Abstract: Aspects of the present disclosure provide methods, apparatuses, and systems for closed-loop respiration entrainment based on spatialized audio signals. According to an aspect, based on a determined rate of a subject's respiration, spatialization of a virtual sound source is altered to simulate a distance or directionality to the subject. Simulating the distance, directionality, and/or source characteristics comprises processing sounds of the virtual sound source to generate a perception of the sounds being heard from one or more distances or directions with reference to the subject. The altered virtual sound source attempts to regulate the rate of respiration of the subject. An audio device outputs the sounds of the altered virtual sound source.

Abstract: An electronic device having a device housing includes a loudspeaker and several microphones within the device housing. A control circuit is electrically coupled to the loudspeaker and microphones. The loudspeaker produces speech and/or music. The control circuit determines a statistical measure for a first data set representing individual impulse responses from the plurality of microphones and compares that to a predetermined statistical measure for a second data set representing individual object-free impulse responses from the plurality of microphones to determine if an object is near the device. The statistical measure may be variance and may be computed in the time domain. Variance may be calculated using differences between the individual impulse responses and a mean impulse response that is a linear combination of the impulse responses for the plurality of microphones. The control circuit may include echo cancellers to mitigate common signals and/or other acoustic sources.

Abstract: A wireless music microphone comprises a central processing unit module, a sound source receiving module, a sound source signal processing module, a sound collecting module, a sound signal processing module, a reverberation processing module and a signal transmitting module. The sound source receiving module receives a song transmitted by devices such as a mobile phone and a tablet computer via a wireless network, the sound source signal processing module removes the original sound from the received music and retains the accompaniment according to a user setting; and a user sound, music or accompaniment is polished and mixed by the reverberation processing module, and is then sent to a loudspeaker box by the signal transmitting module via the wireless network. The multifunctional wireless music microphone is applicable to various wireless communication modes; moreover, the song and user's sound can be effectively mixed, amplified and polished.

Abstract: An example implementation may involve a control device displaying a prompt to initiate a calibration sequence that involves calibration of a playback device for a given environment in which the playback device is located. The example implementation may also involve the control device displaying (i) a prompt to prepare a playback device for calibration within a given environment, (ii) a prompt to prepare the given environment for calibration of the playback device, and/or (iii) a prompt to prepare the control device for calibration of the playback device. The example implementation may also involve a control device displaying a selectable control, that, when selected, initiates calibration of the playback device. The example implementation may further involve a control device initiating calibration of the playback device.

Abstract: Examples described herein involve providing playback device calibration user interfaces to guide a calibration process for one or more playback devices in a playback environment. In one example, a network device receives audio samples continuously from a microphone of the network device for a predetermined duration of time, wherein the predetermined duration of time comprises a plurality of periodic time increments. At each time increment within the predetermined duration of time, the network device dynamically updating on a graphical display of the network device, (i) a representation of a frequency response based on audio samples that have been received between a beginning of the predetermined duration of time and the respective time increment, and (ii) a representation of the respective time increment relative to the predetermined duration of time.

Abstract: A noise reduction device includes: a noise detector that detects noise in a space at least partially surrounded by a shell; a noise controller that generates a control sound signal for reducing the noise detected by the noise detector; and a control sound outputter that outputs control sound to the space based on the generated control sound signal. A seat that is reclinable including a recliner having a seat surface at a variable angle with respect to a reference plane is provided in the space. When L (cm) represents the length of the recliner in the front-to-back direction, the control sound outputter is provided at a height of L×sin ?+8 (cm) above the reference plane where ? is between two degrees and eight degrees, inclusive.

Abstract: The present invention relates to an acoustic signal supply circuit of at least one loudspeaker (HP), this circuit comprising a filtering device of the resonance peak occurring at a given frequency of the supply current, characterized in that the filtering device of the peak is incorporated in a first branch bypassing the intermediate circuit between at least two converters (A, A0), this filtering device being purely electrical in the form of an impedance (Z3) connected, on the one hand, at a point on the intermediate circuit and, on the other hand, to a ground instrumentation, the impedance being called RLC (Z3) for comprising at least one first resistor (R3), at least one first capacitor (C3) and at least one first inductor (L3) arranged in series, the parameters of the first resistor (R3), the first capacitor (C3) and the first inductor (L3) being predetermined as a function of the resonance peak to be filtered.

Abstract: Aspects of the disclosure pertain to a system and method for providing temperature limiting for a voice coil of a speaker. The system and method provide the aforementioned temperature limiting based upon monitoring (e.g., measurement) of an amplifier output signal provided to the speaker. Providing the aforementioned temperature limiting promotes improved protection for the speaker.

Abstract: A method for controlling interference in an audio service and a terminal, where the method includes sending, by a first terminal, a first message using a short-range wireless communications technology when executing an audio service, determining, by a second terminal according to the first message, whether the second terminal supports interference control, and adjusting, by the second terminal, a setting of the second terminal according to the interference control parameter when the second terminal supports the interference control, and the second terminal falls within a controlled range. It can be learned that in a process in which the first terminal performs interference control on the second terminal, the first terminal may send the first message to the second terminal without a need of a noise reduction device such that the second terminal adjusts the setting of the second terminal.

Abstract: A sound source apparatus for receiving an audio signal and displaying the audio signal is provided. The sound source apparatus includes at least two sound source output devices, at least two distance sensors, and a processor. Each distance sensor is installed respectively corresponding to each sound source output device, and is configured to respectively detect at least one target to generate sensing signals. The processor receives the audio signal, and when the processor determines that the at least one target meets a sensing range through respectively receiving the sensing signals generated by each distance sensor, the processor calculates output volume of each sound source output device respectively according to a relative distance between each distance sensor and the at least one target.

Abstract: A security latching mechanism attached to a first device for removably securing the first device to a second device is provided, comprising an upper latching plate, a lower latching plate, and a base plate adapted to retain the upper latching plate and the lower latching in a sliding cooperative engagement with respect to each other and the base plate, and wherein each of the upper and lower latching plates further comprise a latch portion adapted to cause a sliding movement for its respective latching plate from a first position to a second position when the respective latching portions are in sliding contact with a security latching mechanism receptacle in the second device, such that the security latching mechanism can pass through the security latching mechanism receptacle when the second position has been reached by each of the upper and lower latching plate, and upon passing through the security latching mechanism receptacle, each of the upper latching plate and lower latching plate returns to the first

Abstract: A touch screen control device is described herein with integrated speakers. Each of the left and right speakers are mounted on respective side walls such that their long axis is substantially vertical, and the bulk of the speaker, which is contained within the frame of touch screen control device, faces inwardly, in a substantially horizontal direction, so that a minimal amount of depth is needed to mount touch screen control device.

Abstract: A turkey sensor that is used to scout an area without the hunter being present. The device locates wildlife by recording their vocals primarily turkey and duck roosts in the first 30 minutes of daylight in the early morning.

Abstract: A speaker system includes a case, an audio input, speakers, an accelerometer, and a computer processor. The audio input is structured to receive a program audio signal from an audio device. The speakers are configured to play an audio output based on the program audio signal, the audio output causing a vibration of the case. The accelerometer is configured to detect the vibration of the case as well as a user tap on the case. The computer processor is configured to identify a user gesture that includes the tap on the case, to identify the tap apart from the case vibration by processing the detected vibration of the case and the detected user tap on the case based on information from the program audio signal to separate the detected user tap from the detected vibration, and to commence a particular function associated with the user gesture.

Abstract: The present disclosure regards a hearing device configured to receive acoustical sound signals and to generate output sound signals comprising spatial cues.

Abstract: A processor-implemented method for foreground signal suppression. The method includes: capturing a plurality of input signals using a plurality of sensors within a sound field; subjecting each input signal to a short-time Fourier transform to transform each signal into a plurality of non-overlapping subband regions; estimating the diffuseness of the sound field based on the plurality of input signals; decomposing each of the plurality of input signals into a diffuse component and a directional component based on the diffuseness estimate; applying a spatial analysis operation to filter the directional component of each of the plurality of input signals, wherein the spatial analysis operation includes applying a set of beamformers to the directional components to produce a plurality of beamformer signals; and processing the plurality of beamformer signals to decompose the signal into a foreground channel and a background channel.

Abstract: A small speaker for panel mounting in a hole as small as seventy five millimeters diameter in which access to the rear of the panel is not required. The small speaker uses spring-loaded flip dogs that are actuated by rotating a sound disperser that is connected to a hollow cylindrical mechanism having gear teeth on its external surface which engage flip dog gears on the top surfaces of the flip dogs. Internally, the hollow cylindrical mechanism has an indentation that supports a speaker baffle which, in turn supports a speaker bushing. A loudspeaker is attached to the speaker bushing near the speaker baffle. Two housing sections attach, spaced apart, to an annular frame on opposing sides and receive the hollow cylindrical mechanism rotationally therein. The flip dogs operate in the spaces between the housing sections. Methods of mounting and dismounting are described.

Abstract: A processor-implemented method for capturing and reproducing spatial sound. The method includes: capturing a plurality of input signals using a plurality of sensors within a sound field; subjecting each input signal to a short-time Fourier transform to transform each signal into a transformed signal in the time-frequency domain; decomposing each of the transformed signals into a directional component and a diffuse component; optimizing beamformer weights using vector based amplitude panning to determine an optimal directivity pattern for the diffuse component of each transformed signal; constructing a set of diffuse sound channels using the diffuse components of the transformed signals and the optimized beamformer weights; constructing a set of directional sound channels using the directional components of the transformed signals; and reproducing the sound field by distributing the directional and diffuse sound channels to a plurality of output devices.

Abstract: A first microphone is operated in a low power sensing mode, and a buffer at the first microphone is used to temporarily store at least some of the phrase. Subsequently the first microphone is deactivated, then the first microphone is re-activated to operate in normal operating mode where the buffer is no longer used to store the phrase. The first microphone forms first data that does not include the entire phrase. A second microphone is maintained in a deactivated mode until the trigger portion is detected in the first data, and when the trigger portion is detected, the second microphone is caused to operate in normal operating mode where no buffer is used. The second microphone forms second data that does not include the entire phrase. A first electronic representation of the phrase as received at the first microphone and a second electronic representation of the phrase as received at the second microphone are formed from selected portions of the first data and the second data.

Abstract: A selective audio source enhancement system includes a processor and a memory, and a pre-processing unit configured to receive audio data including a target audio signal, and to perform sub-band domain decomposition of the audio data to generate buffered outputs. In addition, the system includes a target source detection unit configured to receive the buffered outputs, and to generate a target presence probability corresponding to the target audio signal, as well as a spatial filter estimation unit configured to receive the target presence probability, and to transform frames buffered in each sub-band into a higher resolution frequency-domain. The system also includes a spectral filtering unit configured to retrieve a multichannel image of the target audio signal and noise signals associated with the target audio signal, and an audio synthesis unit configured to extract an enhanced mono signal corresponding to the target audio signal from the multichannel image.