Abstract: An audio session management method may involve: determining, by an audio session manager, one or more first media engine capabilities of a first media engine of a first smart audio device, the first media engine being configured for managing one or more audio media streams received by the first smart audio device and for performing first smart audio device signal processing for the one or more audio media streams according to a first media engine sample clock; receiving, by the audio session manager and via a first application communication link, first application control signals from the first application; and controlling the first smart audio device according to the first media engine capabilities, by the audio session manager, via first audio session management control signals transmitted to the first smart audio device via a first smart audio device communication link and without reference to the first media engine sample clock.

Abstract: A system and method are described herein for configuring an audio distribution system, comprising a Redis server, the Redis server adapted to store Redis data to be used in configuring the audio distribution system; a plurality of audio devices, the plurality of audio devices and Redis server interconnected to form the audio distribution system, wherein each of the plurality of audio devices comprises—at least one processor; an electronic communications interface operatively connected to the at least one processor and adapted to receive data from a user and transfer the data to the at least one processor; and a memory operatively connected with the at least one processor, wherein the memory stores computer-executable instructions that, when executed by the at least one processor, causes the at least one processor in a first audio device to execute a method for configuring the audio distribution system that comprises: establishing communications using the electronic communications interface between the user an

Abstract: A beamformer, for a differential microphone array (DMA) including a number M of microphones, is constructed based on a specified target directivity factor (DF) value for the DMA. An N order beampattern is generated for the DMA, wherein N is an integer and a first DF value corresponding to the N order beampattern is greater than the target DF value. An N?1 order beampattern is generated for the DMA, wherein a second DF value corresponding to the N?1 order beampattern is greater than the target DF value. A fractional order beampattern is generated for the DMA, wherein a third DF value corresponding to the fractional order beampattern matches the target DF value and the fractional order beampattern comprises a first fractional contribution from the N order beampattern and a second fractional contribution from the N?1 order beampattern.

Abstract: A speaker includes: a housing including a ring-shaped frame, a front cover and a bottom shell; a vibration system and a magnetic circuit system, the vibration system including a diaphragm fixed on the frame, a voice coil support fixed on the diaphragm and a voice coil supported by the voice coil support, the magnetic circuit system including a first magnetic yoke fixed on the front cover and a magnet assembly fixed on the first magnetic yoke. A first opening is provided to penetrate a center portion of the diaphragm, and the diaphragm includes an inner edge enclosing the first opening and an outer edge fixed to the frame, the magnetic circuit system extends through the first opening, the housing further includes an extension portion extending from the front cover toward the diaphragm and surrounding the magnetic circuit system, and the inner edge of the diaphragm is fixed to the extension.

Abstract: A device may be configured to: obtain at least a first audio signal and a second audio signal, wherein the first audio signal and the second audio signal are captured with a microphone array comprising at least two microphones; identify a first direction for a plurality of frequency bands; and identify a second direction for the plurality of frequency bands; wherein the first direction and the second direction are identified based, at least partially, on spatial analysis of the first audio signal and the second audio signal.

Abstract: A receiver is provided in the present invention. The receiver includes: a housing having a hollow inner cavity; a diaphragm mechanism disposed in the hollow inner cavity, partitioning the hollow inner cavity into a first cavity and a second cavity, and including a vibration plate; and an electromagnetic driving mechanism disposed in the hollow inner cavity and including a coil assembly and at least one magnetic field generation member. Each magnetic field generation member is disposed in the first cavity or the second cavity, and the magnetic field generation member is close to a free end of the vibration plate. The coil assembly is disposed in the second cavity and includes a coil and a magnetic core. The coil and the vibration plate are placed in the same direction. The magnetic core is inserted in a hollow inner hole of the coil.

Abstract: An audio device includes a first earbud and a second earbud, each of which is configured to establish respective first and second wireless links with an audio source, and receive, from the audio source, audio information over the respective first and second wireless links. The first earbud and the second earbud are further configured to communicate with one another over a third wireless link. At least one of the first earbud and the second earbud includes a measurement circuit configured to measure respective signal strengths of the first, second and third wireless links, a computation circuit configured to compute a difference in signal strengths of the first and second wireless links, and a determination circuit configured to determine an operating environment of the audio device based on the computed difference and further based on the signal strength of the third wireless link.

Abstract: Embodiments described herein provide a combined multi-source time difference of arrival (TDOA) tracking and voice activity detection (VAD) mechanism that is applicable for generic array geometries, e.g., a microphone array that lies on a plane. The combined multi-source TDOA tracking and VAD mechanism scans the azimuth and elevation angles of the microphone array in microphone pairs, based on which a planar locus of physically admissible TDOAs can be formed in the multi-dimensional TDOA space of multiple microphone pairs. In this way, the multi-dimensional TDOA tracking reduces the number of calculations that was usually involved in traditional TDOA by performing the TDOA search for each dimension separately.

Abstract: An audio device can sense sound in a physical environment using a plurality of microphones to generate a plurality of microphone signals. Clean speech can be extracted from microphone signals. Ambience can be extracted from the microphone signals. The clean speech can be encoded at a first compression level. The ambience can be encoded at a second compression level that is higher than the first compression level. Other aspects are also described and claimed.

Abstract: A display apparatus includes: a display panel configured to be rolled or unrolled, at least one vibration module on a rear surface of the display panel, the at least one vibration module being configured to: be rolled or unrolled with the display panel, and vibrate the display panel, and a partition on the rear surface of the display panel, the partition being spaced apart from and surrounding the at least one vibration module, wherein the at least one vibration module includes: a plurality of first portions having a piezoelectric property, and a plurality of second portions between the plurality of first portions, the plurality of second portions having flexibility.

Abstract: An audio headset receives one or more audio signals carrying one or more audio channels and processes the audio channels to generate stereo signals for output to a left and a right speaker of the audio headset. The processing determines a number of the audio channels carried in the received audio signal(s), adjusts level(s) of the audio channels based on the determined number of audio channels and/or adjusts gain and/or phase of the audio channels to control a perceived location of a listener wearing the headset relative to a source of sounds carried in the stereo signals.

Abstract: An information processing device includes a signal acquisition unit that acquires a voice signal of an object person outputted from a mic array and a control unit that acquires at least one of noise level information indicating a noise level of noise and first information as information indicating whether or not an obstructor is speaking while obstructing speech of the object person and changes a beam width as a width of a beam corresponding to an angular range of acquired sound, centering at the beam representing a direction in which voice of the object person is inputted to the mic array, and dead zone formation intensity as a degree of suppressing at least one of the noise and voice of the obstructor inputted to the mic array based on at least one of the noise level information and the first information.

Abstract: A system for generating a virtual engine sound may include a server having information on at least one review of the virtual engine sound, and a virtual engine sound generator to generate “my sound” by tuning the virtual engine sound based on information on customized design settings, to update a preset driver emotion model for each emotion evaluation area of the virtual engine sound, based on the information on the at least one review of the virtual engine sound, which is received from the server, and to update the virtual engine sound based on the “my sound” and the driver emotion model updated for each emotion evaluation area.

Abstract: Volume limiting systems and methods are operable to limit volume output from media presentation devices. An exemplary embodiment detects a sound using a microphone, wherein the sound corresponds to an audio output of at least one controlled media presentation device, and wherein the microphone is remotely located from the at least one controlled media presentation device; compares a level of the detected sound with a predefined maximum volume limit; generates a volume output limit command in response to the detected sound exceeding the predefined maximum volume limit; and communicates the volume output limit command to the media presentation device. The media presentation device then reduces a volume level of its audio output. In some instances, volume may be limited during user specified periods.

Abstract: Embodiments of the present disclosure disclose an audio processing method, an audio processing apparatus, and storage medium. The audio processing apparatus obtains first audio data of a target speaker acquired by the audio acquisition device on the target speaker, and obtains second audio data acquired by the electronic device on where the target speaker is located. According to a position of the target speaker relative to the electronic device, the first audio data is combined with the second audio data to obtain target audio data, where, when the target audio data is output, voice of the target speaker aligns with the spatial location. Accuracy of the spatial effect of the target speaker's voice in the target audio data is relatively ensured.

Abstract: An integrated system for hearing protection and enhancement having earpieces including microphones and receivers to provide for user control of ambience level (mic gain), monitor input, control over limiting of excessive levels, and frequency equalization, all with very wide input dynamic range capability and controlled output dynamic range. The user control comprises a wired or wireless smartphone app. Also, the wide input dynamic range is achieved by placement of limiter circuitry capable of handling the entire input dynamic range prior to the most dynamic range limited circuit blocks: digital EQ block, output amplifier, earpiece receiver, and the users' ears. Switches allow combining left and right signals to be directed to a user's favored ear, if necessary.

Abstract: Audio rendering devices comprising at least one audio rendering unit for playing audio streams, processing circuit and a wireless communication unit for establishing wireless links and related method are disclosed. One audio rendering device is configured to receive a remote audio stream prioritization policy comprising one or more lists of associations between audio stream attributes and priority values, update a local audio stream prioritization policy based on the remote audio stream prioritization policy, identify audio streams available via wireless links, retrieve audio stream attributes of the available audio streams, select an available audio stream to be played among the available audio streams, based on the updated local audio stream prioritization policy and based on the audio stream attributes of the available audio streams, and play the selected audio stream, by the audio rendering unit of the audio rendering device.

Abstract: The present disclosure relates to a device for processing an audio signal. The device may include a first acoustic-electric transducer and a second acoustic-electric transducer. The first acoustic-electric transducer may have a first frequency response, and may be configured to detect the audio signal and generate a first sub-band signal according to the detected audio signal. The second acoustic-electric transducer may have a second frequency response, the second frequency response being different from the first frequency response. The second acoustic-electric transducer may be configured to detect the audio signal and generate a second sub-band signal according to the detected audio signal.

Abstract: An appropriate level index is obtained from an input audio waveform. An audio signal processing method includes detecting one or more level values in each of a plurality of attack sections included in an audio signal including the plurality of attack sections and a plurality of non-attack sections different from the plurality of attack sections; and generating a histogram of the detected level values in each of the plurality of attack sections, the generated histogram excluding level values of the plurality of non-attack sections.

Abstract: Herein disclosed is a system that may be implemented within a headphone to facilitate hearing testing. Implementation of the system in the headphone may include implementation of tone generation circuitry and sound pressure level measurement circuitry being implemented in the headphone limiting the amount of calibration for accurate measurement when performing a hearing test.