Dereverberators Patents (Class 381/66)
  • 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: 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: 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: 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
  • Patent number: 10586551
    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: August 30, 2017
    Date of Patent: March 10, 2020
    Assignee: TENCENT TECHNOLOGY (SHENZHEN) COMPANY LIMITED
    Inventor: Haolei Yuan
  • Patent number: 10582299
    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: December 11, 2018
    Date of Patent: March 3, 2020
    Assignee: Amazon Technologies, Inc.
    Inventors: Mohamed Mansour, Guangdong Pan
  • Patent number: 10573301
    Abstract: Techniques are provided for pre-processing enhancement of a speech signal. A methodology implementing the techniques according to an embodiment includes performing de-reverberation processing on signals received from an array of microphones, the signals comprising speech and noise. The method also includes generating time-frequency masks (TFMs) for each of the signals. The TFMs indicate the probability that a time-frequency component of the signal associated with that TFM element includes speech. The TFM generation is based on application of a recurrent neural network to the signals. The method further includes generating steering vectors based on speech covariance matrices and noise covariance matrices. The TFMs are employed to filter speech components of the signals, for calculation of the speech covariance, and noise components of the signals for calculation of the noise covariance.
    Type: Grant
    Filed: June 29, 2018
    Date of Patent: February 25, 2020
    Assignee: Intel Corporation
    Inventors: Adam Kupryjanow, Kuba Lopatka
  • Patent number: 10575110
    Abstract: An improved method and system for varying an amount of mechanical coupling in a speakerphone is disclosed. Solutions and implementations provided vary the amount of mechanical coupling between one or more speakers and one or more microphones of the speakerphone to generate high-quality sounds. Implementations include receiving an input signal, sending a copy of the input signal to a first speaker, performing a signal transformation on the input signal to produce a transformed input signal, and transmitting the transformed input signal to a second speaker, where the first speaker generates a first vibration force in response to the input signal, and the second speaker generates a second vibration force in response to the transformed input signal, the second vibration force being in an opposite direction to that of the first vibration force and offsetting at least part of the first vibration force.
    Type: Grant
    Filed: March 26, 2019
    Date of Patent: February 25, 2020
    Assignee: MICROSOFT TECHNOLOGY LICENSING, LLC
    Inventors: Antti Pekka Kelloniemi, Ross Garrett Cutler, Sailaja Malladi, Tommi Antero Raussi
  • Patent number: 10575116
    Abstract: An audio system provides for spatial enhancement, crosstalk processing, and crosstalk compensation of an input audio signal. The crosstalk compensation compensates for spectral defects caused by the application of the crosstalk processing to a spatially enhanced signal. The crosstalk compensation may be performed prior to the crosstalk processing, after the crosstalk processing, or in parallel with the crosstalk processing. The crosstalk compensation includes applying filters to the mid and side components of the left and right input channels to compensate for spectral defects from crosstalk processing of the audio signal. The crosstalk processing may include crosstalk simulation or crosstalk cancellation. In some embodiments, the crosstalk compensation may be integrated with a subband spatial processing that spatially enhances the audio signal.
    Type: Grant
    Filed: June 20, 2018
    Date of Patent: February 25, 2020
    Assignee: LG Display Co., Ltd.
    Inventor: Zachary Seldess
  • Patent number: 10559317
    Abstract: An apparatus includes a beamformer, an echo suppression control unit, and a residual echo cancellation unit. The beamformer is configured to pass desired portions of audio signals and to suppress undesired portions of the audio signals. The beamformer includes a speech blocking filter to prevent suppression of near-end desired talker speech in the audio signals and an echo suppression filter to suppress echo in the audio signals. An echo suppression control unit is coupled to the beamformer and receives signals and determines whether to dynamically adapt the speech blocking filter or to dynamically adapt the echo suppression filter. The speech blocking filter remains unchanged during dynamic adaptation of the echo suppression filter, and the echo suppression filter remains unchanged during dynamic adaptation of the speech blocking filter. The residual echo cancellation unit is coupled to the beamformer and receives output audio signals from the beamformer and further suppresses residual echo.
    Type: Grant
    Filed: June 29, 2018
    Date of Patent: February 11, 2020
    Assignee: Cirrus Logic International Semiconductor Ltd.
    Inventors: Justin L. Allen, Narayan Kovvali
  • Patent number: 10490204
    Abstract: A system, article, and method of acoustic dereverberation factoring the actual non-ideal acoustic environment.
    Type: Grant
    Filed: November 20, 2018
    Date of Patent: November 26, 2019
    Assignee: Intel IP Corporation
    Inventors: Shmuel Markovich Golan, Alejandro Cohen
  • Patent number: 10482894
    Abstract: A dereverberation device includes an input instantaneous value calculation unit configured to calculate an input instantaneous value based on an input signal; a reverberation estimation unit configured to calculate a moving average of the input instantaneous value as a reverberation component; a gain calculation unit configured to calculate, with the input instantaneous value and the reverberation component, a first gain as a basic gain for the input signal; a gain suppression control unit configured to calculate, according to a ratio between the input instantaneous value and the reverberation component, a second gain changing within a range between a predetermined lower limit and a predetermined upper limit, thereby outputting a larger one of the first gain or the second gain as a third gain; and a gain processing unit configured to multiply the input signal by the third gain.
    Type: Grant
    Filed: February 19, 2019
    Date of Patent: November 19, 2019
    Assignee: RION Co., Ltd.
    Inventors: Masatoshi Osawa, Masahiro Sunohara, Yoichi Fujisaka, Yoko Fujishima
  • Patent number: 10482868
    Abstract: A method of operating a playback device includes receiving source audio content that includes a first and second channel stream of audio. The method also includes playing back, via a first and second speaker driver of the playback device, the first and second channel streams of audio, thereby producing a first and second channel audio output. A captured stream of audio is received by a microphone of the playback device, and portions of the captured stream of audio correspond to the first and second channel audio outputs. The first and second channel streams of audio are combined into a compound audio signal, and acoustic echo cancellation is performed on the compound audio signal to produce an acoustic echo cancellation output, which is then applied to the captured stream of audio to increase the signal-to noise ratio of the captured stream of audio.
    Type: Grant
    Filed: September 28, 2017
    Date of Patent: November 19, 2019
    Assignee: Sonos, Inc.
    Inventors: Saeed Bagheri Sereshki, Romi Kadri
  • Patent number: 10454442
    Abstract: Processes and devices for equalizing an audio system that is adapted to use a loudspeaker to transduce test audio signals into test sounds. The processes and devices can involve the use of infrared signals to convey information in one or both directions between the audio system and a portable computer device that captures test sounds, calculates audio parameters that can be used in the equalization process, and transmits these audio parameters back to the audio system for its use in equalizing audio signals that are played by the audio system.
    Type: Grant
    Filed: June 1, 2018
    Date of Patent: October 22, 2019
    Assignee: Bose Corporation
    Inventors: Wontak Kim, Michael J. Daley, Laszlo Drimusz, Matthew S. Walsh
  • Patent number: 10438604
    Abstract: A speech intelligibility enhancing system for enhancing speech, the system comprising: a speech input for receiving speech to be enhanced; an enhanced speech output to output the enhanced speech; and a processor configured to convert speech received from the speech input to enhanced speech to be output by the enhanced speech output, the processor being configured to: i) extract a frame of the speech received from the speech input; ii) calculate a measure of the frame importance; iii) estimate a contribution due to late reverberation to the frame power of the speech when reverbed; iv) calculate a prescribed frame power, the prescribed frame power being a function of the power of the extracted frame, the measure of the frame importance and the contribution due to late reverberation, the function being configured to decrease the ratio of the prescribed frame power to the power of the extracted frame as the contribution due to late reverberation increases above a critical value, {tilde over (l)}; and v) apply
    Type: Grant
    Filed: March 1, 2017
    Date of Patent: October 8, 2019
    Assignee: KABUSHIKI KAISHA TOSHIBA
    Inventors: Petko Petkov, Ioannis Stylianou
  • Patent number: 10425718
    Abstract: Disclosed are an electronic device and a method of processing an audio signal by the electronic device. The electronic device includes: a processor functionally connected to a speaker and a microphone; and a memory electrically connected to the processor. The memory includes instructions to cause the processor, when executed, to output a first audio signal through the speaker; identify that a second audio signal detected by the microphone corresponds to the first audio signal when the first audio signal is outputted through the speaker; and control output of the first audio signal based on the identification. Further, various other embodiments may be possible.
    Type: Grant
    Filed: December 12, 2017
    Date of Patent: September 24, 2019
    Assignee: Samsung Electronics Co., Ltd.
    Inventors: A-Ran Cha, Byeong-Jun Kim, Jae-Hyun Kim, Nam-Il Lee, Hyun Jo
  • Patent number: 10425730
    Abstract: A technique for controlling a loudspeaker system with an artificial neural network includes filtering, with a deconvolution filter, a measured system response of a loudspeaker and a reverberant environment in which the loudspeaker is disposed to generate a filtered response, wherein the measured system response corresponds to an audio input signal applied to the loudspeaker while the loudspeaker is disposed in the reverberant environment. The techniques further include generating, via a neural network model, an initial neural network output based on the audio input signal, comparing the initial neural network output to the filtered response to determine an error value, and generating, via the neural network model, an updated neural network output based on the audio input signal and the error value.
    Type: Grant
    Filed: April 14, 2016
    Date of Patent: September 24, 2019
    Assignee: HARMAN INTERNATIONAL INDUSTRIES, INCORPORATED
    Inventors: Ajay Iyer, Douglas J. Button
  • Patent number: 10424317
    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: January 11, 2017
    Date of Patent: September 24, 2019
    Assignee: Nuance Communications, Inc.
    Inventors: Pablo Peso Parada, Dushyant Sharma, Patrick Naylor
  • 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