Abstract: 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.

Abstract: A method of determining a near optimal forward error correction scheme for the transmission of audio data over a lossy packet switched network having preallocated estimated bandwidth, delay and packet losses, between at least a first and second communications devices, the method including the steps of: determining a first coding rate for the audio data; determining a peak redundancy coding rate for redundant versions of the audio data; determining an average redundancy coding rate over a period of time for redundant versions of the audio data; determining an objective function which maximizes a bitrate-perceptual audio quality mapping of the transmitted audio data including a playout function formulation; and optimizing the objective function to produce a forward error correction scheme providing a high bitrate perceptual audio quality.

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: 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: The present invention relates to methods of treating and/or ameliorating the severity of inflammation and autoimmunity in the central nervous system (CNS). In one embodiment, the present invention provides a method of treating multiple sclerosis by administering a therapeutically effective dosage of nicotine, or a pharmaceutical equivalent, analog, derivative, or salt thereof.

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: A noise-level estimator for a noise suppressor includes a power smoother filter providing smoothed power estimates in timeslices, a minimum follower that represents the lowest smoothed input power, and a maximum follower that represents the highest smoothed input power, the followers subject to leakage factors. The estimator has a speech probability detector receiving outputs of the power smoother and minimum follower; a nonstationary noise detector receiving outputs of both followers; and an estimator receiving outputs of the nonstationary noise detector, power smoother, and speech probability detector and providing a noise estimate.

Abstract: A noise suppressor has a band extractor to separate signal by frequency band; and per-band units for each of band including noise estimator and SNR computation units. The per-band unit has a histogrammer to give histograms of current and past SNRs, and a gain-curve updater computes gain curves from the histogram. Gain curves are used to determine raw gains from current SNRs, raw gain is filtered and controls a variable gain unit to provide band-specific gain-adjusted, signals that are recombined into a noise-reduced frequency-domain output. Raw gain filtering may include finite-impulse-response filtering and weighted averaging of intermediate gains of a current and adjacent-band per-band unit. The method includes separating an input into frequency bands, estimating in-band noise, and deriving a band SNR. Then, histogramming the SNR and updating a gain curve from the histogram, and finding a raw gain using the gain curve and current SNR.

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: 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: The present invention relates to a transparent organosilicon gel adhesive. The transparent organosilicon gel adhesive is prepared by mixing component A with component B in accordance with the mass proportion of 1:1. The component A is composed of 60-95 parts by weight of base material, 0.1-0.5 part by weight of catalyst and 5-40 parts by weight of auxiliary material, and the component B is composed of 60-95 parts by weight of base material, 5-20 parts by weight of crosslinker, 0.1-0.5 part by weight of inhibitor and 1-20 parts by weight of auxiliary material. The present invention relates to a two-component addition type transparent AB silicone gel adhesive, which has good adhesion to glass, a PC board, PMMA. The present invention has a refractive index of 1.40 to 1.43, is a colorless transparent gel, and has a light transmittance of more than 92%.

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.

Abstract: An echo canceller includes a fast Fourier transform (FFT) unit to provide frequency domain representation (FD) of an input. A multiband adaptive filter receives the FD of the input and provides an FD filter output, the adaptive filter is a finite input response (FIR) digital filter. Another FFT unit provides an FD of a microphone signal, and a summer adds the FD filter output to the FD of the microphone signal to provide echo-canceller FD output. A feedback subsystem uses the echo-canceller FD output to adjust filter coefficients of at least a first, a second, and a third frequency band of the multiband adaptive filter to minimize uncancelled output in the echo-canceller FD output. The feedback subsystem adjusts the filter coefficients of the second frequency band of the adaptive filter according to uncancelled output in the first, second, and third frequency bands of the echo-canceller FD output.

Abstract: The present application provides an acoustic echo mitigation apparatus and method, an audio processing apparatus and a voice communication terminal. According to an embodiment, an acoustic echo mitigation apparatus is provided, including: an acoustic echo canceller for cancelling estimated acoustic echo from a microphone signal and outputting an error signal; a residual echo estimator for estimating residual echo power; and an acoustic echo suppressor for further suppressing residual echo and noise in the error signal based on the residual echo power and noise power. Here, the residual echo estimator is configured to be continuously adaptive to power change in the error signal. According to the embodiments of the present application, the acoustic echo mitigation apparatus and method can, at least, be well adaptive to the change of power of the error signal after the AEC processing, such as that caused by change of double-talk status, echo path properties, noise level and etc.

Abstract: Disclosed is a system and computer program product of encoding audio content and corresponding method. The method includes determining a characteristic of the audio content, the characteristic of the audio content including at least one of a type or a property of the audio content. Also the method includes classifying the audio content based on the characteristic of the audio content and determining probabilities for multiple predefined audio coding symbols associated with the audio content by calculating a probability for each of the audio coding symbols based on the result of the classification, the probability for an audio coding symbol indicating a frequency at which the audio coding symbol occurs in the audio content. Further, the method encoded the audio content based on the audio coding symbols and the corresponding probabilities to obtain a code value, the code value representing a compression coding format of the audio content.

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 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: 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.