Abstract: A method performed by a first hearing device comprising microphone(s) configured to generate a first input signal, a communication unit configured to receive a second input signal from a second hearing device, an output unit, and a processor, the method comprising: generating a first intermediate signal including or based on a first weighted combination of the first input signal and the second input signal, wherein the first weighted combination is based on a first gain value and/or a second gain value; and generating an output signal for the output unit based on the first intermediate signal; wherein one or both of the first gain value and the second gain value are determined in accordance with an objective of making a power of the first input signal and a power of the second input signal differ by a preset power level difference greater than 2 dB in the weighted combination.

Abstract: An audio signal processing apparatus includes a first bus, a second bus, and a controller that assigns a first input channel to the first bus, assigns a second input channel to the second bus, receives an amount of feed from the first input channel to the first bus, and receives an amount of feed from the second input channel to the second bus.

Abstract: A computer-implemented method comprises identifying an audio playback device configured to reproduce audio content based on an audio input signal, wherein the audio playback device has a previously measured characteristic frequency response; retrieving the previously measured characteristic frequency response associated with the audio playback device; generating, based on the previously measured characteristic frequency response and a preferred frequency response curve for a plurality of users, a frequency response filter that modifies the previously measured characteristic frequency response of the audio playback device to a modified frequency response; processing the audio input signal with the frequency response filter to generate a filtered audio signal that differs in relative amplitude from the audio input signal in one or more frequency bands; and causing the audio playback device to play back the filtered audio signal.

Abstract: The description relates to rendering directional sound. One implementation includes receiving directional impulse responses corresponding to a scene. The directional impulse responses can correspond to multiple sound source locations and a listener location in the scene. The implementation can also include encoding the directional impulse responses to obtain encoded departure direction parameters for individual sound source locations. The implementation can also include outputting the encoded departure direction parameters, the encoded departure direction parameters providing sound departure directions from the individual sound source locations for rendering of sound.

Abstract: Systems and methods for controlling audio output in a telematics system are provided. In on method a telematics device determines that a parameter has exceeded a first threshold and sends an automatic volume control trigger message to an audio output device. The telematics device determines that a second parameter has exceeded a second threshold and sends an alert message to the audio output device. The audio output device performs an automatic volume control procedure then plays back and audible alert message.

Abstract: Embodiments of the present invention provide a vehicle privacy system (700), comprising audio input means (130, 190, 720) for receiving an external audio signal (725) indicative of audio from within a vehicle (900), audio source means (710, 910) for receiving the external audio signal (725) and determining an output audio signal (735) based thereon for reducing an external intelligibility of speech within the vehicle (900), and audio output means (145, 146, 147, 730, 920) for receiving the output audio signal (735) and outputting audio (925) corresponding thereto to be at least partly audible external to the vehicle (900).

Abstract: An audio apparatus comprises a receiver (201) receiving data describing an audio scene. The data comprises audio data for a set of audio elements corresponding to audio sources in the scene and further includes metadata comprising at least an audio rendering property indicator for a first audio element of the set of audio elements. A first renderer (205) renders audio elements by generating a first set of audio signals for a set of loudspeakers and a second renderer (207) renders audio elements by generating a second set of audio signals for a headphone. Further, a selector (209) is arranged to select between the first renderer and the second renderer for rendering of at least a first part of the first audio element in response to the first audio rendering property indicator. The approach may for example provide improved virtual reality experiences using loudspeakers and headphone hybrid rendering.

Abstract: An electronic module is provided. The electronic module includes a first transducer and a second transducer. The first transducer is configured to radiate a first ultrasonic wave. The second transducer is configured to radiate a second ultrasonic wave. The first transducer and the second transducer are disposed on noncoplanar surfaces.

Abstract: An integration unit IU for tuning an OEM sound system augmented with aftermarket components, such as an amplifier or a subwoofer. The integration unit is contained in a small housing that can be installed inconspicuously in most vehicles without removing the original head unit. The unit may comprise a DSP-based active line output converter that is configured to automatically detect the frequency response and phase of the OEM head unit. The unit performs various algorithms to detect and analyze crossovers, EQ, time delay, and allpass filters and then matches left and right audio channels. The IU may allow for low bass correction detecting and defeating the factory high pass filter. Still further, the IU can correct the various filters to ensure a clean, uncolored audio signal for the augmented components.

Abstract: An audio system includes an equatorial acoustic sensor array (EASA) that may be coupled to an object. The audio system is configured to detect, via the EASA, signals corresponding to a portion of a sound field in a local area. The detected signals are converted into a plurality of corresponding abstract representations that describe the portion of the sound field. Effects of scattering of the object are removed from the abstract representations to create adjusted abstract representations. A set of spherical harmonic (SH) coefficients is determined using the adjusted abstract representations. The set of SH coefficients describe an entirety of the sound field. And the set of SH coefficients and head related transfer functions of a user are used for binaural rendering of the reconstructed sound field to the user.

Abstract: One embodiment provides a computer-implemented method that includes acquiring, via at least one microphone, sound pressure data at one or more discrete frequencies obtained from a frequency response of a loudspeaker in a room. The sound pressure data is input into an artificial intelligence (AI) model that analyses and processes information, and that incorporates a relationship between the frequency response and at least one of an energy average (EA) in a listening area or a total sound power (TSP) produced by the loudspeaker. The AI model automatically estimates, without user interaction, the at least one of the EA in the listening area or the TSP produced by the loudspeaker.

Abstract: A signal processing device according to aspects of the present disclosures comprises a measuring section configured to measure an impulse response between each of a plurality of speakers and a predetermined listening position, a Fourier transformer configured to obtain a frequency spectrum corresponding to each of the plurality of speakers by applying a Fourier transform, a phase adjustment amount calculator configured to calculate a phase adjustment amount for each frequency, a band detector configured to detect a leading phase band, a phase converter configured to convert a phase of the leading phase band to a lagging phase, and a filter coefficient generator configured to generate a filter coefficient based on the phase adjustment amount after conversion by the phase converter.

Abstract: Method for operating a vehicle-mounted acoustic signal-generating device (8), which is configured to generate an acoustic compensation signal (2) for compensating acoustic interference signals (3), which result from a stream of vehicles (5) coming towards a vehicle (4) in question, comprising the steps: —detection of a stream of vehicles (5) coming towards a vehicle (4) in question, and generation of detection information describing at least one parameter of the stream of vehicles, —determination of an acoustic interference signal (3) resulting from the stream of vehicles (5) coming towards the vehicle (4) in question on the basis of the detection information, —generation of an acoustic compensation signal (2) for compensating the acoustic interference signal (3) resulting from the stream of vehicles (5) coming towards the vehicle (4) in question.

Abstract: Provided are a directional acoustic sensor that detects a direction of sound, a method of detecting a direction of sound, and an electronic device including the directional acoustic sensor. The directional acoustic sensor includes a sound inlet through which a sound is received, a sound outlet through which the sound received through the sound inlet is output, and a plurality of vibration bodies arranged between the sound inlet and the sound outlet, in which one or more of the plurality of vibration bodies selectively react to the sound received by the sound inlet according to a direction of the received sound.

Abstract: Systems and methods for optimizing network microphone devices using noise classification are disclosed herein. In one example, individual microphones of a network microphone device (NMD) detect sound. The sound data is analyzed to detect a trigger event such as a wake word. Metadata associated with the sound data is captured in a lookback buffer of the NMD. After detecting the trigger event, the metadata is analyzed to classify noise in the sound data. Based on the classified noise, at least one performance parameter of the NMD is modified.

Abstract: A neural network is trained to digitally model a reference audio system. Training is carried out by repeatedly performing a set of operations. The set of operations includes predicting by the neural network, a model output based upon an input, where the output approximates an expected output of the reference audio system, and the prediction is carried out in the time domain. The set of operations also includes applying a perceptual loss function to the neural network based upon a determined psychoacoustic property, wherein the perceptual loss function is applied in the frequency domain. Moreover, the set of operations includes adjusting the neural network responsive to the output of the perceptual loss function. A neural model file is output that can be loaded to generate a virtualization of the reference audio system.

Abstract: Determination of a set of acoustic parameters for a headset is presented herein. The set of acoustic parameters can be determined based on a virtual model of physical locations stored at a mapping server. The virtual model describes a plurality of spaces and acoustic properties of those spaces, wherein the location in the virtual model corresponds to a physical location of the headset. A location in the virtual model for the headset is determined based on information describing at least a portion of the local area received from the headset. The set of acoustic parameters associated with the physical location of the headset is determined based in part on the determined location in the virtual model and any acoustic parameters associated with the determined location. The headset presents audio content using the set of acoustic parameters received from the mapping server.

Abstract: Some implementations may involve receiving, via an interface system, personnel location data indicating a location of at least one person and receiving, from an orientation system, headset orientation data corresponding with the orientation of a headset. First environmental element location data, indicating a location of at least a first environmental element, may be determined. Based at least in part on the headset orientation data, the personnel location data and the first environmental element location data, headset coordinate locations of at least one person and at least the first environmental element in a headset coordinate system corresponding with the orientation of the headset may be determined. An apparatus may be caused to provide spatialization indications of the headset coordinate locations. Providing the spatialization indications may involve controlling a speaker system to provide environmental element sonification corresponding with at least the first environmental element location data.

Abstract: Methods and apparatus to analyze microphone placement for watermarks and signatures are disclosed. An example instructions cause one or more processors to at least determine a variance of a magnitude spectrum of a frequency band corresponding to a frequency spectrum of a first audio signal sensed with an audio sensor. The example instructions further cause the one or more processors to determine, based on the variance, a recovery rate associated with at least one of watermark detection or signature generation to be performed on a second audio signal to be sensed with the audio sensor.

Abstract: A processor, an out-of-head localization filter generation method, and a program capable of performing appropriate processing are provided. An out-of-head localization filter determination device according to this embodiment includes a built-in microphone embedded in an output unit, a measurement unit configured to output a measurement signal to the output unit and measure a sound pickup signal output from the built-in microphone, a frequency characteristics acquisition unit configured to convert the sound pickup signal into a frequency domain and acquire first frequency characteristics, a conversion unit configured to convert the first frequency characteristics into second frequency characteristics by applying a representative conversion function to the first frequency characteristics, and an inverse filter calculation unit configured to calculate an inverse filter based on the second frequency characteristics.