Abstract: A Bluetooth device controlling a plurality of wireless audio devices, includes: a wireless communication circuit, to receive a wireless audio device identifier from each wireless audio device; a sound processing circuit to apply independent audio effects on a main audio stream such that the sound processing circuit outputs as many modified audio streams as the wireless communication circuit has received identifiers of wireless audio devices, on the basis of the characteristics of each wireless audio device; an allocation circuit to allocate each modified audio stream to a respective wireless audio device; and a transmission circuit to wirelessly transmit through a Bluetooth usage each modified audio stream to the respective wireless audio device for emission by the wireless audio devices.

Abstract: An acoustic system can include an acoustic processor that is configured to analyze and output acoustic signals to each of first and second loudspeakers corresponding to each of first and second acoustic zones of a shared acoustic space. The system can determine a first measure of loudness associated with a first acoustic signal. The system can be configured to output the first acoustic signal as sound on the first loudspeaker in the first acoustic zone. The system can determine a second measure of loudness associated with a second acoustic signal. The system can be configured to output the second acoustic signal as sound on the second loudspeaker in the second acoustic zone. The system can also modify the second acoustic signal based on the first acoustic signal.

Abstract: An apparatus for decoding an encoded multichannel signal includes: a base channel decoder for decoding an encoded base channel to obtain a decoded base channel; a decorrelation filter for filtering at least a portion of the decoded base channel to obtain a filling signal; and a multichannel processor for performing a multichannel processing using a spectral representation of the decoded base channel and a spectral representation of the filling signal, wherein the decorrelation filter is a broad band filter and the multichannel processor is configured to apply a narrow band processing to the spectral representation of the decoded base channel and the spectral representation of the filling signal.

Abstract: Techniques for improving microphone noise suppression are provided. As wind noise may disproportionately impact a subset of microphones, a method for processing audio data using two adaptive reference algorithm (ARA) paths in parallel is provided. For example, first ARA processing performs noise cancellation using all microphones, while second ARA processing performs noise cancellation using only a portion of the microphones. As the first ARA processing and the second ARA processing are performed in parallel, beam merging can be performed using beams from the first ARA, the second ARA, and/or a combination of each. In addition, beam merging can be performed using beam sections instead of individual beams to further improve performance and reduce attenuation to speech.

Abstract: In some embodiments, a system comprises a host computing device and an audio device including at least one speaker and a plurality of microphones, the audio device being wirelessly and communicatively coupled to the host computing device. The host computing device can include one or more processors and one or more machine-readable, non-transitory storage mediums that include instructions configured to cause the one or more processors of the host computing device to perform operations including: receiving user environment data by one or more sensors of the host computing device; determining a characterization profile of a surrounding environment of the user based on the user environment data; and sending the characterization profile to the audio device, the characterization profile configured to cause the audio device to adapt an audio cardioid pattern of the plurality of microphones on the audio device based on the characterization profile.

Abstract: Array microphone systems and methods that can automatically focus and/or place beamformed lobes in response to detected sound activity are provided. The automatic focus and/or placement of the beamformed lobes can be inhibited based on a remote far end audio signal. The quality of the coverage of audio sources in an environment may be improved by ensuring that beamformed lobes are optimally picking up the audio sources even if they have moved and changed locations.

Abstract: A method of providing virtual sound in an electric vehicle, includes: collecting vehicle driving information for outputting the virtual sound; determining characteristics of the virtual sound based on the collected vehicle driving information; generating a virtual effect signal for outputting the virtual sound, based on the determined characteristics; generating a sound signal containing the characteristics, based on the virtual effect signal; collecting the generated sound signal to obtain a final sound signal corresponding to a vehicle rating selected according to setting information based on the preset vehicle rating and a sound correction algorithm; and outputting the virtual sound corresponding to the selected vehicle rating, according to the final sound signal.

Abstract: A system comprises a sound source to output an audio signal to a plurality of speakers arranged in a plurality of locations in a vehicle. A laser device is disposed on an object representing a human head placed in a seat of the vehicle to scan the locations of the speakers. A microphone is disposed in each ear of the object. A controller routes the audio signal to the speakers one speaker at a time, receives audio signals received via the microphones, and compiles binaural acoustic data for the vehicle based on the audio signals received via the microphones. The controller receives scan data from the laser device and generates geometric data for the vehicle based on the scan data. The controller generates head-related transfer functions (HRTFs) for the object based on the binaural acoustic data and the geometric data for the vehicle.

Abstract: 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.

Abstract: An automobile audio system for an automobile includes an audio processing unit, a set of external microphones to record external sound and a set of internal loudspeakers to internally reproduce audio. Furthermore, the automobile audio system includes a transmission switching mechanism. The transmission switching mechanism is manually configurable between at least two manual transmission configurations including a manually activated configuration wherein the set of external microphones is communicatively coupled to the set of internal loudspeakers such that the external sound is reproducible by at least a subset of the set of internal loudspeakers and a manually deactivated configuration wherein the set of external microphones is communicatively decoupled to the set of internal loudspeakers.

Abstract: Example techniques involve a calibration state variable. An example implementation receives, via a network interface, an indication that the first playback device is calibrated. Based on receiving the indication that the first playback device is calibrated, the example implementation updates a calibration state variable to indicate that the first playback device is calibrated, wherein the calibration state variable is stored in the data storage. The example implementation sends, via the network interface, an indication of the updated calibration state variable to a second device.

Abstract: A computer may segment a noisy audio signal into audio frames and execute a deep neural network (DNN) to estimate an instantaneous function of clean speech spectrum and noisy audio spectrum in the audio frame. This instantaneous function may correspond to a ratio of an a-priori signal to noise ratio (SNR) and an a-posteriori SNR of the audio frame. The computer may add estimated instantaneous function to the original noisy audio frame to output an enhanced speech audio frame.

Abstract: A wireless controller that allows a musician to wirelessly switch between audio effect presets, loops, and/or songs which are stored in a specialized audio/instrument cable or adapter. The wireless controller can be located in the same place as (or incorporated into) a digital guitar tuner on a guitar headstock allowing the musician to operate the wireless controller in a familiar location while switching between audio effect presets, loops, and/or songs by tapping on small buttons.

Abstract: Presented herein are techniques for generating a combinatory microphone signal from sounds captured at a microphone array. More specifically, sound signals captured by a microphone array are used to generate first and second directional signals. A cross-power signal is computed from the first and second directional signals. The cross-power signal is converted into an amplitude domain output signal, and a phase of the amplitude domain output signal is reconstructed in order to generate a combinatory microphone signal that is useable for subsequent sound processing operations.

Abstract: A sound quality evaluation method and a sound quality evaluation system using same are provided. The sound quality evaluation system records playback of a test audio file on a plurality of playback devices to generate a plurality of pieces of audio data, and divides the audio data into a plurality of frequency bands. The sound quality evaluation system calculates the frequency bands to obtain a plurality of evaluation scores of the playback devices. The sound quality evaluation system captures sound quality ranking information corresponding to the playback devices from a reference source, and adjusts the evaluation scores according to the sound quality ranking information, to further obtain a reference model. The sound quality evaluation system adjusts correspondingly evaluation scores of a plurality of to-be-tested playback devices according to the reference model, to obtain sound quality ranking information of the to-be-tested playback devices.

Abstract: An audio processing system includes a microphone array, a speech detection system, and a neural network noise reduction module. The microphone array includes at least two microphones and provides an audio signal from an environment surrounding the microphone array. The speech detection system receives the audio signal, and processes the audio signal to a) detect that a first user is speaking, b) determine a first direction relative to the audio array when the first user is located at a first location within the environment, and c) provide beamforming processing on the audio signal in the first direction, and to provide a processed audio signal based upon the beamforming processing. The neural network noise reduction module reduces noise in the processed audio signal.

Abstract: A method provides binaural sound to a person through electronic earphones. The binaural sound localizes to a sound localization point (SLP) in empty space that is away from but proximate to the person. When an event occurs, the binaural sound switches or changes to stereo sound, to mono sound, or to altered binaural sound.

Abstract: Three-dimensional (3D) audio can be presented during an augmented reality, mixed reality, or virtual reality simulation to notify a user that a real-world object is approaching or has crossed a boundary for an interaction space within which the user is to interact with the simulation. The 3D audio may indicate the current location of the intrusion or boundary while still leaving the user immersed in the simulation.

Abstract: A method for noise filtering including receiving directional data corresponding to the relative directions of at least one noise source and at least one target audio source; capturing noise data from the at least one noise source; capturing target audio data from the at least one target audio source; using the directional data to filter the noise data from the target audio data; and outputting filtered target audio.

Abstract: A device includes one or more processors configured to obtain audio signals representing sound captured by at least three microphones and determine spatial audio data based on the audio signals. The one or more processors are further configured to determine a metric indicative of wind noise in the audio signals. The metric is based on a comparison of a first value and a second value. The first value corresponds to an aggregate signal based on the spatial audio data, and the second value corresponds to a differential signal based on the spatial audio data.