Abstract: This disclosure describes a conferencing system with spatial rendering of audio data. The conferencing system includes a local conferencing device that includes a plurality of microphones, an audio encoder, and a spatial encoder that are associated with a local endpoint. The conferencing system further includes a remote conferencing device that includes a plurality of speakers, an audio decoder, and a spatial renderer that are associated with the remote endpoint. The spatial renderer is configured to superimpose a coordinate system for the plurality of speakers and a coordinate system for the plurality of microphones during spatial rendering of the audio data.

Abstract: A system for ultrasonic leak detection employs a number of ultrasonic signal generators, each of which include an ultrasonic transducer generating an ultrasonic signal and an audio transducer generating an audio signal in the frequency range of human hearing that is a replica of the ultrasonic waveform. The ultrasonic signal generators are activated and placed at selected locations within the enclosure to be tested. The operator can listen for the audio signal to verify operation and the output waveform type of the ultrasonic signal generators. A mobile ultrasonic leak detector can then be employed outside the enclosure to detect any leakage of the ultrasonic signal through the enclosure wall to identify any leak paths.

Abstract: A interface circuit for an acoustic transducer provided with a first detection structure and a second detection structure has: a first input and a second input; a first processing path and a second processing path coupled, respectively, to the first input and second input and supply a first processed signal and a second processed signal; and a recombination stage, which supplies a mixed signal by combining the first processed signal and the second processed signal with a respective weight that is a function of a first level value of the first processed signal. The first and second inputs receive a respective detection signal associated, respectively, to the first detection structure and to the second detection structure of the acoustic transducer; and an output stage the first processed signal, the second processed signal or the mixed signal, on the basis of a second level value of the first processed signal.

Abstract: Provided are, among other things, systems, methods and techniques for processing an audio signal to add virtual bass. In one representative embodiment, an apparatus includes: (a) an input line that inputs an original audio signal in the time domain; (b) a bass extraction filter that extracts a bass portion of the original audio signal, which also is in the time domain; (c) an estimator that estimates a fundamental frequency of a bass sound within the bass portion; (d) a frequency translator that shifts the bass portion by a positive frequency increment that is an integer multiple of the fundamental frequency estimated by the estimator, thereby providing a virtual bass signal; (e) an adder having (i) inputs coupled to the original audio signal and to the virtual bass signal and (ii) an output; and (f) an audio output device coupled to the output of the adder.

Abstract: A system can include a hardware processor that can receive left and right audio signals and process the left and right audio signals to generate three or more processed audio signals. The three or more processed audio signals can include a left audio signal, a right audio signal, and a center audio signal. The processor can also filter each of the left and right audio signals with one or more first virtualization filters to produce filtered left and right signals. The processor can also filter a portion of the center audio signal with a second virtualization filter to produce a filtered center signal. Further, the processor can combine the filtered left signal, filtered right signal, and filtered center signal to produce left and right output signals and output the filtered left and right output signals.

Abstract: Provided are, among other things, systems, methods and techniques for processing an audio signal to add virtual bass. In one representative embodiment, an apparatus includes: (a) an input line that inputs an original audio signal; (b) a transform module that transforms the original audio signal into a set of frequency components; (c) a bass extraction filter that extracts a bass portion of such frequency components; (d) an estimator that estimates a fundamental frequency of a bass sound within such bass portion; (e) a frequency translator that shifts the bass portion by a frequency that is an integer multiple of the fundamental frequency estimated by the estimator, thereby providing a virtual bass signal; (f) an adder having (i) inputs coupled to the original audio signal and to the virtual bass signal and (ii) an output; and (g) an audio output device coupled to the output of the adder.

Abstract: An electronic apparatus comprises a housing exposed outside, a piezoelectric vibration element located inside the housing a vibration part having the piezoelectric vibration element in a main surface thereof to be vibrated by the piezoelectric vibration element to generate a sound, and a shock-absorbing part being in contact with the main surface and away from a peripheral edge of the main surface, the piezoelectric vibration element intervening between the shock-absorbing part and the peripheral edge.

Abstract: A system is provided that produces haptic effects. The system receives an audio signal that includes a low-frequency effects audio signal. The system further extracts the low-frequency effects audio signal from the audio signal. The system further converts the low-frequency effects audio signal into a haptic signal by shifting frequencies of the low-frequency effects audio signal to frequencies within a target frequency range of a haptic output device. The system further sends the haptic signal to the haptic output device, where the haptic signal causes the haptic output device to output one or more haptic effects.

Abstract: Examples described herein involve calibrating a playback device. An example implementation involves causing a network device to display a guide to calibrate at least one playback device, the guide comprising an indication that the network device is to be moved during a given time. The example implementation also involves detecting, via the network device during the given time, an audio signal played by the at least one playback device. The example implementation further involves causing, via the network device, identification of an audio processing algorithm based on data indicating the detected audio signal.

Abstract: A method of encoding a multi-channel 3-dimensional (3D) audio signal mixed with a multi-channel 3D object signal is provided. The method includes: obtaining a location parameter indicating a virtual location of the multi-channel 3D object signal on a multi-channel speaker layout based on a gain value of the multi-channel 3D object signal for each channel; and encoding the multi-channel 3D audio signal and the location parameter.

Abstract: Examples described herein involve calibration of a microphone of a network device. An example network device identifies, within a database of microphone acoustic characteristics, an acoustic characteristic of the microphone which corresponds to a particular characteristic of the network device. The network device calibrates a playback device based on at least the identified acoustic characteristic of the microphone.

Abstract: A method of interference suppression is provided that includes receiving a first audio signal from a first audio capture device and a second audio signal from a second audio capture device wherein the first audio signal includes a first combination of desired audio content and interference and the second audio signal includes a second combination of the desired audio content and the interference, performing blind source separation using the first audio signal and the second audio signal to generate an output interference signal and an output audio signal including the desired audio content with the interference suppressed, estimating interference remaining in the output audio signal using the output interference signal, and subtracting the estimated interference from the output audio signal to generate a final output audio signal with the interference further suppressed.

Abstract: Reducing feedback in an input signal that contains a signal component based on an output signal from a proximate output. The input signal is separated into a plurality of frequency bands by band pass filters. The power of signal in each band is determined, and the band signal with the greatest power is selected. That band's signal is sampled at a sampling rate, and at regular intervals one of the samples is selected. Blind signal separation is used to estimate signal sources from the selected samples. The estimated signals are compared to the output signal, and the estimated signal most similar to the output signal is subtracted from the input signal.

Abstract: A sound volume is adjusted in accordance with a volume adjustment coefficient ka calculated in accordance with a coefficient map with respect to an accelerator opening angle APO and another volume adjustment coefficient kv calculated in accordance with the coefficient map with respect to a vehicle speed V. The coefficient map with respect to vehicle speed V is set such that, as accelerator opening angle APO becomes higher, volume adjustment coefficient ka becomes larger in a sigmoid configuration. In addition, the coefficient map with respect to accelerator opening angle APO is set such that, as vehicle speed V becomes higher, volume adjustment coefficient kv becomes smaller in the sigmoid configuration.

Abstract: A device is provided. The device includes: a sensor adapted to receive an acoustic wave and generate an electrical signal in response to receipt of the acoustic wave; clock frequency detection circuitry adapted to receive a clock signal, detect a frequency of the clock signal and generate information representative of the frequency; and digital filter circuitry coupled to the clock frequency detection circuitry. The digital filter circuitry is adapted to: receive the clock signal and the information representative of the frequency; access a first set of one or more digital filter coefficient values; and selectively change the first set of the one or more digital filter coefficient values to a second set of one or more digital filter coefficient values based on the information representative of the frequency, wherein the one or more digital filter coefficient values are included within a plurality of digital filter coefficient values.

Abstract: Presented is a method and associated system for suppression of linear and nonlinear echo. The method includes dividing an input signal into several frequency bands in each of a several of time frames. The input signal may include an echo signal. The method further includes multiplying the input signal in each of the several frequency bands by a corresponding echo suppression signal. Calculating the corresponding echo suppression signal may include estimating a power of the echo signal in a particular frequency band as a sum of several component echo powers, each of the several component echo powers due to an excitation from a far-end signal in a corresponding one of the several frequency bands. Calculating the corresponding echo suppression signal may further include subtracting the power of the echo signal in the particular frequency band from a power of the input signal in the particular frequency band.

Abstract: A system for identifying a periodicity-related attribute in a multi-frequency signal. The system is configured to produce phase and delay modulated signals from a complex input signal. The system is designed to combine these phase and delay modulated signals to identify a periodicity-related attribute associated with the multi-frequency signal. The system is configured to produce a physically perceptible output based on the identified periodicity-related attribute.

Abstract: According to an example aspect of the present invention, an apparatus is provided comprising at least one processing core and at least one memory including computer program code, the at least one memory and the computer program code being configured to, with the at least one processing core, cause the apparatus at least to present a graphical user interface comprising a spatial representation and at least one element, the element being associated with at least one specific physical loudspeaker, and receive input concerning moving of the at least one element within the spatial representation, activate a sensory signal in a physical loudspeaker associated with the first element, determine a location in the spatial representation where the first element is moved to, and based at least in part on the determined location, assign a name to at least the first element and the physical loudspeaker associated with the first element.

Abstract: Examples described herein involve calibrating a playback device for a playback environment based on audio signals detected by a microphone of a network device as the network device moves about the playback environment. While the playback device is playing a first audio signal and the network device is moving within the playback environment from a first physical location to a second physical location, the network device may detect by a microphone of the network device, a second audio signal. The network device may then identify an audio processing algorithm based on data indicating the second audio signal, and transmit to the playback device, data indicating the identified audio processing algorithm. Similar functions may also be performed by the playback device being calibrated or a computing device, such as a server to coordinate calibration of the playback device.

Abstract: A self calibrating dipole microphone formed from two omni-directional acoustic sensors. The microphone includes a sound source acoustically coupled to the acoustic sensors and a processor. The sound source is excited with a test signal, exposing the acoustic sensors to acoustic calibration signals. The responses of the acoustic sensors to the calibration signals are compared by the processor, and one or more correction factors determined. Digital filter coefficients are calculated based on the one or more correction factors, and applied to the output signals of the acoustic sensors to compensate for differences in the sensitivities of the acoustic sensors. The filtered signals provide acoustic sensor outputs having matching responses, which are subtractively combined to form the dipole microphone output.