Abstract:
A method for controlling acoustic echo cancellation and an audio processing apparatus are described. In one embodiment, the audio processing apparatus includes an acoustic echo canceller for suppressing acoustic echo in a microphone signal, a jitter buffer for reducing delay jitter of a received signal, and a joint controller for controlling the acoustic echo canceller by referring to at least one future frame in the jitter buffer.
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:
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:
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:
Example embodiments disclosed herein relate to spatial congruency adjustment. A method for adjusting spatial congruency in a video conference is disclosed. The method includes detecting spatial congruency between a visual scene captured by a video endpoint device and an auditory scene captured by an audio endpoint device that is positioned in relation to the video endpoint device, the spatial congruency being a degree of alignment between the auditory scene and the visual scene, comparing the detected spatial congruency with a predefined threshold and in response to the detected spatial congruency being below the threshold, adjusting the spatial congruency. Corresponding system and computer program products are also disclosed.
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:
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 audio signal processing based on remote user control. A method of processing an audio signal in an audio sender device is disclosed. The method includes receiving, at a current device, a control parameter from a remote device, the control parameter being generated based on a user input of the remote device and specifying a user preference for an audio signal to be transmitted to the remote device. The method also includes processing the audio signal based on the received control parameter and transmitting the processed audio signal to the remote device. Corresponding computer program product of processing an audio signal and corresponding device are also disclosed. Corresponding method in an audio receiver device and computer program product of processing an audio signal as well as corresponding device are also disclosed.
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:
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.