Abstract: A method and circuit for testing an acoustic sensor are disclosed. In a first aspect, the method comprises using electro-mechanical features of the acoustic sensor to measure characteristic of the acoustic sensor. In a second aspect, the method comprises utilizing an actuation signal to evaluate mechanical characteristics of the acoustic sensor. In a third aspect, the method comprises using a feedthrough cancellation system to measure a capacitance of the acoustic sensor. In the fourth aspect, the circuit comprises a mechanism for driving an electrical signal into a signal path of the acoustic sensor to cancel an electrical feedthrough signal provided to the signal path, wherein any of the electrical signal and the electrical feedthrough signal are within or above an audio range.

Abstract: A signal processing device includes an acoustic echo cancellation (AEC) circuit, a blocking matrix circuit, a controller, a subtractor, and a filter. The AEC circuit performs an AEC operation based on a far-end signal and a first input signal to generate a processed signal. The blocking matrix circuit suppresses a target signal component of the first input signal and a second input signal, to generate a reference signal. The controller generates a control coefficient based on the processed signal and the second input signal. The subtractor generates an output signal based on the filtered signal and the processed signal. The filter generates the filtered signal in response to the control coefficient, the reference signal, and the output signal.

Abstract: The invention relates to a method for processing customized data representative of the directivity of an customized audio system, the method comprising the following steps: —obtaining (101), for each individual of an initial set of individuals, at least one customized suite of filters; —determining (103) N independent components common to the suites of filters obtained; —decomposing (104) each of the suites obtained into a first base constructed from the N independent components with a view to obtaining, for each suite of filters, a first suite of weighting coefficients; —decomposing (105) each first suite of weighting coefficients into a second base of P independent components, so as to obtain a second suite of weighting coefficients; —storing (106) each second suite of weighting coefficients obtained in association with an identifier of an individual from among the initial set of individuals.

Abstract: Sound waves cause pressure changes in the air, and the pressure changes cause changes in the dielectric constant of air. Capacitive sensor measurements indicative of the changes in the dielectric constant of air can be processed to extract features associated with sound waves in the air. The features can include sound pressure levels represented and recordable as audio samples. Furthermore, the features can help identify types of sounds, determine direction of travel of the sound waves, and/or determine the source location of the audio. Instead of relying on movement of a mechanical member to transduce sound waves through a port into an electrical signal, an improved microphone uses capacitive sensing to directly sample and sense static pressure as well as dynamic pressure or pressure changes in the air to derive audio samples. The resulting microphone avoids disadvantages of the conventional microphone having the moving mechanical member and port.

Abstract: A method in a computing device configured to perform volume-levelling processing on input audio data by at least applying one or more filters to the input audio data, the computing device being configured to obtain an estimated difference between a target output loudness level and a loudness level associated with the input audio data, and to adapt the filter coefficients of the one or more filters based on the estimated difference. The method involves starting, or stopping, the volume-levelling processing. The method comprises gradually increasing, or decreasing, a weighting applied to the estimated difference, in response to obtaining an indication to start, or stop, the volume-levelling processing.

Abstract: A method for the acoustic analysis of a machine (M) including the acquisition of at least one acoustic signal supplied by at least one microphone (7) positioned in the machine, characterized in that it comprises the following steps: separation of at least one acoustic signal into a plurality of sound sources, the signal being modelled as a mixture of components, each one corresponding to a sound source; for at least one separated sound source, determination of a characteristic acoustic signature; comparison of at least one characteristic acoustic signature with at least one reference acoustic signature recorded in a reference database (5).

Abstract: A method and apparatus for outputting an audio signal and a method of adjusting the volume of an audio signal are provided. An embodiment of the present invention provides a method for outputting an audio signal that includes: (a) determining an output level of a modulated signal to be outputted for each respective frequency band by using a hearing threshold measured for the respective frequency band; and (b) outputting the modulated signal and an audio signal simultaneously. An embodiment of the present invention can improve hearing ability by outputting an audio signal and a modulated signal together. Also, according to an embodiment of the present invention, a user can use visual signals to personally set the output level of audio signals that is optimal for each frequency band in accordance with changes in the user's dynamic hearing threshold for each respective frequency band, even while listening to audio signals of which the output level for each frequency band was determined previously.

Abstract: A pre-distortion system for improved mobile device communications via cancellation of nonlinear distortion is disclosed. The pre-distortion system may transmit an acoustic signal from a network to a device, wherein the acoustic signal includes a linear signal and a nonlinear cancellation signal that cancels at least a portion of nonlinear distortions created once a loudspeaker in the device emits the linear signal. Thus, when a loudspeaker of a mobile device is operating and nonlinear distortions are generated by the loudspeaker or adjacent components of the mobile device in close proximity to the loudspeaker, the pre-distortion system may create one or more nonlinear cancellation signals in the network. The nonlinear cancellation signal may be combined with the linear signal sent to the loudspeaker to cancel the nonlinear distortion signal created by the loudspeaker emitting acoustic sounds from the linear signal. Thus, the nonlinear cancellation signal becomes a pre-distortion signal.

Abstract: An audio processing apparatus includes an audio input unit configured to input a plurality of audio signals, a reduction unit configured to reduce the amount of wind noise in the plurality of input audio signals, and a controller configured to control the reduction amount of the wind noise by the reduction unit based on a difference between an in-phase component and an antiphase component of the plurality of audio signals and control the speed of change of the reduction amount of the wind noise by the reduction unit based on a magnitude of the in-phase component of the plurality of audio signals. The audio processing apparatus reduces the adverse effects of the wind noise reduction process on the audio quality of the audio signals.

Abstract: In at least one embodiment, an apparatus for generating a simulated vehicle sound is provided. At least one parameter module receives a first driver throttle signal and to transmit a first input signal indicative of a first simulated vehicle sound. An engine sound generator block audibly generates a first simulated vehicle sound in response to the first input signal. An engine sound removal block is operably coupled to a microphone that receives a first microphone input signal. The first microphone input signal includes first environmental noise associated with a vehicle exterior environment and the first simulated vehicle sound. The engine sound removal block is configured to filter the first simulated vehicle sound from the first microphone input signal to determine the first environmental noise. The engine sound generator generates a second simulated vehicle sound that is louder than the first environmental noise.

Abstract: Embodiments of active noise control (ANC) headphones and operating methods thereof are disclosed herein. In one example, a headphone for ANC includes a speaker, a microphone, an echo-cancel module, and an ANC module. The speaker is configured to play an audio based on a first audio source signal. The microphone is configured to obtain a mixed audio signal including a noise signal and a second audio source signal based on the audio played by the speaker. The echo-cancel module is configured to reduce the second audio source signal from the mixed audio signal based on the first audio source signal to generate an echo-cancel audio signal. The ANC module is operatively coupled to the echo-cancel module and configure to generate a noise-controlled audio source signal to be played by the speaker based on the echo-cancel audio signal and the first audio source signal.

Abstract: Provided are an actuator fixing device and a panel vibration type sound-generating display device including the same. A display device includes: a display panel configured to display an image, a cover bottom configured to cover the display panel, and a plurality of sound-generating actuators supported by the cover bottom, the sound-generating actuators being configured to vibrate the display panel to generate sound, at least two of the sound-generating actuators being adjacent to each other.

Abstract: Embodiments of wireless audio systems and methods for operation mode switch of wireless headphones are disclosed herein. In one example, a wireless audio system includes a first wireless headphone and a second wireless headphone. The first wireless headphone is configured to establish a first short-range wireless link with an audio source; transmit, to a second wireless headphone, link information associated with the first short-range wireless link; and remove the first short-range wireless link with the audio source in response to the second wireless headphone successfully establishing a second short-range wireless link with the audio source based on the link information. The second wireless headphone is configured to receive, from the first wireless headphone, the link information associated with the first short-range wireless link; and establish the second short-range wireless link with the audio source based on the link information.

Abstract: Encoding of Higher Order Ambisonics (HOA) signals commonly results in high data rates.

Abstract: The disclosure includes a system and method for sonically customizing an audio reproduction device. The system includes a processor and a memory storing instructions that when executed cause the system to determine an application environment associated with an audio reproduction device associated with a user; determine one or more sound profiles based on the application environment; provide the one or more sound profiles to the user; receive a selection of a first sound profile from the one or more sound profiles; and generate tuning data based on the first sound profile, the tuning data configured to sonically customize the audio reproduction device.

Abstract: An electronic device whose enclosure or housing panel is used as part of an acoustic system is described. The panel is divided into several sub-panels. For each sub-panel, the device includes one or more actuators attached to vibrate the sub-panel. The actuator and its attached sub-panel convert an audio signal to acoustic output. Each actuator and sub-panel combination may receive a separate audio signal. The device includes a digital signal processor for controlling each of the sub-panel driving audio signals. The device may further include one or more backing frames that are attached to the panel to provide boundary conditions to the sub-panels. The boundary conditions define a resonance frequency for each sub-panel.

Abstract: Provided are an actuator fixing device and a panel vibration type sound-generating display device including the same. A display device includes: a display panel configured to display an image, a cover bottom configured to cover the display panel, and a plurality of sound-generating actuators supported by the cover bottom, the sound-generating actuators being configured to vibrate the display panel to generate sound, at least two of the sound-generating actuators being adjacent to each other.

Abstract: Embodiments are described for designing a filter in a magnitude domain performing an impedance filtering function over a frequency domain to compensate for directional cues for the left and right ears of the listener as a function of virtual source angles during headphone virtual sound reproduction. The filter is derived by obtaining blocked ear canal and open ear canal transfer functions for loudspeakers placed in a room, obtaining an open ear canal transfer function for a headphone placed on a listening subject, and dividing the loudspeaker transfer functions by the headphone transfer function to invert a headphone response at the entrance of the ear canal and map the ear canal function from the headphone to free field.

Abstract: Described are techniques for distributing audio data to multiple audio devices for generation of a synchronized audio output. A master device may receive audio data from a content server or other remote data source and store the audio data in a local cache. The master device may provide the audio data to multiple slave devices using a single multicast transmission. In some cases, the master device may also provide the audio data to individual slave devices unable to receive the multicast transmission using a network connection. Each slave device may store the audio data in a local cache. To generate the audio output, each audio device may retrieve the audio data from the associated local cache, which may enable the audio data to be distributed without modifying the commands used by audio applications to generate the audio output.

Abstract: The present disclosure relates to a signal processing device, a signal processing method, and a program that allows for generating an input signal suitable for a multi-way speaker. A band dividing unit divides an audio signal into signals in a plurality of bands corresponding to respective bands of a plurality of speaker units of the multi-way speaker. A filter processing unit performs wave front synthesis filter processing on each of the audio signals in the respective bands having been divided into. The audio signal in each of the bands after the wave front synthesis filter processing is supplied to the speaker unit of the corresponding band in the multi-way speaker. The present disclosure is applicable, for example, to a signal processing device or other devices.