Abstract: A method for sound source localization in a digital system having at least two audio capture devices is provided that includes receiving audio signals from the two audio capture devices, computing a signal-to-noise ratio (SNR) for each frequency band of a plurality of frequency bands in a processing frame of the audio signals, determining a frequency band weight for each frequency band of the plurality of frequency bands based on the SNR computed for the frequency band, computing an estimated time delay of arrival (TDOA) of sound for the processing frame using the frequency band weights, and converting the estimated TDOA to an angle representing sound direction.

Abstract: A noise reduction system is provided. The noise reduction system may include a sub-band noise sensor, a plurality of sub-band noise reduction modules, and an output module. The sub-band noise sensor may be configured to detect a noise and generate a plurality of sub-band noise signals in response to the detected noise. Each of the plurality of sub-band noise signals may have a distinctive sub-band of the frequency band of the noise. Each of the sub-band noise reduction modules may be configured to receive one of the sub-band noise signals from the sub-band noise sensor and generate a sub-band noise correction signal for reducing the received sub-band noise signal. The output module may be configured to receive the sub-band noise correction signals and output a noise correction signal for reducing the noise based on the sub-band noise correction signals.

Abstract: A first system includes first and second buck-boost amplifiers. The first amplifier is connected to a battery, includes a first inductor and a first plurality of switches connected to the first inductor, and drives first and second loads. The second amplifier is connected to the battery, includes a second inductor and a second plurality of switches connected to the second inductor, and drives the first and second loads. A controller drives the first and second plurality of switches to operate each of the first and second amplifiers in a single inductor multiple output mode. A second system includes multiple buck-boost amplifiers connected to a battery and driving respective loads. Each amplifier includes inductors and switches connected to the inductors. A controller drives the switches to utilize one or more inductors based on an amount of power used by each amplifier to drive the respective loads.

Abstract: In a device or method for rendering a sound program for headphones, a location is received for placing the sound program with respect to first and second ear pieces. If the location is between the first ear piece and the second ear piece, the sound program is filtered to produce low-frequency and high-frequency portions. The high-frequency portion is panned according to the location to produce first and second high-frequency signals. The low-frequency portion and the first high-frequency signal are combined to produce a first headphone driver signal to drive the first ear piece. A second headphone driver signal is similarly produced. The sound program may be a stereo sound program. The device or method may also provide for a location that is between the first ear piece and a near-field boundary. Other aspects are also described.

Abstract: An audio system can be configured to generate an audio heatmap for the audio emission potential profiles for one or more speakers, in specific or arbitrary locations. The audio heatmap maybe based on speaker location and orientation, speaker acoustic properties, and optionally environmental properties. The audio heatmap often shows areas of low sound density when there are few speakers, and areas of high sound density when there are a lot of speakers. An audio system may be configured to normalize audio signals for a set of speakers that cooperatively emit sound to render an audio object in a defined audio object location. The audio signals for each speaker can be normalized to ensure accurate rendering of the audio object without volume spikes or dropout.

Abstract: Systems, apparatuses, and methods are described for a privacy blocking device configured to prevent receipt, by a listening device, of video and/or audio data until a trigger occurs. A blocker may be configured to prevent receipt of video and/or audio data by one or more microphones and/or one or more cameras of a listening device. The blocker may use the one or more microphones, the one or more cameras, and/or one or more second microphones and/or one or more second cameras to monitor for a trigger. The blocker may process the data. Upon detecting the trigger, the blocker may transmit data to the listening device. For example, the blocker may transmit all or a part of a spoken phrase to the listening device.

Abstract: Methods, apparatus, systems and articles of manufacture are disclosed for wear noise audio signature suppression. An example method disclosed herein includes generating an audio signature based on a media audio signal during a first time period, collecting acceleration data during the first time period, determining whether the acceleration data corresponds to wear noise having occurred during the first time period, and in response to determining the acceleration data corresponds to wear noise during the first time period, inhibiting transmission of the audio signature to a central facility.

Abstract: A technique for producing audio includes providing multiple audio tracks of respective sound sources of an audio performance and rendering a sound production of the audio performance at a listening venue by playing back the audio tracks on respective playback units.

Abstract: An active noise reduction device includes a first adaptive filter, a first filter coefficient updater, and a controller that determines, based on a first parameter of the first adaptive filter, whether first noise control based on a first cancelling sound is in a stable state or an unstable state. The controller transitions the first filter coefficient updater to a restriction state in which an effect of reducing first noise is smaller than in a normal state when it is determined that the first noise control is in the unstable state while the first filter coefficient updater is updating a coefficient of the first adaptive filter in the normal state, and transitions the first filter coefficient updater back to the normal state when it is determined that the first noise control is in the stable state while the first filter coefficient updater is in the restriction state.

Abstract: A three-dimensional (3D) sound localization method, comprising: evaluating distances between a target object and multiple microphones; distinguishing a quadrant in which the target object is located; evaluating multiple elevation angles and azimuth angles of the target object according to spatial coordinates of each of the microphones; setting searching intervals of distance variables between the target object and each of the microphones; generating multiple test points according to the elevation angles, the azimuth angles and the searching intervals of the distance variables; calculating fitness of each of the test points; obtaining fitness values of each of the test points and comparing the fitness values between each of the test points; and, when a convergence condition is reached according to the fitness values, generating a positioning result of the target object.

Abstract: There is provided a mechanism that enables cooperation of a plurality of audio processing devices. An earphone device includes a sensor unit that detects an overlapping state of the earphone device and a headphone device and a wireless communication unit that wirelessly communicates with the headphone device which is worn to overlap an outer side of the earphone device worn by a user.

Abstract: An audio decoder device for decoding a compressed input audio signal having at least one core decoder having one or more processors for generating a processor output signal based on a processor input signal, wherein a number of output channels of the processor output signal is higher than a number of input channels of the processor input signal, wherein each of the one or more processors has a decorrelator and a mixer, wherein a core decoder output signal having a plurality of channels has the processor output signal, and wherein the core decoder output signal is suitable for a reference loudspeaker setup; at least one format converter device configured to convert the core decoder output signal into an output audio signal, which is suitable for a target loudspeaker setup; and a control device configured to control at least one or more processors in such way that the decorrelator of the processor may be controlled independently from the mixer of the processor, wherein the control device is configured to contro

Abstract: Media input audio data corresponding to a media stream and microphone input audio data from at least one microphone may be received. A first level of at least one of a plurality of frequency bands of the media input audio data, as well as a second level of at least one of a plurality of frequency bands of the microphone input audio data, may be determined. Media output audio data and microphone output audio data may be produced by adjusting levels of one or more of the first and second plurality of frequency bands based on the perceived loudness of the microphone input audio data, of the microphone output audio data, of the media output audio data and the media input audio data. One or more processes may be modified upon receipt of a mode-switching indication.

Abstract: An electronic stethoscope device can be integrated into a conventional stethoscope to digitize auscultated sounds from the body of a patient. The device can be switched off so that the conventional stethoscope can be used as a standard stethoscope. When the device is switched on, the digitized auscultated sounds can be modified to remove the noise. Such modified sounds can be sent wirelessly from the electronic stethoscope device to a peripheral device that can receive such wireless signals, such as computer, cell phone, or cloud application, where the data can be viewed and manipulated further as desired.

Abstract: One or more embodiments include techniques for generating immersive audio for an acoustic system. The techniques include determining an apparent location associated with a portion of audio; calculating, for each speaker included in a plurality of speakers of the acoustic system, a perceptual distance between the speaker and the apparent location; selecting a subset of speakers included in the plurality of speakers based on the perceptual distances between the plurality of speakers and the apparent location; generating a set of filters based on the subset of speakers and one or more target characteristics of the acoustic system; and generating, for each speaker included in the subset of speakers, a speaker signal using one or more filters included in the set of filters.

Abstract: The present invention provides a judging circuit including a comparison module and a control module for positive and negative terminals of an earphone MIC and a method thereof, and an electronic device; the control module controls any one first terminal of two terminals of the MIC to inject an excitation current, and control the other second terminal of the two terminals to be connected to the ground via an analog switch, also to judge whether the first terminal and the second terminal is the positive terminal or the negative terminal according to comparison result information; the comparison module compares the current voltage with a preset reference voltage value after excitation current is injected into the first terminal and the second terminal is grounded to obtain comparison result information and to send it to the connected control module. The present invention improves the flow and process without two sets of detection device.

Abstract: The present technology relates to a signal processing apparatus and method, and a program that can easily determine a localization position of a sound image. A signal processing apparatus includes: an acquisition unit configured to acquire information associated with a localization position of a sound image of an audio object in a listening space specified in a state where the listening space viewed from a listening position is displayed; and a generation unit configured to generate a bit stream on the basis of the information associated with the localization position. The present technology can be applied to the signal processing apparatus.

Abstract: An active noise cancellation system (1) for cancelling environment noise perceived by a driver or passenger seated in a seat (3) mounted in a cabin of a vehicle, in combination with said seat, the seat comprising a seat cushion (19), a seat back (21) coupled to the seat cushion at a bottom end and extending upwards to a seat shoulder (23), and a headrest (22) coupled to the seat back, extending upwardly from the seat shoulder, the active noise cancellation system comprising an active noise cancellation circuit (ANC) (30), a plurality of microphones (10) mounted in the headrest and connected electrically to the ANC, and a plurality of speakers (16) mounted in the seat and connected electrically to the ANC circuit.

Abstract: An audio system for an ear mountable playback device comprises a speaker, an error microphone predominantly sensing sound being output from the speaker and a feed-forward microphone predominantly sensing ambient sound. The audio system further comprises a voice activity detector which is configured to record a feed-forward signal from the feed-forward microphone. Furthermore, an error signal is recorded from the error microphone. A detection parameter is determined as a function of the feed-forward signal and the error signal. The detection parameter is monitored and a voice activity state is set depending on the detection parameter.

Abstract: There is provided encoding and decoding methods for encoding and decoding of object based audio. An exemplary encoding method includes inter alia calculating M downmix signals by forming combinations of N audio objects, wherein M?N, and calculating parameters which allow reconstruction of a set of audio objects formed on basis of the N audio objects from the M downmix signals. The calculation of the M downmix signals is made according to a criterion which is independent of any loudspeaker configuration.