Dereverberators Patents (Class 381/66)
  • Patent number: 10412489
    Abstract: A method for auralizing a multi-microphone device. Path information for one or more sound paths using dimensions and room reflection coefficients of a simulated room for one of a plurality of microphones included in a multi-microphone device is determined. An array-related transfer functions (ARTFs) for the one of the plurality of microphones is retrieved. The auralized impulse response for the one of the plurality of microphones is generated based at least on the retrieved ARTFs and the determined path information.
    Type: Grant
    Filed: June 1, 2018
    Date of Patent: September 10, 2019
    Assignee: GOOGLE LLC
    Inventors: Rajeev Conrad Nongpiur, Ananya Misra, Chanwoo Kim
  • Patent number: 10404299
    Abstract: Described is a cognitive signal processor (CSP) for signal denoising. In operation, the CSP receives a noisy signal as a time-series of data points from a mixture of both noise and one or more desired waveform signals. The noisy signal is linearly mapped to reservoir states of a dynamical reservoir. A high-dimensional state-space representation is then generated of the noisy signal by combining the noisy signal with the reservoir states. A delay-embedded state signal is generated from the reservoir states. The reservoir states are denoised by removing noise from each reservoir state signal, resulting in a real-time denoised spectrogram of the noisy signal. A denoised waveform signal is generated combining the denoised reservoir states. Additionally, the signal denoising process is implemented in software or digital hardware by converting the state-space representation of the dynamical reservoir to a system of delay difference equations and then applying a linear basis approximation.
    Type: Grant
    Filed: March 2, 2018
    Date of Patent: September 3, 2019
    Assignee: HRL Laboratories, LLC
    Inventors: Peter Petre, Bryan H. Fong, Shankar R. Rao
  • Patent number: 10380062
    Abstract: Described is a system for signal denoising. The system linearly maps a noisy input signal into a high-dimensional reservoir, where the noisy input signal is a time-series of data points from a mixture of waveforms. A high-dimensional state-space representation of the mixture of waveforms is created by combining the noisy input signal with reservoir states. A delay embedded state signal is generated from the reservoir states, and a denoised spectrogram of the noisy input signal is generated.
    Type: Grant
    Filed: August 21, 2018
    Date of Patent: August 13, 2019
    Assignee: HRL Laboratories, LLC
    Inventors: Shankar R. Rao, Peter Petre
  • Patent number: 10367948
    Abstract: Acoustic echo cancellation systems and methods are provided that can cancel and suppress acoustic echo from the output of a mixer that has mixed audio signals from a plurality of acoustic sources, such as microphones. The microphones may have captured speech and sound from a remote location or far end, such as in a conferencing environment. The acoustic echo cancellation may generate an echo-cancelled mixed audio signal based on a mixed audio signal from a mixer, information gathered from the audio signal from each of the plurality of acoustic sources, and a remote audio signal. The systems and methods may be computationally efficient and resource-friendly.
    Type: Grant
    Filed: January 13, 2017
    Date of Patent: July 30, 2019
    Assignee: Shure Acquisition Holdings, Inc.
    Inventors: Sean Wells-Rutherford, Mathew T. Abraham, John Casey Gibbs
  • Patent number: 10347272
    Abstract: Provided are a de-reverberation control method and apparatus for a device equipped with a microphone. The method includes: reverberation parameters which indicate, at respective moments, reverberation levels of a room environment where the device is located are acquired from an audio signal played by the device; and a de-reverberation mode adopted by the device is dynamically adjusted according to the reverberation levels indicated by the reverberation parameters at different moments and preset correspondences between reverberation levels and de-reverberation modes. By adopting a dynamic de-reverberation mode, the method and the apparatus disclosed herein significantly improve the rate of the recognition of a device for the voice of the user.
    Type: Grant
    Filed: December 20, 2017
    Date of Patent: July 9, 2019
    Assignee: Beijing Xiaoniao Tingting Technology Co., LTD.
    Inventors: Bo Li, Shasha Lou
  • Patent number: 10339951
    Abstract: The present invention relates to a method for audio signal processing in a vehicle. In order to allow simple and reliable echo cancellation for voice recognition during simultaneous reproduction of a multichannel audio source signal in a vehicle, a mono audio signal is generated on the basis of a multichannel audio source signal. The mono audio signal is limited to a frequency range between a prescribed lower frequency and a prescribed upper frequency, for example to a range from 100 Hz to 8 kHz. The limited mono audio signal is output via multiple loudspeakers in the vehicle. An influence of the limited mono audio signal that is output via the multiple loudspeakers on a voice audio signal received in the vehicle via a microphone is compensated for by means of the limited mono audio signal in an echo canceller.
    Type: Grant
    Filed: October 26, 2016
    Date of Patent: July 2, 2019
    Assignee: VOLKSWAGEN AKTIENGESELLSCHAFT
    Inventor: David Scheler
  • Patent number: 10299279
    Abstract: The invention pertains to a communication system for use in a railway vehicle, comprising: a first communication network (10) and a second communication network (20), the communication networks (10, 20) using physically separate communication media; and a plurality of communication terminals (100, 200), each connected to both communication networks (10, 20). The communication system is adapted to prioritize communications from the communication terminals (100, 200) over the communication networks (10, 20) according to at least two levels of service. A first communication terminal (100) comprises a first functional module (110) and a second functional module (120), the functional module (110, 120) being functionally equivalent, the first functional module (110) being adapted to interface with the first communication network (10) and the second functional module (120) being adapted to interface with the second communication network (20).
    Type: Grant
    Filed: May 18, 2015
    Date of Patent: May 21, 2019
    Assignee: Televic Rail NV
    Inventors: John Gesquiere, Luc Claeys, Bart Vercoutter, Kristof Boerjan, Jan Van Den Oudenhoven
  • Patent number: 10276181
    Abstract: A method, computer program product, and computer system for addressing acoustic signal reverberation is provided. Embodiments may include receiving, at one or more microphones, a first audio signal and a reverberation audio signal. Embodiments may further include processing at least one of the first audio signal and the reverberation audio signal. Embodiments may also include limiting a model based reverberation equalizer using a temporal constraint for direct sound distortions, the model based reverberation equalizer configured to generate one or more outputs, based upon, at least in part, at least one of the first audio signal and the reverberation audio signal.
    Type: Grant
    Filed: September 5, 2017
    Date of Patent: April 30, 2019
    Assignee: Nuance Communications, Inc.
    Inventors: Tobias Wolff, Lars Tebelmann
  • Patent number: 10244303
    Abstract: An in-ear BLUETOOTH® headset antenna for single-ear and double-ear BLUETOOTH® headset. The antenna includes a radiation unit and a ground unit, both utilizing components that make up the headset. The radiation unit is composed of a horn of the BLUETOOTH® headset and a conductive foil attached to the horn surface. One end of the conductive foil is attached to the surface of the headset horn and the other end is connected to the feed point of the RF circuit antenna of the BLUETOOTH® headset. The ground unit includes copper pouring on a main printed circuit board connected to copper pouring a key printed circuit board by a cable. No additional antennas are required, and the in-ear BLUETOOTH® headset antenna reduces costs, saves space, and improves the radiation efficiency of the antenna due to the increased effective radiation area of the antenna.
    Type: Grant
    Filed: April 24, 2018
    Date of Patent: March 26, 2019
    Assignee: SHENZHEN ATX TECHNOLOGY CO., LTD.
    Inventors: Mingqiang Xu, Yuechun Huang
  • Patent number: 10177894
    Abstract: A method and an apparatus for tuning an FIR filter in an in-band full duplex transceiver. The method for tuning an FIR filter may include: setting attenuation of the FIR filter to be a first value and then estimating input information of the FIR filter; estimating a delta response using the estimated input information of the FIR filter; and updating the attenuation of the FIR filter to a second value using the estimated delta response.
    Type: Grant
    Filed: June 21, 2016
    Date of Patent: January 8, 2019
    Assignee: ELECTRONICS AND TELECOMMUNICATIONS RESEARCH INSTITUTE
    Inventors: Kapseok Chang, Hyung Sik Ju
  • Patent number: 10177805
    Abstract: A method and an apparatus for tuning an FIR filter in an in-band full duplex transceiver. The method for tuning an FIR filter includes: converting an input signal of the FIR filter into a first signal that is a baseband signal; converting a signal obtained by subtracting an output signal of the FIR filter from the self-transmitted interference signal into a second signal that is the baseband signal; and calculating attenuation of the FIR filter using the first signal and the second signal.
    Type: Grant
    Filed: June 21, 2016
    Date of Patent: January 8, 2019
    Assignee: ELECTRONICS AND TELECOMMUNICATIONS RESEARCH INSTITUTE
    Inventors: Kapseok Chang, Hyung Sik Ju
  • Patent number: 10162378
    Abstract: Described is a neuromorphic processor for signal denoising and separation. The neuromorphic processor generates delay-embedded mixture signals from an input mixture of pulses. Using a reservoir computer, the delay-embedded mixture signals are mapped to reservoir states of a dynamical reservoir having output layer weights. The output layer weights are adapted based on short-time linear prediction, and a denoised output of the mixture of input signals us generated. The denoised output is filtered through a set of adaptable finite impulse response (FIR) filters to extract a set of separated narrowband pulses.
    Type: Grant
    Filed: June 23, 2017
    Date of Patent: December 25, 2018
    Assignee: HRL Laboratories, LLC
    Inventors: Shankar R. Rao, Peter Petre, Charles E. Martin
  • Patent number: 10152986
    Abstract: An acoustic processing apparatus includes a storage, an estimation unit, and a removal unit. The storage stores therein a reference signal indicating a signal obtained by completing removal of reverberation from a first observation signal included in a first processing section. The estimation unit estimates, on the basis of a model representing an observation signal as a signal obtained by adding a signal obtained by applying a reverberation removal filter to an acoustic signal that is input with a delay and the acoustic signal, a filter coefficient of the reverberation removal filter by using a second observation signal and the reference signal. The removal unit determines an output signal indicating a signal obtained by removing reverberation from the second observation signal by using the second observation signal, the reference signal, and the reverberation removal filter having the estimated filter coefficient.
    Type: Grant
    Filed: July 10, 2017
    Date of Patent: December 11, 2018
    Assignee: Kabushiki Kaisha Toshiba
    Inventors: Takehiko Kagoshima, Toru Taniguchi
  • Patent number: 10153806
    Abstract: Described is a cognitive signal processor that can denoise an input signal that contains a mixture of waveforms over a large bandwidth. Delay-embedded mixture signals are generated from a mixture of input signals. The delay-embedded mixture signals are mapped with a reservoir computer to reservoir states of a dynamical reservoir having output layer weights. The output layer weights are adapted based on short-time linear prediction. Finally, a denoised output of the mixture of input signals is generated.
    Type: Grant
    Filed: March 7, 2017
    Date of Patent: December 11, 2018
    Assignee: HRL Laboratories, LLc
    Inventors: Peter Petre, Shankar R. Rao
  • Patent number: 10128820
    Abstract: Described is a cognitive signal processor for signal denoising and blind source separation. During operation, the cognitive signal processor receives a mixture signal that comprises a plurality of source signals. A denoised reservoir state signal is generated by mapping the mixture signal to a dynamic reservoir to perform signal denoising. At least one separated source signal is identified by adaptively filtering the denoised reservoir state signal.
    Type: Grant
    Filed: November 20, 2017
    Date of Patent: November 13, 2018
    Assignee: HRL Laboratories, LLC
    Inventors: Peter Petre, Bryan H. Fong, Shankar R. Rao, Charles E. Martin
  • Patent number: 10110994
    Abstract: A method, apparatus and computer program product enhance audio quality during a voice communication session, such as by enhancing audio quality for a remote participant in a meeting. In a method and for each of two or more microphones of a first device at a first location, a target audio signal is generated that has been steered in a direction of a target audio source in order to provide at least partial isolation from a second audio source in the same environment. The method also produces a filtered audio signal based on the target audio source at least from a respective one of the two or more microphones. The method also includes mixing the filtered audio signal from at least the first device to create an audio output signal associated with an audio playback format and causing the audio output signal to be output by a second device.
    Type: Grant
    Filed: November 21, 2017
    Date of Patent: October 23, 2018
    Assignee: NOKIA TECHNOLOGIES OY
    Inventor: Ian Davis
  • Patent number: 10079028
    Abstract: Embodiments of the present invention relate to enhancing sound through reverberation matching. In sonic implementations, a first sound recording recorded in a first environment is received. The first sound recording is decomposed to a first clean signal and a first reverb kernel. A second reverb kernel corresponding with a second sound recording recorded in a second environment is accessed, for example, based on a user indication to enhance the first sound recording to sound as though recorded in the second environment. An enhanced sound recording is generated based on the first clean signal and the second reverb kernel. The enhanced sound recording is a modification of the first sound recording to sound as though recorded in the second environment.
    Type: Grant
    Filed: December 8, 2015
    Date of Patent: September 18, 2018
    Assignee: Adobe Systems Incorporated
    Inventors: Ramin Anushiravani, Paris Smaragdis, Gautham Mysore
  • Patent number: 10062392
    Abstract: A method for estimating an instantaneous phase of dereverberated acoustic signal, the method comprising the following steps: measurement of an acoustic signal reverberated by propagation in a medium, estimation of a one short-term Fourier transform of the reverberated acoustic signal with a window function, calculation of an instantaneous frequency of dereverberated signal from said short-term Fourier transform and from an influencing factor of the medium, said influencing factor being a function of a reverberation time of said medium, determination of an instantaneous phase of dereverberated signal by integrating the instantaneous frequency of dereverberated signal over time.
    Type: Grant
    Filed: May 25, 2017
    Date of Patent: August 28, 2018
    Assignee: INVOXIA
    Inventors: Arthur Belhomme, Roland Badeau, Yves Grenier, Eric Humbert
  • Patent number: 10013964
    Abstract: A system and method for controlling noise originating from a source external to a vehicle is disclosed. The method includes determining, by an active noise controller of a vehicle, characteristics of an unwanted noise. The unwanted noise originates from a source external to the vehicle. The method also includes determining an inverted noise based on the characteristics of the unwanted noise. The method also includes projecting the inverted noise. The projected inverted noise destructively interferes with the unwanted noise. The method also includes receiving a residual noise via an error microphone. The error microphone is configured to generate a signal based on the received residual noise.
    Type: Grant
    Filed: August 22, 2017
    Date of Patent: July 3, 2018
    Assignee: GM GLOBAL TECHNOLOGY OPERATIONS LLC
    Inventors: Eli Tzirkel-Hancock, Ilan Malka, Scott M. Reilly, Frank C. Valeri
  • Patent number: 9986331
    Abstract: A terminal monitors a working state parameter of a main microphone arranged on the terminal; judges whether the working state parameter satisfies a preset rule; and prompts a user that an audio input signal is abnormal or selects a main microphone satisfying the preset rule as an audio input according to a judgment result.
    Type: Grant
    Filed: August 7, 2014
    Date of Patent: May 29, 2018
    Assignee: ZTE Corporation
    Inventor: Jiuxing Li
  • Patent number: 9936320
    Abstract: A transfer function calculation unit is configured to calculate a transfer function from a sound source installed in a predetermined target direction to each microphone of a microphone array, and a determination unit is configured to determine whether or not the microphone array is normal on the basis of a difference amount between a transfer function to each microphone and a predetermined ideal transfer function to each microphone.
    Type: Grant
    Filed: March 1, 2017
    Date of Patent: April 3, 2018
    Assignee: HONDA MOTOR CO., LTD.
    Inventors: Takeshi Mizumoto, Keisuke Nakamura, Kazuhiro Nakadai
  • Patent number: 9881630
    Abstract: Provided are methods and systems for acoustic keystroke transient cancellation/suppression for user communication devices using a semi-blind adaptive filter model. The methods and systems are designed to overcome existing problems in transient noise suppression by taking into account some less-defective signal as side information on the transients and also accounting for acoustic signal propagation, including the reverberation effects, using dynamic models. The methods and systems take advantage of a synchronous reference microphone embedded in the keyboard of the user device, and utilize an adaptive filtering approach exploiting the knowledge of this keybed microphone signal.
    Type: Grant
    Filed: December 30, 2015
    Date of Patent: January 30, 2018
    Assignee: GOOGLE LLC
    Inventors: Herbert Buchner, Simon J. Godsill, Jan Skoglund
  • Patent number: 9877134
    Abstract: In one embodiment, a recording optimizer included in a recorder optimizes the fidelity of output signals. In operation, the recorder receives and records two sets of input signals—one set of input signals received via microphones included in the recorder and another set of input signals received as line-inputs from a front of house console mixer. The recording optimizer analyzes the recorded signals to identify discrepancies, such as unamplified instruments that are underrepresented in the line-inputs. The recording optimizer then performs compensation operations that adjust one or more recorded signals to mitigate the discrepancies. Subsequently, the recording optimizer combines the compensated recorded signals, generating output signals that leverage the strengths of both sets of input signals to accurately represent the experienced sound.
    Type: Grant
    Filed: July 28, 2015
    Date of Patent: January 23, 2018
    Assignee: HARMAN INTERNATIONAL INDUSTRIES, INCORPORATED
    Inventor: James M. Kirsch
  • Patent number: 9817100
    Abstract: An array of microphones placed on a mobile robot provides multiple channels of audio signals. A received set of audio signals is called an audio segment, which is divided into multiple frames. A phase analysis is performed on a frame of the signals from each pair of microphones. If both microphones are in an active state during the frame, a candidate angle is generated for each such pair of microphones. The result is a list of candidate angles for the frame. This list is processed to select a final candidate angle for the frame. The list of candidate angles is tracked over time to assist in the process of selecting the final candidate angle for an audio segment.
    Type: Grant
    Filed: August 19, 2016
    Date of Patent: November 14, 2017
    Assignee: Microsoft Technology Licensing, LLC
    Inventors: Shankar Regunathan, Kazuhito Koishida, Harshavardhana Narayana Kikkeri
  • Patent number: 9813811
    Abstract: At a microphone array, a soundfield is detected to produce a set of microphone signals each from a corresponding microphone in the microphone array. The set of microphone signals represents the soundfield. The detected soundfield is decomposed into a set of sub-soundfield signals based on the set of microphone signals. Each sub-soundfield signal is processed, such that each sub-soundfield signal is separately dereverberated to remove reverberation therefrom, to produce a set of processed sub-soundfield signals. The set of processed sub-sound field signals are mixed into a mixed output signal.
    Type: Grant
    Filed: June 1, 2016
    Date of Patent: November 7, 2017
    Assignee: Cisco Technology, Inc.
    Inventor: Haohai Sun
  • Patent number: 9749007
    Abstract: Described is a cognitive blind source separator (CBSS). The CBSS includes a delay embedding module that receives a mixture signal (the mixture signal being a time-series of data points from one or more mixtures of source signals) and time-lags the signal to generate a delay embedded mixture signal. The delay embedded mixture signal is then linearly mapped into a reservoir to create a high-dimensional state-space representation of the mixture signal. The state-space representations are then linearly mapped to one or more output nodes in an output layer to generate pre-filtered signals. The pre-filtered signals are passed through a bank of adaptable finite impulse response (FIR) filters to generate separate source signals that collectively formed the mixture signal.
    Type: Grant
    Filed: March 17, 2016
    Date of Patent: August 29, 2017
    Assignee: HRL Laboratories, LLC
    Inventors: Charles E Martin, Shankar R. Rao, Peter Petre
  • Patent number: 9729967
    Abstract: Reducing feedback in an input signal that contains a signal component based on an output signal from a proximate output. The input signal is separated into a plurality of frequency bands by band pass filters. The power of signal in each band is determined, and the band signal with the greatest power is selected. That band's signal is sampled at a sampling rate, and at regular intervals one of the samples is selected. Blind signal separation is used to estimate signal sources from the selected samples. The estimated signals are compared to the output signal, and the estimated signal most similar to the output signal is subtracted from the input signal.
    Type: Grant
    Filed: March 7, 2014
    Date of Patent: August 8, 2017
    Assignee: Board of Trustees of Northern Illinois University
    Inventor: Mansour Tahernezhadi
  • Patent number: 9723152
    Abstract: Presented is a method and associated system for suppression of linear and nonlinear echo. The method includes dividing an input signal into several frequency bands in each of a several of time frames. The input signal may include an echo signal. The method further includes multiplying the input signal in each of the several frequency bands by a corresponding echo suppression signal. Calculating the corresponding echo suppression signal may include estimating a power of the echo signal in a particular frequency band as a sum of several component echo powers, each of the several component echo powers due to an excitation from a far-end signal in a corresponding one of the several frequency bands. Calculating the corresponding echo suppression signal may further include subtracting the power of the echo signal in the particular frequency band from a power of the input signal in the particular frequency band.
    Type: Grant
    Filed: June 1, 2015
    Date of Patent: August 1, 2017
    Assignee: Conexant Systems, LLC
    Inventors: Youhong Lu, Trausti Thormundsson
  • Patent number: 9711164
    Abstract: An embodiment of the invention provides a noise cancellation method for an electronic device. The method comprises: receiving an audio signal; applying a Fast Fourier Transform operation on the audio signal to generate a sound spectrum; acquiring a first spectrum corresponding to a noise and a second spectrum corresponding to a human voice signal from the sound spectrum; estimating a center frequency according to the first spectrum and the second spectrum; and applying a high pass filtering operation to the sound spectrum according to the center frequency.
    Type: Grant
    Filed: January 21, 2016
    Date of Patent: July 18, 2017
    Assignee: HTC CORPORATION
    Inventors: Lei Chen, Yu-Chieh Lai, Chun-Ren Hu, Hann-Shi Tong
  • Patent number: 9699552
    Abstract: An apparatus for computing filter coefficients for an adaptive filter is disclosed. The adaptive filter is used for filtering a microphone signal so as to suppress an echo due to a loudspeaker signal. The apparatus has: an echo decay modeling means for modeling a decay behavior of an acoustic environment and for providing a corresponding echo decay parameter; and computing means for computing the filter coefficients of the adaptive filter on the basis of the echo decay parameter. A corresponding method has: providing echo decay parameters determined by means of an echo decay modeling means; and computing the filter coefficients of the adaptive filter on the basis of the echo decay parameters.
    Type: Grant
    Filed: April 22, 2013
    Date of Patent: July 4, 2017
    Assignee: Faunhofer-Gesellschaft zur Foerderung der angewandten
    Inventors: Fabian Kuech, Markus Schmidt, Alexis Favrot, Christof Faller
  • Patent number: 9672076
    Abstract: A system includes a CPU; an accelerator; a comparing unit that compares a first value that is based on a first processing time period elapsing until the CPU completes a first process and a second processing time period elapsing until the accelerator completes the first process, and a second value that is based on a state of use of a battery driving the CPU and the accelerator; and a selecting unit that selects any one among the CPU and the accelerator, based on a result of comparison by the comparing unit.
    Type: Grant
    Filed: September 16, 2013
    Date of Patent: June 6, 2017
    Assignee: FUJITSU LIMITED
    Inventors: Takahisa Suzuki, Koichiro Yamashita, Hiromasa Yamauchi, Koji Kurihara, Fumihiko Hayakawa, Naoki Odate, Tetsuo Hiraki, Toshiya Otomo
  • Patent number: 9654860
    Abstract: The present invention provides a soundproof housing for an earset, comprising: a housing main body, coupled to the inside of a front surface case having a protrusion portion inserted into the ear, and provided with a speaker accommodation groove and a microphone accommodation groove; a speaker output hole, penetratingly formed in the speaker accommodation groove so as to communicate with the front surface case, and adjacent to the output end of the speaker; and a microphone input hole, formed in a recessed manner inside the microphone accommodation groove so as to communicate with the front surface case, and adjacent to the input end of the microphone, wherein the housing main body is protrudingly formed as a long protrusion toward the inside of the protrusion portion of the front surface case, so as to be tightly coupled with the inside of the protrusion portion of the front surface case.
    Type: Grant
    Filed: November 9, 2012
    Date of Patent: May 16, 2017
    Assignee: HAEBORA
    Inventor: Doo Sik Shin
  • Patent number: 9646629
    Abstract: In accordance with an embodiment of the present invention, a noise/interference reduction method for speech enhancement processing includes selecting one of the microphones as a main microphone wherein the signal from the main microphone is used as a target signal, the selection of the main microphone is adaptive for mono output case, and the selection of the main microphone is fixed for stereo output case. The noise/interference component signal is estimated by subtracting voice component signal from a first microphone input signal wherein the voice component signal is evaluated as a first replica signal produced by passing a second microphone input signal through a first adaptive filter. A noise/interference reduced signal is output by subtracting a second replica signal from the target signal, wherein the second replica signal is produced by passing the estimated noise/interference component signal through a second adaptive filter.
    Type: Grant
    Filed: May 2, 2015
    Date of Patent: May 9, 2017
    Inventor: Yang Gao
  • Patent number: 9647705
    Abstract: The present application a digital self-interference residual cancellation method that adjusts a magnitude of a sampled transmit signal based on compared magnitude and phases associated with tones. The digital self-interference residual cancellation method may follow an analog carrier cancellation stage where the digital self-interference residual cancellation is based on a determination of the channel circuit response used to control an infinite impulse response filter which can compensate using both poles and zeroes.
    Type: Grant
    Filed: December 29, 2015
    Date of Patent: May 9, 2017
    Assignee: LGS INNOVATIONS LLC
    Inventors: Riley Nelson Pack, Alan Scott Brannon, Benjamin Joseph Baker
  • Patent number: 9635457
    Abstract: An audio processing unit having a first and second input for receiving output signals of a microphone with a first and a second physically symmetrically structured microphone capsule. The audio processing unit further has a first filter unit and a first delay unit which is coupled to the first input, and a second filter unit and a second delay unit which is coupled to the second input. The audio processing unit further has an adding unit for adding signals from the first and second filter units and a control unit for influencing the filter parameters of the first and second filters and/or the delay times of the first and second delay units depending on an amplitude of an audio signal received via the first and/or second input. A directivity factor of the output signal of the microphone is controlled depending on the amplitude of the output signal of the microphone.
    Type: Grant
    Filed: March 25, 2015
    Date of Patent: April 25, 2017
    Assignee: Sennheiser electronic GmbH & Co. KG
    Inventor: Alexander Nowak
  • Patent number: 9613028
    Abstract: Broadly speaking, the embodiments disclosed herein describe replacing a current hearing aid profile stored in a hearing aid. In one embodiment, the hearing aid profile is updated by sending a hearing aid profile update request to a hearing aid profile service, receiving the updated hearing aid profile from the hearing aid profile service, and replacing the current hearing aid profile in the hearing aid with the updated hearing aid profile.
    Type: Grant
    Filed: January 19, 2011
    Date of Patent: April 4, 2017
    Assignee: Apple Inc.
    Inventors: Edwin W. Foo, Gregory F. Hughes
  • Patent number: 9596534
    Abstract: Methods and systems are provided for controlling bone conduction, in which a bone conduction element may be used to output acoustic signals when it is in contact with a user. A bone conduction sensor may also be made in contact with the user, and used to obtain feedback relating to the outputting of the acoustic signals via the bone conduction element. The outputting of the acoustic signals may then be adaptively controlled based on processing of the feedback. The adaptive controlling may comprise adjusting components and/or functions related to or used in the outputting of the acoustic signals. For example, the adaptive controlling may comprise adjusting gain, frequency response, and/or equalization associated with a drive amplifier driving the bone conduction element.
    Type: Grant
    Filed: December 30, 2013
    Date of Patent: March 14, 2017
    Assignee: DSP Group Ltd.
    Inventors: Arie Heiman, Moshe Haiut, Uri Yehuday
  • Patent number: 9584910
    Abstract: A sound gathering system is disclosed herein and includes a plurality of microphones each configured to sample sound coming from a sound source. A plurality of processors are arranged in a processor chain. Each processor is coupled to at least one of the microphones and is configured to store sound samples received from the at least one microphone to a memory. A controller is terminally connected to the processor chain via a first processor. The controller is configured to calculate at least one time delay for each microphone, wherein the at least one time delay for each microphone is provided to the processor coupled thereto and is used by the processor to determine a memory position from which to begin reading sound samples.
    Type: Grant
    Filed: December 17, 2014
    Date of Patent: February 28, 2017
    Assignee: Steelcase Inc.
    Inventor: Scott Edward Wilson
  • Patent number: 9538297
    Abstract: The invention is directed to a single channel mask estimation method capable of improving reverberant speech identification for CI users. The method is based on the energy of the reverberant signal and the residual signal computed from linear prediction (LP) analysis. The mask is estimated by comparing the energy ratio of the two signals at different frequency bins with an adaptive threshold. As the threshold is updated for each frame of speech based on the energy ratios of the reverberant and LP residual signals computed from previous frames, it is amenable for real-time implementation. It can thus be used as a specialized (for reverberant environments) sound coding strategy used for cochlear implant applications.
    Type: Grant
    Filed: November 7, 2014
    Date of Patent: January 3, 2017
    Assignee: The Board of Regents of the University of Texas System
    Inventors: Oldooz Hazrati, Philipos C. Loizou
  • Patent number: 9530428
    Abstract: An echo cancellation device includes: a full-band echo canceller that generates a pseudo-echo signal; a downsample processor that downsamples a received signal and extracts a low-band component delayed by a delay amount D1; a delay controller that delays the low-band component by a delay amount D2; a delay controller that delays an output signal of the delay controller by a delay amount D3; a low-band echo canceller that generates a pseudo-echo signal delayed by a delay amount D1+D2; and an upsample processor that upsamples the low-band pseudo-echo signal to generate a full-band pseudo-echo signal delayed by the delay amount 2D1+D2. The delay controllers control the delay amounts D2 and D3 such that a tap length LA satisfies a condition of LA?2D1+D2=D2+D3, the tap length LA indicating a response time of the adaptive filter in the full-band echo canceller.
    Type: Grant
    Filed: May 14, 2013
    Date of Patent: December 27, 2016
    Assignee: Mitsubishi Electric Corporation
    Inventors: Tomoharu Awano, Takashi Sudo, Atsuyoshi Yano, Atsushi Hotta
  • Patent number: 9520137
    Abstract: A method for suppressing the late reverberation of an audio signal. A plurality of prediction vectors are calculated. A plurality of observation vectors from the modulus of the complex time-frequency transform of an input signal is generated. A plurality of synthesis dictionaries from the plurality of observation vectors are constructed. A late reverberation spectrum from the plurality of synthesis dictionaries and the plurality of prediction vectors are estimated. A plurality of observation vectors are filtered to eliminate the late reverberation spectrum and obtain a dereverberated signal modulus.
    Type: Grant
    Filed: July 21, 2014
    Date of Patent: December 13, 2016
    Assignee: ARKAMYS
    Inventors: Nicolas Lopez, Gaël Richard, Yves Grenier
  • Patent number: 9497544
    Abstract: A method for echo reduction by an electronic device is described. The method includes nulling at least one speaker. The method also includes mixing a set of runtime audio signals based on a set of acoustic paths to determine a reference signal. The method also includes receiving at least one composite audio signal that is based on the set of runtime audio signals. The method further includes reducing echo in the at least one composite audio signal based on the reference signal.
    Type: Grant
    Filed: July 1, 2013
    Date of Patent: November 15, 2016
    Assignee: QUALCOMM Incorporated
    Inventors: Asif I. Mohammad, Lae-Hoon Kim, Erik Visser
  • Patent number: 9485572
    Abstract: A sound processing device includes a first calculation unit configured to calculate a suppression gain of noise by using respective input signals input from a plurality of microphones; an integration unit configured to obtain an integration gain by using a suppression gain of an acoustic echo and the suppression gain of the noise; an application unit configured to apply the integration gain to one input signal among the plurality of input signals; and a second calculation unit configured to calculate the suppression gain of the acoustic echo by using signals to which the integration gain is applied, output signals that are output to a replay device, and the one input signal.
    Type: Grant
    Filed: March 6, 2014
    Date of Patent: November 1, 2016
    Assignee: FUJITSU LIMITED
    Inventors: Kaori Endo, Yoshiteru Tsuchinaga
  • Patent number: 9443530
    Abstract: A method and system for acoustic echo cancellation stores received far-end data in a first buffer. When the far-end data in the first buffer exceeds a predefined length, the stored far-end data is used to calculate echo estimate data. The echo estimate data is stored in a second buffer. Whenever microphone data is received the error data is calculated independent of echo estimate data availability. In particular, subsequent to sufficient echo estimate data being stored in the second buffer and responsive to the reception of the microphone data, the error data is calculated by subtracting, from the microphone data, corresponding echo estimate data stored in the second buffer.
    Type: Grant
    Filed: September 11, 2014
    Date of Patent: September 13, 2016
    Assignee: Imagination Technologies Limited
    Inventors: Senthil Kumar Mani, Srinivas Akella
  • Patent number: 9418678
    Abstract: A sound processing device includes: a nonlinear processing unit that outputs a plurality of sound signals including sound sources existing in predetermined areas by performing a nonlinear process for a plurality of observed signals that are generated by a plurality of sound sources and are observed by a plurality of sensors; a signal selecting unit that selects a sound signal including a specific sound source from among the plurality of sound signals output by the nonlinear processing unit and the observed signal including the plurality of sound sources; and a sound separating unit that separates a sound signal including the specific sound source that is selected by the signal selecting unit from the observed signal selected by the signal selecting unit.
    Type: Grant
    Filed: July 14, 2010
    Date of Patent: August 16, 2016
    Assignee: SONY CORPORATION
    Inventors: Toshiyuki Sekiya, Mototsugu Abe
  • Patent number: 9398374
    Abstract: In accordance with embodiments of the present disclosure, an audio processing circuit for use in an audio device may perform non-linear acoustic echo cancellation by predicting a displacement associated with an audio speaker, wherein such prediction takes into account a nonlinear response of the audio speaker with a mathematical model that calculates the predicted displacement of the audio speaker as a function of a current signal associated with the audio speaker using a time-varying difference equation, wherein coefficients of the difference equation are based on a set of physical parameters of the audio speaker. From the predicted displacement, the processing circuit may calculate a predicted acoustic output of the audio speaker, which may be used to generate a reference signal to an acoustic echo canceller.
    Type: Grant
    Filed: August 12, 2014
    Date of Patent: July 19, 2016
    Assignee: Cirrus Logic, Inc.
    Inventors: Khosrow Lashkari, Jie Su
  • Patent number: 9384757
    Abstract: A desired signal is extracted with a higher accuracy from a mixed signal wherein a plurality of signals are mixed. At the time of extracting a first signal from a first mixed signal and a second mixed signal, said first mixed signal and second mixed signal having the first signal and second signal mixed therein, an estimate value of the first signal in the past is obtained as a first estimate value, and an estimate value of the second signal in the past is obtained as a second estimate value. Then, a first isolation signal is generated by subtracting the second estimate value from the first mixed signal, and a second isolation signal is generated by subtracting the first estimate value from the second mixed signal. Then, the signal generated using the first isolation signal and the second isolation signal is outputted as the first signal.
    Type: Grant
    Filed: September 30, 2010
    Date of Patent: July 5, 2016
    Assignee: NEC CORPORATION
    Inventor: Akihiko Sugiyama
  • Patent number: 9386373
    Abstract: A system and method for estimating a reverberation time is provided. The method includes estimating at least one room response of an audio capture environment with an acoustic echo canceller and generating an estimate of the reverberation time of the audio capture environment based on the at least one room response from the acoustic echo canceller.
    Type: Grant
    Filed: June 20, 2013
    Date of Patent: July 5, 2016
    Assignee: DTS, INC.
    Inventors: Changxue Ma, Guangji Shi, Jean-Marc Jot
  • Patent number: 9361874
    Abstract: A method and system for acoustic echo cancellation varies a step size of an adaptive filter in an acoustic echo canceller. Far-end data is received and echo estimate data is calculated using the received far-end data. Microphone data is received and error data is calculated using the received microphone data and the echo estimate data. A first average of the microphone data and a second average of the error data are computed over a predefined number of samples. An echo leakage is estimated using the first average and the second average wherein the echo leakage indicates an extent to which the far-end data is present in the error data, and the step size of the adaptive filter is varied based on the echo leakage and a maximum allowed step size.
    Type: Grant
    Filed: September 11, 2014
    Date of Patent: June 7, 2016
    Assignee: Imagination Technologies Limited
    Inventors: Senthil Kumar Mani, Srinivas Akella
  • Patent number: 9336767
    Abstract: An audio device may be configured to produce output audio and to capture input audio for speech recognition. In some cases, a second device may also be used to capture input audio to improve isolation of input audio with respect to the output audio. In addition, acoustic echo cancellation (AEC) may be used to remove components of output audio from input signals of the first and second devices. AEC may be implemented by an adaptive filter based on dynamically optimized filter coefficients. The filter coefficients may be analyzed to detect situations in which the first and second devices are too close to each other, and the user may then be prompted to increase the distance between the two devices.
    Type: Grant
    Filed: March 28, 2014
    Date of Patent: May 10, 2016
    Assignee: Amazon Technologies, Inc.
    Inventors: William Folwell Barton, Kavitha Velusamy, Philip Ryan Hilmes