A sound signal processing method includes: obtaining a plurality of sound signals respectively collected by a plurality of microphones arranged in a space; adjusting respective levels of the plurality of sound signals in accordance with respective positions of the plurality of microphones; mixing the plurality of sound signals having the adjusted respective levels to thereby obtain a mixed signal; and generating a reflected sound by using the obtained mixed signal.

A mobile electronic device (device) operating with other devices form a group for wireless coordinated playout of audio media content. The processor performs operations that determine a common time sync shared with the other devices, and determine a timing of occurrence of a sound transient sensed relative to the common time sync. The operations receive timing reports from the other devices, where each of the timing reports indicates a timing of occurrence of the sound transient sensed at a respective one of the other devices relative to the common time sync. The sound transient sensed by the devices is generated at a preferred listening location. The operations coordinate timing of playout of audio media content by the group of devices responsive to comparison of the timing of occurrence of the sound transient sensed by the microphone of the device and the timing of occurrences indicated by the timing reports.

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.

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.

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.

Systems and methods are disclosed for networked audio automixing using array microphones and an aggregator unit that participate in making a common gating decision to determine which channels to gate on and off. Through the use of such a network of array microphones having the capability to generate submix audio signals and reduced bandwidth metrics, as well as AEC processing capability, array microphone lobe selection can be enhanced while maximizing signal-to-noise ratio, increasing intelligibility, and increasing user satisfaction.

An example may include detecting, via a controller, one or more microphones and one or more speakers in an area, measuring, via the one or more microphones, an initial frequency response of an audio signal generated by the one or more speakers inside the area and generating an initial room performance rating, comparing the initial frequency response to a target frequency response, creating audio compensation values to apply to the one or more speakers based on the comparison, and applying the audio compensation values to the one or more speakers.

The present embodiments generally relate to enabling participants in an online gathering with networked audio to use a cancelling auralizer at their respective locations to create a common acoustic space or set of acoustic spaces shared among subgroups of participants. For example, there are a set of network connected nodes, and the nodes can contain speakers and microphones, as well as participants and node mixing-processing blocks. The node mixing-processing blocks generate and manipulate signals for playback over the node loudspeakers and for distribution to and from the network. This processing can include cancellation of loudspeaker signals from the microphone signals and auralization of signals according to control parameters that are developed locally and from the network. A network block can contain network routing and processing functions, including auralization, synthesis, and cancellation of audio signals, synthesis and processing of control parameters, and audio signal and control parameter routing.

A method and a device for operating an arrangement of sound transducers according to the wave-field synthesis principle. In order to supply an extended audience region with the same signal, the same signal content is generated by at least two virtual sound sources, which are arranged such that the wavefronts thereof are directed only onto a part audience area, rather than generating only a single beam extending over the entire audience area. The wavefronts of the distributed virtual sound sources add up vectorially in the plane of the arrangement of sound transducers, whereby the effectiveness of the sound generation is increased.

Example techniques may involve performing aspects of a spatial calibration. An example implementation may include detecting a trigger condition that initiates calibration of a media playback system including multiple audio drivers that form multiple sound axes, each sound axis corresponding to a respective channel of multi-channel audio content The implementation may also include causing the multiple audio drivers to emit calibration audio that is divided into constituent frames, the multiple sound axes emitting calibration audio during respective slots of each constituent frame. The implementation may further include recording the emitted calibration audio. The implementation may include causing delays for each sound axis of the multiple sound axes to be determined, the determined delay for each sound axis based on the slots of recorded calibration audio corresponding to the sound axes and causing the multiple sound axes to be calibrated.