Abstract: Example techniques involve determining a direction of an NMD. An example implementation includes causing two audio drivers to output audio content, where the audio drivers are positioned at a known distance apart in a given azimuthal plane and recording the audio content output by the two audio drivers. The implementation also includes causing a computing device to identify a direction of the NMD relative to a center point of the two audio drivers in the given azimuthal plane. Identifying the direction of the NMD may involve determining a range of possible azimuthal angles between the NMD and the center point, the range of possible angles dependent upon the known distance between the two audio drivers and a phase offset between the two audio drivers indicated by the recorded audio content output. The implementation further includes causing a playback device to adjust calibration based on the identified direction of the NMD.

Abstract: An example embodiment may involve determining that a client device (such as a smartphone, tablet, or in-automobile audio device) is in an automobile and that the client device has access to a playlist of audio content. Possibly based on the client device being in the automobile and having access to the playlist of audio content, the client device may request a stream of the audio content. As a consequence of making the request, the client device may receive the stream of the audio content and begin audible playout of the audio content.

Abstract: Methods and systems for audio source separation in real-time are described. In an embodiment, the present disclosure describes reading and decoding an audio source into PCM samples, fragmenting Pulse Code Modulation (PCM) samples into fragments, transforming fragments into spectrograms, performing audio source separation using a deep neural network (DNN) to generate an estimated magnitude spectrogram of the component(s) of the audio source, reconstructing the estimated time domain component signals, and streaming the component signals to a playback engine. In an embodiment, a semantic equalizer graphical user allows for real-time mixing of individual component signals.

Abstract: An acoustic echo cancellation (AEC) system that detects and compensates for differences in delay times between the AEC system and a set of wireless speakers. The filter coefficients used for AEC are adjusted based on the determined delay time to correct for frequency domain signal rotation.

Abstract: The present disclosure discloses a signal sending method and device. The method includes: receiving, by a base station, an uplink pilot signal sent by authorized user equipment, and determining a direction vector parameter and a first channel fading parameter of a channel calculating, according to the direction vector parameter and the first channel fading parameter, a first signal beamformer parameter, determining a transmission area of an artificial noise signal according to the direction vector parameter, and calculating a second signal beamformer parameter; and processing a to-be-transmitted signal by using the first signal beamformer parameter and the second signal beamformer parameter, and transmitting the processed signal. In this way, in a non-target direction, energy leakage of the secrecy signal to the authorized user equipment is relatively small, and transmitted artificial noise signals are concentrated in an area with a relatively high secrecy signal leakage risk.

Abstract: A seat integrating a loudspeaker system includes a seat having a seat body with a horizontally oriented seat bottom and a seat back secured thereto. The seat back includes an upper end and a lower end, wherein the lower end is positioned adjacent the seat bottom. The loudspeaker system includes a sound assembly mounted within the seat for selective movement between a storage orientation in which the sound assembly is housed and hidden within a support housing mounted along the seat and a use orientation extending from the support housing in which the sound assembly is positioned for use.

Abstract: The present disclosure provides methods, devices and computer program products for encoding and decoding of a vector of parameters in an audio coding system. The disclosure further relates to a method and apparatus for reconstructing an audio object in an audio decoding system. According to the disclosure, a modulo differential approach for coding and encoding a vector of a non-periodic quantity may improve the coding efficiency and provide encoders and decoders with less memory requirements. Moreover, an efficient method for encoding and decoding a sparse matrix is provided.

Abstract: In some embodiments, a microphone system includes a multi-bit delta-sigma modulator configured to be coupled to a microphone and configured to covert an output of the microphone into a first digital signal with a multi-bit resolution at a first output of the multi-bit delta-sigma modulator. The microphone system also includes a digital noise shaper coupled to the first output of the multi-bit delta-sigma modulator, and configured to convert a multi-bit digital input of the digital noise shaper into a one-bit digital signal.

Abstract: A transfer function calculation unit is configured to calculate a transfer function from a sound source installed in a predetermined target direction to each microphone of a microphone array, and a determination unit is configured to determine whether or not the microphone array is normal on the basis of a difference amount between a transfer function to each microphone and a predetermined ideal transfer function to each microphone.

Abstract: An electric fader drive unit includes a drive portion, a fader position control portion, and an electric fader control portion. The drive portion drives an electric fader by supplying drive power to the electric fader. The fader position control portion indicates a target position of the electric fader. The electric fader control portion receives an indication of the target position from the fader position control portion, and, when controlling the drive portion to move the electric fader to the target position, moves the electric fader to the target position by changing the drive power maintained at preparation drive power less than the minimum drive power that is the lowest drive power required for moving the electric fader to a predetermined drive power not less than the minimum drive power.

Abstract: A mixing console includes: a plurality of operating regions provided by dividing an operation panel of the mixing console into a plurality of regions, wherein a plurality of channels are allocated to each of the plurality of operating regions. Each of the operating regions includes each of channel selectors, respectively. Each of the channel selectors is configured to select one of the plurality of channels allocated to the operating region corresponding to the channel selector as a selected channel, and to set a plurality of parameters for signal processing on audio signals of the selected channel. Thus, a plurality of selected channels can be selected, and values of the parameters of the plurality of selected channels can be adjusted promptly and in real-time.

Abstract: An apparatus comprising: at least one element generator configured to determine at least one hearing aid display element; at least one element locator configured to determine a location of the at least one hearing aid display element based on a location of at least one hearing aid compatible coil; and a display configured to display the at least one hearing aid compatible display element based on the location determined.

Abstract: A prosthetic support, including a structure configured to apply a clamping force to a head of a recipient while extending about a back of at least one of a head or neck of the recipient such that output generated by a device supported by the structure is directed into skin of the recipient at a location behind an ear canal of the recipient that covers the mastoid bone of the recipient.

Abstract: Disclosed herein, among other things, are methods and apparatus for own-voice sensing in hearing assistance devices. One aspect of the present subject matter includes an in-the-ear (ITE) hearing assistance device adapted to process sounds, including sounds from a wearer's mouth. According to various embodiments, the device includes a hollow plastic housing adapted to be worn in the ear of the wearer and a differential sensor mounted to an interior surface of the housing in an ear canal of the wearer. The differential sensor includes inlets located within the housing and the differential sensor is configured to improve speech intelligibility of sounds from the wearer's mouth, in various embodiments.

Abstract: A method, including determining a change in an actuator impedance based on a change in an electrical property of a system of which the actuator is apart, and determining one or more system characteristics based on the change in the actuator impedance.

Abstract: There is provided encoding and decoding methods for encoding and decoding of object based audio. An exemplary encoding method includes inter alia calculating M downmix signals by forming combinations of N audio objects, wherein M?N, and calculating parameters which allow reconstruction of a set of audio objects formed on basis of the N audio objects from the M downmix signals. The calculation of the M downmix signals is made according to a criterion which is independent of any loudspeaker configuration.

Abstract: In an impedance converter using an electron tube as an active element, output impedance can be made sufficiently low, and the number of circuit elements therefor is decreased and a circuit configuration therefor is made simple. Provided is an impedance converter having an electron tube cathode-follower connected. The impedance converter includes a bias diode that provides a bias voltage to a cathode of the electron tube, high resistance elements that provide a voltage of the bias diode to a grid of the electron tube, a load circuit connected to the electron tube, and a complementary emitter output circuit including two transistors, respective bases of which are connected to one end and the other end of the bias diode.

Abstract: An electronic device includes a housing and a user interface. The electronic device also includes an acoustic detector and one or more processors operable with the acoustic detector. The one or more processors can receive, from the user interface, user input corresponding to an operation of the electronic device. The one or more processors can then optionally initiate a timer in response to receiving the user input and monitor the acoustic detector for a predefined acoustic marker, one example of which is acoustic data indicating detection of one or more finger snaps. Where the one or more finger snaps occur prior to expiration of the timer, the one or more processors can perform the operation of the electronic device. Otherwise ignore the user input. The acoustic confirmation of user input helps to eliminate false triggers, thereby conserving battery power and extending run time.

Abstract: In general, techniques are described for obtaining one or more first vectors describing distinct components of a soundfield and one or more second vectors describing background components of the soundfield, both the one or more first vectors and the one or more second vectors generated at least by performing a transformation with respect to a plurality of spherical harmonic coefficients.

Abstract: The sound collection and reproduction system reproduces comprise: a microphone array selection unit which selects a microphone arrays; an area sound collection unit which collects sounds of all areas by using the microphone arrays; an area sound selection unit which selects an area sound of an area corresponding to a specified listening position, and area sounds of surrounding areas of this area corresponding to a listening direction, in accordance with a sound reproduction environment; an area volume adjustment unit which adjusts a volume of each area sound unit in accordance with a distance from the specified listening position; and a stereophonic sound processing unit which performs a stereophonic sound process, for each area sound to which volume adjustment has been performed by the area volume adjustment unit, by using a transfer function corresponding to a sound reproduction environment.