Abstract: Disclosed is an artificial intelligence hub including: a main body having a speaker for outputting sound; a communication module for wirelessly communicating with a surround ding device; a base arranged below the main body to support the main body; a cover coupled to a top of the main body, the cover having an upper surface for displaying a screen based on information exchanged via the communication module; and a grille having a vertically elongated cylindrical shape and provided with a plurality of through-holes therein, the main body being arranged within the grille, the grille being coupled at an upper end thereof to the cover housing and coupled at a lower end thereof to the base.

Abstract: The present disclosure provides a terminal device, a wireless headset, and an electronic device component. The terminal device includes: a body, a magnetic component, and a first controller. An accommodating groove for accommodating the wireless headset is disposed on the body. The magnetic component is at least partially disposed in the accommodating groove. The first controller is disposed in the body and electrically connected to the magnetic component, and is configured to control the magnetic component to connect to the wireless headset by attraction or to separate from the wireless headset by repulsion.

Abstract: An audio processing system is described for an audio event detection (AED) system. The system includes a feature extraction block configured to derive at least one feature which represents a spectral feature of the input signal.

Abstract: Systems and methods are described for compensating for inaccurate echo prediction in audio systems. A signal may be received at a microphone of an audio system, the signal including audio rendered using a spatial audio renderer across a multi-channel audio output. A signal may be received from the spatial audio renderer that indicates a change in rendering of audio. The audio system may then determine if there is echo power within the received signal greater than an expected echo power. After the signal from the spatial audio renderer has been received, the echo suppression applied to the received signal may be modified in response to a determination that the echo power is greater than the expected echo power, the echo suppression attenuating pre-selected frequency bands of the received signal.

Abstract: An automatic speech recognition (ASR) triggering system, and a method of providing an ASR trigger signal, is described. The ASR triggering system can include a microphone to generate an acoustic signal representing an acoustic vibration and an accelerometer worn in an ear canal of a user to generate a non-acoustic signal representing a bone conduction vibration. A processor of the ASR triggering system can receive an acoustic trigger signal based on the acoustic signal and a non-acoustic trigger signal based on the non-acoustic signal, and combine the trigger signals to gate an ASR trigger signal. For example, the ASR trigger signal may be provided to an ASR server only when the trigger signals are simultaneously asserted. Other embodiments are also described and claimed.

Abstract: A display device includes a display panel including a substrate and a pixel array layer disposed on a first surface of the substrate; a first sound generator disposed on a second surface of the substrate, which is opposite to the first surface of the substrate, the first sound generator being configured to vibrate the display panel in accordance with a first sound signal to output first sound; and a second sound generator disposed on the second surface of the substrate, the second sound generator configured to vibrate the display panel in accordance with a second sound signal to output second sound, wherein the first sound includes both sound in a low-frequency range and sound in a high-frequency range, which is higher than the low-frequency range, and the second sound includes sound in the high-frequency range.

Abstract: A system for digitally modelling analog devices having a processor executing an executable code to select a baseline calibration signal, receive a digitized audio from an analog device having parameters with incremental controls for the parameters, wherein the digitized audio is the baseline calibration signal processed with the incremental control of the first parameter set at a neutral setting, determine a baseline impulse response of the analog device, select an incremental signal, receive a plurality of incremental audios, wherein each incremental audio corresponds to the incremental signal processed by the analog device with a unique setting of the incremental control for the first parameter, derive formant-based filters for each of the incremental audios, and construct a digital model of the analog device based on the baseline impulse response of the analog device and the plurality of formant-based filter representations of the plurality of incremental audios.

Abstract: In at least one embodiment, a loudspeaker assembly for a vehicle is provided. The assembly includes a diaphragm; a loudspeaker, and a carrier. The loudspeaker includes a first end for being positioned in a first area of the vehicle that is exposed to external ambient noise to the vehicle and a second end for being positioned in a second area of the vehicle that is exposed directly within an interior cabin of the vehicle to provide desired audio along a first axis to the interior cabin. The carrier is attached to the second end of the loudspeaker and includes noise absorption material to prevent the external ambient noise from entering into the vehicle. The carrier defines a plurality of openings positioned on an outer perimeter thereof to enable the desired audio to enter into the interior cabin along a second axis that is different than the first axis.

Abstract: A speaker apparatus includes a first wireless communicator connected with a first source apparatus to receive a first audio signal from the first source apparatus or connected with an external speaker apparatus to transmit the first audio signal to the external speaker apparatus; a second wireless communicator configured to be connected with a second source apparatus to receive a second audio signal from the second source apparatus; and a controller configured to control the first wireless communicator and the second wireless communicator. The controller controls the first wireless communicator to switch from a reception mode to a transmission mode to transmit the second audio signal to the external speaker apparatus in response to the second wireless communicator being connected with the second source apparatus.

Abstract: The technology described in this document can be embodied in a method that includes receiving an input signal representing audio captured by a microphone of an active noise reduction (ANR) headphone, processing, by one or more processing devices, a portion of the input signal to determine a noise level in the input signal, and determining that the noise level satisfies a threshold condition. The method also includes, in response to determining that the noise level satisfies the threshold condition, generating an output signal in which ANR processing on the input signal is controlled in accordance with a target loudness level of the output signal, and driving an acoustic transducer of the ANR headphone using the output signal.

Abstract: An adaptive active noise cancellation apparatus performs a filtering operation in a first digital domain and performs adaptation of the filtering operation in a second digital domain.

Abstract: A method and audio apparatus for processing an audio signal are provided. The audio apparatus includes at least one microphone to acquire ambient sound of the audio apparatus, a speaker to output the audio signal, an air pressure regulator including a fluid tube connecting an external space of a housing of the audio apparatus to an internal space of the housing, and configured to adjust a change in an air pressure of the internal space of the housing and an audio signal processor configured to generate an anti-noise signal for canceling noise in the ambient sound by using the acquired ambient sound and output the generated anti-noise signal and the audio signal through the speaker.

Abstract: A method and system directed to controlling audio devices with active noise reduction. The system detects an instability condition in a first headphone; generates one or more control signals to adjust one or more ANR parameters of the first headphone using a first controller, wherein the one or more ANR parameters are adjusted to change the first headphone from a first ANR state to a second ANR state to mitigate the instability condition; and synchronizes the one or more ANR parameters of the first headphone with second headphone. In an example, the system returns the first headphone to the first ANR state after detecting that the first headphone was removed from an ear of a user at the first time and detecting that the first headphone was engaged with the ear at the second time.

Abstract: Embodiments described herein relate to earphone device supports and earphone support cases. In embodiments described herein, an earphone support is configured to be securely positioned over an earphone device. The earphone support includes a main body with a supporting element extending therefrom. The supporting element has a curvature shaped to follow a contour of the earphone device such that it clips onto to the earphone device. In additional embodiments described herein, an earphone support case is configured to securely store the earphone supports and mate with an earphone device case. The earphone support case has a main body coupled to a lid by a hinge. The main body includes a plurality of earphone support mounts. Each of the earphone support mounts are configured to support one of a plurality of earphone supports such that each earphone support is securely positioned within the earphone support case.

Abstract: A system that incorporates teachings of the present disclosure may include, for example, a computing device having a controller to modify sound produced by a gaming application according to an otological profile of a user. Additional embodiments are disclosed.

Abstract: The present disclosure provides a volume adjusting method, a device, and a terminal device. In the method, audio information of an audio source may be obtained when a terminal device is playing the audio source. An audio amplitude corresponding to the audio information is determined basing on the audio information. The audio amplitude is adjusted according to an amplitude adjustment coefficient, and the amplitude adjustment coefficient is assigned according to the value of the audio amplitude.

Abstract: An audio signal processing graph is automatically recomposed to optimize its properties by reassigning and instantiating its nodes on available locales such that the number of inter-locale connections is minimized and latency between inputs, such a performer's live input, and outputs, such as a monitored recording mix, is minimized. The recomposition exploits associative, commutative, and decomposition properties of certain node types, including mixer nodes. A graph recomposition may decompose a mixer node into a first stage instantiated by a software plug-in hosted by a computer running a digital audio workstation in a first locale, and a second stage assigned to an audio processing device in a second locale. Automatic signal graph recomposition occurs when the system is initialized, the graph is reconfigured, a new desired behavior of the graph is specified, or the available network resources are changed.

Abstract: The application relates to HFR (High Frequency Reconstruction/Regeneration) of audio signals. In particular, the application relates to a method and system for performing HFR of audio signals having large variations in energy level across the low frequency range which is used to reconstruct the high frequencies of the audio signal. A system configured to generate a plurality of high frequency subband signals covering a high frequency interval from a plurality of low frequency subband signals is described.

Abstract: An adjusting system and an adjusting method for equalization processing are provided. Frequency band energies of sound receiving signals are obtained. The frequency band energies correspond to different frequency bands, respectively. Target gains corresponding to frequency bands are determined according to the frequency band energies. Frequency responses of filtering processing with respect to a plurality of center frequencies are obtained. Equalization gains corresponding to the frequency bands and having the least gain error are determined. The gain error is related to a difference between the amplitude obtained after the equalization gains are reflected on the frequency responses corresponding to the filtering processing and the target gains. The equalization gains are inputted into the filtering processing according to the corresponding frequency bands. Accordingly, the impact of the filtering processing can be reduced.

Abstract: A front left speaker cover for a front left speaker includes a first hole forming portion in which holes each having a bottom are formed. The first hole forming portion is a section including a first edge of the front left speaker cover. The holes include a first hole and a second hole. The first hole is formed in the first edge of the front left speaker cover and opens in a surface direction of the front left speaker cover. The second hole is formed in a section different from the first edge and a fourth edge. The distance from the surface of the front left speaker cover to the bottom of the first hole is shorter than the distance from the surface of the front left speaker cover to the bottom of the second hole.