An acoustic analysis device and the like that can separate acoustic signals of a target sound source at a higher speed are provided. The acoustic analysis device includes: an acquiring unit configured to acquire acoustic signals; a first generating unit configured to generate acoustic signals of diffuse noise using a first model which includes a spatial correlation matrix related to frequency, a first parameter related to the frequency, and a second parameter related to the frequency and time; a second generating unit configured to generate acoustic signals emitted from a target sound source using a second model which includes a steering vector related to the frequency, and a third parameter related to the frequency and the time; and a determining unit configured to determine the first parameter, the second parameter and the third parameter so that the likelihood of the first parameter, the second parameter and the third parameter is maximized. The determining unit decomposes an inverse matrix of the matrix related to the frequency and the time into an inverse matrix of the matrix related to the frequency, and determines the first parameter, the second parameter and the third parameter so that the likelihood is maximized.

An acoustic analysis device and the like that can separate acoustic signals of a target sound source at a higher speed are provided. The acoustic analysis device includes: an acquiring unit configured to acquire acoustic signals; a first generating unit configured to generate acoustic signals of diffuse noise using a first model which includes a spatial correlation matrix related to frequency, a first parameter related to the frequency, and a second parameter related to the frequency and time; a second generating unit configured to generate acoustic signals emitted from a target sound source using a second model which includes a steering vector related to the frequency, and a third parameter related to the frequency and the time; and a determining unit configured to determine the first parameter, the second parameter and the third parameter so that the likelihood of the first parameter, the second parameter and the third parameter is maximized. The determining unit decomposes an inverse matrix of the matrix related to the frequency and the time into an inverse matrix of the matrix related to the frequency, and determines the first parameter, the second parameter and the third parameter so that the likelihood is maximized.

An audio spotting system configured for various operating modes including a regular mode and sensitivity mode is described. An example cascade audio spotting system may include a high-power subsystem including a high-power trigger and a transfer module. This high-power trigger includes one or more detection models used to detect whether a target sound activity is included in the one or more audio streams. The one or more detection models are associated with a first set of hyperparameters when the cascade audio spotting system is in a regular mode, and the one or more detection models are associated with a second set of hyperparameters when the cascade audio spotting system is in a sensitivity mode. The transfer module provides at least one of one or more processed audio streams for further processing in response to the high-power trigger detecting the target sound activity in the one or more audio streams.

An audio spotting system configured for various operating modes including a regular mode and sensitivity mode is described. An example cascade audio spotting system may include a high-power subsystem including a high-power trigger and a transfer module. This high-power trigger includes one or more detection models used to detect whether a target sound activity is included in the one or more audio streams. The one or more detection models are associated with a first set of hyperparameters when the cascade audio spotting system is in a regular mode, and the one or more detection models are associated with a second set of hyperparameters when the cascade audio spotting system is in a sensitivity mode. The transfer module provides at least one of one or more processed audio streams for further processing in response to the high-power trigger detecting the target sound activity in the one or more audio streams.

A method, computer program product, and computing system for receiving a signal from each microphone of a plurality of microphones, thus defining a plurality of signals. One or more inter-microphone gain-based augmentations may be performed on the plurality of signals, thus defining one or more inter-microphone gain-augmented signals.

An audio system includes: a speaker; a microphone that generates a microphone signal based on sound output from the speaker; a mixer module configured to generate a mixed signal by mixing the microphone signal with an audio signal; a filter module configured to filter the mixed signal to produce a filtered signal and to apply the filtered signal to the speaker; and a detector module configured to determine a howling frequency in the microphone signal attributable to sound output from the speaker, where the filter module is configured to decrease a magnitude of the filtered signal at the howling frequency.

In one aspect, a first playback device is configured to (i) receive a set of voice signals, (ii) process the set of voice signals using a first set of audio processing algorithms, (iii) identify, from the set of voice signals, at least two voice signals that are to be further processed, (iv) determine that the first playback device does not have a threshold amount of computational power available, (v) receive an indication of an available amount of computational power of a second playback device, (vi) send the at least two voice signals to the second playback device, (vii) cause the second playback device to process the at least two voice signals using a second set of audio processing algorithms, (viii) receive, from the second playback device, the processed at least two voice signals, and (ix) combine the processed at least two voice signals into a combined voice signal.

Systems, apparatuses, and methods are described for controlling source tracking and delaying beamforming in a microphone array system. A source tracker may continuously determine a direction of an audio source. A source tracker controller may pause the source tracking of the source tracker if a user may continue to speak to the system. The source tracker controller may resume the source tracking of the source tracker if the user may cease to speak to the system, or when one or more pause durations have been reached.

Method includes receiving, at a server device, from a plurality of input devices, audio data. The audio data of each input device corresponds to a time-related portion of the audio data. The method determines a speech energy level for each input device by providing the time-related audio portion as input to a trained model. For each input device, a statistical value associated with the speech energy level is determined. A strongest input device is identified based on the statistical value. The statistical value associated with the speech energy level of each input device other than the strongest input device is compared to the statistical value of the strongest input device. Depending on the comparison, the method determines whether to update the gain value of an input device to an estimated target gain value based on the statistical value of the speech energy level of the respective input device.

An electronic device is provided. The electronic device includes, for the purpose of determining a customized beamformer filter, an input module including a plurality of microphones configured to receive an external sound signal, a memory configured to store computer-executable instructions and an initial value of a voice parameter used to perform beamforming on the external sound signal, and a processor configured to execute the instructions by accessing the memory. The instructions may be configured to estimate a feature value of the external sound signal, calculate the initial value of the voice parameter used to perform beamforming based on the external sound signal received by the plurality of microphones, determine whether to store the calculated initial value according to the feature value, determine which one of the calculated initial value or an initial value stored in the memory used according to the feature value, and obtain a target voice parameter.