Abstract: 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.

Abstract: 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.

Abstract: 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.

Abstract: 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.

Abstract: In an audio processing system (300), a filtering section (350, 400): receives subband signals (410, 420, 430) corresponding to audio content of a reference signal (301) in respective frequency subbands; receives subband signals (411, 421, 431) corresponding to audio content of a response signal (304) in the respective subbands; and forms filtered inband references (412, 422, 432) by applying respective filters (413, 423, 433) to the subband signals of the reference signal.

Abstract: 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.

Abstract: 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.

Abstract: 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.

Abstract: Embodiments of the present disclosure provide for single crystal organometallic halide perovskites, methods of making, methods of use, devices incorporating single crystal organometallic halide perovskites, and the like.

Abstract: 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.

Abstract: 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.

Abstract: In an audio processing system (300), a filtering section (350, 400): receives subband signals (410, 420, 430) corresponding to audio content of a reference signal (301) in respective frequency subbands; receives subband signals (411, 421, 431) corresponding to audio content of a response signal (304) in the respective subbands; and forms filtered inband references (412, 422, 432) by applying respective filters (413, 423, 433) to the subband signals of the reference signal.

Abstract: 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.

Abstract: Example embodiments disclosed herein relate to a estimation of reverberant energy components from audio sources. A method of estimating a reverberant energy component from an active audio source (100) is disclosed. The method comprises determining a correspondence between the active audio source and a plurality of sample sources by comparing one or more spatial features of the active audio source with one or more spatial features of the plurality of sample sources, each of the sample sources being associated with an adaptive filtering model (101); obtaining an adaptive filtering model for the active audio source based on the determined correspondence (102); and estimating the reverberant energy component from the active audio source over time based on the adaptive filtering model (103). Corresponding system (800) and computer program product (900) are also disclosed.

Abstract: Apparatuses and Methods of double talk detection for echo suppression in the power domain are disclosed herein. An example method includes comparing a power echo estimate signal to a microphone power signal, determining if the power echo estimate signal is greater than or less than the microphone power signal by a first threshold, based on the power echo estimate signal being greater than or less than the microphone power signal by the first threshold, disabling adaptation of filter coefficients of an adaptive filter of an echo suppressor, and based on the power echo estimate signal not being greater than or less than the microphone power signal by the first threshold, enabling adaptation of filter coefficients of an adaptive filter of an echo suppressor.

Abstract: A speaker system featuring automatic volume adjustment to avoid excess distortion includes a first converter for converting an input signal to an input frequency domain signal; and a second converter for converting a microphone signal to an environmental frequency domain signal. A distortion analyzer receives output of the first and second converter and determines at least one distortion level and provides at least one gain adjustment signal used to control a corrector to adjust gain of a speaker driver. The system performs a method of reducing distortion by automatically adjusting volume in each of a plurality of frequency bands including converting an input signal to a frequency domain input signal; converting a microphone signal to an environmental frequency domain signal; determining a distortion level; generating a gain adjustment signal; and adjusting gain of a signal provided to a speaker driver.

Abstract: Systems and methods are described for analyzing and resolving feedback caused by having multiple audio links in a conference room. An audio system may detect the presence of duplicated audio caused by multiple audio links in the conference room. Marker signals may be injected into the conference room or over the network in response to detecting the duplicated audio. Echoes of the marker signals may be received, and the system may determine which case corresponds to the detected duplicated audio based on the received echo of the marker signals. Based on the determined case, operation of at least one of the speaker and the microphone may be modified. After the modification, audio playback in the conference room may be monitored to verify that far end audio playback is taking place and that the duplicated audio has been resolved.

Abstract: 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 linear subband of the audio signal. The method also includes determining a 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.

Abstract: Example embodiments disclosed herein relate to impulsive noise suppression. A method of impulsive noise suppression in an audio signal is disclosed. The method includes determining an impulsive noise related feature from a current frame of the audio signal. The method also includes detecting an impulsive noise in the current frame based on the impulsive noise related feature, and in response to detecting the impulsive noise in the current frame, applying a suppression gain to the current frame to suppress the impulsive noise. Corresponding system and computer program product of impulsive noise suppression in an audio signal are also disclosed.

Abstract: 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.