Abstract: A method and apparatus for processing audio signals in an entertainment system from at least one audio source modify audio signals during playback by the entertainment system for adjustment to a psychoacoustic loudness set value, wherein this modification is performed in each case on the basis of an average psychoacoustic loudness maximum determined over a predefined time interval for the audio source concerned.

Abstract: A voice enhancement device includes: a sound receiving unit configured to receive an audio signal; a vehicle state monitor unit configured to monitor a vehicle state; a noise estimation unit configured to estimate a noise component for each frequency component using a cumulative histogram created by accumulating frequency of power of the audio signal received by the sound receiving unit for each frequency component; and a voice enhancer configured to suppress the noise component for each frequency component estimated by the noise estimation unit in the received audio signal, wherein the noise estimation unit resets the cumulative histogram on the basis of a monitoring result of the vehicle state monitor unit.

Abstract: The derivation of personalized HRTFs for a human subject based on the anthropometric feature parameters of the human subject involves obtaining multiple anthropometric feature parameters and multiple HRTFs of multiple training subjects. Subsequently, multiple anthropometric feature parameters of a human subject are acquired. A representation of the statistical relationship between the plurality of anthropometric feature parameters of the human subject and a subset of the multiple anthropometric feature parameters belonging to the plurality of training subjects is determined. The representation of the statistical relationship is then applied to the multiple HRTFs of the plurality of training subjects to obtain a set of personalized HRTFs for the human subject.

Abstract: A binaural healing system includes a left-ear hearing device, a right-ear hearing device and an auxiliary device. The auxiliary device has a connected mode wherein it transmits data messages and a disconnected mode wherein it does not transmit data messages, and the auxiliary device is adapted to enable the connected mode in dependence on receiving beacon messages, to synchronize its transmission of data messages with received beacon messages and to enable the disconnected mode in dependence on not receiving beacon messages. To avoid disconnection of the auxiliary device S during gaps, the left-ear hearing device and the right-ear hearing device are adapted to alternatingly transmit the beacon messages.

Abstract: A loudspeaker baffle that provides variable sound patterns is described. The baffle may support non-low frequency sound sources and a waveguide to provide varying sound beam patterns. The baffle may include a center mount adapted to receive a plurality of audio outputs and a plurality of low frequency apertures to receive a plurality low frequency output. The waveguide may be formed from a front face of the baffle. The front face may be intermediate the center mount and the low frequency apertures. The front face may include a continuously varying waveguide surface with a first waveguide portion adjacent a first audio output of the plurality of audio outputs providing a first sound pattern and a second waveguide portion adjacent a second audio output of the plurality of audio outputs providing a second sound pattern that is different than the first sound pattern.

Abstract: An electronic stethoscope includes an acoustic sensor assembly having a first microphone to detect biological sounds within a body, a detection system in communication with the first microphone to receive an auscultation signal from the first microphone, the auscultation signal including information of the biological sounds detected by the first microphone. The stethoscope also includes a second microphone in communication with the detection system to detect noise from an environment of the body. The detection system receives a noise signal from the second microphone, and provides a resultant signal based on the auscultation signal and the noise signal. The detection system subtracts information from the auscultation signal to produce the resultant signal, where the subtracted information is based on the noise signal such that the subtracted information is based more on higher frequency ranges of the noise signal compared to a lower frequency range corresponding to the biological sounds.

Abstract: A portable audio input/output device may include one or more openings that extend through a cover of the device and allow acoustic signals outside a housing of the device to reach a microphone disposed within the housing. The opening(s) may be illuminated by a light guide disposed within the housing, which scatters light emitted from lights disposed within the housing. In some instances, a hole may pass through a printed circuit board to allow acoustic signals to be received by the microphone disposed below the printed circuit board. An input/output (I/O) interface module with multiple buttons and inputs may be installed in the hole. The multiple buttons and I/O ports of the I/O interface module may be aligned along an axis vertical relative to the housing and centered with respect to each other.

Abstract: An acoustic chamber includes a loudspeaker, which includes at least two membranes that each reproduce a different frequency band, and a filter that makes it possible to generate a plurality of activation signals from an audio signal source. The activation signals are each applied to an actuator of one of the membranes. The acoustic chamber has an operating range having a variable and controlled directivity, each frequency of which belongs to at least two frequency bands reproduced by the membranes. The acoustic chamber obtains a directivity control signal, and the filter makes it possible to dose, for each frequency of the operating range and depending on the directivity control signal, the contribution of each one of the at least two membranes reproducing the frequency.

Abstract: A detachable audio system with an interface plate mounted on a head strap, and a mounting plate assembly removably attachable to the interface plate with a portion of the strap between the plates to provide independent electrical and mechanical connections between the plates. The mounting plate assembly connects to an earbud assembly via a flexible audio cable. The mounting plate assembly has a threaded attachment member that mates with a threaded attachment member on the interface plate. A pair of electrical connectors on opposing sides of the threaded attachment member protrude from spaced apart cylindrical bosses, such that when the mounting plate is attached to the interlace plate, independent mechanical and electrical connections are provided for detachable retention and operation of the earbud assembly.

Abstract: A matrix is generated that stores sinusoidal components evaluated for a given sample rate corresponding to the matrix. The matrix is then used to convert an audio signal to chroma vectors representing of a set of “chromae” (frequencies of interest). The conversion of an audio signal portion into its chromae enables more meaningful analysis of the audio signal than would be possible using the signal data alone. The chroma vectors of the audio signal can be used to perform analyzes such as comparisons with the chroma vectors obtained from other audio signals in order to identify audio matches.

Abstract: An audio playback path of an audio apparatus includes a digital modulator, a digital-to-analog converter (DAC), and a power amplifier. The digital modulator receives a playback signal corresponding to playback audio content and generates a digital input signal in accordance with the playback signal. The DAC receives the audio input signal and generates an analog preamplifier signal. The power amplifier generates an audio output signal in accordance with the preamplifier signal and an analog attenuation determined by the analog attenuation signal.

Abstract: An apparatus for realizing an active noise cancellation control law transfer function between a sensing microphone and a speaker. The apparatus includes a multiplicity of filters. Each filter is operable over a different frequency range. At least one filter has an adjustable parameter whereby the filter can be adjusted such that the filters cumulatively realize a required control law transfer function. The adjustable parameter may in one embodiment be the amplitude. In other embodiments, it may be other parameters.

Abstract: In one embodiment, the present invention provides a sound processing device with a binaural input and binaural output, where “binaural input” means at least one microphone mounted in or near each ear of the device user, and “binaural output” means at least one output signal directed to each ear. The device may be comprised of two parts connected by a wired or wireless link.

Abstract: A device includes an input for receiving an audio signal, a speaker to convert the audio signal into an audible sound, and a memory for storing remediation instructions and detection instructions. The device further includes a processor coupled to the input, the speaker, and the memory. The processor is configured to process the audio signal according to the detection instructions and the remediation instructions to modulate amplitude of the audio signal based on the remediation instructions.

Abstract: Embodiments are directed a method of rendering object-based audio comprising determining an initial spatial position of objects having object audio data and associated metadata, determining a perceptual importance of the objects, and grouping the audio objects into a number of clusters based on the determined perceptual importance of the objects, such that a spatial error caused by moving an object from an initial spatial position to a second spatial position in a cluster is minimized for objects with a relatively high perceptual importance. The perceptual importance is based at least in part by a partial loudness of an object and content semantics of the object.

Abstract: In general, the present invention relates to a method and system for synthesizing artificial reverberation using modal analysis of a room or resonating object. In one embodiment of the inventive system, a collection of resonant filters is employed, each driven by the source signal, and their outputs summed. With filter resonance frequencies and dampings tuned to the modal frequencies and decay times of the acoustic space or resonating object being simulated, and filter gains set according to the source and listener positions within the space or object, any number of acoustic spaces and resonant objects may be simulated.

Abstract: Technologies are described to DC-couple an integrated amplifier system to a source that provides a signal with an unknown DC component, for example to DC-couple an integrated audio codec to an analog microphone. In one aspect, methods include receiving, by an amplifier, a signal having an unknown DC component, and issuing an amplified signal; low pass filtering, with respect to a cutoff frequency, by a feedback circuit coupled between an output of the amplifier and an input of the amplifier, the amplified signal issued at the output of the amplifier to generate a filtered signal having frequencies lower than the cutoff frequency; and injecting, by the feedback circuit, the filtered signal into the input of the amplifier to cancel the unknown DC component below the cutoff frequency.

Abstract: A method generating a virtual audio signal for a listener. The method includes estimating spherical harmonic coefficients based on an individual character of the listener. The estimated spherical harmonic coefficients are compared to a distribution of known spherical harmonic coefficients. The estimated spherical harmonic coefficients are iteratively updated and compared to the distribution of known spherical harmonic coefficients until convergence. The individual character and the converged spherical harmonic coefficients are then applied to a mono-channel sound.

Abstract: Techniques are disclosed for synchronizing gain adjustments across a cascaded network of audio gain stages having variant operating delays. In particular, a delay-synchronized volume adjustment system configured in accordance with an embodiment of the present disclosure includes a controller operatively coupled to the cascaded network of audio and configured to apply gain adjustments in a synchronized manner that accounts for operating delays that are inherent to each gain stage. In an embodiment, the controller synchronously adjusts each gain stage relative to a corresponding operating delay such that gain adjustments fully propagate at substantially a same point in time within a given acceptable tolerance, and thus, eliminates or otherwise mitigates perceivable volume shifts when mixing audio from two or more audio sources.

Abstract: A MEMS microphone assembly includes a MEMS transducer element having a back plate and a diaphragm displaceable relative to the back plate. A bias voltage generator is adapted to provide a DC bias voltage applicable between the diaphragm and the back plate. An amplifier receives an electrical signal from the MEMS transducer element and provides an output signal. The amplifier is adapted to amplify the electrical signal from the MEMS transducer element according to an amplifier gain setting. A processor is adapted to carry out a calibration routine at power-on of the microphone assembly determining information regarding the DC bias voltage and/or the amplifier gain setting.