Abstract: The present invention provides a speaker, which includes a basket, a diaphragm fixed to the basket, and a voice coil located below the diaphragm and used for driving the diaphragm to vibrate and sound. The diaphragm includes a dome and a folded ring. The dome includes two first straight edges arranged oppositely and two second straight edges connected with the two first straight edges, the dome includes a central portion and an edge portion, the edge portion includes a first region arranged along the first straight edge and a second region arranged along the second straight edge, and both the first region and the second region are provided with an opening portion penetrating through the dome along a vibration direction.

Abstract: A reproduction device according to an embodiment is a reproduction device configured to output a reproduction signal to a speaker system including a first speaker unit group and a second speaker unit group disposed behind the first speaker unit group, the reproduction device including: first delay circuits configured to delay reproduction signals output to speaker units included in the first speaker unit group; and second delay circuits configured to delay reproduction signals output to speaker units included in the second speaker unit group. In the second speaker unit group, a delay time in accordance with a distance difference between a distance and a distance and a longitudinal distance is set for the second delay circuit.

Abstract: A personal sound amplification seating solution including integrated microphones, speakers, and a hand-operated-remote is disclosed. By detecting audio inputs from up to three (3) meters away, passing the audio inputs through a printed circuit board, and broadcasting the audio through personalized speakers proximate the user's ears, conversations and entertainment options such as television shows and movies can be accessed by those with reduced hearing functionality without need for increased volume.

Abstract: An apparatus configured to use captured spatial audio content captured by a rotatable spatial audio capture microphone array, the captured spatial audio content defining the three-dimensional arrangement of the audio in the scene, an orientation of the aural scene defined relative to the microphone array at the time of audio capture, and use captured direction data representative of the direction of a reference direction of said microphone array relative to the scene over a time during which said spatial audio content was captured; to generate modified spatial audio content in which the aural scene is decoupled from any rotation of the rotatable spatial audio capture microphone array that occurred during capture of the spatial audio content for presentation to the recipient user relative to a space in which the recipient user is located by virtue of modification of the orientation of the aural scene using the captured direction data.

Abstract: A stereo sound encoding method and system for encoding left and right channels of a stereo sound signal, down mix the left and right channels of the stereo sound signal to produce primary and secondary channels, encode the primary channel, and encode the secondary channel. Encoding the secondary channel comprises analyzing coherence between coding parameters calculated during the secondary channel encoding and coding parameters calculated during the primary channel encoding to decide if the coding parameters calculated during the primary channel encoding are sufficiently close to the coding parameters calculated during the secondary channel encoding to be re-used during the secondary channel encoding.

Abstract: A signal processing method performs echo reduction processing on at least one of a collected sound signal of a first microphone, a collected sound signal of a second microphone, or both the collected sound signal of the first microphone and the collected sound signal of the second microphone, and calculates a correlated component between the collected sound signal of the first microphone and the collected sound signal of the second microphone, using a collected sound signal of which echo has been reduced by the an echo reduction processing.

Abstract: The present disclosure discloses a method and an apparatus for testing a speaker, an electronic device and a storage medium. A specific implementation includes: obtaining first audio data recorded by a microphone integrated with the speaker in ambient white noise; analyzing the first audio data to derive a first analysis result; and determining whether there is a defect in the microphone according to the first analysis result. Hence, these allow for testing a completed set on an assembled speaker to ensure the consistency of a microphone test and improve the accuracy of the test result.

Abstract: A computing device is configured to perform functions comprising: receiving via a network microphone device of a media playback system, a voice command detected by at least one microphone of the network microphone device, wherein the media playback system comprises a plurality of zones, and the network microphone device may be a member of a default playback zone. The computing device may be further configured to perform functions comprising: dynamically selecting an audio response zone from the plurality of zones to play an audio response to the voice input and foregoing selection of the default playback zone. The selected zone may comprise a playback device, and the dynamically selecting may comprise determining that the network microphone device is paired with the playback device. The computing device may cause the playback device of the selected zone to play the audio response.

Abstract: In general, techniques are described for adapting audio streams for rendering. A device comprising a memory and one or more processors may be configured to perform the techniques. The memory may store a plurality of audio streams that include one or more sub-streams. The one or more processors may determine, based on the plurality of audio streams, a total number of the one or more sub-streams for all of the plurality of audio streams, and adapt, when the total number of the sub-streams is greater than a render threshold, the plurality of audio streams to decrease the number of the one or more sub-streams and obtain an adapted plurality of audio streams. The one or more processors may also apply the renderer to the adapted plurality of audio streams to obtain the one or more speaker feeds, and output the one or more speaker feeds to one or more speakers.

Abstract: Embodiments provide techniques for controlling an audio output for a headphone system. In one embodiment, a headphone system comprising a speaker, a communication interface configured to receive a first audio signal from a first audio source hosted on a first computing device and a master audio signal from a master audio source hosted on a second computing device separate from the first computing device, wherein the first audio source and master audio source are separate from the headphone system, and an audio control module, is configured to output the first audio signal on the speaker using a first volume level, receive the master audio signal, and in response, decrease a volume of the first audio signal relative to the first volume level, output the master audio signal on the speaker at a second volume level greater than the decreased volume of the first audio signal, and after determining the master audio signal is finished, increase the volume of the first audio signal.

Abstract: Systems and methods herein utilize psychoacoustic principles to process audio signals to achieve a desired result. In some embodiments, an exemplary process utilizes a combination of re-harmonization and auditory processing techniques in which a plurality of filtered audio cues are sampled, created and down-mixed into a final music output, which may be designed to trigger specific effects on the function of the body and the brain. In some embodiments, enhanced audio material may be altered during the stream or fixed for a desired result at the beginning of the stream. In some embodiments, a signal enhancement process may be embedded into silicon or a media player, decoder or device with software. In other embodiments, audio or video material may be encoded or finalized into the media file.

Abstract: Systems and methods of presenting an output audio signal to a listener located at a first location in a virtual environment are disclosed. According to embodiments of a method, an input audio signal is received. For each sound source of a plurality of sound sources in the virtual environment, a respective first intermediate audio signal corresponding to the input audio signal is determined, based on a location of the respective sound source in the virtual environment, and the respective first intermediate audio signal is associated with a first bus. For each of the sound sources of the plurality of sound sources in the virtual environment, a respective second intermediate audio signal is determined. The respective second intermediate audio signal corresponds to a reverberation of the input audio signal in the virtual environment.

Abstract: A mixing device includes: processing units each provided for a set of two microphones, each processing unit being configured to process acoustic signals output by the corresponding set of two microphones to output a first acoustic signal and a second acoustic signal; a first adding units configured to add up the first acoustic signals; and a second adding unit configured to add up the second acoustic signals. Each processing unit processes the acoustic signals output by the corresponding set of two microphones, based on a scaling factor that determines a scaling up/down rate of a sound field, a shift factor that determines a shift amount of the sound field, and attenuation factors that determine attenuation amounts of the acoustic signals output by the microphones.

Abstract: The present disclosure provides an audio playback system adapted to automatically switch an active microphone back and forth between two or more devices. For example, where the system is a pair of earbuds, where each earbud is worn by a separate user, the system may switch the active microphone to the device worn by the user that is speaking at a given time. While that device holds the active microphone, the other device may wait until a particular event that frees up the microphone, such as if the user wearing the device with the active microphone stops talking. According to some examples, a notification may be provided through one or more of the devices in the system to let the user know, for example, that he does not have the active microphone, that the active microphone is free, that the active microphone has been switched, etc.

Abstract: One or more earbuds comprise one or more external connectors for removably coupling the earbuds with an additional article. The one or more earbuds are additionally able to comprise magnets for removably coupling with each other. In some embodiment, the one or more earbuds are configured to removably couple with a base unit. Based on a coupling and decoupling of the one or more earbuds with the base unit, a signal is sent to control a remotely located electronic device.

Abstract: A signal processing method performs echo reduction processing on at least one of a collected sound signal of a first microphone, a collected sound signal of a second microphone, or both the collected sound signal of the first microphone and the collected sound signal of the second microphone, and calculates a correlated component between the collected sound signal of the first microphone and the collected sound signal of the second microphone, using a collected sound signal of which echo has been reduced by the an echo reduction processing.

Abstract: A head-wearable hearing device includes: a housing; a microphone arrangement configured to generate a microphone signal in response to incoming sound; a digital signal processor configured to generate a processed output signal based on the microphone signal in accordance with one or more audio processing algorithms; an impact sensor configured to generate an impact pulse or impact signal in response to mechanical impact of the housing; and a reset circuit configured to apply a reset signal to the digital signal processor to place the digital signal processor in a predetermined logic state; wherein the reset circuit is configured to generate the reset signal in response to the impact pulse or the impact signal generated by the impact sensor.

Abstract: Disclosed are a method and a device for controlling a terminal speaker, and a computer readable storage medium. A display screen of a terminal is divided into at least two displaying areas, each displaying area corresponding to at least one speaker, and the method includes: detecting whether a video to be displayed is processed by an artificial intelligence learning, after the video to be displayed sent by a server is received; if yes, acquiring a frame identification code in the video to be displayed; if a frame currently displayed by the terminal is a target frame corresponding to the frame identification code, determining position information of a target sound source according to the frame identification code corresponding to the target frame; and controlling the at least one speaker in the displaying area corresponding to the position information of the target sound source to output a sound corresponding to the target frame.

Abstract: A far-field, marine sound system including a coaxial speaker horn composed of a pair of horn bodies, each having an opening, a mouth and a sound pathway extending between the opening and the mouth. The pair of horns bodies include a first horn body contained within a second horn body, the horn bodies having aligned longitudinal axes. A first speaker is coupled to the first horn body and arranged to project sound into the first horn through the mouth, through the sound pathway and out of the first horn body through the opening, and a second speaker coupled to the second horn body and arranged to project sound into the second horn body through the mouth, through the sound pathway and out of the second horn body through the opening. The speaker horn bodies and speakers are contained with an aerodynamically-contoured housing.

Abstract: A system for detection of a target sound in an environment of a vehicle, includes an audio sensor, a computer processor, and a memory storing a digital target sound template produced by converting a sample of the target sound in accordance with conversion parameters. The computer processor receives a sound signal from the audio sensor, digitizes the sound signal in accordance with the conversion parameters, and determines a degree of similarity between the digitized signal and the digital target sound template. The sound signal may be logarithmically amplified before being digitized. The sound signal may be received from two audio sensors, and the direction of the target sound may be determined based on a difference between time indices for detection of the target sound for each audio sensor.