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: [Problem] To propose a mechanism capable of performing a noise cancellation process in an ear hole opening type audio processing device. [Solution] An audio processing device including: an audio information acquisition unit that acquires audio information; a holding unit that abuts on a cavum concha or an inner wall of an ear canal and holds the audio information acquisition unit in a space closer to an eardrum side than a tragus, in a state of being worn by a user; an opening portion that opens an ear hole to an outside; and a signal processing unit that generates a noise cancellation signal based on the audio information acquired by the audio information acquisition unit.

Abstract: Disclosed are techniques and devices which include a memory configured to store audio data within a first audio zone, or a second audio zone in a layered soundfield. The memory is coupled to one or more processors and the memory is configured to store the audio data in the first audio zone and the second audio data in the layered soundfield. The one or more processors are configured to receive an interaction command to control the audio data in the first audio zone and the second audio zone in the layered soundfield, and generate one or more indicators that the interaction command was received to control the audio data, in the first audio zone or the second audio zone of the layered soundfield.

Abstract: A downmixer for providing a downmix signal on the basis of a plurality of input signals is configured to determine a magnitude value of a spectral domain value of the downmix signal on the basis of a loudness information of the input signals. The downmixer is configured to determine a phase value of the spectral domain value of the downmix signal and the downmixer is configured to apply the phase value in order to obtain a complex valued number representation of the spectral domain value of the downmix signal on the basis of the magnitude value of the spectral domain value of the downmix signal. An audio encoder uses such a downmixer. A method for downmixing and a computer program are also described.

Abstract: A method of detecting singing of a user of a personal audio device, the method comprising: receiving a first audio signal comprising bone-conducted speech of the user from a first transducer of the personal audio device; monitoring a second audio signal output to a speaker of the personal audio device; and determining whether the user is singing based on the first audio signal and the second audio signal.

Abstract: A method includes obtaining a listening position associated with a user and obtaining audio and metadata corresponding to a rendering at the listening position. The method also includes obtaining a listening environment and determining an effect of the listening environment on the rendering at the listening position. The method further includes detecting audio interaction at the listening position by comparing an audio rendering level against a corresponding level threshold, and applying, by a processing device, an audio modification according to the audio interaction detection. Audio is rendered at the listening position based on the applied audio modification.

Abstract: To enable channel setting to be performed easily and accurately to speakers. Thus, provided is a speaker system including: N number of speakers, the N being three or more; and a signal processing device capable of communicating with each speaker. The signal processing device performs processing of recognizing two arrangement reference speakers with reception of notification that a designation operation has been received from a user, from two speakers among the N number of speakers, and processing of acquiring distance information between each speaker. The signal processing device recognizes a relative-position relationship between the N number of speakers, with the two arrangement reference speakers and the distance information between each speaker. Then, the signal processing device automatically sets a channel to each speaker, on the basis of the relative-position relationship recognized.

Abstract: A mixing apparatus includes: a first speaker set processing unit to a P-th speaker set processing unit. K-th speaker set processing unit (K being an integer from 1 to P) includes: a mic set processing unit configured to process acoustic signals output by two microphones of a corresponding microphone set and to output a first acoustic signal and a second acoustic signal. The mic set processing unit configured to process acoustic signals output by two microphones of a corresponding microphone set based on an expansion/contraction coefficient for determining an expansion/contraction rate of a sound field, a shift coefficient for determining a shift amount of a sound field, and an attenuation coefficient for determining an attenuation amount of an acoustic signal output by a microphone.

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: A method and system for encoding data in audio are provided. A sequence of time deltas is generated at least partially based on a set of data. At least some of the time deltas are less than a threshold at which a human naturally detects an echo. A second audio channel is generated from a first audio channel, the second audio channel being temporally shifted relative to the first audio channel using the sequence of time deltas. The first and second audio channels are played back simultaneously via at least one audio transducer. The composite audio channel is registered via at least one microphone and processed to identify the first and second audio channels that are at least partially relatively temporally shifted. A sequence of time deltas by which the second audio channel is shifted temporally relative to the first audio channel is determined, and a set of data is decoded at least partially therefrom.

Abstract: A headrest (1, 101, 102) is disclosed, with a support structure (2) and with at least one speaker basket (7.1, 7.2) for a speaker (6.1, 6.2) provided on the support structure (2). In order to achieve a simply designed and durable headrest (1, 101, 102), the proposed support structure (2) transitions integrally into the speaker basket (7.1, 7.2) and forms the latter.

Abstract: A portable speaker system includes a tunnel with at least one internal speaker facing inwardly into the tunnel such that acoustic resonance amplifies desirable frequencies of sounds emitted by the internal speaker. The portable speaker system includes solar panels and is waterproof. A display touch screen permits operation of the speaker system. The portable speaker system is adapted to receive input of audio files from a variety of sources. A non-transitory storage medium permits storage of the audio files and a CPU is programmed to permit playback of audio files in a desired sequence or from a desired live-streaming source. The smart speaker can utilize Bluetooth, Wi-Fi, and/or telecommunication connectivity to receive and/or play audio files, and can queue files from multiple sources for playback.

Abstract: Exemplary road and engine noise control systems and methods include directly picking up road noise from a structural element of a vehicle to generate a first sense signal representative of the road noise, directly picking up engine noise from an engine of the vehicle to generate a second sense signal representative of the engine noise, and combining the first sense signal and the second sense signal to provide a combination signal representing the combination of the first sense signal and the second sense signal. The systems and methods further include broadband active noise control filtering to generate a filtered combination signal from the combination signal, converting the filtered combination signal provided by the active noise control filtering into anti-noise and radiating the anti-noise to a listening position in an interior of the vehicle. The filtered combination signal is configured so that the anti-noise reduces the noise at the listening position.

Abstract: An earpiece includes an earpiece housing, a processor disposed within the earpiece, a speaker operatively connected to the processor, a microphone operatively connected the processor, and a global navigation satellite system (GNSS) receiver disposed within the earpiece. A system may include a first earpiece having a connector with earpiece charging contacts, a charging case for the first earpiece, the charging case having contacts for connecting with the earpiece charging contacts, and a global navigation satellite system (GNSS) receiver disposed within the charging case.

Abstract: An auxiliary sensing method and system based on sensory substitution includes: establishing a discrete horizontal plane included angle and a normal vector included angle in three-dimensional space as well as response functions corresponding to different discrete distances, and establishing an acoustic coding library of an object; obtaining a surrounding environmental video; obtaining object information and category of current environment; establishing a three-dimensional sound field of the object according to the image information, the object information and the three-dimensional coordinate information obtained in combination with the acoustic coding library of the object; and adjusting, for objects of different categories, the three-dimensional sound field of each object according to movement information and variations of images, and playing the three-dimensional sound field of each object in real time.

Abstract: The present technology relates to a signal processing device and method, and a program for improving reproducibility of a sound image with a small amount of calculation. A signal processing device includes a rendering method selection unit configured to select one or more methods of rendering processing of localizing a sound image of an audio signal in a listening space from among a plurality of methods, and a rendering processing unit configured to perform the rendering processing for the audio signal by the method selected by the rendering method selection unit. The present technology can be applied to a signal processing device.

Abstract: A signal processing method for an earphone includes: a motion state of a wearer of the earphone is detected by using an acceleration sensor arranged inside the earphone; a first microphone and a second microphone both arranged outside the earphone detect wind noise conditions corresponding to different frequency bands; and according to the motion state of the wearer of the earphone and the wind noise conditions corresponding to different frequency bands, operating modes of a feedforward filter and a feedback filter inside the earphone are adjusted, herein the feedforward filter and the feedback filter are configured for active noise cancellation of the earphone.

Abstract: The present disclosure discloses a sound receiving that includes an air conduction sound receiving circuit, a bone conduction sound receiving circuit, an adaptive filter, a crossover frequency control circuit and a synthesis circuit. The air conduction sound receiving circuit generates an air conduction sound signal. The bone conduction sound receiving circuit generates a bone conduction sound signal. The adaptive filter performs calculation according to a minimum of an error function in real time to generate a transferring filter function to filter the bone conduction sound signal to generate a transferred bone conduction sound signal. The crossover frequency control circuit determines a crossover frequency according to a maximum energy frequency point of the transferring filter function on a frequency domain.

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 method of calibrating a feedback-based noise cancellation system of an ear device may involve obtaining a measured plant response of the ear device, obtaining a reference plant response value and determining a plant response variation between the reference plant response value and a value corresponding to the measured plant response. The method may involve obtaining a measured a coupler response of the ear device, obtaining a reference coupler response value and determining a coupler response variation between the reference coupler response value and a value corresponding to the measured coupler response. The method may involve determining, based at least in part on the plant response variation and the coupler response variation, a microphone signal gain correction factor and applying the microphone signal gain correction factor to ear device microphone signals that are input to a feedback loop of the feedback-based noise cancellation system.