Abstract: A method, apparatus, and computer program product are provided for encoding audio events as geo-referenced audio events for use in location establishment. Methods may include: receiving first audio data from a first sensor; establishing a baseline frequency profile; identifying, within the first audio data, a first audio event, where the first audio event satisfies at least one predefined criteria relative to the baseline frequency profile; identify a location corresponding to the first audio event; encoding the first audio event in a database to correspond to the location; receiving second audio data from a second sensor; identifying, within the second audio data, a second audio event; correlating the second audio event with the first audio event; and providing the location in response to the second audio data. The at least one criteria may include a statistically significant change in a prominent frequency in the audio data.

Abstract: Microphones, microphone systems, and methods for capturing and processing sound are described. The microphones and microphone systems may adaptively change the direction from which sound is captured. The microphones and microphone systems avoid the need to provide arrays of microphones, while providing adaptive beamforming without a time delay between each channel of information, and multi-directional sound capture. A dependency between the frequency response and system size is also avoided.

Abstract: Validation of an activation event of a device based on analyzing audio data is disclosed. The device detects an activation event, e.g., an input to a soft-touch button or a wake word detected by a microphone array. Responsive to detecting the activation event, the device captures, via a microphone array on the device, sound from a local area of the device. The device stores the captured sound as audio data in an audio buffer. The device performs a validation of the activation event by analyzing the captured sound. The device performs an action based on a result of the validation.

Abstract: Disclosed are a method of controlling a display by a headset with inline control, and related devices. The method includes: detecting whether an headset jack of the display is plugged with the headset; after the headset jack of the display is determined to be plugged with the headset, inputting a detection bias voltage to the headset, and monitoring a feedback signal of the detection bias voltage; in response to detecting a change in the feedback signal of the detection bias voltage, identifying a user's control instruction to the display using the headset according to a control signal sent by the headset; and controlling the display to perform a corresponding display operation according to the control instruction.

Abstract: Provided is a sensor interface including a first cantilever beam bundle including at least one resonator and a first output terminal, a second cantilever beam bundle including at least one resonator and a second output terminal, and a differential amplifier including a first input terminal electrically connected to the first output terminal of the first cantilever beam bundle and a second input terminal electrically connected to the second output terminal of the second cantilever beam bundle.

Abstract: Playback devices can include touch sensor assemblies with one or more integrated antennas. Such touch sensor assemblies can be incorporated into a playback device such as a headphone device. The playback device can include an electrode comprising a first conductor, a second conductor, and a filter coupled between the first and second conductors, a capacitive-touch circuit coupled to the electrode, and a wireless radio coupled to the second conductor. The capacitive-touch circuit is configured to deliver a capacitive sensing signal to the electrode. The wireless radio is configured to facilitate communication over at least one wireless network via the second conductor.

Abstract: In some embodiments, a method for performing enhancement on an audio signal to generate an enhanced audio signal in response to feedback indicative of amount of compression applied to at least one frequency band of the enhanced audio signal. In typical embodiments, the enhancement is or includes bass enhancement. Examples of other types of enhancement performed in other embodiments include dialog enhancement, upmixing, frequency shifting, harmonic injection or transposition, subharmonic injection, virtualization, and equalization. Other aspects are systems (e.g., programmed processors) and devices (e.g., devices having physically-limited bass reproduction capabilities, such as, for example, a notebook, tablet, mobile phone, or other device with small speakers) configured to perform any embodiment of the method.

Abstract: Disclosed herein is a sound synthesis system for generating a user defined synthesised sound effect, the system comprising: a receiver of user defined inputs for defining a sound effect; a generator of control parameters in dependence on the received user defined inputs; a plurality of sound effect objects, wherein each sound effect object is arranged to generate a different class of sound and each sound effect object comprises a sound synthesis model arranged to generate a sound in dependence on one or more of the control parameters; a plurality of audio effect objects, wherein each audio effect object is arranged to receive a sound from one or more sound effect objects and/or one or more other audio effect objects, process the received sound in dependence on one or more of the control parameters and output the processed sound; a scene creation function arranged to receive sound output from one or more sound effect objects and/or audio effect objects and to generate a synthesised sound effect in dependence o

Abstract: A distributed audio processing system is disclosed, for providing users with the capability of producing a personalized audio mix of a plurality of signals from a plurality of audio sources. The system includes a wireless transmitter for each audio source and, for each user, a wireless receiver. The receiver comprises a programmable audio signal processor configured to process and mix a plurality of audio tracks received via a radio broadcast of a multi-track audio signal comprising the audio signals from the plurality of sources, said processing and mixing being programmable via received commands, instructions and/or parameters. The transmitters are configured to process the audio signals received from their respective sources, according to received commands, instructions and/or parameters. According to an embodiment, a user may provide commands, instructions and/or parameters to any of the receivers and/or transmitters of the system.

Abstract: Systems and methods to augment audio output in an electric vehicle (EV) include obtaining inputs from one or more sensors. The inputs include information about the EV and about one or more persons outside the EV. A current scenario is defined based on the inputs. Whether the current scenario matches a predefined scenario among a set of predefined scenarios is determined, and augmented audio output is produced according to the predefined scenario.

Abstract: A microphone may comprise a microphone element for detecting sound, and a digital signal processor configured to process a first audio signal that is based on the sound in accordance with a selected one of a plurality of digital signal processing (DSP) modes. Each of the DSP modes may be for processing the first audio signal in a different way. For example, the DSP modes may account for distance of the person speaking (e.g., near versus far) and/or desired tone (e.g., darker, neutral, or bright tone). At least some of the modes may have, for example, an automatic level control setting to provide a more consistent volume as the user changes their distance from the microphone or changes their speaking level, and that may be associated with particular default (and/or adjustable) values of the parameters attack, hold, decay, maximum gain, and/or target gain, each depending upon which DSP is being applied.

Abstract: An audio output device is provided. The audio output device includes a buffering member including a biometric sensor and a first terminal connected to the biometric sensor, a housing including a portion to which the buffering member is mounted, a second terminal disposed in the portion of the housing and electrically connected to the first terminal of the buffering member, and a control circuit positioned inside the housing and operatively connected to the biometric sensor, the first terminal, and the second terminal, wherein the control circuit may be configured to supply power to the biometric sensor through the second terminal if the first terminal and the second terminal are connected, and to receive at least one piece of biometric signal data obtained from the biometric sensor of the buffering member through the second terminal.

Abstract: The following coding scenario is addressed: A number of audio source signals need to be transmitted or stored for the purpose of mixing wave field synthesis, multi-channel surround, or stereo signals after decoding the source signals. The proposed technique offers significant coding gain when jointly coding the source signals, compared to separately coding them, even when no redundancy is present between the source signals. This is possible by considering statistical properties of the source signals, the properties of mixing techniques, and spatial hearing. The sum of the source signals is transmitted plus the statistical properties of the source signals, which mostly determine the perceptually important spatial cues of the final mixed audio channels. Source signals are recovered at the receiver such that their statistical properties approximate the corresponding properties of the original source signals. Subjective evaluations indicate that high audio quality is achieved by the proposed scheme.

Abstract: Microphone arrays comprise several microphone capsules, the outputs of which being electronically combined for directional recording of sound. The directional and frequency properties of the microphone array depend on the number and positions of the microphone array. In order to obtain the smallest possible microphone array with only few microphone capsules, which, however, has an essentially uniform directional and frequency dependence over a speech frequency range, is scalable and robust against small incorrect positioning of the capsules, fifteen or twenty-one microphone capsules (K15,11-K15,35, K21,11-K21,37) are arranged on a carrier such that they lie on three similar branches, each with the same number of microphone capsules, which are rotated against each other by 120°. Each of the microphone capsules lies on a corner of a triangle of a grid in a flat isometric coordinate system with three axes rotated by 120° against each other and forming the grid of equilateral triangles.

Abstract: A vehicle selects various sound sources and outputs driving sound based on information received from the navigation system. A database stores sound sources classified as first and second emotions. A controller outputs a landmark name based on a route guidance text and outputs a ratio of the first and second emotions corresponding to the landmark name. The controller selects a first sound source from the sound sources classified as the first emotion based on the first emotion ratio, selects a second sound source from the sound sources classified as the second emotion based on the second emotion ratio and determines first and second playback periods based on the ratios of the first and second emotions. The controller outputs a first sound generated based on the first sound source during the first playback period and a second sound generated based on the second sound source during the second playback period.

Abstract: Audio systems and methods for obtaining location information are disclosed. A portable, wireless speaker may include a beacon that a user may activate to remotely trigger a request for shot information, which may include location information of the user or speaker. The beacon may be wirelessly coupled to the speaker. Notably, the beacon may comprise an earpiece such as an earbud or pair of earbuds. The speaker may receive location information from a remote device, such as a GPS-enabled mobile phone. The shot information may be converted into an audio format for playback by the speaker. The speaker may pause the playback of the speaker to play out the shot information. The system may also comprise a visual display.

Abstract: Technology described in this document can be embodied in a method that includes receiving a first input signal representing audio captured by a first sensor disposed in a signal path of an active noise reduction (ANR) device, and receiving a second input signal representing audio captured by a second sensor disposed in the signal path of the ANR device. The method also includes processing, by at least one compensator, the first input signal and the second input signal to generate a drive signal for an acoustic transducer of the ANR device. A gain applied to the signal path is at least 3 dB less relative to an ANR signal path having a single sensor.

Abstract: A vibration transducer for sensing vibrations includes a first flexible triboelectric member, a second flexible triboelectric member, a plurality of attachment points, a first electrode and a second electrode. The first flexible triboelectric member includes a first triboelectric layer and a material being on a first position on a triboelectric series. A conductive layer is deposited on the second side thereof. The second flexible triboelectric member includes a second triboelectric layer and a material being on a second position on the triboelectric series that is different from the first position on the triboelectric series. The second triboelectric member is adjacent to the first flexible triboelectric member. When the first triboelectric member comes into and out of contact with the second triboelectric member as a result of the vibrations, a triboelectric potential difference having a variable intensity corresponding to the vibrations can be sensed between the first and second triboelectric members.

Abstract: An acoustic system includes: a first terminal device including a first-terminal sound emitting unit, an earhole sound collecting unit configured to collect earhole echo sound, and a first-terminal control unit configured to perform authentication between the first terminal device and a user based on the echo sound collected by the earhole sound collecting unit; an onboard device including an onboard-device sound emitting unit and an onboard-device control unit configured to perform authentication between the onboard device and the user based on communication with the first terminal device; and a second terminal device includes a second-terminal control unit configured to set a destination of sound data to one of the first terminal device and the onboard device when the onboard-device control unit detects that a predetermined condition including a condition that the user has been authenticated is satisfied.

Abstract: The technology described in this document can be embodied in a method that includes receiving an input signal captured by one or more sensors associated with an active noise reduction (ANR) headphone, and determining one or more characteristics of a first portion of the input signal. Based on the one or more characteristics of the first portion of the input signal, a gain of a variable gain amplifier (VGA) disposed in an ANR signal flow path can be adjusted, and accordingly, a set of coefficients for a tunable digital filter disposed in the ANR signal flow path can be selected. The method further includes processing a second portion of the input signal in the ANR signal flow path using the adjusted gain and selected set of coefficients to generate a second output signal for the electroacoustic transducer of the ANR headphone.