Abstract: Disclosed is an active-noise control device including two passive attenuation earphones each provided with an external microphone, an internal microphone and a loudspeaker. The control device includes a first processing chain having a feedforward filter, a feedback filter and a reconstruction module. The control device includes a second processing chain having a sound source identification module, said second processing chain being implemented in parallel with the first processing chain, and being suitable for parameterizing the first processing chain.

Abstract: Boomless-microphones are described for a wireless helmet communicator with siren signal detection and classification capabilities. An acoustic component receives an audio signal and comprises a left acoustic sensor and a right acoustic sensor. The left acoustic sensor is mountable or attachable to the surface of a left wall of a helmet and the right acoustic sensor is mountable or attachable to the surface of a right wall. A speaker component can generate an echoless audio signal via signal inversion of the audio signal, outputs to a left speaker mountable or attachable to a left ear area of the helmet and a right speaker mountable or attachable to a right ear area of the helmet. A signal enhancement component can increase an intensity of the first audio signal associated with an emergency siren based on a determined proximity of an emitting emergency vehicle or emergency object to the device.

Abstract: A virtual-reality device displays a virtual scene. The scene includes an area sound source, which is located within a predefined near-field distance from the listener (e.g., less than one meter). The device selects sample point sources from the area source and projects audio data from each sample onto a virtual sphere surrounding the listener. The virtual sphere comprises multiple concentric spherical shells that extend from the listener. The device determines, for each sample, energy contributions of the sample to two respective successive shells that enclose the sample. The device determines a head-related impulse response (HRIR) for each shell by combining energy contributions that are associated with the respective shell. The device determines an overall HRIR for the virtual scene by combining the determined HRIRs for the shells. The device convolves the audio data with the overall HRIR and transmits the convolved audio data to sound-producing devices of the virtual-reality device.

Abstract: An audio event detection system that subsamples input audio data using a series of recurrent neural networks to create data of a coarser time scale than the audio data. Data frames corresponding to the coarser time scale may then be upsampled to data frames that match the finer time scale of the original audio data frames. The resulting data frames are then scored with a classifier to determine a likelihood that the individual frames correspond to an audio event. Each frame is then weighted by its score and a composite weighted frame is created by summing the weighted frames and dividing by the cumulative score. The composite weighted frame is then scored by the classifier. The resulting score is taken as an overall score indicating a likelihood that the input audio data includes an audio event.

Abstract: Personalized HRTFs for each of plural computer game participants are convolved with functions representing the acoustics of a physical room in which each participant is located or representing the acoustics of a virtual space in the game environment. Head-tracking of each participant may be employed as the game is played and the HRTFs established accordingly. As the game progresses, the location of each player's character within the virtual game space with respect to the other players is tracked and audio adjusted accordingly. If desired, audio can be down-mixed from, e.g., 5.1 audio and then up-rendered for each player by the game console or by the players' local game interface, e.g., a computer game head-worn visor assembly.

Abstract: A method and system for a passive headset with dynamically controlled LEDs, where the method comprises, in a passive headset comprising speakers, light emitting diodes (LEDs), and LED driver circuitry: receiving an electrical signal that includes an audio signal and an LED control signal, filtering out the LED control signal from the received electrical signal and communicating a resulting output audio signal to the speakers, filtering out the audio signal from the received electrical signal and communicating a resulting output LED control signal to the LED driver circuitry, generating a bias voltage for each of the one or more LEDs utilizing the LED driver circuitry and the output LED control signal, and configuring a light output of the LED utilizing the generated bias voltage. The amplifier may include a mixer that sums an audio signal with a control signal.

Abstract: Embodiments are provided for equalizing audio data for output by a speaker of an electronic device based on a local position or orientation of the electronic device. According to certain aspects, the electronic device can determine (858, 868) its local position based on various sensor data, and identify (870, 872) an appropriate equalization setting. In some cases, the electronic device can modify (876, 880) the equalization setting based on acoustic and/or optical data. The electronic device can apply (882) the modified or unmodified equalization setting to an audio signal and cause the speaker to output (886) the audio signal with the applied equalization setting.

Abstract: Systems, devices, and methods are described for providing loudspeaker protection and distortion suppression. An upstream loudspeaker model estimation component receives sensed electrical characteristics of a loudspeaker and generates an impedance model from which an excursion model of the loudspeaker and a gain change parameter may be generated. The impedance components are fitted to features of an estimated impedance, based on the sensed characteristics, to generate the estimated impedance model that is converted to an excursion model of the loudspeaker. A downstream audio signal processing component, based on the excursion model, or parameters thereof, limits a predicted excursion of the loudspeaker utilizing excursion-constraining processing circuitry that includes a non-linear constraint filter. Processed audio signals associated with the limited excursion are subject to distortion suppression prior to releasing the output audio signals for playback on the loudspeaker.

Abstract: A method clusters audio sources in virtual environments. The method is performed at a virtual-reality device displaying a virtual environment. The device identifies two audio sources in the virtual environment. For each of the two audio sources, the device determines a bounding box in the virtual environment. Each bounding box includes termination points for a respective plurality of rays emanating from a point in the virtual environment corresponding to the audio source. The device applies an overlap test to the bounding boxes to determine whether the two audio sources are in a same room. The device forms an angle according to rays from the location of the listener to the audio source points. When the two audio sources are in the same room and the angle is less than a predetermined threshold angle, the device clusters the two audio sources together, including rendering combined audio for the two audio sources.

Abstract: Volume limiting systems and methods are operable to limit volume output from media presentation devices. An exemplary embodiment detects a sound using a microphone, wherein the sound corresponds to an audio output of at least one controlled media presentation device, and wherein the microphone is remotely located from the at least one controlled media presentation device; compares a level of the detected sound with a predefined maximum volume limit; generates a volume output limit command in response to the detected sound exceeding the predefined maximum volume limit; and communicates the volume output limit command to the media presentation device. The media presentation device then reduces a volume level of its audio output. In some instances, volume may be limited during user specified periods.

Abstract: At least one exemplary embodiment is directed to a method and device for voice operated control. The method can include measuring a first sound received from a first microphone, measuring a second sound received from a second microphone, detecting a spoken voice based on an analysis of measurements taken at the first and second microphone, mixing the first sound and the second sound to produce a mixed signal, and controlling the production of the mixed signal based on one or more aspects of the spoken voice.

Abstract: Disclosed herein are systems and methods for establishing a metadata exchange channel between a media playback system and a networked microphone system, wherein the networked microphone system comprises a networked microphone device and a networked microphone server. After establishing the metadata exchange channel, the media playback system experiences a metadata exchange triggering event. In response to experiencing the metadata exchange triggering event, the media playback system provides metadata to the networked microphone system via the metadata exchange channel.

Abstract: A MEMS may include a backplate comprising first and second electrodes electrically isolated from one another and mechanically coupled to the backplate in a fixed relationship relative to the backplate, and a diaphragm configured to mechanically displace relative to the backplate as a function of sound pressure incident upon the diaphragm. The diaphragm may comprise third and fourth electrodes electrically isolated from one another and mechanically coupled to the diaphragm in a fixed relationship relative to the diaphragm such that the third and fourth electrodes mechanically displace relative to the backplate as the function of the sound pressure. The first and third electrodes may form a first capacitor, the second and fourth electrodes may form a second capacitor, and the first capacitor may be configured to sense a displacement of the diaphragm responsive to which the second capacitor may be configured to apply an electrostatic force to the diaphragm to return the diaphragm to an original position.

Abstract: Loudness signal processors and methods for processing an input audio signal in order to control a resulting integrated loudness level and a resulting loudness range of an output audio signal by a predetermined target loudness level and by a predetermined target loudness range, the processors and methods comprising level detection and level distribution analysis; transfer function generation based on the level distribution, the predetermined target loudness level and the predetermined target loudness range; and calculation of a gain to apply to said input audio signal, resulting in said output audio signal.

Abstract: A technique for auditory masking for a coherence-controlled calibration system includes generating a first auditory masking pattern based on a frequency spectrum of at least a first frame of a first audio signal. The technique continues by generating a first calibration signal based on the first auditory masking pattern. The technique then includes producing a combined input signal based on the first audio signal and the first calibration signal. The technique also includes performing one or more calibration operations based on the combined input signal.

Abstract: A mechanism is described for facilitating context-based cancellation and amplification of acoustical signals in acoustical environments according to one embodiment. An apparatus of embodiments, as described herein, includes detection and recognition logic to detect an acoustical signal being emitted by an acoustical signal source; evaluation, estimation, and footprint logic to classify the acoustical signal as an emergency acoustical signal or a non-emergency acoustical signal, wherein the classification is based on a footprint or a footprint identification (ID) associated with the acoustical signal; acoustical signal cancellation logic to cancel the acoustical signal if the acoustical signal is regarded as the non-emergency acoustical signal based on the footprint or the footprint ID; and acoustical signal amplification logic to amplify the acoustical signal if the acoustical signal is classified as the emergency acoustical signal based on the footprint or the footprint ID.

Abstract: A method for voice recording and an electronic device thereof are provided. The method includes determining a voice recording mode from among a plurality of voice recording modes, determining a voice beamforming direction according to the determined voice recording mode, and recording voice signals based on the determined voice beamforming direction.

Abstract: The present invention relates to an apparatus for detachably engaging wireless headsets (100), and more particularly to detachably engage with individual wireless earpiece modules. Accordingly, the apparatus includes: a) a neck-band or neck-loop module (110) with at least one battery pack (112); b) at least one earpiece holders (116); wherein the at least one earpiece holder (116) is adapted to be detachably engaged with at least one wireless earpiece module (120); and wherein the at least one battery pack (112) of the neck-band or neck-loop module (110) is configured to supply power to the at least one wireless earpiece module (120) via the at least one earpiece holder (116).

Abstract: An audio system is disclosed that contains speakers that are configured to transmit unique parameters to an audio source. The system includes a hub that is configured to be connected with one or more audio sources. At least one speaker is included having one or more filter parameters stored internally that are specific to the speaker. The speaker is operable to transmit the one or more filter parameters to the hub. The hub is operable to use the one or more filter parameters to filter an audio channel associated with the speaker and an associated subwoofer channel.

Abstract: A method and system for communicating audio, video, and/or control signals within a home entertainment system. One or more signals are communicated between an input device and one or more output devices via one or more networks. The output device can include loudspeakers, display devices, and headphones. In some embodiments an output device, for example a center channel loudspeaker, transmits signals to other output devices. For example, the center channel loudspeaker can transmit a combined audio signal and control signal to a remote loudspeaker over a first network and transmit a video signal to a display device over a second network. The display device displays the video signal. The networks can be wireless, wired, infrared, RF, and powerline.