Abstract: The present disclosure provided a speaker monomer with a pair of dynamic speakers thereof includes a housing, an electro-acoustic conversion member received in the housing, and a PCB. The electro-acoustic conversion member includes a first dynamic speaker and a second dynamic speaker, and the housing is a double-barrel-shaped configuration and includes a first receiving room for receiving the first dynamic speaker therein, and a second receiving room arranged in parallel and separated from the first receiving room for receiving the second dynamic speaker therein. The PCB is installed in an end portion of the housing and electrically connected to the first and second dynamic speakers. The present disclosure not only can improve sound quality of an ear-in earphone, but also can reduce a volume of the ear-in earphone, improve assembly efficiency and reduce a defective rate of the ear-in earphone.

Abstract: Audio reproduction systems include a plurality of loudspeakers that are actuated in accordance with a multi-channel audio format. The loudspeakers can be configuratable by way of a network interface The loudspeakers are registered on the network.

Abstract: Methods, systems, and apparatuses are described herein for improved processing audio in a video stream. A system may split audio in a frame of video content into multiple bands based on their audio levels. The system may then dynamically compress and dynamically normalize the audio level in each band. When dynamically compressing the bands, the system may determine, based on stored information, what audio level range is acceptable for an end user and may smooth and maintain the ranges of the audio to be within the acceptable range. The system may include the dynamically normalized and dynamically compressed frames as a second audio track in the video content. A computing device receiving the video content may select the second audio track during playback. If an end user selects the second audio track, the video is delivered with the modified sound of the second audio track.

Abstract: Embodiments of systems and methods are described for reducing undesired leakage energy produced by a non-front-facing speaker in a multi-speaker system. For example, the multi-speaker system can include an array of forward-facing speakers, one or more upward-facing speakers, and/or one or more side-facing speakers. Filters coupled to any two of the speakers in the multi-speaker system can generate audio signals output by the coupled speakers to reduce, attenuate, or cancel a portion of an audio signal output by one or more non-front-facing speakers that acoustically propagates along a direct path from the respective non-front-facing speaker to a listening position in a listening area in front of the multi-speaker system.

Abstract: A frame for a loudspeaker cabinet including a connection device for connecting to another loudspeaker cabinet, the connection device including a eyelet and a locking system, the locking system including a front leg and a back leg, a slide disposed between the two legs and configured to slide between a high position and a low position under the action of a first elastic member connected to the slide, and an actuator configured to take a pulled position and a pushed position. The locking system being configured to take a locked configuration and a cocked configuration, the locking system passing from the locked configuration to the cocked configuration by movement of the actuator and from the cocked configuration to the locked configuration by movement of the slide from the low position to the high position.

Abstract: A connector includes a pushing member having a projection, and a contact made of a conductive material and having a projection accommodating portion of recess shape into which the projection is to be inserted, the projection having a protruding portion that is situated in a lateral surface of the projection with protruding from the lateral surface of the projection and approaches an inner surface of the projection accommodating portion when the projection is inserted into the projection accommodating portion, and a gap forming portion that is situated in the lateral surface of the projection around the protruding portion and forms a predetermined gap between the gap forming portion and the inner surface of the projection accommodating portion when the projection is inserted into the projection accommodating portion, the predetermined gap being wider than a distance between the protruding portion and the inner surface of the projection accommodating portion.

Abstract: A method for scaling audio content using reverberant energy for the placement of virtual audio sources in an artificial reality experience. The method comprises obtaining a set of critical distance parameters. The set of critical distance parameters include at least a reverberation time of a local area and a geometry of the local area. The method further comprises determining a critical distance for the local area based on the set of critical distance parameters and scaling an amplitude of audio content based in part on the critical distance and a distance between a target location in the local area and an artificial reality headset in the local area. The method further comprises presenting the audio content by the headset in accordance with the scaled amplitude.

Abstract: A connector includes a pushing member having a projection, and a contact having a projection accommodating portion, the projection accommodating portion having a protruding portion that is situated in an inner surface of the projection accommodating portion with protruding from the inner surface of the projection accommodating portion toward an inside of the projection accommodating portion and approaches a lateral surface of the projection when the projection is inserted into the projection accommodating portion, and a gap forming portion that is situated in the inner surface of the projection accommodating portion around the protruding portion and forms a predetermined gap between the gap forming portion and the lateral surface of the projection when the projection is inserted into the projection accommodating portion, the predetermined gap being wider than a distance between the protruding portion and the lateral surface of the projection.

Abstract: A front assembly sliding structure includes: a front slider provided with a first magnetic body, and a second magnetic body disposed on a panel. The front slider is slidable relative to the panel, and the first magnetic body is opposite to the second magnetic body when the front slider is in a first position. The front slider is configured to slide from the first position to a second position when a force in a first direction is present between the first magnetic body and the second magnetic body, and configured to slide from the second position to the first position when a force in a second direction is present between the first magnetic body and the second magnetic body, the first direction being opposite to the second direction.

Abstract: Determination of an acoustic filter for incorporating local effects of room modes within a target area is presented herein. A model of the target area is determined based in part on a three-dimensional virtual representation of the target area. In some embodiments, the model is selected from a group of candidate models. Room modes of the target area are determined based on a shape and/or dimensions of the model. The room mode parameters are determined based on at least one of the room modes and the position of a user within the target area. The room mode parameters describe an acoustic filter that as applied to audio content, simulates acoustic distortion at the position of the user and at frequencies associated with the at least one room mode. The acoustic filter is generated at a headset based on the room mode parameter and is used to present audio content.

Abstract: A technique for dynamic sweet spot calibration. The technique includes receiving an image of a listening environment, which may have been captured under poor lighting conditions, and generating a crowd-density map based on the image. The technique further includes setting at least one audio parameter associated with an audio system based on the crowd-density map. At least one audio output signal may be generated based on the at least one audio parameter.

Abstract: A method of fabricating an information handling system chassis may comprise operably connecting an A-cover housing a digital display to a chassis base, via a hinge, forming an opening into a space between the A-cover and the digital display, inserting a first interchangeable vibration actuator housing a first vibration actuator, selected from a plurality of different types of vibration actuators into the opening of the A-cover, such that a first surface of the first vibration actuator housing lies substantially coplanar with the rear surface of the A-cover, and a second surface of the first vibration actuator housing is situated adjacent the rear surface of the digital display. The method may also include transmitting code instructions to the first vibration actuator, and transducing the code instructions, via the first vibration actuator, to cause a first corresponding movement of at least a portion of the digital display to produce a vibration.

Abstract: The present disclosure provides systems and methods associated with acoustic transmitters, receivers, and antennas. Specifically, the present disclosure provides a transducer system for transmitting and receiving acoustic energy according to a determined acoustic emission/reception pattern. In various embodiments, an acoustic transducer system may include an array of sub-wavelength transducer elements each configured with an electromagnetic resonance at one of a plurality of electromagnetic frequencies. Each sub-wavelength transducer element may generate an acoustic emission in response to the electromagnetic resonance. A beam-forming controller may cause electromagnetic energy to be transmitted at select electromagnetic frequencies to cause a select subset of the sub-wavelength transducer elements to generate acoustic emissions according to a selectable acoustic transmission pattern. A common port may facilitate electromagnetic communication with each of the sub-wavelength transducer elements.

Abstract: According to an exemplary embodiment, a display apparatus includes: a display device configured to display a video data; and a plurality of integrated speakers configured to output a plurality of audio signals in a low-pitched tone band and a middle and high-pitched tone band. The plurality of integrated speakers may each include: a woofer speaker configured to output a first audio signal of the audio signals in the low-pitched tone band; and at least one front speaker which is adjacent to the woofer speaker and configured to output a second audio signal of the audio signals in the middle and high-pitched tone band.

Abstract: A garment connector includes a connector body, one or more contacts held by the connector body, one or more conductive members held by the connector body and corresponding to the one or more contacts, and a connector fixation member in a sheet-like shape held by the connector body and extending to an exterior of the connector body, the garment connector being attached to a garment by fixing the connector fixation member to the garment at the exterior of the connector body.

Abstract: An audio playback device comprises a microphone, a speaker, and a processor. The processor is arranged to output by the speaker first audio content and receive by the microphone an indication of the first audio content. A first acoustic response of a room in which the audio playback device is located is determined based on the received indication of first audio content. A mapping is applied to the first acoustic response to determine a second acoustic response. The second acoustic response is indicative of an approximated acoustic response of the room at a spatial location different from a spatial location of the microphone. The second audio content output by the speaker is adjusted based on the second response.

Abstract: Described herein is a technology related to a mobile device with a pair of stereo speakers. The mobile device has an orientation detection system that detects the orientation of the mobile device and a crosstalk cancellation system that performs crosstalk cancellation with complementary cardioid beams in response to the detected orientation of the mobile device. The mobile device also has an audio system that emits complementary cardioid sound beams from the speakers.

Abstract: Methods and apparatus for decoding a compressed Higher Order Ambisonics (HOA) representation of a sound or soundfield. The method may include receiving a bit stream containing the compressed HOA representation and decoding, based on a determination that there are multiple layers, the compressed HOA representation from the bitstream to obtain a sequence of decoded HOA representations. A first subset of the sequence of decoded HOA representations is determined based only on corresponding ambient HOA components. A second subset of the sequence of decoded HOA representations is determined based on corresponding ambient HOA components and corresponding predominant sound components.

Abstract: An apparatus for generating one or more audio channels is provided. The apparatus comprises a metadata decoder for generating one or more reconstructed metadata signals from one or more processed metadata signals depending on a control signal, wherein each of the one or more reconstructed metadata signals indicates information associated with an audio object signal of one or more audio object signals, wherein the metadata decoder is configured to generate the one or more reconstructed metadata signals by determining a plurality of reconstructed metadata samples for each of the one or more reconstructed metadata signals. The apparatus comprises an audio channel generator for generating the one or more audio channels depending on the one or more audio object signals and depending on the one or more reconstructed metadata signals. The metadata decoder is configured to receive a plurality of processed metadata samples of each of the one or more processed metadata signals.

Abstract: An apparatus for generating one or more audio channels is provided. The apparatus comprises a metadata decoder for generating one or more reconstructed metadata signals from one or more processed metadata signals depending on a control signal, wherein each of the one or more reconstructed metadata signals indicates information associated with an audio object signal of one or more audio object signals, wherein the metadata decoder is configured to generate the one or more reconstructed metadata signals by determining a plurality of reconstructed metadata samples for each of the one or more reconstructed metadata signals. The apparatus comprises an audio channel generator for generating the one or more audio channels depending on the one or more audio object signals and depending on the one or more reconstructed metadata signals. The metadata decoder is configured to receive a plurality of processed metadata samples of each of the one or more processed metadata signals.

Abstract: An apparatus for generating a plurality of audio channels for a first speaker setup is characterized by an imaginary speaker determiner, an energy distribution calculator, a processor and a renderer. The imaginary speaker determiner is configured to determine a position of an imaginary speaker not contained in the first speaker setup to obtain a second speaker setup containing the imaginary speaker. The energy distribution calculator is configured to calculate an energy distribution from the imaginary speaker to the other speakers in the second speaker setup. The processor is configured to repeat the energy distribution to obtain a downmix information for a downmix from the second speaker setup to the first speaker setup. The renderer is configured to generate the plurality of audio channels using the downmix information.

Abstract: Systems and methods are described for an adaptive audio system that renders reflected sound for adaptive audio systems in different ways depending on the orientation of at least one speaker in a set of speakers. A speaker of the system may comprise an integrated speaker having front-firing and upward-firing drivers, a sensor to determine the orientation of the speaker (e.g., horizontal or vertical) and a transceiver and control unit that transmits the orientation to a decoder and receives updated speaker feeds from the renderer based on the orientation.

Abstract: Various embodiments of the present disclosure provide an electronic gaming machine (EGM) and methods of operating an EGM that provide 3D audio synced with 3D objects or player gestures. In various embodiments, the EGM includes a display device configured to display 3D images to a player without requiring the player to wear 3D glasses, and a binaural audio system configured to provide 3D audio synced with 3D objects or player gestures in association with one or more plays of a primary or secondary game or other EGM functionality.

Abstract: Various embodiments of the present invention relate to providing an audible route sequence for guiding a user along a route. A user apparatus determines a user location, distance and directional aspects for one or more waypoint of the route, and a cadence. A distance aspect of a waypoint is a distance between the waypoint and the user location. A directional aspect of a waypoint is the angle between a reference heading and straight line connecting the waypoint to the user location. The cadence is determined based on the distance between consecutive waypoints. The user apparatus determines a route sequence based on the distance and directional aspects for the waypoints and the cadence. The route sequence comprises a sequence of audible pitches that each correspond to a waypoint of the route. The user apparatus causes the route sequence to by audibly provided via a sound device.

Abstract: Some disclosed methods may involve receiving audio reproduction data, including audio objects, differentiating near-field audio objects and far-field audio objects in the audio reproduction data, and rendering the far-field audio objects into speaker feed signals for room speakers of a reproduction environment. Each speaker feed signal may correspond to at least one of the room speakers. The near-field audio objects may be rendered into speaker feed signals for near-field speakers and/or headphone speakers of the reproduction environment. Reverberant audio objects may be generated based on physical microphone data from physical microphones in the reproduction environment and from virtual microphone data that is calculated for near-field audio objects. The reverberant audio objects may be rendered into speaker feed signals for the room speakers.

Abstract: Disclosed herein is an audio lighting device, which includes a casing, a circuit board located within the casing, an audio guiding device located below the casing, a speaker, a bass tube, a heat dissipation device, a plurality of audio signal outlets and a light source part. The audio guiding device has a top ring sheet and a bottom audio shield located therebelow; the top ring sheet and the casing are sealed to form an audio box. The speaker is located in a central opening of the top ring sheet. The bass tube is disposed on the top ring sheet of the audio guiding device, so that the audio box has open-end conduit that facilitates the propagation and output of the low-frequency audio signal.

Abstract: Embodiments relate to audio processing for opposite facing speaker configurations that results in multiple optimal listening regions around the speakers. A system includes a left speaker and a right speaker in an opposite facing speaker configuration, and a crosstalk cancellation processor connected with the left speaker and the right speaker. The crosstalk cancellation processor applies a crosstalk cancellation to an input audio signal to generate left and right output channels. The left output channel is provided to the left speaker and the right output channel is provided to the right speaker to generate sound including multiple crosstalk cancelled listening regions that are spaced apart.

Abstract: An apparatus for processing soundfield data is provided. The soundfield data defines a soundfield within a spatial reproduction region comprising at least one bright zone and at least one quiet zone. The apparatus comprises an applicator configured to apply a spatially continuously varying weighting function to the soundfield data in order to obtain weighted soundfield data defining a weighted soundfield, wherein the spatially continuously varying weighting function is configured to enhance the soundfield in at least one of the bright zone and the quiet zone.

Abstract: An apparatus for generating a filtered audio signal from an audio input signal includes a filter information determiner being configured to determine filter information depending on input height information wherein the input height information depends on a height of a virtual sound source. Moreover, the apparatus includes a filter unit being configured to filter the audio input signal to obtain the filtered audio signal depending on the filter information. The filter information determiner is configured to determine the filter information using selecting, depending on the input height information, a selected filter curve from a plurality of filter curves, or the filter information determiner is configured to determine the filter information using determining a modified filter curve by modifying a reference filter curve depending on the elevation information.

Abstract: Some implementations may involve receiving, via an interface system, personnel location data indicating a location of at least one person and receiving, from an orientation system, headset orientation data corresponding with the orientation of a headset. First environmental element location data, indicating a location of at least a first environmental element, may be determined. Based at least in part on the headset orientation data, the personnel location data and the first environmental element location data, headset coordinate locations of at least one person and at least the first environmental element in a headset coordinate system corresponding with the orientation of the headset may be determined. An apparatus may be caused to provide spatialization indications of the headset coordinate locations. Providing the spatialization indications may involve controlling a speaker system to provide environmental element sonification corresponding with at least the first environmental element location data.

Abstract: The present invention relates to an apparatus and a method for processing an audio signal, and more particularly, to an apparatus and a method for efficiently rendering a higher order ambisonics signal. To this end, provided are an audio signal processing apparatus, including: a pre-processor configured to separate an input audio signal into a first component corresponding to at least one object signal and a second component corresponding to a residual signal and extract position vector information corresponding to the first component from the input audio signal; a first rendering unit configured to perform an object-based first rendering on the first component using the position vector information; and a second rendering unit configured to perform a channel-based second rendering on the second component and an audio signal processing method using the same.

Abstract: An audio synchronization method includes: receiving a first audio signal from a first recording device; receiving a second audio signal from a second recording device; performing a correlation operation upon the first audio signal and the second audio signal to align a first pattern of the first audio signal and the first pattern of the second audio signal; after the first patterns of the first audio signal and the second audio signal are aligned, calculating a difference between a second pattern of the first audio signal and the second pattern of the second audio signal; and obtaining a starting-time difference between the first audio signal and the second audio signal for audio synchronization according to the difference between the second pattern of the first audio signal and the second pattern of the second audio signal.

Abstract: A method and an apparatus for determining an inter-channel time difference parameter are provided, so that precision of a determined ITD parameter can adapt to channel quality. The method includes: determining a target search complexity from plurality of search complexities, where the plurality of search complexities are in a one-to-one correspondence with plurality of channel quality values; and performing search processing on a signal on a first sound channel and a signal on a second sound channel according to the target search complexity so as to determine a first inter-channel time difference ITD parameter corresponding to the first sound channel and the second sound channel.

Abstract: Producing a virtual sound field may include receiving an audio signal associated with a remote sound source within a remote environment. The audio signal may be defined or recorded as a binaural recording and recorded from a remote set of binaural microphones. The audio signal may be indicative of a position of the remote sound source relative to the remote set of binaural microphones within the remote environment. Producing the virtual sound field may include determining a virtual position relative to the position of the remote sound source within the remote environment, generating a virtual sound field audio signal which simulates audio representing the remote sound source perceived from the virtual position within the remote environment relative to the position of the remote sound source within the remote environment, and playing back the virtual sound field to simulate the remote sound source as perceived from the virtual position.

Abstract: A device for calculating loudspeaker signals for a plurality of loudspeakers while using a plurality of audio sources, an audio source including an audio signal, includes a forward transform stage for transforming each audio signal, block-by-block, to a spectral domain so as to obtain for each audio signal a plurality of temporally consecutive short-term spectra, a memory for storing a plurality of temporally consecutive short-term spectra for each audio signal, a memory access controller for accessing a specific short-term spectrum among the plurality of short-term spectra for a combination consisting of a loudspeaker and an audio signal on the basis of a delay value, a filter stage for filtering the specific short-term spectrum for the combination of the audio signal and the loudspeaker by using a filter provided for the combination of the audio signal and the loudspeaker, so that a filtered shot-term spectrum is obtained for each combination of an audio signal and a loudspeaker, a summing stage for summing

Abstract: Disclosed are a main speaker, sub speaker and system including the same. The present invention includes a main speaker configured to receive a first audio signal from a first source device and output the received first audio signal and at least one sub speaker configured communicate with the main speaker by wire or wireless. Particularly, if the communication with the main speaker is connected, the sub speaker outputs the first audio signal. If the sub speaker is separated from the main speaker, the sub speaker outputs the second audio signal.

Abstract: Provided is an audio device, including: at least one processor; and at least one memory device that stores a plurality of instructions, which when executed by the at least one processor, cause the at least one processor to: play, with a given length of delay, a first-channel audio signal included in an audio signal; obtain one of operation information about operation of the audio device and setting information about playing of the audio signal; and transmit one of the operation information and the setting information, and at least a second-channel audio signal included in the audio signal, to another audio device.

Abstract: Methods, systems, and apparatuses are described herein for determining distances and angles between speakers and other home theater components for performance of calibration operations and functions. Distances are determined through timing playback of sweep signals from a reference speaker to the speakers, and also by timing playback the sweep signals between the speakers and a capture device at a desired listening position. Angles between the speakers and other theater components are determined based on the distances, and locations or mappings of the speakers and other theater components are determined from the angles and distances. The locations/mappings are utilized to perform different calibration operations. The calibration operations improve the user audio experience for speakers that are located in non-optimal positions for speaker configurations.

Abstract: An audio/video (A/V) hub may measure the sound, corresponding to audio content, output by electronic devices. Then, the A/V hub may compare the measured sound to a desired acoustic characteristic at a first location in the environment based on the first location, a second location of the A/V hub, and an acoustic transfer function of the environment in at least a band of frequencies. For example, the comparison may involve calculating the acoustic transfer function at the first location based on the acoustic transfer function at other locations in the environment and correcting the measured sound based on the calculated the acoustic transfer function at the first location. Moreover, the A/V hub may determine equalized audio content based on the comparison and the audio content.

Abstract: An audio playback device comprises a microphone, a speaker, and a processor. The processor is arranged to output by the speaker first audio content and receive by the microphone an indication of the first audio content. A first acoustic response of a room in which the audio playback device is located is determined based on the received indication of first audio content. A mapping is applied to the first acoustic response to determine a second acoustic response. The second acoustic response is indicative of an approximated acoustic response of the room at a spatial location different from a spatial location of the microphone. The second audio content output by the speaker is adjusted based on the second response.

Abstract: An apparatus for generating one or more audio channels is provided. The apparatus comprises a metadata decoder for generating one or more reconstructed metadata signals from one or more processed metadata signals depending on a control signal, wherein each of the one or more reconstructed metadata signals indicates information associated with an audio object signal of one or more audio object signals, wherein the metadata decoder is configured to generate the one or more reconstructed metadata signals by determining a plurality of reconstructed metadata samples for each of the one or more reconstructed metadata signals. The apparatus comprises an audio channel generator for generating the one or more audio channels depending on the one or more audio object signals and depending on the one or more reconstructed metadata signals. The metadata decoder is configured to receive a plurality of processed metadata samples of each of the one or more processed metadata signals.

Abstract: An audio device includes a casing configured for operation in a first orientation and a second orientation different from the first orientation, an orientation input device disposed on the casing to detect an orientation of the casing relative to the direction of the force of gravity, and a plurality of acoustic drivers disposed on the casing and operable to form a plurality of acoustic interference arrays, each of which is associated with one of a plurality of audio channels. The acoustic drivers of the audio device operate in a first frequency range and modify at least one of the acoustic interference arrays in response to a change in the orientation detected by the orientation input device.

Abstract: Audio encoder for encoding audio input data to obtain audio output data includes an input interface for receiving a plurality of audio channels, a plurality of audio objects and metadata related to one or more of the plurality of audio objects; a mixer for mixing the plurality of objects and the plurality of channels to obtain a plurality of pre-mixed channels, each pre-mixed channel including audio data of a channel and audio data of at least one object; a core encoder for core encoding core encoder input data; and a metadata compressor for compressing the metadata related to the one or more of the plurality of audio objects, wherein the audio encoder is configured to operate in at least one mode of the group of two modes.

Abstract: A method, apparatus, and medium for rendering an audio program to a number of loudspeaker feed signals are provided. The audio program may include one or more audio objects, and metadata associated with each of the one or more audio objects. The metadata may include position information indicating a time-varying position of the audio object and a parameter indicating whether the audio object should be reproduced at the time-varying position, or at one of a plurality of fixed positions. In response to the position and the parameter, a position at which to reproduce each audio object may be determined. The determined position may be one of the plurality of fixed positions that is nearest to the time-varying position indicated by the position information. Each audio object may be reproduced at the determined position by rendering the audio object into one or more of the loudspeaker feed signals.

Abstract: A multimedia system includes at least a first pair of satellite loudspeakers for radiating left and right channel signals, the satellite loudspeakers having a first lower cutoff frequency. The loudspeaker system also includes a separate subwoofer having a first upper cutoff frequency. The first lower cutoff frequency is higher in frequency than the first upper cutoff frequency. The loudspeaker system also includes a center loudspeaker array. The center loudspeaker array radiates center channel signals, and also directionally radiates left and right channel signals in the frequency range between the first lower cutoff frequency and the first upper cutoff frequency. The multi-channel loudspeaker system may also incorporate additional satellite loudspeakers for reproduction of surround channel signals.

Abstract: Provided is an audio device, including: at least one processor; and at least one memory device that stores a plurality of instructions, which when executed by the at least one processor, cause the at least one processor to: play, with a given length of delay, a first-channel audio signal included in an audio signal; obtain one of operation information about operation of the audio device and setting information about playing of the audio signal; and transmit one of the operation information and the setting information, and at least a second-channel audio signal included in the audio signal, to another audio device.

Abstract: A method of managing communications between communication devices within a digital intercom system includes providing input and output tables for each communication device. The tables for each device are made up of cells formed from linear arrays corresponding to communication channels and the other devices. When two devices communicate on a given channel, a cell in the appropriate input and output table is identified based on the channel and the devices communicating. The cells provide volume control information which can be relied upon to determine if or to what degree audio signals should be transferred between the devices.

Abstract: A method for determining an inter-channel time difference (ITD) parameter includes determining a reference parameter according to a time-domain signal on a first sound channel and a time-domain signal on a second sound channel, where the reference parameter corresponds to a sequence of obtaining the time-domain signal on the first sound channel and the time-domain signal on the second sound channel, determining a search range according to the reference parameter and a limiting value (Tmax), where the Tmax is determined according to a sampling rate of the time-domain signal on the first sound channel, and performing search processing within the search range based on a frequency-domain signal on the first sound channel and a frequency-domain signal on the second sound channel to determine a first ITD parameter corresponding to the first sound channel and the second sound channel.

Abstract: The present invention relates to an acoustic signal supply circuit of at least one loudspeaker (HP) incorporating a filtering device of the resonance peak of said at least one loudspeaker (HP) occurring at a given frequency, characterized in that the filtering device of the resonance peak of said at least one loudspeaker (HP) is incorporated either into the first instrumentation ground circuit or in the feedback loop, this filtering device being purely electrical in the form an impedance incorporated in the first instrumentation ground circuit or in the feedback loop, the impedance parameters being predetermined as a function of the resonance peak to be filtered of said at least one loudspeaker (HP).

Abstract: A method for compressing a HOA signal being an input HOA representation with input time frames (C(k)) of HOA coefficient sequences comprises spatial HOA encoding of the input time frames and subsequent perceptual encoding and source encoding. Each input time frame is decomposed (802) into a frame of predominant sound signals (XPS(k?1)) and a frame of an ambient HOA component (CAMB(k?1)). The ambient HOA component (C˜AMB(k?1)) comprises, in a layered mode, first HOA coefficient sequences of the input HOA representation (cn(k?1)) in lower positions and second HOA coefficient sequences (cAMB,n(k?1)) in remaining higher positions. The second HOA coefficient sequences are part of an HOA representation of a residual between the input HOA representation and the HOA representation of the predominant sound signals.