Abstract: A method for processing a signal includes receiving a first frame of an input audio bitstream at a decoder. The first frame includes at least one signal associated with a frequency range. The method also includes decoding the at least one signal to generate at least one decoded signal having an intermediate sampling rate. The intermediate sampling rate is based on coding information associated with the first frame. The method further includes generating a resampled signal based at least in part on the at least one decoded signal. The resampled signal has an output sampling rate of the decoder.

Abstract: The present disclosure describes a multi-mode headset incorporating both bone conduction and aural transducers. Bone conduction plates of the headset rest on a users' temples, just in front of each ear in one implementation, and create sound through vibrations in a first mode, allowing the user to listen to the headset while still leaving environmental or external sounds audible. In a second mode, the user detaches aural transducers or earbuds from an attachment point and inserts them into the user's ears, blocking external noise and providing passive and/or active noise isolation. In a third mode, both the bone conduction plates and aural transducers may be used simultaneously, providing additional amplitude and potentially wider frequency response. The headset may include a switch, such as a magnetic proximity sensor detecting the position of the aural transducers, to control a mixer or amplifier to re-route audio through the aural transducers.

Abstract: The methods described herein are configured to accurately propagate sound through a virtual environment. During runtime of an application using the virtual environment, a shortest path is calculated from a sound source in a first zone to a sound destination in a second zone that passes through at least one portal. The direction of the calculated shortest path from the portal to the sound destination is determined. Then, a virtual sound source is generated based on the length and determined direction of the calculated path. Further, obstructions in the virtual environment are processed to determine an attenuation of audio data associated with the sound source. The generated virtual sound source and the attenuated audio data are provided to an audio engine for rendering to a user. Propagating sound using a combination of these methods provides an accurate, realistic sound experience in the virtual environment.

Abstract: One embodiment provides a sound apparatus comprising a plurality of driver units arranged linearly in an end-fire array, and for each driver unit, a corresponding digital filter for individual digital signal processing of signals received by the driver unit.

Abstract: Disclosed herein are methods, systems, and devices for discontinuously recording audio input representative of an audio environment. The audio environment can be an environment of a recipient of a hearing prosthesis and the discontinuously recorded audio input can be used to characterize the audio environment of the recipient, to control settings of the hearing prosthesis, or to provide some other application. Discontinuously recording the audio input can provide a recording that characterizes the audio environment of the recipient while protecting the privacy of individuals speaking nearby the recipient. Discontinuously recording the audio input may include recording discontinuous time segments of the audio input such that the semantic content of speech within the audio input, the identity of speakers in the audio input, or other private information present in the audio input is masked.

Abstract: An audio headset may receive a plurality of audio signals corresponding to plurality of surround sound channels. The headset may determine, via its audio processing circuitry, context and content of the audio signals. The audio processing circuitry may process the audio signals to generate a pair of stereo signals carrying one or more virtual surround channels, wherein the processing comprises automatically controlling, based on the context and the content of the audio signals, a simulated acoustic environment of the virtual surround channels.

Abstract: A feedback signal is employed to facilitate enhancing an acoustic overload point and electrostatic discharge protection of a sensor component and associated circuit. The sensor component comprises a backplate component and a diaphragm component. The backplate component is biased to a low-level voltage associated with a ground. The diaphragm component is biased to a defined high-voltage associated with a charge pump. The diaphragm component generates a signal based on movement of the diaphragm component in relation to the backplate component in response to the input signal. A feedback component receives the signal from the diaphragm component and generates an inverted signal based on the signal. The inverted signal or a processed inverted signal, which can be derived from a filter component that filters the inverted signal, is transmitted to the backplate component.

Abstract: Techniques for acoustic voice activity detection (AVAD) is described, including detecting a signal associated with a subband from a microphone, performing an operation on data associated with the signal, the operation generating a value associated with the subband, and determining whether the value distinguishes the signal from noise by using the value to determine a signal-to-noise ratio and comparing the value to a threshold.

Abstract: A headphone detector including a headphone and a processor. The headphone has a microphone and a speaker, and the microphone is configured to generate an audio signal based on an output of the speaker. The processor is configured to receive the audio signal, determine a characteristic of the audio signal, and assess whether the headphone is on ear or off ear based on a comparison of the characteristic to a threshold. The threshold corresponds to one or more of an audio response of the audio signal at a corresponding frequency and an audio response of a feedback microphone signal at a corresponding frequency, under one or more known conditions.

Abstract: Techniques for noise suppression systems coupled to one or more microphone arrays are described, including a housing, a first microphone, a second microphone, and a third microphone, where the third microphone functions as a common rear vent for the first and the second microphones.

Abstract: A noise cancelling headset includes an earpiece, the earpiece including a feedback microphone, a feed-forward microphone, and an output driver. A first feedback filter receives an input from at least the first feedback microphone and produces a first filtered feedback signal. A first feed-forward filter receives an input from at least the first feed-forward microphone and produces a first filtered feed-forward signal. A first summer combines the first filtered feedback signal and the first filtered feed-forward signal and produces a first output signal. An output interface provides the first output signal as an output from the headset.

Abstract: Systems, methods, and computer hardware storage media are provided for variably reducing, but not eliminating, the ambient volume that reaches the eardrum in a particular setting. One or more noise signatures of sound from an ambient environment are received at a mobile device. A profile associated with a suppression or amplification control of the one or more noise signatures of sound from the ambient environment are created and stored at the mobile device. The suppression or amplification control is communicated from the mobile device to a set of variable audio ear plugs. The suppression control causes the variable audio ear plugs to variably reduce, but not eliminate, the sound from the ambient environment and the amplification control does not amplify sound from any source other than the ambient environment.

Abstract: A system and method to perform an estimation of a location of an active speaker in real time includes designating a microphone of an array of microphones as a reference microphone. The method includes storing a relative transfer function (RTF) for each microphone of the array of microphones other than the reference microphone associated with each potential location among potential locations as a set of stored RTFs, and obtaining a voice sample of the active speaker and obtaining a speaker RTF for each microphone of the array of microphones other than the reference microphone. The method also includes performing an RTF projection of the speaker RTF for each microphone on the set of stored RTFs, and determining one of the potential locations as the location of the active speaker based on the performing the RTF projection.

Abstract: The disclosure is directed to a device that includes a multi-layered porous display that permits the passage of sound through the display, while avoiding dead-spots that produce no light. Embodiments of the invention include a display that has first porous layer overlaying a second porous layer. The pores of each layer do not align so that light cannot pass directly through pores of each layer. In this way, a user may not view the pores of the second layer through the pores of the first layer. Further, because the first layer is backed by a second layer, the image displayed will be viewable through the pores of the first layer and, thus, a user will not perceive the pores of the first layer.

Abstract: A device for processing audio signals includes an interchannel temporal mismatch analyzer, an interchannel phase difference (IPD) mode selector and an IPD estimator. The interchannel temporal mismatch analyzer is configured to determine an interchannel temporal mismatch value indicative of a temporal misalignment between a first audio signal and a second audio signal. The IPD mode selector is configured to select an IPD mode based on at least the interchannel temporal mismatch value. The IPD estimator is configured to determine IPD values based on the first audio signal and the second audio signal. The IPD values have a resolution corresponding to the selected IPD mode.

Abstract: Some methods involve receiving an input audio signal that includes N input audio channels, the input audio signal representing a first soundfield format having a first soundfield format resolution, N being an integer ?2. A first decorrelation process may be applied to two or more of the input audio channels to produce a first set of decorrelated channels, the first decorrelation process maintaining an inter-channel correlation of the set of input audio channels. A first modulation process may be applied to the first set of decorrelated channels to produce a first set of decorrelated and modulated output channels. The first set of decorrelated and modulated output channels may be combined with two or more undecorrelated output channels to produce an output audio signal that includes O output audio channels representing a second and relatively higher-resolution soundfield format than the first soundfield format, O being an integer ?3.

Abstract: This application relates to methods and apparatus for loudspeaker protection. A loudspeaker protection system (100) receives a digital audio signal comprising a plurality of samples at an input node (IN). A delay block (15) delays the digital audio signal and a gain block (14) applies a controlled gain to the delayed digital audio signal. An excursion predictor (12) is configured to receive a version of the audio signal from the signal path upstream of the delay block and determine a predicted excursion for a loudspeaker based on the audio signal. A gain controller (23) controls a gain setting (g) of the gain block in response to the predicted excursion and a first loudspeaker impulse response model and a predetermined excursion limit. The gain controller (23) is configured to determine at least one gain setting {ga gi} to be applied to a set of samples {Va . . .

Abstract: A system, method and wireless earpieces for determining a status of the wireless earpiece. Sensor measurements are made utilizing sensors of the wireless earpieces at a first location. The sensor measurements are analyzed. Data is extrapolated for one or more other locations utilizing the sensor measurements from the first location within the wireless earpiece.

Abstract: A class-D amplifier includes measurement of speaker current via the low-side drive transistors of the amplifier. In one embodiment, a class-D amplifier includes two high-side transistors, two low-side transistors, a first sense resistor, a second sense resistor, and a sigma delta analog to digital converter (?? ADC). The two high-side transistors and two low-side transistors are connected as a bridge to drive a bridge tied speaker. The first sense resistor is connected between a first of the low-side transistors and a low-side reference voltage. The second sense resistor is connected between a second of the low-side transistors and the low-side reference voltage. The ?? ADC is coupled to the bridge to measure voltage across the first sense resistor and the second sense resistor.

Abstract: Improving noise rejection of a micro-electro-mechanical system (MEMS) microphone by utilizing a membrane sandwiched between oppositely biased backplates is presented herein. The MEMS microphone can comprise a diaphragm that converts an acoustic pressure into an electrical signal; a first backplate capacitively coupled to a first side of the diaphragm—the first backplate biased at a first direct current (DC) voltage; a second backplate capacitively coupled to a second side of the diaphragm—the second backplate biased at a second DC voltage; and an electronic amplifier that buffers the electrical signal to generate a buffered output signal representing the acoustic pressure.