Dereverberators Patents (Class 381/66)
  • Patent number: 11399100
    Abstract: A loudspeaker is driven with a loudspeaker signal to generate sound, and sound is converted to one or more microphone signals with one or more microphones. The microphone signals are concurrently transformed into far-field beam signals and near-field beam signals. The far-field beam signals and the near-field beam signals are concurrently processed to produce one or more far-field output signals and one or more near-field output signals, respectively. Echo is detected and canceled in the far-field beam signals and in the near-field beam signals. When the echo is not detected above a threshold, the one or more far-field output signals are outputted. When the echo is detected above the threshold, the one or more near-field output signals are outputted. A signal based on the one or more output signals is transmitted.
    Type: Grant
    Filed: June 28, 2019
    Date of Patent: July 26, 2022
    Assignee: CISCO TECHNOLOGY, INC.
    Inventors: Haohai Sun, Johan Ludvig Nielsen
  • Patent number: 11395067
    Abstract: A microphone-loudspeaker integrated apparatus includes a plurality of microphones, a loudspeaker, and a synthesis unit. The loudspeaker is disposed between the plurality of microphones. The synthesis unit synthesizes sounds collected from the plurality of microphones. A distance between the plurality of microphones is set so as to fall within a range within which sound at a frequency needed for recognizing voice is collectable and within a range within which a noise amount due to vibration of the loudspeaker is allowable, when voice recognition is performed based on sound obtained through synthesis by the synthesis unit.
    Type: Grant
    Filed: March 18, 2019
    Date of Patent: July 19, 2022
    Assignees: TOYOTA JIDOSHA KABUSHIKI KAISHA, DENSO CORPORATION, YAMAHA CORPORATION
    Inventors: Keizoh Kawaguchi, Ichiro Shigetomi, Kunito Takahashi
  • Patent number: 11381906
    Abstract: A conference system is provided that includes a microphone array unit having a plurality of microphone capsules arranged in or on a board mountable on or in a ceiling of a conference room. The microphone array unit has a steerable beam and a maximum detection angle range. The conference system comprises a processing unit which is configured to receive the output signals of the microphone capsules and to steer the beam based on the received output signal of the microphone array unit. The processing unit is configured to control the microphone array to limit the detection angle range to exclude at least one predetermined exclusion sector in which a noise source is located.
    Type: Grant
    Filed: October 1, 2020
    Date of Patent: July 5, 2022
    Assignee: Sennheiser electronic GmbH & Co. KG
    Inventors: J. Douglas Rollow, IV, Lance Reichert, Daniel Voss
  • Patent number: 11356765
    Abstract: The present disclosure provides a microphone apparatus. The microphone apparatus may include a microphone and a vibration sensor. The microphone may be configured to receive a first signal including a voice signal and a first vibration signal. The vibration sensor may be configured to receive a second vibration signal. And the microphone and the vibration sensor are configured such that the first vibration signal may be offset with the second vibration signal.
    Type: Grant
    Filed: October 24, 2020
    Date of Patent: June 7, 2022
    Assignee: SHENZHEN SHOKZ CO., LTD.
    Inventors: Lei Zhang, Fengyun Liao, Xin Qi
  • Patent number: 11350205
    Abstract: The present disclosure provides a microphone apparatus. The microphone apparatus may include a microphone and a vibration sensor. The microphone may be configured to receive a first signal including a voice signal and a first vibration signal. The vibration sensor may be configured to receive a second vibration signal. And the microphone and the vibration sensor are configured such that the first vibration signal may be offset with the second vibration signal.
    Type: Grant
    Filed: February 8, 2021
    Date of Patent: May 31, 2022
    Assignee: SHENZHEN SHOKZ CO., LTD.
    Inventors: Lei Zhang, Fengyun Liao, Xin Qi
  • Patent number: 11349525
    Abstract: A double talk detection method, a double talk detection apparatus and an echo cancellation system are provided. The double talk detection method comprises: determining, according to an energy ratio between a far-end digital voice signal and a near-end digital voice signal, and a frequency coherence value between the near-end digital voice signal and the far-end digital voice signal, whether a near-end speaker's digital voice signal is present in the near-end digital voice signal. The double talk detection method avoids missing detection and false detection, improves the accuracy of double talk detection, cancels the echo in the near-end voice signal thoroughly when applied in the field of echo cancellation, and improves the communication experience of both talk parties.
    Type: Grant
    Filed: September 24, 2020
    Date of Patent: May 31, 2022
    Assignee: SHENZHEN GOODIX TECHNOLOGY CO., LTD.
    Inventors: Wenkai Han, Guoliang Li, Xinshan Wang, Hongjing Guo, Hu Zhu
  • Patent number: 11330383
    Abstract: A method and an apparatus detect an echo delay. If an electronic device plays an audio and a counter is in an on state, an Nth recording data block received after the counter is turned on is acquired, where the Nth recording data block is a currently received recording data block, and N is greater than 1. Matching is performed between the Nth recording data block and a reference data block that is in a buffer, where the recording data block and the reference data block are data blocks with target duration. After the counter is on, the buffer buffers an audio data block played by the electronic device. If the Nth recording data block matches the reference data block in the buffer, a counting value is acquired. The counter performs counting if the recording data block does not match the reference data block. An echo delay is determined based on the target duration and the counting value.
    Type: Grant
    Filed: September 29, 2020
    Date of Patent: May 10, 2022
    Assignee: Apollo Intelligent Connectivity (Beijing) Technology Co., Ltd.
    Inventors: Zhengbin Song, Danqing Yang, Junfei Bu
  • Patent number: 11315540
    Abstract: A system for reducing noise for a user includes a first detector configured to generate a first noise signal, wherein the first noise signal is a representation of a first noise that is transmitted to the user through a first sound pathway, and a second detector configured to generate a second noise signal, wherein the second noise signal indicates a second noise perceived by the user. The system also includes a processor configured to determine a noise correction signal based on the first noise signal and/or the second noise signal, and a speaker configured to generate a sound for reducing the noise based on the noise correction signal.
    Type: Grant
    Filed: January 6, 2021
    Date of Patent: April 26, 2022
    Assignee: SHENZHEN SHOKZ CO., LTD.
    Inventors: Chengqian Zhang, Fengyun Liao, Xin Qi
  • Patent number: 11310609
    Abstract: A data system for handling user data for a hearing aid user and at least one hearing aid (10, 11) having a memory for storing personal settings for alleviating a hearing loss for the hearing aid user includes a remote server (25) accessible by means of an Internet enabled computer device. A personal communication device (13) is Internet enabled and is able to act as a gateway to the Internet for the at least one hearing aid. The user may create a user account on the remote server, enter personal information into the user account and store the entered personal information on the remote server, and enter the gateway information to the user account. The personal communication device and the at least one hearing aid are provided with respective transceivers for establishing a wireless connection under guidance of said application software. The least one hearing aid uploads the personal settings for alleviating the hearing loss via the gateway to the remote server for storing in the user account.
    Type: Grant
    Filed: November 27, 2019
    Date of Patent: April 19, 2022
    Assignee: Widex A/S
    Inventors: Soren Erik Westermann, Svend Vitting Andersen, Anders Westergaard, Niels Erik Boelskift Maretti
  • Patent number: 11303991
    Abstract: A beamformer system includes an in-ear audio device, such as an earbud, that has three microphones. Two microphones may be disposed on an external face of the audio-device; one microphone may be disposed in or near the ear canal of a user. Data from two microphones is phase-adjusted and combined; a reference signal is generated by phase-adjusting and removing data from the two microphones from data from a third microphone. The combined data is filtered using the reference signal to remove any residual echo, and the resulting data may be used for communications, speech processing, or other uses.
    Type: Grant
    Filed: November 18, 2019
    Date of Patent: April 12, 2022
    Assignee: Amazon Technologies, Inc.
    Inventor: Ludger Solbach
  • Patent number: 11295752
    Abstract: A method and a device of sustainably updating a coefficient vector of a finite impulse response FIRfilter. The method includes obtaining (21) a time-varying regularization factor used for iteratively updating the coefficient vector of the FIR filter in a case that the coefficient vector of the FIR filter is used for processing a preset signal; updating (22) the coefficient vector of the FIR filter according to the time-varying regularization factor.
    Type: Grant
    Filed: August 21, 2018
    Date of Patent: April 5, 2022
    Assignee: CHINA ACADEMY OF TELECOMMUNICATIONS TECHNOLOGY
    Inventor: Min Liang
  • Patent number: 11226396
    Abstract: Methods, apparatus, systems and articles of manufacture are disclosed to improve detection of audio signatures. An example apparatus includes a TDOA determiner to determine a first time difference of arrival for a first audio sensor of a meter and a second audio sensor of the meter, and a second time difference of arrival for the first audio sensor and a third audio sensor of the meter, a TDOA matcher to determine a match by comparing the first time difference of arrival to a first virtual source time difference of arrival and a second virtual source time difference of arrival, responsive to determining that the first time difference of arrival matches the first virtual source time difference of arrival, identify a first virtual source location as the location of a media presentation device, and remove an audio recording of the second audio sensor to reduce a computational burden on the processor.
    Type: Grant
    Filed: June 27, 2019
    Date of Patent: January 18, 2022
    Assignee: Gracenote, Inc.
    Inventor: Zafar Rafii
  • Patent number: 11197093
    Abstract: An echo suppression device includes an echo canceller which suppresses a linear echo signal from an input signal acquired by a microphone; a nonlinear echo estimation unit which, by using a nonlinear echo model indicative of a relationship between at least one of a call reception signal to be output to a speaker and the input signal, and a nonlinear echo signal, estimates the nonlinear echo signal included in the input signal from at least one of the call reception signal and the input signal; a nonlinear echo suppression unit which, by using the estimated nonlinear echo signal, suppresses the nonlinear echo signal from an output signal of the echo canceller; and an echo suppressor which suppresses a residual linear echo signal not suppressed by the echo canceller from an output signal of the nonlinear echo suppression unit.
    Type: Grant
    Filed: November 6, 2020
    Date of Patent: December 7, 2021
    Assignee: PANASONIC INTELLECTUAL PROPERTY CORPORATION OF AMERICA
    Inventors: Yuki Terashima, Shinichi Yuzuriha
  • Patent number: 11189297
    Abstract: A multi-channel acoustic echo cancellation (AEC) system that includes a residual echo suppressor (RES) that dynamically controls an amount of attenuation to reduce distortion of local speech during double-talk conditions. The RES determines when double-talk conditions are present based on an echo return loss enhancement (ERLE) value. When the ERLE value is above a first threshold value but below a second threshold value, the RES reduces an amount of attenuation applied while generating an RES mask to pass local speech without distortion. When the ERLE value is below the first threshold value or above the second threshold value, the RES applies full attenuation while generating the RES mask in order to suppress a residual echo signal. To further improve RES processing, the RES may apply smoothing across time, smoothing across frequencies, or apply extra echo suppression processing to further attenuate the residual echo signal.
    Type: Grant
    Filed: June 8, 2020
    Date of Patent: November 30, 2021
    Assignee: Amazon Technologies, Inc.
    Inventors: Carlos Renato Nakagawa, Carlo Murgia, Berkant Tacer
  • Patent number: 11133019
    Abstract: A signal processor for providing one or more processed audio signals on the basis of one or more input audio signals is configured to estimate coefficients of an autoregressive reverberation model using the input audio signals and the delayed noise-reduced reverberant signals obtained using a noise reduction. The signal processor is configured to provide noise-reduced reverberant signals using the input audio signals and the estimated coefficients of the autoregressive reverberation model. The signal processor is configured to derive noise-reduced and reverberation-reduced output signals using the noise-reduced reverberant signals and the estimated coefficients of the autoregressive reverberation model. A method and a computer program include a similar functionality.
    Type: Grant
    Filed: March 19, 2020
    Date of Patent: September 28, 2021
    Assignee: Fraunhofer-Gesellschaft zur Foerderung der angewandten Forschung e.V.
    Inventors: Sebastian Braun, Emanuel Habets
  • Patent number: 11134331
    Abstract: An audio playback system may be adaptable to various situations for improved user experience and audio playback quality. For example, earbuds may be both worn by a same user, worn by two different users within audio range of one another, worn by two different users outside of audio range of one another, etc. A second earbud with a secondary microphone operates in a second mode in which it captures audio and encodes it for transmission to the first earbud having a primary microphone. The first earbud with the primary microphone operates in a first mode in which it mixes the audio received from the second earbud with audio received through its own microphone for playback. The first earbud in the first mode may also delay its own microphone stream to compensate for wireless transmission delay and correlate the two audio streams to improve audio quality in case there are sounds that can be picked up by both microphones.
    Type: Grant
    Filed: November 14, 2019
    Date of Patent: September 28, 2021
    Assignee: Google LLC
    Inventors: Vitali Lovich, Jeffrey Kuramoto
  • Patent number: 11122366
    Abstract: Regions of the audio frequency spectrum that are most susceptible to howling are identified or determined. When detected, these approaches shape the frequency spectrum of the audio going to loudspeakers so that only those howling frequencies are suppressed.
    Type: Grant
    Filed: February 5, 2020
    Date of Patent: September 14, 2021
    Assignee: Continental Automotive Systems, Inc.
    Inventor: Mike Reuter
  • Patent number: 11100942
    Abstract: Systems and methods are described for compensating for inaccurate echo prediction in audio systems. A signal may be received at a microphone of an audio system, the signal including audio rendered using a spatial audio renderer across a multi-channel audio output. A signal may be received from the spatial audio renderer that indicates a change in rendering of audio. The audio system may then determine if there is echo power within the received signal greater than an expected echo power. After the signal from the spatial audio renderer has been received, the echo suppression applied to the received signal may be modified in response to a determination that the echo power is greater than the expected echo power, the echo suppression attenuating pre-selected frequency bands of the received signal.
    Type: Grant
    Filed: July 13, 2018
    Date of Patent: August 24, 2021
    Assignee: Dolby Laboratories Licensing Corporation
    Inventors: Glenn N. Dickins, Tet Fei Yap, Guodong Li, Paul Holmberg
  • Patent number: 11089420
    Abstract: Speech processing system includes a first sound receiving device, a second sound receiving device, a main controller, and an audio processor. First and second sound receiving devices are configured to generate a main voice signal or a secondary voice signal. A first sensing device in first sound receiving device and a second sensing device in second sound receiving device are configured to output a first sensing signal or a second sensing signal based on a sensing result. Main controller controls first sound receiving device to generate main voice signal and controls second sound receiving device to generate secondary voice signal when receiving first sensing signal. Main controller controls second sound receiving device to generate main voice signal and controls first sound receiving device to generate secondary voice signal when receiving second sensing signal. Audio processor is configured to process main and secondary voice signals into an output voice signal.
    Type: Grant
    Filed: August 5, 2019
    Date of Patent: August 10, 2021
    Assignee: Chicony Electronics Co., Ltd.
    Inventor: Chih-Hsiang Hsu
  • Patent number: 10999692
    Abstract: The audio device according to the present disclosure may include a mixer that adjusts the number of channels of an inputted audio signal based on the number of speakers connected, a transmitter that transmits a test audio signal for speaker setup, to at least one speaker among the plurality of speakers, a feedback receiver that receives a signal of the outputted audio, a controller that determines an output time difference between the plurality of speakers, based on the signal of the outputted audio, and a post-processor that adds an output delay signal to the audio signal of at least one channel of a multi-channel audio signal provided to the plurality of speakers so as to synchronize the outputs of the plurality of speakers, based on the determined output time difference.
    Type: Grant
    Filed: January 8, 2020
    Date of Patent: May 4, 2021
    Assignee: LG ELECTRONICS INC.
    Inventors: Tae Young Kim, Tae Jin Park, Si Jin Kim, Eun Jung Lee, Soon Hyung Hwang, Hyo Rim Kim, Min Jae Kim, Hyo Sung Kim
  • Patent number: 10986444
    Abstract: Techniques for simulating a microphone array and generating synthetic audio data to analyze the microphone array geometry. This reduces the development cost of new microphone arrays by enabling an evaluation of performance metrics (False Rejection Rate (FRR), Word Error Rate (WER), etc.) without building device hardware or collecting data. To generate the synthetic audio data, the system performs acoustic modeling to determine a room impulse response associated with a prototype device (e.g., potential microphone array) in a room. The acoustic modeling is based on two parameters—a device response (information about acoustics and geometry of the prototype device) and a room response (information about acoustics and geometry of the room). The device response can be simulated based on the microphone array geometry, and the room response can be determined using a specialized microphone and a plane wave decomposition algorithm.
    Type: Grant
    Filed: February 24, 2020
    Date of Patent: April 20, 2021
    Assignee: Amazon Technologies, Inc.
    Inventors: Mohamed Mansour, Guangdong Pan
  • Patent number: 10945071
    Abstract: A method for sound collection includes: converting time domain signals with a number of M collected by devices for sound collecting with a number of M into original frequency domain signals with a number of M; performing beam-forming on the M original frequency domain signals at each of preset grid points, to obtain beam-forming frequency domain signals with a number of N in one-to-one correspondence with the preset grid points; determining an average amplitude of frequency components with a number of N corresponding to each of frequency points with a number of K based on the beam-forming frequency domain signals with a number of N, and synthesizing a synthesized frequency domain signal including the frequency points and having an average amplitude as an amplitude at the each of the frequency points with a number of K; and converting the synthesized frequency domain signal into a synthesized time domain signal.
    Type: Grant
    Filed: November 28, 2019
    Date of Patent: March 9, 2021
    Assignee: BEIJING XIAOMI MOBILE SOFTWARE CO., LTD.
    Inventors: Taochen Long, Haining Hou
  • Patent number: 10924614
    Abstract: A speech signal processing method is performed at a terminal device, including: obtaining a recorded signal and a to-be-output speech signal, the recorded signal including a noise signal and an echo signal; calculating a loop transfer function according to the recorded signal and the speech signal; calculating a power spectrum of the echo signal and a power spectrum of the noise signal according to the recorded signal, the speech signal, and the loop transfer function; calculating a frequency weighted coefficient according to the two power spectra of the echo signal and the noise signal; adjusting a frequency amplitude of the speech signal based on the frequency weighted coefficient; and outputting the adjusted speech signal to a speaker electrically coupled to the terminal device. As such, the frequency amplitude of the speech signal is automatically adjusted according to the relative frequency distribution of a noise signal and the speech signal.
    Type: Grant
    Filed: January 28, 2020
    Date of Patent: February 16, 2021
    Assignee: TENCENT TECHNOLOGY (SHENZHEN) COMPANY LIMITED
    Inventor: Haolei Yuan
  • Patent number: 10923138
    Abstract: A sound collection apparatus for far-field voice includes a multi-channel analog sound receiver configured to convert an obtained sound signal into an electrical signal; a first analog-to-digital converter coupled to the multi-channel analog sound receiver and configured to convert the electrical signal into a digital signal; and an interface controller coupled to the analog-to-digital converter and configured to transmit the digital signal to a control device via a preset interface. Using the above solutions, the technical problems of high hardware cost and unguaranteed performance of existing sound collection devices can be solved, and the technical effects of effectively reducing the hardware cost and the difficulties of development are achieved.
    Type: Grant
    Filed: November 9, 2018
    Date of Patent: February 16, 2021
    Assignee: ALIBABA GROUP HOLDING LIMITED
    Inventors: Zhihui Yang, Qiang Fu, Zhijie Yan
  • Patent number: 10916239
    Abstract: Provided is a method for beamforming by using maximum likelihood estimation in a speech recognition apparatus, including: (a) receiving an input signal (Xn,k) at a time frame n and a frequency k where noise is mixed: (b) determining a probability density function for a target signal (Yn,k) obtained by removing the noise from the input signal satisfies a complex generalized Guassian distribution or a complex gamma distribution where an average value is zero in a time-frequency domain; (c) estimating a variance (?n,k) of the target signal so as to maximize log likelihood for the probability density function; (d) estimating a filter (wk) maximizing a cost function so as to maximize the log likelihood for the probability density function; and (e) repeatedly performing the estimation of the steps (c) and (d) until the filter (wk) coverages, and finally acquiring a final filter (wk).
    Type: Grant
    Filed: December 18, 2018
    Date of Patent: February 9, 2021
    Assignee: INDUSTRY-UNIVERSITY COOPERATION FOUNDATION SOGANG UNIVERSITY
    Inventor: Hyung Min Park
  • Patent number: 10911887
    Abstract: This sound system includes a first sound device including a plurality of first speakers and a plurality of first microphones, and a second sound device including a plurality of second speakers and a plurality of second microphones. The first sound device includes: a device detector that detects the second sound device connected to a network; a test audio transmitter/receiver that acquires, via the plurality of first microphones, test audio output from the plurality of second speakers; a speaker sound source localizer that performs sound source localization with respect to the plurality of second speakers based on the test audio acquired by the test audio transmitter/receiver; a position calculator that calculates positions of the plurality of second speakers relative to the first sound device; and a position notifier that notifies the second sound device of position information representing the positions calculated by the position calculator.
    Type: Grant
    Filed: August 23, 2019
    Date of Patent: February 2, 2021
    Assignee: SHARP KABUSHIKI KAISHA
    Inventor: Atsushi Ohnuma
  • Patent number: 10891932
    Abstract: A playback device is configured to receive, via a network interface, a source stream of audio including first and second channel streams of audio, and to produce, via respective first and second speaker drivers, a first channel audio output and a second channel audio output. The playback device is also configured to receive, via one or more microphones, a captured stream of audio including first and second portions corresponding to the respective first and second channel audio outputs. The playback device is also configured to combine at least the first channel stream of audio and the second channel stream of audio into a compound audio signal and perform acoustic echo cancellation on the compound audio signal and thereby produce an acoustic echo cancellation output, then to apply the acoustic echo cancellation output to the captured stream of audio and thereby increase a signal-to noise ratio of the captured stream of audio.
    Type: Grant
    Filed: October 10, 2019
    Date of Patent: January 12, 2021
    Assignee: Sonos, Inc.
    Inventors: Saeed Bagheri Sereshki, Romi Kadri
  • Patent number: 10887692
    Abstract: A microphone array device including microphone capsules and at least one processing unit configured to receive output signals of the microphone capsules, dynamically steer an audio beam based on the received output signal of the microphone capsules, and generate and provide an audio output signal based on the received output signal of the microphone capsules. The processing unit is configured to operate in a dynamic beam mode where at least one focused audio beam is formed that points towards a detected audio source and in a default beam mode where a broader audio beam is formed that covers substantially a default detection area. The microphone array may be incorporated into a conference system.
    Type: Grant
    Filed: July 5, 2019
    Date of Patent: January 5, 2021
    Assignee: Sennheiser electronic GmbH & Co. KG
    Inventors: Eugen Rasumow, Sebastian Rieck, Fabian Logemann, Jens Werner
  • Patent number: 10880427
    Abstract: Method, apparatus, and program code embodied in computer-readable media, for providing enhanced echo suppression in a conferencing system having at least one microphone and at least one speaker. At least one microphone input signal is received, and at least one speaker input signal is provided. At least one processor has at least one primary echo-suppressor and at least one secondary echo-suppressor. The at least one primary echo-suppressor receives (i) the microphone input signal(s) and (ii) the speaker input signal(s). The at least one primary echo-suppressor provides at least one echo-suppressed microphone signal. The at least one secondary echo-suppressor receives the at least one echo-suppressed microphone signal and provides an output signal. The at least one processor provides the at least one echo-suppressed microphone signal to the at least one secondary echo-suppressor without providing the at least one speaker input signal directly to the at least one secondary echo-suppressor.
    Type: Grant
    Filed: May 9, 2019
    Date of Patent: December 29, 2020
    Inventors: Richard Dale Ferguson, Linshan Li, Mahdi Javer, Nicholas Alexander Norrie
  • Patent number: 10873805
    Abstract: A sound processing apparatus and a sound processing method thereof are provided. The following steps are included. Multiple first sound signals corresponding to multiple sound reception sources are obtained. A sound source position of a sound source relative to the sound reception sources is determined. A relationship among multiple sound receiving directions corresponding to the sound reception sources is determined according to the sound source position. The sound receiving directions relate to directionality of the sound reception sources. A second sound signal is outputted from the first sound signals based on the relationship among the sound receiving directions. Accordingly, an optimal sound receiving direction corresponding to the sound source can be adjusted automatically, so as to improve sound quality.
    Type: Grant
    Filed: December 19, 2018
    Date of Patent: December 22, 2020
    Assignee: Wistron Corporation
    Inventors: Tzu-Peng Chang, Chuan-Yen Kao
  • Patent number: 10867617
    Abstract: This disclosure describes, in part, techniques for processing audio data. For instance, an electronic device may include an automatic gain controller (AGC) that determines AGC gains for amplifying or attenuating an audio data. To determine the AGC gains, the AGC uses information from a residual echo suppressor (RES) and/or a noise reductor (NR). The information may indicate RES gains applied to the audio data by the RES and/or NR gains applied to the audio data by the NR. In some instances, to determine the AGC gain, the AGC determines time-constant parameter(s) using the information. The AGC then uses the time-constant parameter(s) to determine an input signal level for the audio data and/or the AGC gain. In some instances, to determine the AGC gain, the AGC operates in an attack mode or a release mode based on the information.
    Type: Grant
    Filed: December 10, 2018
    Date of Patent: December 15, 2020
    Assignee: Amazon Technologies, Inc.
    Inventors: Carlos Renato Nakagawa, Carlo Murgia, Wai Chung Chu, Kuan-Chieh Yen
  • Patent number: 10869140
    Abstract: A hearing prosthesis arrangement is described for a hearing assisted patient. A microphone senses an acoustic environment around the hearing assisted patient and generates a corresponding microphone output signal. An audio signal processor processes the microphone output signal and produces a corresponding prosthesis stimulation signal to the patient for audio perception. The audio signal processor includes a dereverberation process that measures a dedicated reverberation reference signal produced in the acoustic environment to determine reverberation characteristics of the acoustic environment, and reduces reverberation effects in the hearing prosthesis stimulation signal based on the reverberation characteristics.
    Type: Grant
    Filed: September 25, 2015
    Date of Patent: December 15, 2020
    Assignee: MED-EL Elektromedizinische Geraete GmbH
    Inventors: Cornelia Falch, Ernst Aschbacher, Florian Frühauf, Thomas Schwarzenbeck
  • Patent number: 10856079
    Abstract: A system and method for processing and enhancing utility of a sound mask noise signal, including generating, by a signal processor, the sound mask noise signal by modulating a noise signal with embedded additional information, outputting, by a plurality of audio speakers, sound signals comprising the sound mask noise signal with the embedded additional information; and receiving, by one or more microphones, the outputted sound signals comprising the sound mask noise signal, wherein an impulse response between each audio speaker and each microphone is measured in real time based on the embedded additional information.
    Type: Grant
    Filed: December 2, 2019
    Date of Patent: December 1, 2020
    Inventor: Grant Howard McGibney
  • Patent number: 10847171
    Abstract: Disclosed methods and systems are directed to determining a best microphone pair and segmenting sound signals. The methods and systems may include receiving a collection of sound signals comprising speech from one or more audio sources (e.g., meeting participants) and/or background noise. The methods and systems may include calculating a TDOA and determining, based on the TDOA and via robust statistics, the best pair of microphones. The methods and systems may also include segmenting sound signals from multiple sources.
    Type: Grant
    Filed: September 24, 2019
    Date of Patent: November 24, 2020
    Assignee: Nuance Communications, Inc.
    Inventors: Pablo Peso Parada, Dushyant Sharma, Patrick Naylor
  • Patent number: 10832650
    Abstract: A speakerphone system is provide, comprising: at least one mechanical vibration sensor (MVS) adapted to convert mechanical vibrations in a speakerphone enclosure (enclosure) to a mechanical vibration error signal, and output the same as an MVS output signal; at least one microphone (mic) adapted to convert an input sound acoustic signal into an input sound electrical signal and to output the same as a mic output signal; and circuitry adapted to subtract the MVS output signal from the mic output signal and output the resultant signal as a speakerphone output signal.
    Type: Grant
    Filed: September 16, 2019
    Date of Patent: November 10, 2020
    Assignee: Crestron Electronics, Inc.
    Inventors: Alexander Marra, Mark Hrozenchik
  • Patent number: 10834512
    Abstract: A speakerphone calibration system (system) is provided herein, comprising: a calibration unit adapted to generate at least one test signal, and further adapted to determine at least one calibration factor in response to at least one test signal, and wherein a first calibration factor characterizes a speakerphone system under test in regard to mechanical vibrations generated in the speakerphone system under test, the mechanical vibrations caused by a first test signal.
    Type: Grant
    Filed: September 16, 2019
    Date of Patent: November 10, 2020
    Assignee: Crestron Electronics, Inc.
    Inventors: Alexander Marra, Mark Hrozenchik
  • Patent number: 10834513
    Abstract: A method for calibrating a speakerphone system under test is provided herein, comprising: connecting a calibration unit to the speakerphone system under test (SSUT) via a communications interface, and wherein the SSUT further comprises a loudspeaker, microphone (mic), and mechanical vibration sensor (MVS); generating a first test signal by the calibration unit and transmitting the same to the SSUT; generating a first set of mechanical vibrations in response to the first test signal in the SSUT; and determining a calibration factor in regard to the first test signal and first set of mechanical vibrations by the calibration unit.
    Type: Grant
    Filed: September 16, 2019
    Date of Patent: November 10, 2020
    Assignee: Crestron Electronics, Inc.
    Inventors: Alexander Marra, Mark Hrozenchik
  • Patent number: 10820128
    Abstract: Embodiments herein enable fast and easy interconnectivity among multimedia accessories including mobile devices and other devices. There is only limited space on mobile devices yet there are numerous input connectors. The standard TRRS audio jack is one such input that has and remains common, primarily because it is the accepted standard for audio input; namely, headphones and earpieces for listening purposes. Embodiments herein describe an intelligent switch to that audio jack that permits for additional backward and forward compatibility. It transparently allows a user to insert analog or digital audio devices, such as earphones, without the need to manually reconfigure device settings. The device herein automatically converts between input connector types using the same input convention present on their existing mobile devices. Other embodiments are disclosed.
    Type: Grant
    Filed: September 23, 2019
    Date of Patent: October 27, 2020
    Assignee: Staton Techiya, LLC
    Inventors: Koen Weijand, Steven W. Goldstein
  • Patent number: 10811029
    Abstract: A system configured to perform cascade echo cancellation processing to improve a performance when reference signals are asymmetric (e.g., dominant reference signal(s) overshadow weak reference signal(s)). The system may perform cascade echo cancellation processing to separately adapt filter coefficients between the dominant reference signal(s) and the weak reference signal(s). For example, the system may use a dominant reference signal to process a microphone audio signal and generate a residual audio signal, using the residual audio signal to adapt first filter coefficient values corresponding to the dominant reference signal. Separately, the system may use a weak reference signal to process the residual audio signal and generate an output audio signal, using the output audio signal to adapt second filter coefficient values corresponding to the weak reference signal.
    Type: Grant
    Filed: October 31, 2019
    Date of Patent: October 20, 2020
    Assignee: Amazon Technologies, Inc.
    Inventors: Mohamed Mansour, Shobha Devi Kuruba Buchannagari
  • Patent number: 10812902
    Abstract: A method and system for real-time auralization is described in which room sounds are reverberated and presented over loudspeakers, thereby augmenting the acoustics of the space. Room microphones are used to capture room sound sources, with their outputs processed in a canceler to remove the synthetic reverberation also present in the room. Doing so gives precise control over the auralization while suppressing feedback. It also allows freedom of movement and creates a more natural acoustic environment for performers or participants in music, theater, gaming, home entertainment, and virtual reality applications. Canceler design methods are described, including techniques for handling varying loudspeaker-microphone transfer functions such as would be present in the context of a performance or installation.
    Type: Grant
    Filed: June 14, 2019
    Date of Patent: October 20, 2020
    Assignee: The Board of Trustees of the Leland Stanford Junior University
    Inventors: Jonathan S. Abel, Eoin F. Callery, Elliot Kermit Canfield-Dafilou
  • Patent number: 10805575
    Abstract: A non-transitory computer-readable storage medium may include instructions stored thereon. When executed by at least one processor, the instructions may be configured to cause a computing system to determine that a video system is aiming at a single speaker of a plurality of people, receive audio signals from a plurality of microphones, the received audio signals including audio signals generated by the single speaker, based on determining that the video system is aiming at the single speaker, transmit a monophonic signal, the monophonic signal being based on the received audio signals, determine that the video system is not aiming at the single speaker, and based on the determining that the video system is not aiming at the single speaker, transmit a stereophonic signal, the stereophonic signal being based on the received audio signals.
    Type: Grant
    Filed: June 4, 2019
    Date of Patent: October 13, 2020
    Assignee: Google LLC
    Inventors: Tore Rudberg, Christian Schuldt
  • Patent number: 10798483
    Abstract: The disclosure relates to an audio signal processing method, device, and computer-readable medium. The method is applied to an electronic equipment that includes multiple audio acquisition devices with distances between the multiple audio acquisition devices meeting a preset distance condition.
    Type: Grant
    Filed: May 29, 2019
    Date of Patent: October 6, 2020
    Assignee: BEIJING XIAOMI MOBILE SOFTWARE CO., LTD.
    Inventors: Jiongliang Li, Si Cheng
  • Patent number: 10785566
    Abstract: A method and a device for processing an audio signal in a vehicle are provided. The method includes: obtaining an audio signal by a microphone array; performing echo cancellation on the obtained audio signal, to obtain a first processed signal; and performing beamforming on the first processed signal according to sound zones in which microphones of the microphone array are located, to obtain a second processed signal, wherein the vehicle includes at least two sound zones, and each microphone of the microphone array is located in at least one sound zone. With the beamforming, the requirements for isolation degree between different sound zones is not high, and the sound source of the audio signal can be accurately determined.
    Type: Grant
    Filed: August 28, 2019
    Date of Patent: September 22, 2020
    Assignee: Baidu Online Network Technology (Beijing) Co., Ltd.
    Inventor: Lei Geng
  • Patent number: 10777214
    Abstract: A system that performs wall detection, range estimation, and/or corner detection to determine a position of a device relative to acoustically reflective surfaces. The device generates output audio using loudspeaker(s), generates microphone audio data using a microphone array, and generates impulse response data for each of the microphones. The device may generate the impulse response data using an acoustic echo cancellation (AEC) component or multi-channel AEC (MC-AEC). The device may detect a peak in the impulse response data and determine a distance to a reflective surface based on the peak. Based on a number of reflected surfaces detected by the device, the device may classify a position of the device within the room, such as whether the device is in a corner, along one wall, or in an open area. By knowing the position relative to the room surfaces, the device may improve sound equalization and other processing.
    Type: Grant
    Filed: June 28, 2019
    Date of Patent: September 15, 2020
    Assignee: AMAZON TECHNOLOGIES, INC.
    Inventors: Guangji Shi, Trausti Thor Kristjansson, Jan Aage Abildgaard Pedersen, Philip Ryan Hilmes
  • Patent number: 10764703
    Abstract: A device for room geometry analysis comprising: a plurality of segments (101-106) built of acoustic metamaterials, each segment (?, ?) acting as a waveguide with a unique transfer function (B(?, ?, ?)); and a processor configured to calculate delays (?(?, ?)) and respective directions (c(?, ?)) of mirror sound sources (721-725) by decomposing a sound signal (?) obtained from a microphone (110) based on the transfer functions (B(?, ?, ?)) of the segments (101-106) and based on a calibration signal (tsp(t)) emitted by a speaker (420).
    Type: Grant
    Filed: March 25, 2019
    Date of Patent: September 1, 2020
    Assignee: SONY CORPORATION
    Inventors: Franck Giron, Fabien Cardinaux, Thomas Kemp, Stefan Uhlich, Marc Ferras Font, Andreas Schwager, Patrick Putzolu
  • Patent number: 10726857
    Abstract: Audio signal processing techniques are described which are employed within a circuit of a speech dereverberation system. The amount of data or number of samples input to a reverberation coefficient determination unit is determined, taking into account information about the background noise in the acoustic space and information about energy of reverberant sound in the acoustic space.
    Type: Grant
    Filed: February 23, 2018
    Date of Patent: July 28, 2020
    Assignee: Cirrus Logic, Inc.
    Inventor: Tom Birchall
  • Patent number: 10720173
    Abstract: Audio systems and methods are provided that receive a playback signal and produce an acoustic signal based upon the playback signal, and include microphone signal(s) for capturing and processing user voice signals. An echo reference signal is based upon the playback signal, and an echo canceler reduces echo components from the microphone signal(s). Functionality of the echo canceler is modified, such as by freezing an adaptive filter, in response to a non-linear condition in the audio playback, or a likelihood of such a non-linear condition.
    Type: Grant
    Filed: February 21, 2018
    Date of Patent: July 21, 2020
    Assignee: BOSE CORPORATION
    Inventors: Eric J. Freeman, Joseph Gaalaas
  • Patent number: 10679617
    Abstract: A real-time audio signal processing system includes an audio signal processor configured to process audio signals using a modified generalized eigenvalue (GEV) beamforming technique to generate an enhanced target audio output signal. The digital signal processor includes a sub-band decomposition circuitry configured to decompose the audio signal into sub-band frames in the frequency domain and a target activity detector configured to detect whether a target audio is present in the sub-band frames. Based on information related to the sub-band frames and the determination of whether the target audio is present in the sub-band frames, the digital signal processor is configured to use the modified GEV technique to estimate the relative transfer function (RTF) of the target audio source, and generate a filter based on the estimated RTF. The filter may then be applied to the audio signals to generate the enhanced audio output signal.
    Type: Grant
    Filed: December 6, 2017
    Date of Patent: June 9, 2020
    Assignee: SYNAPTICS INCORPORATED
    Inventors: Frederic Philippe Denis Mustiere, Francesco Nesta
  • Patent number: 10667157
    Abstract: A source device can transmit initial streaming content to a playback device (e.g., wireless ear buds) using first settings and measure playback performance of the content at a plurality of times. The measured performance values can relate to a quality of communication of the initial streaming content between the source device and the playback device, e.g., relating to packet loss, retransmission rates and patterns, fluctuations in a playback (jitter) buffer, and/or other values. The measured performance values can be used to determine one or more second settings to be used for a playback of subsequent streaming content between the source device and the playback device. In this manner, each source device can account for variations in communication behavior specific to a user (e.g., due to differences in body type as electromagnetic waves travel through the body when a source device is in a pocket).
    Type: Grant
    Filed: August 21, 2018
    Date of Patent: May 26, 2020
    Assignee: Apple Inc.
    Inventors: Ahmad Rahmati, Natalia A. Fornshell, Aarti Kumar
  • Patent number: 10595126
    Abstract: An apparatus of reducing feedback noise in an acoustic system, the apparatus comprising: a first input for receiving a first signal derived from a first microphone associated with a first channel, the first signal comprising a first set of frequency sub-bands; a second input for receiving a second signal derived from a second microphone associated with a second channel, the second signal comprising second set of frequency sub-bands, the first and second sets of frequency sub-bands having matching frequency ranges, each frequency sub-band of the first and second sets of frequency sub-bands having a frequency of greater than a threshold frequency; and one or more processors configured to: determining feedback at a first speaker associated with the first channel; and responsive to determining feedback, mix each of the first set of frequency sub-bands with a corresponding one of the second set of frequency sub-bands to generate a mixed output signal comprising a mixed set of frequency sub-bands; wherein the mixin
    Type: Grant
    Filed: December 7, 2018
    Date of Patent: March 17, 2020
    Assignee: Cirrus Logic, Inc.
    Inventors: Henry Chen, Tom Harvey, Brenton Steele