Pretransmission Patents (Class 704/227)
  • Patent number: 10431211
    Abstract: An apparatus includes multiple microphones to generate audio signals based on sound of a far-field acoustic environment. The apparatus also includes a signal processing system to process the audio signals to generate at least one processed audio signal. The signal processing system is configured to update one or more processing parameters while operating in a first operational mode and is configured to use a static version of the one or more processing parameters while operating in the second operational mode. The apparatus further includes a keyword detection system to perform keyword detection based on the at least one processed audio signal to determine whether the sound includes an utterance corresponding to a keyword and, based on a result of the keyword detection, to send a control signal to the signal processing system to change an operational mode of the signal processing system.
    Type: Grant
    Filed: December 21, 2016
    Date of Patent: October 1, 2019
    Assignee: Qualcomm Incorporated
    Inventors: Lae-Hoon Kim, Erik Visser, Asif Mohammad, Ian Ernan Liu, Ye Jiang
  • Patent number: 10388264
    Abstract: A frequency domain converter divides an input signal for each predetermined frame, and generates a first signal X(f, ?) for each first frequency division unit. A noise estimation signal generator generates a signal Y(f, ?) for each second frequency division unit wider than the first frequency division unit. A signal comparator calculates a representative value for each second frequency division unit based on the signal Y(f, ?) stored in a storage unit, and compares the representative value and the signal Y(f, ?) with each other for each second frequency division unit. A mask generator generates a mask M(f, ?), which determines a degree of suppression or emphasis for each first frequency division unit, based on a peak range of the signal X(f, ?), and a comparison result by the signal comparator. The mask application unit multiplies the signal X(f, ?) by the mask M(f, ?).
    Type: Grant
    Filed: November 16, 2017
    Date of Patent: August 20, 2019
    Assignee: JVC KENWOOD CORPORATION
    Inventor: Masato Sugano
  • Patent number: 10366706
    Abstract: According to one embodiment, a signal processing apparatus includes a processer. The processor separates a plurality of signals, which are received at different positions and come from different directions, by a separation filter. The processor estimates incoming directions of a plurality of separate signals respectively, and associates the plurality of separate signals with transmission sources of the plurality of signals. The processor associates either one of a first attribute and a second attribute with the separate signals which are associated with the transmission sources of the signals based on results of the estimation of the incoming directions in a first period, and add either one of first label information and second label information.
    Type: Grant
    Filed: September 12, 2017
    Date of Patent: July 30, 2019
    Assignee: Kabushiki Kaisha Toshiba
    Inventors: Makoto Hirohata, Toru Taniguchi, Taro Masuda
  • Patent number: 10347258
    Abstract: A voice communication system is equipped with a voice encoder which classifies respective bits of a voice information bit string in accordance with the degree of importance which is the magnitude of auditory influence when there is an error therein, classifies a group of bits which are high in degree of importance into a core layer and classifies a group of bits which are not high into an extension layer and a voice decoder which decodes a voice by using the bit strings in both of the core layer and the extension layer on the basis of frequency that the error is detected by error detection processing and when the frequency is low and decodes the voice using all bits or only some bets in the core layer when the frequency is high.
    Type: Grant
    Filed: March 4, 2016
    Date of Patent: July 9, 2019
    Assignee: Hitachi Kokusai Electric Inc.
    Inventor: Seishi Sasaki
  • Patent number: 10332537
    Abstract: A set of signal measures is sent, wherein each signal measure in the set of signal measures corresponds to a respective audio signal received by a playback device in a media playback system and is processed based on a first set of audio processing algorithms. A plurality of signal measures is identified in the set of signal measures. Audio signals corresponding to the identified plurality of signal measures are processed by one or more devices in the media playback system to improve a signal measure of each of the audio signals. The audio signals are processed based on a second set of audio processing algorithms. The processed audio signals are combined into a combined audio signal.
    Type: Grant
    Filed: April 23, 2018
    Date of Patent: June 25, 2019
    Assignee: Sonos, Inc.
    Inventor: Shao-Fu Shih
  • Patent number: 10255903
    Abstract: A system and method are presented for forming the excitation signal for a glottal pulse model based parametric speech synthesis system. The excitation signal may be formed by using a plurality of sub-band templates instead of a single one. The plurality of sub-band templates may be combined to form the excitation signal wherein the proportion in which the templates are added is dynamically based on determined energy coefficients. These coefficients vary from frame to frame and are learned, along with the spectral parameters, during feature training. The coefficients are appended to the feature vector, which comprises spectral parameters and is modeled using HMMs, and the excitation signal is determined.
    Type: Grant
    Filed: October 6, 2015
    Date of Patent: April 9, 2019
    Inventors: Rajesh Dachiraju, E. Veera Raghavendra, Aravind Ganapathiraju
  • Patent number: 10114815
    Abstract: A set of core points is aggregated from a set of points extracted from a large document. A point and a core point each is a topic covered in the document. For the core point in the set of core points, a network of associations is constructed, where an association in the network includes an entity that has a relationship with the core point by virtue of having contributed data in the document that relates to the core point. From the contributed data, a sentiment value of the contributed data is computed, the sentiment value being indicative of a sentiment of the entity towards the core point. From a set of sentiment values corresponding to the associations in the network of associations, an overall sentiment value is computed for the core point. The overall sentiment values for each core point in the document is reported.
    Type: Grant
    Filed: October 25, 2016
    Date of Patent: October 30, 2018
    Assignee: INTERNATIONAL BUSINESS MACHINES CORPORATION
    Inventors: James E. Bostick, John M. Ganci, Jr., Martin G. Keen, Sarbajit K. Rakshit
  • Patent number: 10056091
    Abstract: A system that includes a microphone array comprising a plurality of microphones positioned at different locations, where the microphones output microphone signals. A beamformer is applied to the microphone output signals and is configured to control a gain that is applied to the microphone output signals. The gain is frequency dependent and is related to a mismatch in sensitivity between two or more of the microphones.
    Type: Grant
    Filed: January 6, 2017
    Date of Patent: August 21, 2018
    Assignee: Bose Corporation
    Inventors: Marko Orescanin, Mehmet Ergezer
  • Patent number: 10038379
    Abstract: A controller for controlling operation of a switching regulator including a modulator, a discontinuous conduction mode (DCM) controller, an audible DCM (ADCM) controller, and a sub-sonic discontinuous conduction mode (SBDCM) controller. The modulator generally operates in a continuous conduction mode. The DCM controller modifies operation to DCM during low loads. The ADCM controller detects when the switching frequency is less than a super-sonic frequency threshold and modifies operation to maintain the switching frequency at a super-sonic frequency level. The SBDCM controller detects a sub-sonic operating condition during ADCM operation and responsively inhibits operation of the ADCM mode controller to allow a SBDCM mode within a sub-sonic switching frequency range. The SBDCM operating mode allows for efficient connected standby operation. The SBDCM controller allows operation to return to other modes when the switching frequency increases above the subsonic level.
    Type: Grant
    Filed: January 13, 2017
    Date of Patent: July 31, 2018
    Assignee: INTERSIL AMERICAS LLC
    Inventors: M. Jason Houston, Steven P. Laur
  • Patent number: 10026395
    Abstract: A system extracting features from a time-varying signal comprising a computer processor and a computer readable medium having computer executable instructions for providing: a bank of bandpass filters; a module approximating the output of those filters with nonlinear components; a module representing a decorrelated projection of the output of the filters with nonlinear components; and a module representing the temporal derivative of the decorrelated information with nonlinear components.
    Type: Grant
    Filed: January 6, 2017
    Date of Patent: July 17, 2018
    Assignee: Applied Brain Research Inc.
    Inventor: Trevor Bekolay
  • Patent number: 9866947
    Abstract: The present invention discloses a headset, a terminal, and a method for processing an audio signal based on the headset and the terminal. The method for processing an audio signal includes: acquiring, by a first microphone of a headset in a call process, a first audio signal including a call voice and background noise, and acquiring, by a second microphone of the headset in the call process, a second audio signal including the background noise; and transmitting, by using an external plug of the headset, the first audio signal and the second audio signal to a noise reduction chip built in a terminal, so that the noise reduction chip performs noise reduction processing on the first audio signal according to the second audio signal.
    Type: Grant
    Filed: March 14, 2014
    Date of Patent: January 9, 2018
    Assignee: Huawei Device Co., Ltd.
    Inventor: Haiquan Yang
  • Patent number: 9754606
    Abstract: A processing apparatus estimates a noise amplitude spectrum of noise included in a sound signal. The processing apparatus includes an amplitude spectrum calculation part configured to calculate an amplitude spectrum of the sound signal for each one of frames obtained from dividing the sound signal into units of time; and a noise amplitude spectrum estimation part configured to estimate the noise amplitude spectrum of the noise detected from the frame. The noise amplitude spectrum estimation part includes a first estimation part configured to estimate the noise amplitude spectrum based on a difference between the amplitude spectrum calculated by the amplitude spectrum calculation part and the amplitude spectrum of the frame occurring before the noise is detected, and a second estimation part configured to estimate the noise amplitude spectrum based on an attenuation function obtained from noise amplitude spectra of the frames occurring after the noise is detected.
    Type: Grant
    Filed: April 19, 2013
    Date of Patent: September 5, 2017
    Assignee: RICOH COMPANY, LTD.
    Inventors: Akihito Aiba, Junichi Takami
  • Patent number: 9607625
    Abstract: Systems and methods are provided for audio encoding. For example, first quantization encoding is performed on audio data associated with a current frame of an audio data stream to obtain first quantization-encoded data; second quantization encoding is performed on the audio data to obtain second quantization-encoded data; the first quantization-encoded data is coupled to the current frame of the audio data stream; and the second quantization-encoded data is coupled to a next frame of the audio data stream.
    Type: Grant
    Filed: August 3, 2015
    Date of Patent: March 28, 2017
    Assignee: Tencent Technology (Shenzhen) Company Limited
    Inventors: Guoming Chen, Yuanjiang Peng, Wenjun Ou, Chao Peng, Hong Liu, Xingping Long
  • Patent number: 9552803
    Abstract: A communication method is provided. The communication method includes generating a pseudo background sound signal based on a gradient pulse control signal, performing a computation of subtracting the pseudo background sound signal from an acoustic signal having a sound signal and a background sound signal including a gradient coil drive sound signal, the acoustic signal obtained by an input device configured to receive the voice of a subject, and outputting sound based on a result of the computation, wherein a parameter of generating the pseudo background sound signal is controlled to reduce the difference resulting from the subtraction.
    Type: Grant
    Filed: December 17, 2012
    Date of Patent: January 24, 2017
    Assignee: General Electric Company
    Inventors: Yasuyuki Innami, Kiyoshi Murakami, Shohei Kimoto, Yuya Mizobe, Yusuke Asaba
  • Patent number: 9514766
    Abstract: Differing first and second audio signal sample rates from first and second audio signals are matched to each other. If signal sample rates are different, a frame of samples of the first audio signal is duplicated. The duplicate copies are multiplied by a window function and its inverse to produce “windowed frames” first and last samples of which can be deleted or added to increase or decrease a frame rate.
    Type: Grant
    Filed: July 8, 2015
    Date of Patent: December 6, 2016
    Assignee: Continental Automotive Systems, Inc.
    Inventors: Bijal Joshi, Nitu Anil Kumar
  • Patent number: 9418177
    Abstract: Methods, systems, and apparatus, including computer programs encoded on a computer storage medium, for processing spoken query terms. In one aspect, a method includes performing speech recognition on an audio signal to select two or more textual, candidate transcriptions that match a spoken query term, and to establish a speech recognition confidence value for each candidate transcription, obtaining a search history for a user who spoke the spoken query term, where the search history references one or more past search queries that have been submitted by the user, generating one or more n-grams from each candidate transcription, where each n-gram is a subsequence of n phonemes, syllables, letters, characters, words or terms from a respective candidate transcription, and determining, for each n-gram, a frequency with which the n-gram occurs in the past search queries, and a weighting value that is based on the respective frequency.
    Type: Grant
    Filed: August 5, 2013
    Date of Patent: August 16, 2016
    Assignee: Google Inc.
    Inventors: Matthew I. Lloyd, Johan Schalkwyk, Pankaj Risbood
  • Patent number: 9361899
    Abstract: The present disclosure is directed towards a process for estimating the signal to noise ratio of a speech signal. The process may include receiving, at a computing device, a speech signal having a bitstream and a signal-to-noise ratio (“SNR”) associated therewith. The process may further include estimating the SNR directly from the bitstream or using a partial decoder that is configured to extract one or more parameters, the parameters including at least one of a fixed codebook gain, an adaptive codebook gain, a pitch lag, and a line spectral frequency (“LSF”) coefficient.
    Type: Grant
    Filed: July 2, 2014
    Date of Patent: June 7, 2016
    Assignee: Nuance Communications, Inc.
    Inventors: Jose Lainez, Daniel A. Barreda, Dushyant Sharma, Patrick Naylor, Sridhar Pilli
  • Patent number: 9286808
    Abstract: The invention provides an electronic method for assessing student skills on a stringed musical instrument, based on comparing the digital signature of an expert template sound to the digital signature of a student's attempt to replicate the template on the stringed instrument. In particular the method employs in sequential series an audio input device, an analog-to-digital converter, an audio correlation decoder, a digital-to-analog converter, and an audio output device; in certain embodiments they are all under the control of the same central control processor, which also has access to a first addressable data memory including unprocessed emulations from a student, a logic filter circuit element, a second addressable data memory including emulations from the student that have been filtered to remove background noise and other extraneous elements, and an addressable data memory including expert templates for incremental components of stringed musical performance.
    Type: Grant
    Filed: June 9, 2011
    Date of Patent: March 15, 2016
    Assignee: PRA Audio Systems, LLC
    Inventors: K. Paul Raley, Howard A. Carnes
  • Patent number: 9247346
    Abstract: Systems, apparatuses and methods for integrating adaptive noise cancellation (ANC) with communication features in an enclosure, such as an incubator, bed, and the like. Utilizing one or more error and reference microphones, a controller for a noise cancellation portion reduces noise within a quiet area of the enclosure. Voice communications are provided to allow external voice signals to be transmitted to the enclosure with minimized interference with noise processing. Vocal communications from within the enclosure may be processed to determine certain characteristics/features of the vocal communications. Using these characteristics, certain emotive and/or physiological states may be identified.
    Type: Grant
    Filed: March 15, 2013
    Date of Patent: January 26, 2016
    Assignee: Northern Illinois Research Foundation
    Inventors: Sen M. Kuo, Lichuan Liu
  • Patent number: 9196261
    Abstract: Acoustic noise suppression is provided in multiple-microphone systems using Voice Activity Detectors (VAD). A host system receives acoustic signals via multiple microphones. The system also receives information on the vibration of human tissue associated with human voicing activity via the VAD. In response, the system generates a transfer function representative of the received acoustic signals upon determining that voicing information is absent from the received acoustic signals during at least one specified period of time. The system removes noise from the received acoustic signals using the transfer function, thereby producing a denoised acoustic data stream.
    Type: Grant
    Filed: February 28, 2011
    Date of Patent: November 24, 2015
    Assignee: ALIPHCOM
    Inventors: Gregory C. Burnett, Eric F. Breitfeller
  • Patent number: 9191752
    Abstract: A method of signal processing an input signal in a hearing aid to avoid entrainment, the hearing aid including a receiver and a microphone, the method comprising using an adaptive filter to measure an acoustic feedback path from the receiver to the microphone and adjusting an adaptation rate of the adaptive filter using an output from a filter having an autoregressive portion, the output derived at least in part from a ratio of a predictive estimate of the input signal to a difference of the predictive estimate and the input signal.
    Type: Grant
    Filed: March 24, 2014
    Date of Patent: November 17, 2015
    Assignee: Starkey Laboratories, Inc.
    Inventors: Lalin Theverapperuma, Harikrishna P. Natarajan, Arthur Salvetti, Jon S. Kindred
  • Patent number: 9106196
    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: Grant
    Filed: June 20, 2013
    Date of Patent: August 11, 2015
    Assignee: 2236008 Ontario Inc.
    Inventor: Phillip Alan Hetherington
  • Patent number: 9043203
    Abstract: An encoder for providing an audio stream on the basis of a transform-domain representation of an input audio signal includes a quantization error calculator configured to determine a multi-band quantization error over a plurality of frequency bands of the input audio signal for which separate band gain information is available. The encoder also includes an audio stream provider for providing the audio stream such that the audio stream includes information describing an audio content of the frequency bands and information describing the multi-band quantization error. A decoder for providing a decoded representation of an audio signal on the basis of an encoded audio stream representing spectral components of frequency bands of the audio signal includes a noise filler for introducing noise into spectral components of a plurality of frequency bands to which separate frequency band gain information is associated on the basis of a common multi-band noise intensity value.
    Type: Grant
    Filed: January 11, 2011
    Date of Patent: May 26, 2015
    Assignee: Fraunhofer-Gesellschaft zur Foerderung der angewandten Forschung e.V.
    Inventors: Nikolaus Rettelbach, Bernhard Grill, Guillaume Fuchs, Stefan Geyersberger, Markus Multrus, Harald Popp, Juergen Herre, Stefan Wabnik, Gerald Schuller, Jens Hirschfeld
  • Patent number: 9026433
    Abstract: A voice quality measurement device that measures voice quality of a decoded voice signal outputted from a voice decoder unit. The voice quality measurement device includes a packet buffer unit and a voice information monitoring unit. The packet buffer unit accumulates voice packets that arrive non-periodically as voice information, and outputs the voice information to the voice decoder unit periodically. The voice information monitoring unit monitors continuity of the voice information inputted to the voice decoder unit, and calculates an index of voice quality of the decoded voice signal that reflects acceptability of this continuity.
    Type: Grant
    Filed: November 25, 2011
    Date of Patent: May 5, 2015
    Assignee: Oki Electric Industry Co., Ltd.
    Inventor: Hiromi Aoyagi
  • Patent number: 9026434
    Abstract: An audio coding terminal and method is provided. The terminal includes a coding mode setting unit to set an operation mode, from plural operation modes, for input audio coding by a codec configured to code the input audio based on the set operation mode such that when the set operation mode is a high frame erasure rate (FER) mode the codec codes a current frame of the input audio according to a select frame erasure concealment (FEC) mode of one or more FEC modes. Upon the setting of the operation mode to be the High FER mode the one FEC mode is selected, from the one or more FEC modes predetermined for the High FER mode, to control the codec by incorporating of redundancy within a coding of the input audio or as separate redundancy information separate from the coded input audio according to the selected one FEC mode.
    Type: Grant
    Filed: April 10, 2012
    Date of Patent: May 5, 2015
    Assignee: Samsung Electronic Co., Ltd.
    Inventors: Steven Craig Greer, Hosang Sung
  • Patent number: 9015042
    Abstract: Embodiments are described of a multi-block coding scheme for an audio signal to prevent partial collapse conditions from causing pre-echo compression artifacts. An audio codec includes a segmentation component partitioning the audio signal into a plurality of tiles, wherein each tile comprises data from a particular segment of time and a particular set of frequencies of the audio signal; a band energy component determining an energy value for each tile corresponding to a signal component in a respective tile; an encoder flag tracking component marking a tile as not collapsed or collapsed based on the energy value in that tile; and a decoder flag tracking component filling all tiles marked as collapsed with pseudorandom noise at an estimated energy level.
    Type: Grant
    Filed: March 7, 2012
    Date of Patent: April 21, 2015
    Inventors: Jean-Marc Valin, Timothy B. Terriberry
  • Patent number: 8996365
    Abstract: A howling canceller which suppresses occurrence of howling even when an open loop gain exceeds “1” in the whole reproduction band. In the howling canceller, an adaptive filter (107) operates a digital received voice signal with a tap coefficient to generate a pseudo echo; a subtractor (108) subtracts the pseudo echo from a digital transmitted voice signal to generate a residual signal; and an amplitude limiting circuit (110) limits the absolute value of the amplitude of the digital received voice signal to be equal to or smaller than a predetermined threshold which ensures that all of a D/A converter (101), a power amplifier (102), a speaker (103), a microphone (104), a microphone amplifier (105), and an A/D converter (106) operate in a linear operation area, and outputs the amplitude-limited digital received voice signal to the D/A converter (101) and the adaptive filter (107).
    Type: Grant
    Filed: March 19, 2010
    Date of Patent: March 31, 2015
    Assignee: Yugengaisya Cepstrum
    Inventor: Akio Yamaguchi
  • Patent number: 8977545
    Abstract: Described herein are multi-channel noise suppression systems and methods that are configured to detect and suppress wind and background noise using at least two spatially separated microphones: at least one primary speech microphone and at least one noise reference microphone. The multi-channel noise suppression systems and methods are configured, in at least one example, to first detect and suppress wind noise in the input speech signal picked up by the primary speech microphone and, potentially, the input speech signal picked up by the noise reference microphone. Following wind noise detection and suppression, the multi-channel noise suppression systems and methods are configured to perform further noise suppression in two stages: a first linear processing stage that includes a blocking matrix and an adaptive noise canceler, followed by a second non-linear processing stage.
    Type: Grant
    Filed: November 14, 2011
    Date of Patent: March 10, 2015
    Assignee: Broadcom Corporation
    Inventors: Huaiyu Zeng, Jes Thyssen, Nelson Sollenberger, Juin-Hwey Chen, Xianxian Zhang
  • Patent number: 8972250
    Abstract: The invention relates to audio signal processing. More specifically, the invention relates to enhancing multichannel audio, such as television audio, by applying a gain to the audio that has been smoothed between portions of the audio. The invention relates to methods, apparatus for performing such methods, and to software stored on a computer-readable medium for causing a computer to perform such methods.
    Type: Grant
    Filed: August 10, 2012
    Date of Patent: March 3, 2015
    Assignee: Dolby Laboratories Licensing Corporation
    Inventor: Hannes Muesch
  • Patent number: 8965756
    Abstract: Systems and methods to automatically equalize coloration in speech recordings is provided. In example embodiments, a reference spectral shape based on a reference signal is determined. An estimated spectral shape for an input signal is derived. Using the estimated spectral shape and the reference spectral shape a comparison is performed to determine gain settings. The gain settings comprise a gain value for each filter of a filter system. Using gain values associated with the gain setting, automatic equalization is performed on the input signal.
    Type: Grant
    Filed: March 14, 2011
    Date of Patent: February 24, 2015
    Assignee: Adobe Systems Incorporated
    Inventors: Sven Duwenhorst, Martin Schmitz
  • Patent number: 8965757
    Abstract: Multi-channel noise suppression systems and methods are described that omit the traditional delay-and-sum fixed beamformer in devices that include a primary speech microphone and at least one noise reference microphone with the desired speech being in the near-field of the device. The multi-channel noise suppression systems and methods use a blocking matrix (BM) to remove desired speech in the input speech signal received by the noise reference microphone to get a “cleaner” background noise component. Then, an adaptive noise canceler (ANC) is used to remove the background noise in the input speech signal received by the primary speech microphone based on the “cleaner” background noise component to achieve noise suppression. The filters implemented by the BM and ANC are derived using closed-form solutions that require calculation of time-varying statistics of complex frequency domain signals in the noise suppression system.
    Type: Grant
    Filed: November 14, 2011
    Date of Patent: February 24, 2015
    Assignee: Broadcom Corporation
    Inventors: Jes Thyssen, Huaiyu Zeng, Juin-Hwey Chen, Nelson Sollenberger, Xianxian Zhang
  • 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
  • Patent number: 8942976
    Abstract: The present invention provides a noise reduction control method using a microphone array and a noise reduction control device using a microphone array wherein the method comprises the steps of: S1: collecting, by the microphone array, acoustic signals; S2: estimating incidence angles of all acoustic signals of the microphone array; S3: conducting a statistics on signal components according to incidence angles; S4: determining a parameter ? from a ratio of noise components according to the statistical result and using the parameter ? as a control parameter for controlling an adaptive filter. With the present invention, space position information of the sound is obtained directly with the microphone array to control update of the adaptive filter more accurately, so as to eliminate noise, enhance SNR and protect speech quality well at the same time.
    Type: Grant
    Filed: December 15, 2010
    Date of Patent: January 27, 2015
    Assignee: Goertek Inc.
    Inventors: Bo Li, Shasha Lou, Song Li
  • Patent number: 8935164
    Abstract: A non-spatial speech detection system includes a plurality of microphones whose output is supplied to a fixed beamformer. An adaptive beamformer is used for receiving the output of the plurality of microphones and one or more processors are used for processing an output from the fixed beamformer and identifying speech from noise though the use of an algorithm utilizing a covariance matrix.
    Type: Grant
    Filed: May 2, 2012
    Date of Patent: January 13, 2015
    Assignee: Gentex Corporation
    Inventors: Robert R. Turnbull, Michael A. Bryson
  • Patent number: 8935159
    Abstract: 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 musical noises.
    Type: Grant
    Filed: April 17, 2013
    Date of Patent: January 13, 2015
    Assignees: SK Telecom Co., Ltd, Transono Inc.
    Inventors: Seong-Soo Park, Seong Il Jeong, Dong Gyung Ha, Jae Hoon Song
  • Patent number: 8934641
    Abstract: Systems and methods for reconstructing decomposed audio signals are presented. In exemplary embodiments, a decomposed audio signal is received. The decomposed audio signal may include a plurality of frequency sub-band signals having successively shifted group delays as a function of frequency from a filter bank. The plurality of frequency sub-band signals may then be grouped into two or more groups. A delay function may be applied to at least one of the two or more groups. Subsequently, the groups may be combined to reconstruct the audio signal, which may be outputted accordingly.
    Type: Grant
    Filed: December 31, 2008
    Date of Patent: January 13, 2015
    Assignee: Audience, Inc.
    Inventors: Carlos Avendano, Ludger Solbach
  • Patent number: 8930197
    Abstract: A method comprising receiving at a user equipment encrypted content. The content is stored in said user equipment in an encrypted form. At least one key for decryption of said stored encrypted content is stored in the user equipment.
    Type: Grant
    Filed: May 9, 2008
    Date of Patent: January 6, 2015
    Assignee: Nokia Corporation
    Inventors: Anssi Ramo, Mikko Tammi, Adriana Vasilache, Lasse Laaksonen
  • Patent number: 8930186
    Abstract: 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.
    Type: Grant
    Filed: November 14, 2012
    Date of Patent: January 6, 2015
    Assignee: 2236008 Ontario Inc.
    Inventors: Phillip A. Hetherington, Shreyas Paranjpe, Xueman Li
  • Patent number: 8924199
    Abstract: A voice correction device includes a detector that detects a response from a user, a calculator that calculates an acoustic characteristic amount of an input voice signal, an analyzer that outputs an acoustic characteristic amount of a predetermined amount when having acquired a response signal due to the response from the detector, a storage unit that stores the acoustic characteristic amount output by the analyzer, a controller that calculates an correction amount of the voice signal on the basis of a result of a comparison between the acoustic characteristic amount calculated by the calculator and the acoustic characteristic amount stored in the storage unit, and a correction unit that corrects the voice signal on the basis of the correction amount calculated by the controller.
    Type: Grant
    Filed: December 20, 2011
    Date of Patent: December 30, 2014
    Assignee: Fujitsu Limited
    Inventors: Chisato Ishikawa, Takeshi Otani, Taro Togawa, Masanao Suzuki, Masakiyo Tanaka
  • Patent number: 8924206
    Abstract: An electrical apparatus a voice signal receiving method thereof are disclosed. The electrical apparatus includes a plurality of voice receivers, a voice activity detector, a voice channel switch and a noise eliminator. The voice receivers are used to receive the voice signals. The voice activity detector receives and detects the voice signals, and obtains a main voice signal from the voice signals. The voice channel switch transports the main voice signal to a voice transporting channel and transports a plurality of other voice signals of the voice signals other than the main voice signal to a noise transporting channel according to a detecting result of the voice activity detector. The noise eliminator reduces the noise in the main voice according to the voice signals from the noise transporting channel.
    Type: Grant
    Filed: November 4, 2011
    Date of Patent: December 30, 2014
    Assignee: HTC Corporation
    Inventors: Ting-Wei Sun, Hann-Shi Tong
  • Publication number: 20140372110
    Abstract: A Voice Call Enhancement Method for wireless telephonic communication devices includes providing an input voice audio source, enhancing the voice audio input in multiple harmonic and dynamic ranges and outputting the voice enhanced audio. The Voice Call Enhancement method is suitable for use of wireless telephony devices, such as cellular phones. The enhancement includes resynthesizing audio to an increased harmonic and dynamic range than original values.
    Type: Application
    Filed: February 17, 2014
    Publication date: December 18, 2014
    Applicant: Max Sound Corporation
    Inventor: Lloyd Trammell
  • Publication number: 20140372111
    Abstract: A Voice Recognition Enhancement Method for wireless telephonic communication devices includes providing an input voice audio source, enhancing the voice audio input in one or more of harmonic and dynamic ranges and outputting the voice enhanced audio. The Voice Recognition Enhancement method is suitable for use of wireless telephony devices, such as cellular phones. The enhancement includes resynthesizing audio to an increased harmonic and dynamic range than original values.
    Type: Application
    Filed: February 17, 2014
    Publication date: December 18, 2014
    Applicant: Max Sound Corporation
    Inventor: Lloyd Trammell
  • Patent number: 8914281
    Abstract: A method and an apparatus for processing an audio signal in a mobile terminal, in which an audio signal that is received from a counterpart mobile terminal is classified into a voice signal and a noise signal according to respective energy. A frequency of the classified voice signal and an energy of the classified noise signal is controlled according to a predetermined criteria, then the controlled voice signal and the controlled noise signal are coupled and output to a speaker.
    Type: Grant
    Filed: October 4, 2011
    Date of Patent: December 16, 2014
    Assignee: Samsung Electronics Co., Ltd.
    Inventors: Gun-Hyun Yoon, Dong-Won Lee, Ju-Hee Chang, Koong-Hoon Nam
  • Patent number: 8886526
    Abstract: 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 separation described herein involves mixed multivariate probability density functions that are mixtures of component density functions having different parameters corresponding to frequency components of different sources, different time segments, or some combination thereof.
    Type: Grant
    Filed: May 4, 2012
    Date of Patent: November 11, 2014
    Assignee: Sony Computer Entertainment Inc.
    Inventors: Jaekwon Yoo, Ruxin Chen
  • Patent number: 8880395
    Abstract: 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.
    Type: Grant
    Filed: May 4, 2012
    Date of Patent: November 4, 2014
    Assignee: Sony Computer Entertainment Inc.
    Inventors: Jaekwon Yoo, Ruxin Chen
  • Patent number: 8880394
    Abstract: In response to a first envelope within a kth frequency band of a first channel, a speech level within the kth frequency band of the first channel is estimated. In response to a second envelope within the kth frequency band of a second channel, a noise level within the kth frequency band of the second channel is estimated. A noise suppression gain for a time frame n is computed in response to the estimated speech level for a preceding time frame, the estimated noise level for the preceding time frame, the estimated speech level for the time frame n, and the estimated noise level for the time frame n. An output channel is generated in response to multiplying the noise suppression gain for the time frame n and the first channel.
    Type: Grant
    Filed: August 20, 2012
    Date of Patent: November 4, 2014
    Assignee: Texas Instruments Incorporated
    Inventors: Devangi Nikunj Parikh, Muhammad Zubair Ikram, Takahiro Unno
  • Patent number: 8868416
    Abstract: Disclosed in the present invention is a method for cancelling echo in joint time domain and frequency domain.
    Type: Grant
    Filed: December 22, 2010
    Date of Patent: October 21, 2014
    Assignee: Goertek Inc.
    Inventors: Shasha Lou, Song Liu
  • Patent number: 8868415
    Abstract: A method and system is disclosed for control of discontinuous transmission based on vocoder and voice activity. An access terminal (AT) may engage in a communication session via an encoder-decoder in a network device in a wireless network. During silence intervals of the communication session, when the AT has no data to transmit, the AT may transmit periodic silence frames at a silence-frame rate to the encoder-decoder. The silence frames may contain parameters for generation of audio noise by the network device. Upon determining that the encoder-decoder has ceased transmitting data to the AT in response to a prolonged absence of transmissions from the AT, the AT may increase the silence-frame rate so as to reduce the duration of the absence of transmissions from the AT, and correspondingly cause the encoder-decoder to begin transmitting audio data to the AT.
    Type: Grant
    Filed: May 22, 2012
    Date of Patent: October 21, 2014
    Assignee: Sprint Spectrum L.P.
    Inventors: Deveshkumar Rai, Sachin R. Vargantwar, Maulik K. Shah, Jasinder P. Singh
  • Patent number: 8868432
    Abstract: A method for decoding an audio signal having a bandwidth that extends beyond a bandwidth of a CELP excitation signal in an audio decoder including a CELP-based decoder element. The method includes obtaining a second excitation signal having an audio bandwidth extending beyond the audio bandwidth of the CELP excitation signal, obtaining a set of signals by filtering the second excitation signal with a set of bandpass filters, scaling the set of signals using a set of energy-based parameters, and obtaining a composite output signal by combining the scaled set of signals with a signal based on the audio signal decoded by the CELP-based decoder element.
    Type: Grant
    Filed: September 28, 2011
    Date of Patent: October 21, 2014
    Assignee: Motorola Mobility LLC
    Inventors: Jonathan A. Gibbs, James P. Ashley, Udar Mittal
  • Patent number: 8849657
    Abstract: In an apparatus and method for isolating a multi-channel sound source, the probability of speaker presence calculated when noise of a sound source signal separated by GSS is estimated is used to calculate a gain. Thus, it is not necessary to additionally calculate the probability of speaker presence when calculating the gain, the speaker's voice signal can be easily and quickly separated from peripheral noise and reverb and distortion are minimized. As such, if several interference sound sources, each of which has directivity, and speakers are simultaneously present in a room with high reverb, a plurality of sound sources generated from several microphones can be separated from one another with low sound quality distortion, and the reverb can also be removed.
    Type: Grant
    Filed: December 14, 2011
    Date of Patent: September 30, 2014
    Assignee: Samsung Electronics Co., Ltd.
    Inventor: Ki Hoon Shin