An online conversation management apparatus includes a processor. The processor acquires, across a network, reproduction environment information from at least one terminal that reproduces a sound image via a reproduction device. The reproduction environment information is information of a sound reproduction environment of the reproduction device. The processor acquires azimuth information. The azimuth information is information of a localization direction of the sound image with respect to a user of the terminal. The processor performs control for reproducing a sound image of each terminal based on the reproduction environment information and the azimuth information.

Electronic glasses track head movement of a user with respect to a location in empty space on top of a physical object. One or more processors process sound that externally localizes as binaural sound to the location in empty space on top of the physical object. Speakers in the electronic glasses provide the binaural sound to the user.

An audio cancellation system includes a voice enabled computing system that is connected to an audio output device using a wired or wireless communication network. The voice enabled computing device can provide media content to a user and receive a voice command from the user. The connection between the voice enabled computing system and the audio output device introduces a time delay between the media content being generated at the voice enabled computing device and the media content being reproduced at the audio output device. The system operates to determine a calibration value adapted for the voice enabled computing system and the audio output device. The system uses the calibration value to filter the user's voice command from a recording of ambient sound including the media content, without requiring significant use of memory and computing resources.

The present technology relates to a signal processing device, method, and program that can improve encoding efficiency.

A signal processing device includes: an acquisition unit that acquires reverb information including at least one of space reverb information specific to a space around an audio object or object reverb information specific to the audio object and an audio object signal of the audio object; and a reverb processing unit that generates a signal of a reverb component of the audio object on the basis of the reverb information and the audio object signal. The present technology can be applied to a signal processing device.

One example includes a digital twin of a microphone array. The digital twin acts as a digital copy of a physical microphone array. The digital array allows the microphone array to be analyzed, simulated and optimized. Further, the microphone array can be optimized for performing sound quality operations such as noise suppression and speech intelligibility.

One example method includes generating sound information regarding sound sources in an environment. The sound information is generated by separating and localizing the sound sources. The sound information is then presented as guidance to a user. The guidance may be presented graphically in a user interface, haptically, or in another manner.

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.

Systems and methods of presenting an output audio signal to a listener located at a first location in a virtual environment are disclosed. According to embodiments of a method, an input audio signal is received. A first intermediate audio signal corresponding to the input audio signal is determined, based on a location of the sound source in the virtual environment, and the first intermediate audio signal is associated with a first bus. A second intermediate audio signal is determined. The second intermediate audio signal corresponds to a reverberation of the input audio signal in the virtual environment. The second intermediate audio signal is determined based on a location of the sound source, and further based on an acoustic property of the virtual environment. The second intermediate audio signal is associated with a second bus. The output audio signal is presented to the listener via the first and second buses.

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.

A method is provided that includes: obtaining initial acoustic characteristics of a spatial sound field corresponding to a venue; adjusting the initial acoustic characteristics based on at least one adjustment parameter to obtain adjusted acoustic characteristics; and applying the adjusted acoustic characteristics to audio data to obtain audio data with sound effect restored.