An example method of operation may include initializing a microphone array in a defined space to receive one or more sound instances based on a preliminary beamform tracking configuration, detecting the one or more sound instances within the defined space via the microphone array, modifying the preliminary beamform tracking configuration, based on a location of the one or more sound instances, to create a modified beamform tracking configuration, and saving the modified beamform tracking configuration in a memory of a microphone array controller.

A method (S100) for optimizing speech pickup in a speakerphone system, wherein the speakerphone system comprises a microphone system placed in a specific configuration, wherein the method comprising receiving (S10) acoustic input signals (12) by the microphone system, processing (S20) said acoustic input signals (12) by using an algorithm (100) for focusing and steering a selected target sound signal towards a desired direction, and transmitting (S30) an output signal (13) based on said processing.

Audio systems and methods are provided that receive a sound management input signal as a reference signal to remove related content from a microphone signal in, e.g., an automotive hands-free system. The sound management signal may provide an acoustic augmentation to reduce, enhance, or create an acoustic effect, e.g., of an engine, motor, or other operating components. A signal processor receives the sound management signal and the microphone signal, and reduces or removes the sound management signal components from the microphone signal.

The disclosed computer-implemented method may include receiving a signal for each channel of an audio transducer array on a wearable device. The method may also include calculating a beamformed signal for each beam direction of a set of beamforming filters for the wearable device. Additionally, the method may include classifying a first beamformed signal from the calculated beamformed signals into a first class of sound and a second beamformed signal from the calculated beamformed signals into a second class of sound. The method may also include adjusting, based on the classifying, a gain of the first beamformed signal relative to the second beamformed signal. Furthermore, the method may include converting the beamformed signals into spatialized binaural audio based on a position of a user. Finally, the method may include transmitting the spatialized binaural audio to a playback device. Various other methods, systems, and computer-readable media are also disclosed.

A device includes a memory configured to store untransformed ambisonic coefficients at different time segments. The device also includes one or more processors configured to obtain the untransformed ambisonic coefficients at the different time segments, where the untransformed ambisonic coefficients at the different time segments represent a soundfield at the different time segments. The one or more processors are also configured to apply one adaptive network, based on a constraint, to the untransformed ambisonic coefficients at the different time segments to generate transformed ambisonic coefficients at the different time segments, wherein the transformed ambisonic coefficients at the different time segments represent a modified soundfield at the different time segments, that was modified based on the constraint.

This disclosure describes a conferencing device with beamforming and echo cancellation that includes: a microphone array that further comprises a plurality of microphones oriented to develop a corresponding plurality of microphone signals; a processor configured to execute the following steps: (1) performing a beamforming operation to combine the plurality of microphone signals from the microphone array into a plurality of combined signals, (2) performing an acoustic echo cancellation operation on the plurality of combined signals to generate a plurality of combined echo cancelled signals, (3) receiving with a voice activity detector the far end signal as an input, (4) selecting one or more of the combined echo cancelled signals for transmission to the far end where a signal selector uses the far end signal as information to inhibit the signal selector from changing the selection of the combined echo cancelled signals while only the far end signal is active.

A method of presenting audio comprises: identifying a first ear listener position and a second ear listener position in a mixed reality environment; identifying a first virtual sound source in the mixed reality environment; identifying a first object in the mixed reality environment; determining a first audio signal in the mixed reality environment, wherein the first audio signal originates at the first virtual sound source and intersects the first ear listener position; determining a second audio signal in the mixed reality environment, wherein the second audio signal originates at the first virtual sound source, intersects the first object, and intersects the second ear listener position; determining a third audio signal based on the second audio signal and the first object; presenting, to a first ear of a user, the first audio signal; and presenting, to a second ear of the user, the third audio signal.

The aircraft intercom employs beam steerable microphone arrays and speaker arrays deployed in each of a plurality of zones within the aircraft. A speech recognizer generates recognition text from utterances picked up by the microphone arrays. A direction control processor analyzes the arrival time of signals from the microphone arrays to identify utterance location, which is then used to control the beam direction of at least one steerable speaker array. A dialogue manager processor coupled to the speech recognizer and to each of the plurality of microphone arrays and plurality of speaker arrays responds to a set of predefined keywords to selectively route a voiced communication from a first selected zone to a second selected zone using the microphone array and speaker array disposed in each of the first and second selected zones.

There is provided an imaging apparatus including: a sound processing unit that performs processing with respect to a sound signal input through a microphone; and a control unit that separately controls a parameter related to processing of the sound signal at a time of recording of a captured image when the sound data processed by the sound processing unit is recorded together with image data obtained by imaging through an imaging unit and a time of recording a sound memo when the sound data processed by the sound processing unit is recorded as the sound memo.

A signal processing apparatus includes a neural network (“NN”), a sorting unit, and a spatial covariance matrix calculation unit. The NN converts a mixed signal, in which sounds of a plurality of sound sources input by a plurality of channels are mixed, into a separated signal separated into a signal for each sound source as a signal in a time domain as it is and outputs the separated signal. The sorting unit sorts, for the separated signal of each channel output from the NN, the separated signal of each channel such that the plurality of sound sources of a plurality of the separated signals are aligned among the plurality of channels. The spatial covariance matrix calculation unit calculates a spatial covariance matrix corresponding to each sound source in accordance with the separated signal for each channel output from the sorting unit and sorted.