An communication system supports communication paths within an environment by receiving speech signals of a speaker and playing it back for one or more listeners. Signal processing tasks are split into a microphone related part and into a loudspeaker related part. A sound processing system suitable for use in an environment having multiple acoustic zones includes a plurality of microphone communication instances coupled and a plurality of loudspeaker instances.

A wireless microphone system with a first wireless microphone comprising a first microphone element arranged to generate a first audio signal, wherein the first microphone is arranged to generate a first output signal that comprises the first audio signal and information identifying the first wireless microphone; a second wireless microphone with a second microphone element arranged to generate a second audio signal, wherein the second microphone is arranged to generate a second output signal that comprises the second audio signal and information identifying the second wireless microphone; and a monitoring device with a display, wherein the monitoring device is arranged to receive the first and second output signals and to use the display to generate a visualisation of at least one aspect of the first and second audio signals, and to associate each visualisation with a representation of the information identifying the corresponding first or second wireless microphone.

A power distribution system including a first and second speaker is provided. The first speaker includes (1) a first input receiving a first power signal; (2) a second input receiving a second power signal phase shifted 120 degrees from the first signal; (3) a third input receiving a third power signal phase shifted 120 degrees from both the first and second signal; (4) a first output transmitting the third power signal; (5) a second output transmitting the first power signal; (6) a third output transmitting the second power signal; and (7) a first load coupled to the first and second input. The second speaker includes (1) a first input receiving the third power signal; (2) a second input receiving the first power signal; (3) a third input receiving the second power signal; and (4) a second load coupled to the first and second input.

In order to estimate the trajectory of a moving wave source, this trajectory estimation device (10) comprises: an acquisition unit (11) that acquires wave motion data based on wave motion detected by a plurality of sensors (100); a generation unit (12) that generates a spectrogram using the wave motion data; an extraction unit (13) that extracts a Doppler shift from the spectrogram; a selection unit (14) that selects, as a sensor pair, two sensors that satisfy a preset selection condition pertaining to the Doppler shift; and an estimation unit (15) that estimates the trajectory of a wave source, which is the source generating the wave motion, on the basis of the positional relationship between the sensors constituting the sensor pair and the relationship of the Doppler shift between the two sensors constituting the sensor pair.

Method and apparatus for automatic gain control utilizing sound source position information in a shared space having a plurality of microphones and a plurality of sound sources. Sound signals are received from the microphones. One or more processors locate position information corresponding to each of the sound sources. The processor(s) determine the distance to each of the sound sources from each of the microphones. The processor(s) define a predetermined gain weight adjustment for each of the microphones. The processor(s) apply the defined weight adjustments to the microphones to achieve a consistent volume of the desired plurality of sound sources. The processor(s) maintain a consistent ambient sound level regardless of the position of the sound sources and the applied gain weight adjustments. The processor(s) output a summed signal of the sound sources at a consistent volume with a constant ambient sound level across the plurality of sound source positions.

Determination of a composite acoustic parameter value for a headset is presented herein. A directionally enhanced audio signal is generated based on audio signals from an acoustic sensor array and a spatial signal enhancement filter that is directed for enhancement of a sound source. A SNR improvement value is determined based on a SNR value of the directionally enhanced audio signal and a SNR value of an audio signal from an acoustic sensor of the acoustic sensor array. The SNR improvement value is input into a model that maps SNR improvement values to spatial acoustic parameters to determine a spatial acoustic parameter. A temporal acoustic parameter is determined based on the audio signals. The composite acoustic parameter value is determined based on the spatial acoustic parameter and a temporal acoustic parameter value. Audio content presented to a user is adjusted based in part on the composite acoustic parameter value.

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.

Audio device transducer arrays and associated systems and methods are disclosed herein. In some examples, the audio device can comprise a housing including a baffle and an acoustically reflective plate spaced apart from and facing at least partially toward the baffle, such that the baffle and the plate define an opening therebetween. The audio device can further comprise an array of electroacoustic transducers disposed within or surrounded by the baffle and configured to emit acoustic waves toward the opening. The array of transducers can be configured to produce acoustic waves having a frequency of about 1.5 kilohertz (kHz) or greater.

Head-wearable apparatus to generate binaural audio content includes a first stem coupled to a first microphone housing that encases first front microphone and first rear microphone that generates acoustic signals, respectively. First microphone housing includes a first front port that faces downward and a first rear port that faces backwards. Apparatus includes second stem coupled to second microphone housing that encases second front microphone and second rear microphone that generate acoustic signals, respectively. Second microphone housing includes second front port that faces downward and second rear port that faces backwards. Apparatus includes binaural audio processor that includes beamformer and storage device. Beamformer generate first beamformer signal based on acoustic signals from first front microphone and first rear microphone, and second beamformer based on acoustic signals from second front microphone and second rear microphone. Storage device stores first and second beamformer signals as a two-channel file.

Methods, apparatus, systems, and articles of manufacture are disclosed. An example apparatus includes instructions that, when executed, cause processor circuitry to at least: obtain audio data channels produced by physical capture devices; calculate: a first plurality of angles corresponding to ones of the physical capture devices, the first plurality of angles to describe how sound produced by an audio source arrives to the physical capture devices, a location of the audio source based on the first plurality of angles, a location of virtual capture arrays, and a second plurality of angles to describe how sound produced by the audio source would arrive to the virtual capture arrays; interpolate between two angles from either of the first plurality or the second plurality of angles; and render a binaural audio signal based on the audio data channels and the interpolated angle.