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61. 发明授权

US11189303B2 Persistent interference detection 有权
公开(公告)号：US11189303B2
公开(公告)日：2021-11-30
申请号：US15714190
申请日：2017-09-25
申请人： Cirrus Logic International Semiconductor Ltd.
发明人： Narayan Kovvali , Seth Suppappola
IPC分类号： G10L25/84 , G01S3/808 , G01S3/80 , H04R1/40 , H04R3/00 , G10L21/0208 , G10L21/0216 , G10L21/0224 , G10L21/0232
摘要： A multi-microphone algorithm for detecting and differentiating interference sources from desired talker speech in advanced audio processing for smart home applications is described. The approach is based on characterizing a persistent interference source when sounds repeated occur from a fixed spatial location relative to the device, which is also fixed. Some examples of such interference sources include TV, music system, air-conditioner, washing machine, and dishwasher. Real human talkers, in contrast, are not expected to remain stationary and speak continuously from the same position for a long time. The persistency of an acoustic source is established based on identifying historically-recurring inter-microphone frequency-dependent phase profiles in multiple time periods of the audio data. The detection algorithm can be used with a beamforming processor to suppress the interference and for achieving voice quality and automatic speech recognition rate improvements in smart home applications.

62. 发明申请

US20210366504A1 ADAPTIVE DYNAMIC AUDIO HUM EXTRACTOR AND EXTRACTION PROCESS 有权
公开(公告)号：US20210366504A1
公开(公告)日：2021-11-25
申请号：US16994297
申请日：2020-08-14
申请人： James K. Waller, JR. , Jon J. Waller
发明人： James K. Waller, JR. , Jon J. Waller
IPC分类号： G10L21/0232 , H04R3/04 , G10L25/51 , G10L25/18 , G10L21/038
摘要： An adaptive dynamic audio hum extractor eliminates line frequency hum components and associated higher harmonics from an audio signal. An audio signal containing line frequency hum can be processed by providing dynamically controlled notch filters at the fundamental line frequency and additional harmonic multiples of the fundamental frequency. The audio signal is detected to provide dynamic control of the depth of the notch filters. Alternatively, an audio signal containing hum can be processed by dividing the spectrum into at least two frequency bands, an unaltered high band combined with a dynamically processed low band. The adaptive dynamically controlled notch filters vary the depth of the notches in relation to the envelope or time averaged level of the bandwidth limited audio signal. This allows masking of the hum components with higher levels of audio, thereby providing transparency devoid of audio path notches.

63. 发明申请

US20210360316A1 SYSTEMS AND METHODS FOR PROVIDING SURVEY DATA 有权
公开(公告)号：US20210360316A1
公开(公告)日：2021-11-18
申请号：US16876914
申请日：2020-05-18
申请人： Mercury Analytics, LLC
发明人： Scott BRICKNER , Matthew Thomas WILLIAMS , Peter VISS , Ivan VICAN
IPC分类号： H04N21/442 , G10L25/51 , G10L21/0232
摘要： A method includes receiving, at a network server, a data package from a user mobile device, the data package comprising real-time survey user input associated with watching a video program and survey audio from the video program, the survey audio being recorded via a microphone of the user mobile device during the real-time survey, receiving an audio file associated with the video program, comparing the audio file with the survey audio to yield a comparison, aligning, based on the comparison, the survey audio with the audio file to yield an modified data package and providing the modified data package.

64. 发明授权

US11172291B2 Millimeter wave sensor used to optimize performance of a beamforming microphone array 有权
公开(公告)号：US11172291B2
公开(公告)日：2021-11-09
申请号：US16802111
申请日：2020-02-26
申请人： Crestron Electronics, Inc.
发明人： Mark LaBosco
IPC分类号： H04R3/00 , H04R1/40 , H04R29/00 , G10L21/02 , G10L21/0208 , G06F3/01 , G10L21/0232
摘要： A method for operating a beamforming microphone array for use in a predetermined area comprising: receiving acoustic audio signals at each of a plurality of microphones, converting the same to an electrical mic audio signal, and outputting each of the plurality of electrical mic audio signals; generating a user location data signal by a wave sensor system, and outputting the user location data signal, wherein the user location data signal includes location information of one or more people within the predetermined area; receiving both the user location data signal and plurality of mic audio signals at an adaptive beamforming device; adapting one or more beams by the adaptive beamforming device based on the user location data signal and plurality of output electrical mic audio signals wherein each of the one or more beams acquires sound from one or more specific locations in the predetermined area; and performing acoustic echo cancellation on each of the one or more beams output from the adaptive beamforming device.

65. 发明授权

US11170799B2 Nonlinear noise reduction system 有权
公开(公告)号：US11170799B2
公开(公告)日：2021-11-09
申请号：US16275126
申请日：2019-02-13
申请人： HARMAN INTERNATIONAL INDUSTRIES, INCORPORATED
发明人： James Lambrick
IPC分类号： G10L21/0232 , G10L15/20 , G10L15/22 , G10L25/18 , G10L25/21 , G10L25/24
摘要： Embodiments of the present disclosure set forth a method of decomposing an audio signal into a set of sub-band signals and detecting a set of signal energy values, where each signal energy value is associated with a sub-band signal. The method also includes generating a noise reduction threshold based on at least one sub-band signal, and, for each sub-band signal, comparing the associated signal energy value to the noise reduction threshold. Based on determining that at least one sub-band signal is associated with a signal energy value below the noise reduction threshold, the method includes attenuating the at least one the sub-band signal to generate a set of attenuated sub-band signals. The method also includes combining at least one sub-band signal included in the set of sub-band signals with at least one attenuated sub-band signal included in the set of attenuated sub-band signals to generate an output audio signal.

66. 发明申请

US20210327451A1 METHOD FOR DEBUGGING NOISE ELIMINATION ALGORITHM, APPARATUS AND ELECTRONIC DEVICE 有权
公开(公告)号：US20210327451A1
公开(公告)日：2021-10-21
申请号：US17361445
申请日：2021-06-29
申请人： APOLLO INTELLIGENT CONNECTIVITY (BEIJING) TECHNOLOGY CO., LTD.
发明人： Tengfei ZHANG
IPC分类号： G10L21/0232 , G10L21/034 , G06F11/36 , G06F11/30
摘要： The application discloses a debugging method for a noise elimination algorithm, an apparatus and an electronic device, which relate to the technical fields of voice, automatic driving and intelligent transportation. An implementation scheme is: when the noise elimination algorithm is debugged, acquiring multiple voice control signals from a vehicle to be debugged, modifying a weight of a configuration parameter of the noise elimination algorithm in a digital signal processing to obtain an updated noise elimination algorithm; then adopting the updated noise elimination algorithm to perform noise elimination processing on the multiple voice control signals; if control results of noise-eliminated voice control signals on the vehicle to be debugged do not meet a preset condition, continuing to modify the weight of the configuration parameter until the preset condition is met, and then sending a noise elimination algorithm that meets the preset condition to the vehicle to be debugged.

67. 发明申请

US20210327450A1 A METHOD AND APPARATUS FOR PROCESSING AN AUDIO SIGNAL STREAM TO ATTENUATE AN UNWANTED SIGNAL PORTION 有权
公开(公告)号：US20210327450A1
公开(公告)日：2021-10-21
申请号：US17271128
申请日：2019-08-19
申请人： Calrec Audio Limited
发明人： Caleb Maynard Price
IPC分类号： G10L21/0232 , G10L19/022 , G06F3/16
摘要： A method of processing an audio signal stream to attenuate an unwanted signal portion, the method comprising the steps of (a) providing a filter block having an input port and an output port, the filter block having an inactive state in which signals pass from the input port to the output port without being filtered and an active state in which signals are filtered to attenuate an unwanted signal portion as they pass from the input port to the output port; (b) providing the audio signal stream to the input port of the filter; and, (c) whilst the audio signal stream is being provided to the input port of the filter— (i) calculating the entropy of at least a portion of the audio signal stream; (ii) comparing the calculated entropy to a threshold value; and, (iii) setting the state of the filter block to be either active or inactive depending on the comparison between the calculated entropy and the threshold value.

68. 发明授权

US11151385B2 System and method for detecting deception in an audio-video response of a user 有权
公开(公告)号：US11151385B2
公开(公告)日：2021-10-19
申请号：US16722083
申请日：2019-12-20
申请人： RTScaleAI Inc
发明人： Vivek Iyer , Peter Walker
IPC分类号： G06K9/00 , G10L15/02 , G10L15/22 , G10L15/18 , G06K9/32 , G06N5/04 , G10L21/0232 , G10L25/63 , G10L25/90 , G06K9/62 , G06N20/00
摘要： A method for (of) detecting deception in an Audio-Video response of a user, using a server, in a distributed computing architecture, characterized in that the method including: enabling an Audio-Video connection with a user device upon receiving a request from a user; obtaining, from the user device, an Audio-Video response of the user corresponding to a first set of questions that are provided to the user by the server; extracting audio signals and video signals from the Audio-Video response; detecting an activity of the user by determining a plurality of Natural Language Processing (NLP) features from the extracted audio signals by (i) performing a speech to text translation and (ii) extracting the plurality of NLP features from the translated text, and determining a plurality of speech features from the extracted audio signals by (i) splitting the extracted audio signals into a plurality of short interval audio signals and (ii) extracting the plurality of speech features from the plurality of short interval audio signals; aggregating (i) the plurality of NLP features to obtain a plurality of temporal NLP features and (ii) the plurality of speech features to obtain a plurality of temporal speech features; aggregating the plurality of temporal NLP features and the plurality of temporal speech features to obtain first temporal aggregated features; detecting a plurality of micro-expressions of the user by splitting extracted video signals into a plurality of short fixed-duration video signals, detecting a plurality of Region Of Interest (ROI) in the plurality of short fixed-duration video signals, and comparing the plurality of detected ROI with video signals annotated with micro-expression labels that are stored in a database to detect the plurality of micro-expressions of the user in the plurality of short fixed-duration video signals; tracking and determining a gesture of the user from the extracted video signals; aggregating the plurality of micro-expressions and the gesture of the user to obtain second temporal aggregated features; aggregating the first temporal aggregated features and the second temporal aggregated features to obtain final temporal aggregated features; and detecting, using a machine learning model, a deception in the Audio-Video response based on the final temporal aggregated features.

69. 发明授权

US11138989B2 Sound quality prediction and interface to facilitate high-quality voice recordings 有权
公开(公告)号：US11138989B2
公开(公告)日：2021-10-05
申请号：US16296122
申请日：2019-03-07
申请人： ADOBE INC.
发明人： Prem Seetharaman , Gautham J. Mysore , Bryan A. Pardo
IPC分类号： G10L25/60 , G10L25/30 , G10L21/0232 , G10L25/84 , G10L21/0208
摘要： Embodiments of the present invention provide systems, methods, and computer storage media for sound quality prediction and real-time feedback about sound quality, such as room acoustics quality and background noise. Audio data can be sampled from a live sound source and stored in an audio buffer. The audio data in the buffer is analyzed to calculate a stream of values of one or more sound quality measures, such as speech transmission index and signal-to-noise ratio. Speech transmission index can be calculated using a convolution neural network configured to predict speech transmission index from reverberant speech. The stream of values can be used to provide real-time feedback about sound quality of the audio data. For example, a visual indicator on a graphical user interface can be updated based on consistency of the values over time. The real-time feedback about sound quality can help users optimize their recording setup.

70. 发明申请

US20210304782A1 FORCED GAP INSERTION FOR PERVASIVE LISTENING 有权
公开(公告)号：US20210304782A1
公开(公告)日：2021-09-30
申请号：US17261884
申请日：2019-07-26
申请人： DOLBY LABORATORIES LICENSING CORPORATION
发明人： Christopher Graham Hines , Glenn N. Dickins
IPC分类号： G10L21/0232 , G10K11/178
摘要： A pervasive listening method including steps of inserting at least one forced gap in a playback signal (thus generating a modified playback signal), and during playback of the modified playback signal, monitoring non-playback content (e.g., including by generating an estimate of background noise) in a playback environment using output of a microphone in the playback environment. Optionally, the method includes generation of the playback signal, including by processing of (e.g., performing noise compensation on) an input signal using a result (e.g., a background noise estimate) of the monitoring of non-playback content. Other aspects are systems configured to perform any embodiment of the pervasive listening method.

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