Abstract: The embodiments of the present application provide an earphone abnormality processing method, an earphone, a system, and a storage medium. In the embodiments of the present application, an abnormal state self-detection function is added to an earphone of an existing dual-microphone, and when the earphone is in a self-detection state, acquiring a sound signal picked up from a specified sound source by a primary microphone and a secondary microphone; and determining the type of abnormal state of the earphone according to a frequency response curve of the sound signal picked up by the primary microphone and the secondary microphone, and further performing abnormality processing on the earphone by using a processing manner adapted to the abnormal state of the earphone, thereby solving the difficulty in the prior art of being unable to process the sound pickup abnormality of an earphone, and improving the usage performance of an earphone and also facilitating the prolonging of the service life of the earphone.

Abstract: An audio output device comprises a loudspeaker and, four stem control inputs substantially even spaced on the upper surface of a housing. The stem control inputs are each divided into four segments, each comprising a touch sensor and an illumination unit. Each stem control input is associated with a different stem in an audio track for playback via the loudspeaker. Touching the segments of each stem control input results in the touch being detected and a touch signal being output by the respective segment. In response to a touch signal, the associated illumination unit is togged on/off and a property of the associated stem is varied during playback by loudspeaker.

Abstract: The following coding scenario is addressed: A number of audio source signals need to be transmitted or stored for the purpose of mixing wave field synthesis, multi-channel surround, or stereo signals after decoding the source signals. The proposed technique offers significant coding gain when jointly coding the source signals, compared to separately coding them, even when no redundancy is present between the source signals. This is possible by considering statistical properties of the source signals, the properties of mixing techniques, and spatial hearing. The sum of the source signals is transmitted plus the statistical properties of the source signals, which mostly determine the perceptually important spatial cues of the final mixed audio channels. Source signals are recovered at the receiver such that their statistical properties approximate the corresponding properties of the original source signals. Subjective evaluations indicate that high audio quality is achieved by the proposed scheme.

Abstract: The following coding scenario is addressed: A number of audio source signals need to be transmitted or stored for the purpose of mixing wave field synthesis, multi-channel surround, or stereo signals after decoding the source signals. The proposed technique offers significant coding gain when jointly coding the source signals, compared to separately coding them, even when no redundancy is present between the source signals. This is possible by considering statistical properties of the source signals, the properties of mixing techniques, and spatial hearing. The sum of the source signals is transmitted plus the statistical properties of the source signals, which mostly determine the perceptually important spatial cues of the final mixed audio channels. Source signals are recovered at the receiver such that their statistical properties approximate the corresponding properties of the original source signals. Subjective evaluations indicate that high audio quality is achieved by the proposed scheme.

Abstract: The following coding scenario is addressed: A number of audio source signals need to be transmitted or stored for the purpose of mixing wave field synthesis, multi-channel surround, or stereo signals after decoding the source signals. The proposed technique offers significant coding gain when jointly coding the source signals, compared to separately coding them, even when no redundancy is present between the source signals. This is possible by considering statistical properties of the source signals, the properties of mixing techniques, and spatial hearing. The sum of the source signals is transmitted plus the statistical properties of the source signals, which mostly determine the perceptually important spatial cues of the final mixed audio channels. Source signals are recovered at the receiver such that their statistical properties approximate the corresponding properties of the original source signals. Subjective evaluations indicate that high audio quality is achieved by the proposed scheme.

Abstract: A method of generating sounds for reducing an effect of tinnitus is disclosed. The method includes the following steps: acquiring a sound frequency of a tinnitus sound of a user, and playing a plurality of pure tones within a frequency range during a playback time, wherein the plurality of pure tones include a plurality of first pure tones and a plurality of second pure tones; the plurality of first pure tones and the sound frequency are located at a first frequency area, and the plurality of second pure tones are located at a second frequency area; the first frequency area covers X Hz, where 100?X?12000, the second frequency area covers the remaining portion of the frequency range excluding the first frequency area, and the plurality of first pure tones accounts for M % of the plurality of pure tones, where 50<M?90.

Abstract: A headphone includes a speaker unit, a baffle plate, an ear pad, and a grille. The baffle plate is configured to support the speaker unit. The ear pad is configured to surround an acoustic space in which a sound emitted from the speaker unit is transmitted toward an ear of a listener. The grille is configured to cover the speaker unit. The grille has a hole and a cavity connected to the hole.

Abstract: The disclosure describes systems and methods for processing audio signals in a vehicle to perform sound source separation. The sound source separation is performed using transfer functions and involves separation of the speech of multiple occupants. The separated speech can be used to isolate and correctly respond to a command to control vehicle systems.

Abstract: Method for setting parameters for individual adaptation of an audio signal, including: performing a first listening test with the substeps: playing a plurality of first audio signals having different levels; obtaining feedback per frequency range from an individual which of the plurality of first acoustic signals is above an individual listening threshold; and using the lowest level of the different levels for which feedback is available as a level for the individual listening threshold per frequency range; performing adaptation of a second audio signal with the substeps: playing the second audio signal according to a total volume level considering a sound adaptation characteristic map; and varying the sound adaptation characteristic wherein the levels for the individual listening thresholds are used as minimum output levels in the sound adaptation characteristic map.

Abstract: A microelectromechanical systems (MEMS) device comprises a diaphragm assembly and a force feedback system. The diaphragm assembly includes a first diaphragm and a second diaphragm facing the first diaphragm, with a low pressure region being defined therebetween. The diaphragm assembly further includes a first plurality of electrodes, a second plurality of electrodes, and a third plurality of electrodes. A solid dielectric is spaced between the first and second diaphragms and includes a plurality of apertures. Each electrode of the first, second, and third pluralities of electrodes is disposed at least partially within an aperture of the plurality of apertures. The force feedback system receives output from the diaphragm assembly and produces a feedback voltage that is applied to the diaphragm assembly to produce an electrostatic force on the diaphragm assembly that counters a low-frequency pressure across the diaphragm assembly.

Abstract: An audiovisual communication system includes a plurality of environment sound sources, a microphone array, an image processing device, an audio processing device and a denoise processing device. Each of the plurality of environment sound sources has an environment sound. A plurality of microphones of the microphone array receive the plurality of environment sounds and output a plurality of receiving audio signals according to the plurality of environment sounds respectively. The image processing device obtains an image including the plurality of environment sound sources and selects one of the plurality of environment sound sources in the image as a target sound source according to a selection command. The image processing device calculates a relative position between the target sound source and the microphone array according to a first coordinate. The audio processing device receives the plurality of receiving audio signals and calculates a target audio signal.

Abstract: Loudspeaker enclosures and loudspeaker devices are provided. In some embodiments, a loudspeaker device is provided, the loudspeaker device comprising: a loudspeaker enclosure, comprising: a rear side; and a front side of the loudspeaker enclosure comprising: a lower portion of the front side comprising a first opening for receiving a first loudspeaker and a second opening for receiving a second loudspeaker; and an upper portion of the front side comprising a third opening for receiving a third loudspeaker, a first port, and a second port, wherein a distance from the lower portion of the front side to the rear side is greater than a distance from the upper portion of the front side to the rear side; the first loudspeaker positioned in the first opening; the second loudspeaker positioned in the second opening; and the third loudspeaker positioned in the third opening.

Abstract: One or more embodiments include an audio processing system for generating an audio scene for an extended reality (XR) environment. The audio processing system determines that a first virtual sound source associated with the XR environment affects a sound in the audio scene. The audio processing system generates a sound component associated with the first virtual sound source based on a contribution of the first virtual sound source to the audio scene. The audio processing system maps the sound component to a first loudspeaker included in a plurality of loudspeakers. The audio processing system outputs at least a first portion of the component for playback on the first loudspeaker.

Abstract: The following coding scenario is addressed: A number of audio source signals need to be transmitted or stored for the purpose of mixing wave field synthesis, multi-channel surround, or stereo signals after decoding the source signals. The proposed technique offers significant coding gain when jointly coding the source signals, compared to separately coding them, even when no redundancy is present between the source signals. This is possible by considering statistical properties of the source signals, the properties of mixing techniques, and spatial hearing. The sum of the source signals is transmitted plus the statistical properties of the source signals, which mostly determine the perceptually important spatial cues of the final mixed audio channels. Source signals are recovered at the receiver such that their statistical properties approximate the corresponding properties of the original source signals. Subjective evaluations indicate that high audio quality is achieved by the proposed scheme.

Abstract: Arrangements described herein relate to systems, apparatuses, and methods for a headset system that includes a transducer configured to collect physiological data of a subject, a device housing configured to support the transducer, an insert portion disposed on the device housing, a support structure comprising a baseplate and a receiving portion, and a head cradle supported by the baseplate. The receiving portion extends in a direction that is oblique with respect to the baseplate. The insert portion is sized and shaped to engage the receiving portion to removably attach the device housing to the baseplate.

Abstract: A method includes: generating a first and second shift signal by using a phase of a sound signal regarding an M-channel or a S-channel, the sound signal of the M-channel and the sound signal of the S-channel being obtained by using a mid-side microphone, the sound signal of the S-channel including a positive channel and a negative channel, the first shift signal being configured to reduce a phase difference caused by a difference between a sound arrival distance to the M-channel and a sound arrival distance to the positive channel of the S-channel, the second shift signal being configured to reduce a phase difference caused by a difference between the sound arrival distance to the M-channel and a sound arrival distance to the negative channel of the S-channel; and approximately converting the first or second shift signal into an L-channel signal and an R-channel signal of an XY-microphone.

Abstract: Audio systems and methods for obtaining location information are disclosed. A portable, wireless speaker that may include a beacon that a user may activate to remotely trigger a request for shot information, which may include location information of the user or speaker. The beacon may be wirelessly coupled to the speaker. The speaker may receive location information from a remote device, such as a GPS-enabled mobile phone. The shot information may be converted into an audio format for playback by the speaker. The speaker may pause the playback of the speaker to play out the shot information.

Abstract: Concentric rings of speakers in a room are activated in sequence to emit test sounds that are picked up by microphones placed in the ears of a listener in the middle of the room. The output of the microphones is used to generate plural HRTF spheres that are concentric, and that can be convolved with each other to render a multi-distance HRTF filter that can be configured to “move” the perceived distance and bearing of audio objects “inside” the outer sphere as desired, without being restricted to placing the audio objects at the outer edges of a single sphere. Any desired placement of an audio object between spheres is done by interpolating between respective coefficients of the closest spheres.

Abstract: This applications relates to methods and apparatus for monitoring a socket (101), to detect a connection status of a mating plug (102), e.g. for monitoring an audio jack socket for connection of an audio jack plug. A monitor (115, 305) is configured to monitor a voltage (VM) at a monitoring node (114), which is coupled to a jack detect contact (112) of the socket and a voltage pull-up element (113). The voltage (VM) at the monitoring node (114) is monitored against a threshold (Vthv) and a threshold module (302) is configured to vary the threshold depending on an indication of signal activity (SACT) of a signal path for a first socket contact (103) which will be electrically connected to the jack detect contact when a plug when inserted in the socket.

Abstract: A first layer of data having a first set of Ambisonic audio components can be decoded where the first set of Ambisonic audio components is generated based on ambience and one or more object-based audio signals. A second layer of data is decoded having at least one of the one or more object-based audio signals. One of the object-based audio signals is subtracted from the first set of Ambisonic audio components. The resulting Ambisonic audio components are rendered to generate a first set of audio channels. The one or more object-based audio signals are spatially rendered to generate a second set of audio channels. Other aspects are described and claimed.