Abstract: Techniques are described for reducing sensitivity to non-acoustic stimuli. In some embodiments, differential beamforming is applied to microphone signals generated based on responses of microphones to an acoustic stimulus and a non-acoustic stimulus. Compensated signals can be generated based on the microphone signals such that the compensated signals are in phase with respect to the acoustic stimulus. The non-acoustic stimulus is detectable by comparing a first signal to a second signal to determine that one signal has a greater instantaneous magnitude. The first signal can be a beamformed signal or signal derived therefrom, and the second signal can be an average of the compensated signals or signal derived therefrom. An output audio signal can be generated by switching or cross fading between the beamformed signal and a noise-reduced signal such that a contribution of the noise-reduced signal is increased and a contribution of the beamformed signal is decreased.

Abstract: A public address method for live broadcast, in a helmet, of an audio signal conditioned from a plurality of raw audio channels, includes a pre-processing phase including the operations that consist of taking into account characteristics of the auditory perception of the listener; correcting each channel as a function of the characteristics of the auditory perception of the listener; a mixing phase including the production, from the channels thus pre-processed, of a mixed audio signal; a post-processing phase including the operations that consist of: measuring the sound level of a background noise; correcting the mixed audio signal as a function of the sound level of the background noise; a phase of reproducing, in the helmet, the conditioned audio signal resulting from post-processing.

Abstract: A noise cancellation filter structure for a noise cancellation enabled audio device, in particular headphone, comprises a noise input for receiving a noise signal and a filter output for providing a filter output signal. A first noise filter produces a first filter signal by filtering the noise signal and a second noise filter produces a second filter signal by filtering the noise signal. The second noise filter has a frequency response with a non-minimum-phase, in particular maximum-phase. A combiner is configured to provide the filter output signal based on a linear combination of the first filter signal and the second filter signal.

Abstract: A media system and a method of using the media system to accommodate hearing loss of a user, are described. The method includes selecting a personal level-and-frequency dependent audio filter that corresponds to a hearing loss profile of the user. The personal level-and-frequency dependent audio filter can be one of several level-and-frequency-dependent audio filters having respective average gain levels and respective gain contours. An accommodative audio output signal can be generated by applying the personal level-and-frequency dependent audio filter to an audio input signal to enhance the audio input signal based on an input level and an input frequency of the audio input signal. The audio output signal can be played by an audio output device to deliver speech or music that the user perceives clearly, despite the hearing loss of the user. Other aspects are also described and claimed.

Abstract: Audio device and method for designing and making a diaphragm, the audio device comprising a diaphragm having a curved profile adapted for radiation of audio signals from a plurality of bending modes and a piston mode, one or more of the plurality of bending modes having coincident nodal line locations, the diaphragm having a frontal side and a rear side, and a transducer coupled to the rear side of the diaphragm, the transducer adapted for driving the diaphragm for radiation of audio signals having reduced audio distortion, wherein the plurality of bending modes each have minima locations throughout the diaphragm, and wherein the transducer is mounted on one of the minima locations of the plurality of bending modes and one or more impedance components are mounted on at least one of the remaining minima locations to inertially balance the diaphragm based on a pre-determined relative mean modal velocity limit.

Abstract: Methods, systems, and computer program products that infer and correct automatic gain compensation (AGC) values over time are described. A device emits a series of inaudible reference audio signals during recording. The reference audio signals have a constant amplitude. A microphone of the device records the reference audio signals while recording audio content. The device may apply AGC during the recording. An AGC inference module receives the recorded signals and extracts a representation of the reference audio signals, which have been subject to the AGC. The AGC inference module determines variations in levels of the extracted representation of the reference audio signals over time. Based on the variations, the AGC inference module infers the AGC applied to the recording over time. The AGC inference module can then provide the AGC for reading, or undo effects of the AGC applied to the audio content.

Abstract: Provided are an audio signal processing method and device. The method can include acquiring audio signals from at least two sound sources through at least two microphones to obtain multiple frames of original noise signals of the at least two microphones in a time domain, and, for each frame in the time domain, acquiring respective frequency-domain estimated signals of the at least two sound sources according to the respective original noise signals. The method can further include, for each sound source, dividing the frequency-domain estimated signal into frequency-domain estimated components which each corresponds to a frequency-domain sub-band and includes multiple frequency point data in a frequency domain, determining a weighting coefficient of each frequency point in the frequency-domain sub-band, and updating a separation matrix of each frequency point according to the weighting coefficient and obtaining the audio signals based on the updated separation matrices and the original noise signals.

Abstract: The present invention provides for an apparatus, system, and method for generating a head related audio transfer function in real time. Specifically, the present invention utilizes unique structural components including a tragus structure and an antihelix structure in connection with a microphone in order to communicate the location of a sound in three-dimensional space to a user. The invention also utilizes an audio processor to digitally process the head related audio transfer function.

Abstract: Methods and apparatus for predictive head-tracked binaural audio rendering in which a rendering device renders multiple audio streams for different possible head locations based on head tracking data received from a headset, for example audio streams for the last known location and one or more predicted or possible locations, and transmits the multiple audio streams to the headset. The headset then selects and plays one of the audio streams that is closest to the actual head location based on current head tracking data. If none of the audio streams closely match the actual head location, two closest audio streams may be mixed. Transmitting multiple audio streams to the headset and selecting or mixing an audio stream on the headset may mitigate or eliminate perceived head tracking latency.

Abstract: Sound output devices, output systems and methods are provided. The sound output device includes a speaker; a communicator configured to perform communication with an external speaker and a content source that provides sound content to the external speaker; and a processor configured to recognize the external speaker in response to the device entering into a predetermined range from the external speaker, to perform pairing with the external speaker, to receive information about the content source from the external speaker, to receive the sound content from the content source based on the information about the content source, and to output the received sound content through the speaker.

Abstract: A receiver for a hearable, said receiver comprising a moveable membrane, and an arrangement for detecting the movements of the moveable membrane during, for example, a fitting process. The arrangement for detecting the movements of the moveable membrane may include one or more electrodes forming one or more capacitors in combination with the moveable membrane. The receiver may further include a moving armature type motor having an inductor being wounded around at least part of a moving armature. This inductor may form part of the arrangement for detecting the movements of the moveable membrane. The present invention further relates to a hearable and an associated method.

Abstract: A panel audio loudspeaker includes a panel and an actuator rigidly coupled to a surface of the panel. The actuator includes: a magnet assembly that includes a permanent magnet arranged within a cup, wherein an air gap exists between sidewalls of the cup and the permanent magnet; and a coil rigidly coupled to the panel, the coil including a length of an electrically conducing wire wound in a coil and extending along an axis. The coil includes a first region having a first winding density and a second region having a second winding density higher than the first winding density, the second region at least partially extending into the air gap of the magnet assembly.

Abstract: Aspects of the present disclosure provide methods, apparatuses, and systems for closed-loop sleep protection and/or sleep regulation. According to an aspect, sleep disturbing noises are predicted and a biosignal parameter is measured to dynamically mask predicted disturbing environmental noises in the sleeping environment with active attenuation. Environmental noises in a sleeping environment of a subject are detected, input, or predicted based on historical data of the sleeping environment collected over a period of time. The biosignal parameter is used to determine sleep physiology of a subject. Based on the environmental noises in the sleeping environment and the determined sleep physiology, the noises are predicted to be disturbing or non-disturbing noises. For predicted disturbing noises, one or more actions are taken to regulate sleep and avoid sleep disruption by using sound masking prior to or concurrently with the occurrence of the predicted disturbing noises.

Abstract: A controller for an audio system. The audio system comprising an audio processor and an amplifier. The controller is configured to: receive an amplifier-operating-condition-signal representative of an operating condition of the amplifier; receive a maximum-threshold-value; and generate control signaling based on the amplifier-operating-condition-signal and the maximum-threshold-value, wherein the control signaling is configured to set an operating parameter of the audio processor.

Abstract: An intelligent speaker adapted to recover vibration energy to generate electrical power is provided, including a housing, a speaker module, a main board, a power generation module, and a battery module. The speaker module is disposed in the first chamber formed by the housing. The main board disposed in the first chamber transmits an audio signal to the speaker module, wherein the speaker module transmits a main sound wave based on the audio signal. The power generation module is disposed in the first chamber and is vibrated in response to the main sound wave to generate an induction current. The battery module is disposed in the first chamber. The battery module is coupled to the main board to supply the electrical power to the main board, wherein the power generation module is coupled to the battery module and changes the battery module by the induction current.

Abstract: A wearable audio device including a magnetic device and a docking or parking magnet that has opposed first and second sides and produces a magnetic field, wherein the flux of the magnetic field from the first side has a greater magnitude than the flux of the magnetic field from the second side, and wherein the second side is closer to the magnetic device than is the first side.

Abstract: An exemplary low-profile loudspeaker can be provided having an enclosure volume defined between a bottom and an outer wall with an outer periphery configured to suspend such membrane in the center, and excite the membrane at the periphery. An exemplary loudspeaker membrane can also be provided with a suspension path along which to suspend the membrane in a loudspeaker and with an excitation path along which to excite the membrane to produce sound when suspended and excited. The membrane can have an outward face for radiating sound and opposite an inward face for facing towards a volume of a loudspeaker, the membrane having a plurality of structure paths each single structure path substantially extending between the excitation path and the suspension path and along each individual structure path being non-flush structures to the otherwise flush outward face.

Abstract: The present invention relates to a method and a device 10 for processing mixed audio data, including decomposing in real-time with low latency, in which a continuous stream of mixed audio data is received from an audio source 14, a first chunk of the stream of mixed audio data is loaded into a buffer, the audio data contained in the buffer is decomposed to obtain first decomposed audio data representing audio signals of a predetermined timbre, and a first chunk of output data is obtained from the first decomposed audio data, preferably for direct playback via speaker 26.

Abstract: An earpiece is configured for providing audiometric testing. The earpiece includes an earpiece housing, an intelligent control system disposed within the earpiece housing, at least one transducer operatively connected to the intelligent control, and at least one speaker operatively connected to the intelligent control. The earpiece is configured to perform audiometric testing of a user by reproducing sounds at the at least one transducer and receiving user feedback regarding the sounds to provide audiometric test data.

Abstract: Example techniques are provided for controlling volume of A/V devices using an improved volume control. The volume control may be implemented as a slider, where the sliding element is temporarily slid in response to user input (e.g., touched and dragged by the user), but then automatically “snaps” back to the center of the slider's range when the user input ends (e.g., is released by the user). Movement of the element from the center is interpreted as a relative volume adjustment, with the rate of volume change being proportional to the distance between the center of the range and the temporary position of the element. An indicator is provided indicating the rate, to provide user feedback. The volume control may be utilized as a master volume control in a home automation system.