA) signals in which incoming microphone signals are transformed from time domain signals to frequency domain signals having separable magnitude and phase information. Processing of the frequency domain signals is performed using the magnitude information, following which phase information is reintroduced using phase information of one of the original frequency domain signals. As a result, high pass filtering effects of conventional differential signal processing of CTDMA signals are substantially avoided.
    Type: Grant
    Filed: January 4, 2013
    Date of Patent: April 5, 2016
    Assignee: NATIONAL SEMICONDUCTOR CORPORATION
    Inventors: Yunhong Li, Lin Sun, Wei Ma
  • Patent number: 9275624
    Abstract: An audio processing apparatus includes an acquisition unit configured to acquire an audio signal, and an audio processing unit configured to reduce noise contained in the audio signal, wherein the audio processing unit complements an audio signal in a section containing noise of the audio signal with a signal generated based on an audio signal in a predetermined section before the section containing noise and an audio signal in a predetermined section after the section containing noise, and wherein, in a case where noise is contained in one of the audio signal in the predetermined section before the section containing noise and the audio signal in the predetermined section after the section containing noise, the audio processing unit complements the audio signal in the section containing noise with a signal generated based on the audio signal in a noise-free section.
    Type: Grant
    Filed: February 27, 2013
    Date of Patent: March 1, 2016
    Assignee: Canon Kabushiki Kaisha
    Inventor: Masafumi Kimura
  • Patent number: 9245530
    Abstract: An apparatus for providing one or more adjusted parameters for a provision of an upmix signal representation on the basis of a downmix signal representation and a parametric side information associated with the downmix signal representation has a parameter adjuster. The parameter adjuster is configured to receive one or more parameters and to provide, on the basis thereof, one or more adjusted parameters. The parameter adjuster is configured to provide the one or more adjusted parameters in dependence on an average value of a plurality of parameter values, such that a distortion of the upmix signal representation caused by the use of non-optimal parameters is reduced at least for parameters deviating from optimal parameters by more than a predetermined deviation.
    Type: Grant
    Filed: April 13, 2012
    Date of Patent: January 26, 2016
    Assignee: Fraunhofer-Gesellschaft zur Foerderung der angewandten Forschung e.V.
    Inventors: Juergen Herre, Cornelia Falch, Leon Terentiv
  • Patent number: 9245538
    Abstract: The present technology provides robust, high quality expansion of the speech within a narrow bandwidth acoustic signal which can overcome or substantially alleviate problems associated with expanding the bandwidth of the noise within the acoustic signal. The present technology carries out a multi-faceted analysis to accurately identify noise within the narrow bandwidth acoustic signal. Noise classification information regarding the noise within the narrow bandwidth acoustic signal is used to determine whether to expand the bandwidth of the narrow bandwidth acoustic signal. By expanding the bandwidth based on the noise classification information, the present technology can expand the speech bandwidth of the narrow bandwidth acoustic signal and prevent or limit the bandwidth expansion of the noise.
    Type: Grant
    Filed: October 19, 2010
    Date of Patent: January 26, 2016
    Assignee: Audience, Inc.
    Inventors: Carlos Avendano, Carlo Murgia
  • Patent number: 9237391
    Abstract: A differential microphone array (DMA) is provided that includes a number (M) of microphone sensors for converting a sound to a number of electrical signals and a processor that is configured to apply linearly-constrained minimum variance filters on the electrical signals over a time window to calculate frequency responses of the electrical signals over a plurality of subbands and sum the frequency responses of the electrical signals for each subband to calculate an estimated frequency spectrum of the sound.
    Type: Grant
    Filed: December 4, 2012
    Date of Patent: January 12, 2016
    Assignee: Northwestern Polytechnical University
    Inventors: Jacob Benesty, Jingdong Chen
  • Patent number: 9153243
    Abstract: 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.
    Type: Grant
    Filed: January 17, 2012
    Date of Patent: October 6, 2015
    Assignee: NIKON CORPORATION
    Inventors: Yoko Yoshizuka, Mitsuhiro Okazaki, Kosuke Okano
  • Patent number: 9134167
    Abstract: Measurement signals for calculating acoustic characteristics of the acoustic space are reproduced in a plurality of periods at sound reproduction intervals. A picked-up signal is acquired by picking up a reproduced signal. The picked-up signal is divided for each period, and the acoustic characteristics of the acoustic space are calculated from an arithmetic unit of the divided periodic signals and the measurement signal. Before sound reproduction, a background noise signal in the acoustic space is measured, and a peak frequency component of a frequency characteristic of the background noise signal is detected. The number of periods and sound reproduction intervals of measurement signals to be reproduced are controlled so as to cancel out a detected peak frequency component at the time of calculating arithmetic unit.
    Type: Grant
    Filed: September 6, 2011
    Date of Patent: September 15, 2015
    Assignee: CANON KABUSHIKI KAISHA
    Inventor: Noriaki Tawada
  • Patent number: 9055367
    Abstract: Psychoacoustic Bass Enhancement (PBE) is integrated with one or more other audio processing techniques, such as active noise cancellation (ANC), and/or receive voice enhancement (RVE), leveraging each technique to achieve improved audio output. This approach can be advantageous for improving the performance of headset speakers, which often lack adequate low-frequency response to effectively support ANC.
    Type: Grant
    Filed: December 15, 2011
    Date of Patent: June 9, 2015
    Assignee: QUALCOMM Incorporated
    Inventors: Ren Li, Pei Xiang
  • Patent number: 9054764
    Abstract: A novel beamforming post-processor technique with enhanced noise suppression capability. The present beamforming post-processor technique is a non-linear post-processing technique for sensor arrays (e.g., microphone arrays) which improves the directivity and signal separation capabilities. The technique works in so-called instantaneous direction of arrival space, estimates the probability for sound coming from a given incident angle or look-up direction and applies a time-varying, gain based, spatio-temporal filter for suppressing sounds coming from directions other than the sound source direction, resulting in minimal artifacts and musical noise.
    Type: Grant
    Filed: July 20, 2011
    Date of Patent: June 9, 2015
    Assignee: Microsoft Technology Licensing, LLC
    Inventors: Ivan Tashev, Alejandro Acero
  • Publication number: 20150139446
    Abstract: An audio separating apparatus obtains a matrix by performing time-frequency conversion on an input audio signal. The audio separating apparatus divides the obtained matrix into at least a basis matrix and an activity matrix, and classifies base spectra that configure the respective columns of the basis matrix into first base spectra corresponding to a target sound and second base spectra corresponding to a non-target sound.
    Type: Application
    Filed: October 29, 2014
    Publication date: May 21, 2015
    Inventor: Noriaki Tawada
  • Patent number: 9008329
    Abstract: Provided are methods and systems for noise suppression within multiple time-frequency points of spectral representations. A multi-feature cluster tracker is used to track signal and noise sources and to predict signal versus noise dominance at each time-frequency point. Multiple features, such as binaural and monaural features, may be used for these purposes. A Gaussian mixture model (GMM) is developed and, in some embodiments, dynamically updated for distinguishing signal from noise and performing mask-based noise reduction. Each frequency band may use a different GMM or share a GMM with other frequency bands. A GMM may be combined from two models, with one trained to model time-frequency points in which the target dominates and another trained to model time-frequency points in which the noise dominates. Dynamic updates of a GMM may be performed using an expectation-maximization algorithm in an unsupervised fashion.
    Type: Grant
    Filed: June 8, 2012
    Date of Patent: April 14, 2015
    Assignee: Audience, Inc.
    Inventors: Michael Mandel, Carlos Avendano
  • Patent number: 9002031
    Abstract: A device, system and method of playing back a digital audio stream wherein large amounts of pre-emphasis of the high frequencies is applied before the digital to analog conversion and before an interpolation or digital filter, followed by de-emphasis in the analog domain in order to yield better audio fidelity.
    Type: Grant
    Filed: March 24, 2012
    Date of Patent: April 7, 2015
    Inventor: Michael Yee
  • Patent number: 8983091
    Abstract: The disclosure provides a network signal receiving system and a network signal receiving method. The network signal receiving system comprises: a high pass filter, a canceller, and an adder. The high pass filter is utilized for performing a high pass filtering operation for an audio data signal to output at least a signal corresponding to transitions of the audio data signal, wherein the audio data signal is synchronized with a network data signal. The canceller is coupled to the high pass filter, and utilized for generating a noise cancelling signal according to the at least a signal output by the high pass filter. The adder is coupled to the canceller, utilized for receiving the network data signal and the noise cancelling signal, so as to use the noise cancelling signal to cancel at least a noise in the network data signal, which is corresponding to the at least a signal.
    Type: Grant
    Filed: August 30, 2012
    Date of Patent: March 17, 2015
    Assignee: Realtek Semiconductor Corp.
    Inventors: Yuan-Jih Chu, Liang-Wei Huang, Hsuan-Ting Ho, Ming-Feng Hsu
  • Publication number: 20150071462
    Abstract: The embodiments herein disclose a device and a method for controlling noise in a wideband communication system. In one embodiment herein, multiple microphones for receiving wideband audio signals are provided. A processor is configured to analyze each wideband audio signal received by each microphone. Further, unique signal patterns are generated based on each analyzed wideband signals for each microphone and the unique signal patterns are compared to detect any identical signal patterns. A controller is also provided for controlling gains of those microphones that are detected to be receiving wideband audio signal of identical signal patterns.
    Type: Application
    Filed: November 11, 2014
    Publication date: March 12, 2015
    Inventor: Alon Konchitsky
  • Publication number: 20150055800
    Abstract: Provided are methods and systems for enhancing the intelligibility of an audio (e.g., speech) signal rendered in a noisy environment, subject to a constraint on the power of the rendered signal. A quantitative measure of intelligibility is the mean probability of decoding of the message correctly. The methods and systems simplify the procedure by approximating the maximization of the decoding probability with the maximization of the similarity of the spectral dynamics of the noisy speech to the spectral dynamics of the corresponding noise-free speech. The intelligibility enhancement procedures provided are based on this principle, and all have low computational cost and require little delay, thus facilitating real-time implementation.
    Type: Application
    Filed: August 22, 2014
    Publication date: February 26, 2015
    Inventors: Willem Bastiaan KLEIJN, Petko N. PETKOV
  • Patent number: 8958571
    Abstract: A personal audio device, such as a wireless telephone, includes noise canceling circuit that adaptively generates an anti-noise signal from a reference microphone signal and injects the anti-noise signal into the speaker or other transducer output to cause cancellation of ambient audio sounds. An error microphone may also be provided proximate the speaker to estimate an electro-acoustical path from the noise canceling circuit through the transducer. A processing circuit uses the reference and/or error microphone, optionally along with a microphone provided for capturing near-end speech, to determine whether one of the reference or error microphones is obstructed by comparing their received signal content and takes action to avoid generation of erroneous anti-noise.
    Type: Grant
    Filed: September 30, 2011
    Date of Patent: February 17, 2015
    Assignee: Cirrus Logic, Inc.
    Inventors: Nitin Kwatra, Jeffrey Alderson, Jon D. Hendrix
  • Patent number: 8958572
    Abstract: Null processing noise subtraction is performed per sub-band and time frame for acoustic signals received from multiple microphones. The acoustic signals may include a primary acoustic signal and one or more additional acoustic signals. A noise component signal may be determined for each additional acoustic signal in each sub-band of signals received by N microphones by subtracting a desired signal component within every other acoustic signal weighted by a complex-valued coefficient ? from the secondary acoustic signal. The noise component signals, each weighted by a corresponding complex-valued coefficient ?, may then be subtracted from the primary acoustic signal resulting in an estimate of a target signal (i.e., a noise subtracted signal).
    Type: Grant
    Filed: August 12, 2010
    Date of Patent: February 17, 2015
    Assignee: Audience, Inc.
    Inventor: Ludger Solbach
  • Patent number: 8958570
    Abstract: A microphone array apparatus includes: an acquisition unit configured to acquire samples from a sound signal inputted from each of a plurality of microphones, at predetermined time intervals; an operation unit configured to calculate a value based on volumes of the sound signal possessed by a plurality of the samples for each of the sound signals inputted from the plurality of microphones; a correlation coefficient calculator configured to calculate a coefficient of correlation between the sound signals, on the basis of the values calculated for the respective sound signals; and a gain calculator configured to calculate reduction gain for the sound signals inputted from the plurality of microphones, on the basis of the coefficient of correlation.
    Type: Grant
    Filed: March 21, 2012
    Date of Patent: February 17, 2015
    Assignee: Fujitsu Limited
    Inventor: Naoshi Matsuo
  • Patent number: 8953818
    Abstract: A listening device for processing an input sound to an output sound, includes an input transducer for converting an input sound to an electric input signal, an output transducer for converting a processed electric output signal to an output sound, a forward path being defined between the input transducer and the output transducer and including a signal processing unit for processing an input signal in a number of frequency bands and an SBS unit for performing spectral band substitution from one frequency band to another and providing an SBS-processed output signal, and an LG-estimator unit for estimating loop gain in each frequency band thereby identifying plus-bands having an estimated loop gain according to a plus-criterion and minus-bands having an estimated loop gain according to a minus-criterion.
    Type: Grant
    Filed: February 6, 2009
    Date of Patent: February 10, 2015
    Assignee: Oticon A/S
    Inventors: Thomas Bo Elmedyb, Jesper Jensen
  • Patent number: 8949120
    Abstract: Systems and methods for controlling adaptivity of noise cancellation are presented. One or more audio signals are received by one or more corresponding microphones. The one or more signals may be decomposed into frequency sub-bands. Noise cancellation consistent with identified adaptation constraints is performed on the one or more audio signals. The one or more audio signals may then be reconstructed from the frequency sub-bands and outputted via an output device.
    Type: Grant
    Filed: April 13, 2009
    Date of Patent: February 3, 2015
    Assignee: Audience, Inc.
    Inventors: Mark Every, Ludger Solbach, Carlo Murgia, Ye Jiang
  • Publication number: 20150030180
    Abstract: A method, an apparatus, and logic to post-process raw gains determined by input processing to generate post-processed gains, comprising using one or both of delta gain smoothing and decision-directed gain smoothing. The delta gain smoothing comprises applying a smoothing filter to the raw gain with a smoothing factor that depends on the gain delta: the absolute value of the difference between the raw gain for the current frame and the post-processed gain for a previous frame. The decision-directed gain smoothing comprises converting the raw gain to a signal-to-noise ratio, applying a smoothing filter with a smoothing factor to the signal-to-noise ratio to calculate a smoothed signal-to-noise ratio, and converting the smoothed signal-to-noise ratio to determine the second smoothed gain, with smoothing factor possibly dependent on the gain delta.
    Type: Application
    Filed: March 21, 2013
    Publication date: January 29, 2015
    Applicant: DOLBY LABORATORIES LICENSING CORPORATION
    Inventors: Xuejing Sun, Glenn N. Dickins
  • Publication number: 20150030181
    Abstract: A device includes a HPF 702 that modifies frequency characteristics of a target signal; a phase correcting unit 701 that corrects the phase characteristics of the target signal to make the phase characteristics nearly equal to phase characteristics of the HPF 702; a first multiplier 705 that adjusts the gain of the signal output from the phase correcting unit 701; a second multiplier 706 that adjusts the gain of the signal output from the HPF 702; a coefficient determining unit that determines the gain coefficients of the first and second multipliers 705 and 706 in such a manner that the sum of the gain coefficient of the first multiplier 705 and the gain coefficient of the second multiplier 706 becomes a fixed value; and an adder 713 that adds the two signals output from the first multiplier 705 and second multiplier 706.
    Type: Application
    Filed: December 14, 2012
    Publication date: January 29, 2015
    Applicant: Mitsubishi Electric Corporation
    Inventors: Masaru Kimura, Takashi Yamazaki
  • Patent number: 8942387
    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: Grant
    Filed: March 9, 2007
    Date of Patent: January 27, 2015
    Assignee: MH Acoustics LLC
    Inventors: Gary W. Elko, Jens M. Meyer, Tomas Fritz Gaensler
  • Patent number: 8924204
    Abstract: Unlike sound based pressure waves that go everywhere, air turbulence caused by wind is usually a fairly local event. Therefore, in a system that utilizes two or more spatially separated microphones to pick up sound signals (e.g., speech), wind noise picked up by one of the microphones often will not be picked up (or at least not to the same extent) by the other microphone(s). Embodiments of methods and apparatuses that utilize this fact and others to effectively detect and suppress wind noise using multiple microphones that are spatially separated are described.
    Type: Grant
    Filed: September 30, 2011
    Date of Patent: December 30, 2014
    Assignee: Broadcom Corporation
    Inventors: Juin-Hwey Chen, Jes Thyssen, Xianxian Zhang, Huaiyu Zeng
  • Patent number: 8923530
    Abstract: A method is disclosed for acoustic feedback attenuation at a telecommunications terminal. A speakerphone equipped with a loudspeaker and two microphones is featured. Signals from the two microphones are subjected to a calibration stage and then to a runtime stage. The purpose of the calibration stage is to match the microphones to each other by advantageously using both magnitude and phase equalization across the frequency spectrum of the microphones. During the runtime stage, the microphones monitor the ambient sounds received from sound sources, such as the speakerphone's users and the loudspeaker itself, during a conference call. The speakerphone applies the generated set of filter coefficients to the optimized microphone's signals. By combining the signal from the reference microphone with the filtered signal from the optimized microphone, the speakerphone is able to attenuate the sounds from the loudspeaker that would otherwise be transmitted back to other conference call participants.
    Type: Grant
    Filed: April 10, 2009
    Date of Patent: December 30, 2014
    Assignee: Avaya Inc.
    Inventors: Eric John Diethorn, Heinz Teutsch
  • Publication number: 20140376744
    Abstract: In a system and method for maintaining the spatial stability of a sound field a balance gain may be calculated for two or more microphone signals. The balance gain may be associated with a spatial image in the sound field. Signal values may be calculated for each of the microphone. The signal values may be signal estimates or signal gains calculated to improve a characteristic of the microphone signals. The differences between the signal values associated with each microphone signal may be limited although some difference between signal values may be allowable. One or more microphone signals are adjusted responsive to the two or more balance gains and the signal gains to maintain the spatial stability of the sound field. The adjustments of one or more microphone signals may include mixing of two or more microphone. The signal gains are applied to the two or more microphone signals.
    Type: Application
    Filed: June 20, 2013
    Publication date: December 25, 2014
    Inventor: Phillip Alan Hetherington
  • Publication number: 20140376742
    Abstract: In a system and method for maintaining the spatial stability of a sound field a balance gain may be calculated for two or more microphone signals. The balance gain may be associated with a spatial image in the sound field. Signal values may be calculated for each of the microphone. The signal values may be signal estimates or signal gains calculated to improve a characteristic of the microphone signals. The differences between the signal values associated with each microphone signal may be limited although some difference between signal values may be allowable. One or more microphone signals are adjusted responsive to the two or more balance gains and the signal gains to maintain the spatial stability of the sound field. The adjustments of one or more microphone signals may include mixing of two or more microphone. The signal gains are applied to the two or more microphone signals.
    Type: Application
    Filed: June 20, 2013
    Publication date: December 25, 2014
    Inventor: Phillip Alan Hetherington
  • Publication number: 20140376743
    Abstract: In a system and method for maintaining the spatial stability of a sound field a balance gain may be calculated for two or more microphone signals. The balance gain may be associated with a spatial image in the sound field. Signal values may be calculated for each of the microphone. The signal values may be signal estimates or signal gains calculated to improve a characteristic of the microphone signals. The differences between the signal values associated with each microphone signal may be limited although some difference between signal values may be allowable. One or more microphone signals are adjusted responsive to the two or more balance gains and the signal gains to maintain the spatial stability of the sound field. The adjustments of one or more microphone signals may include mixing of two or more microphone. The signal gains are applied to the two or more microphone signals.
    Type: Application
    Filed: June 20, 2013
    Publication date: December 25, 2014
    Applicant: QNX Software Systems Limited
    Inventor: Phillip Alan Hetherington
  • Patent number: 8913758
    Abstract: Disclosed herein are systems, methods, and non-transitory computer-readable storage media for suppressing spatial noise based on phase information. The method transforms audio signals to frequency-domain data and identifies time-frequency points that have a parameter (e.g., signal-to-noise ratio) above a threshold. Based on these points, unwanted signals can be attenuated the desired audio source can be isolated. The method can work on a microphone array that includes two microphones or more.
    Type: Grant
    Filed: August 8, 2011
    Date of Patent: December 16, 2014
    Assignee: Avaya Inc.
    Inventors: Avram Levi, Heinz Teutsch
  • Patent number: 8908883
    Abstract: The present invention discloses a microphone array structure able to reduce noise and improve speech quality and a method thereof. The method of the present invention comprises steps: using at least two microphone to receive at least two microphone signals each containing a noise signal and a speech signal; using FFT modules to transform the microphone signals into frequency-domain signals; calculating an included angle between a speech signal and a noise signal of the microphone signal, and selecting a phase difference estimation algorithm, a noise reduction algorithm or both to reduce noise according to the included angle; if the phase difference estimation algorithm is used, calculating phase difference of the microphone signals to obtain a time-space domain mask signal; and multiplying the mask signal and the average of the microphone signals to obtain the speech signals of the microphone signals. Thereby is eliminated noise and improve speech quality.
    Type: Grant
    Filed: August 16, 2011
    Date of Patent: December 9, 2014
    Assignee: National Chiao Tung University
    Inventors: Mingsian R. Bai, Chun-Hung Chen
  • Patent number: 8903098
    Abstract: The present invention relates to a signal processing apparatus and method, a program, and a data recording medium configured such that the playback level of an audio signal can be easily and effectively enhanced without requiring prior analysis. An analyzer 21 generates mapping control information in the form of the root mean square of samples in a given segment of a supplied audio signal. A mapping processor 22 takes a nonlinear function determined by the mapping control information taken as a mapping function, and conducts amplitude conversion on a supplied audio signal using the mapping function. In this way, by conducting amplitude conversion of an audio signal using a nonlinear function that changes according to the characteristics in respective segments of an audio signal, the playback level of an audio signal can be easily and effectively enhanced without requiring prior analysis. The present invention may be applied to portable playback apparatus.
    Type: Grant
    Filed: September 6, 2011
    Date of Patent: December 2, 2014
    Assignee: Sony Corporation
    Inventors: Minoru Tsuji, Toru Chinen