Dolby Laboratories Licensing Corporation

Omnivision Technologies, Inc.

Dobly Laboratories Licensing Corporation

Dobly International Ab

Hyzon Motors USA Inc.

Dolby International Ab

Dong Shi

Glenn N Dickins

Xuejing Sun

David Gunawan

Chung-An Wang

Kai Lin Li

Paul Francis Holmberg

Cheng-Lun Hu

Hannes Muesch

Heiko Purnhagen

Harald Mundt

Shen Huang

Zhiwei Shuang

Lu Chang

Kai Li

Wei Fan

Andrew Border

Craig Johnston

JiaQuan Huo

Qunlong Dong

Jay Andrew De Veny, Iii

Paul Holmberg

Jiaquan Huo

Semi-truck front windshield

Methods for echo estimation or echo management (echo suppression or cancellation) on an input audio signal, with at least one of adaptation of a sparse prediction filter set, modification (for example, truncation) of adapted prediction filter impulse responses, generation of a composite impulse response from adapted prediction filter impulse responses, or use of echo estimation and/or echo management resources in a manner determined at least in part by classification of the input audio signal as being (or not being) echo free. Other aspects are systems configured to perform any embodiment of any of the methods.

Echo estimation and management with adaptation of sparse prediction filter set

Example embodiments disclosed herein relate to filter coefficient updating in time domain filtering. A method of processing an audio signal is disclosed. The method includes obtaining a predetermined number of target gains for a first portion of the audio signal by analyzing the first portion of the audio signal. Each of the target gains is corresponding to a subband of the audio signal. The method also includes determining filter coefficients for time domain filtering the first portion of the audio signal so as to approximate a frequency response given by the target gains. The filter coefficients are determined by iteratively selecting at least one target gain from the target gains and updating the filter coefficient based on the selected at least one target gain. Corresponding system and computer program product for processing an audio signal are also disclosed.

Filter coefficient updating in time domain filtering

In an audio processing system (), a filtering section (): receives subband signals () corresponding to audio content of a reference signal () in respective frequency subbands; receives subband signals () corresponding to audio content of a response signal () in the respective subbands; and forms filtered inband references () by applying respective filters () to the subband signals of the reference signal. For a frequency subband: filtered crossband references () are formed by multiplying, by scalar factors (), filtered inband references of other subbands; a composite filtered reference () is formed by summing the filtered inband reference of the subband () and the filtered crossband references; a residual signal () is computed as a difference between the composite filtered reference and the subband signal of the response signal corresponding to the subband; and the scalar factors and the filter applied to the subband signal of the reference signal corresponding to the subband are adjusted based on the residual signal.

Adaptive audio filtering

Example embodiments disclosed herein relate to audio signal processing. A method of indicating a presence of a nuisance in an audio signal is disclosed. The method includes determining a probability of the presence of the nuisance in a frame of the audio signal based on a feature of the audio signal, the nuisance representing an unwanted sound made by a user, in response to the probability of the presence of the nuisance exceeding a threshold, tracking the audio signal based on a metric over a plurality of frames following the frame, determining, based on the tracking, that the presence of the nuisance is to be indicated to the user, and in response to the determination, presenting to the user a notification of the presence of the nuisance. Corresponding system and computer program product are also disclosed.

Nuisance notification

A surveillance system includes an infrared sensor system coupled to output an infrared signal in response to receiving infrared light, and an audio recording system coupled to output an audio signal in response to recording sound. An image sensor is system coupled to output an image signal in response to receiving image light. A controller is coupled to the infrared sensor system, the audio recording system, and the image sensor system. The controller includes logic that when executed by the controller causes the surveillance system to perform operations including receiving the infrared signal from the infrared sensor system, activating the audio recording system to record the sound, and activating the image sensor system to output the image signal.

Reducing false alarms in surveillance systems

An electronic device to equalize sound includes a microphone coupled to receive the sound and output sound data. The sound has amplitude that changes with respect to time, and the sound includes one or more frequencies. The sound data indexes the amplitude with respect to the time. The electronic device also includes a controller coupled to microphone, and the controller includes logic that when executed by the controller causes the electronic device to perform operations. The operations may include: receiving the sound data from the microphone with the controller; adjusting the amplitude of the sound included in the sound data across the one or more frequencies using a filter disposed in the logic; and outputting filtered sound data.

Low complexity loudness equalization

Example embodiments disclosed herein relate to audio signal processing. A method of processing an audio signal is disclosed. The method includes detecting, based on a power distribution of the audio signal, a type of content of a frame of the audio signal, generating a first gain based on a sound level of the frame for adjusting the sound level, processing the audio signal by applying the first gain to the frame; and in response to the type of content being detected to be a breath sound, generating a second gain for mitigating the breath sound and processing the audio signal by applying the second gain to the frame. Corresponding system and computer program product are also disclosed.

Suppression of breath in audio signals

Example embodiments disclosed herein relate to separated audio analysis and processing. A system for processing an audio signal is disclosed. The system includes an audio analysis module configured to analyze an input audio signal to determine a processing parameter for the input audio signal, the input audio signal being represented in time domain. The system also includes an audio processing module configured to process the input audio signal in parallel with the audio analysis module. The audio processing module includes a time domain filter configured to filter the input audio signal to obtain an output audio signal in the time domain, and a filter controller configured to control a filter coefficient of the time domain filter based on the processing parameter determined by the audio analysis module. Corresponding method and computer program product of processing an audio signal are also disclosed.

Separated audio analysis and processing

Described herein is a method and audio capture system for suppressing noise due to mechanical disturbances in an audio capture system. In one embodiment a method includes the steps of: a) receiving an input audio signal from a microphone; b) receiving an input reference signal separate from the input audio signal; c) processing the input reference signal to generate a control signal for identifying a noise event; d) in response to the control signal, estimating a contribution of the noise event to the input audio signal by applying an adaptive filter to a sliding temporal window of the control signal in the frequency domain; and e) based on the estimated contribution, selectively modifying the input audio signal to generate an output audio signal in which the noise arising from the noise event is at least partially suppressed.

Adaptive suppression for removing nuisance audio

A method of processing data comprises processing first frequency-domain audio or video data using signal processing of a first type, and transforming the processed first frequency-domain audio or video data to processed time-domain audio or video data using a transform which is the inverse of the first transform, and transforming the processed time-domain audio or video data using a second transform which is matched to a second type of signal processing. The method further comprises identifying time-domain audio or video data for which signal processing of the first type, after transformation using the second transform, would yield satisfactory results. The method further comprises transforming the identified time-domain audio or video data to frequency-domain identified audio or video data using the second transform, instead of using the first transform, and processing the identified frequency-domain audio or video data using signal processing of the first type.

Switching between transforms

Example embodiments disclosed herein relate to assessment and adjustment for an audio environment. A computer-implemented method is provided. The method includes obtaining a first audio signal captured by a device located in an environment. The method also includes analyzing a characteristic of the first audio signal to determine an acoustic performance metric for the environment. The method further includes, in response to the acoustic performance metric being below a threshold, providing a first task for a user to perform based on the characteristic of the first audio signal. The first task is related to an adjustment to a setting of the environment. Embodiments in this regard further provide a corresponding computer program product. Corresponding system and computer program product are also disclosed.

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Shanghai, China

Dong Shi