124results about "Loudspeaker enclosure positioning" patented technology

Two speaker units are arranged on the right side from the center of a seat; two speaker units are arranged on the left side from the center of the seat; one of the two speaker units arranged on the right side is arranged at a position closer to the right ear of a user seated on the seat as compared with the other one of the two speaker units arranged on the right side; one of the two speaker units arranged on the left side is arranged at a position closer to the left ear of the user seated on the seat as compared with the other one of the two speaker units arranged on the left side; acoustic signals obtained on the basis of a reproduction target are emitted from the two speaker units arranged on the right side to be opposite in phase to each other; the acoustic signals obtained on the basis of the reproduction target are emitted from the two speaker units arranged on the left side to be opposite in phase to each other; and the acoustic signals are emitted from two speaker units arranged on the positions closer to the ears not to be opposite in phase to each other.

A loudspeaker system includes a three-loudspeaker array and signal processing method for rendering and playing audio content in a listening environment. The three-loudspeaker array includes a left loudspeaker element, a center / height loudspeaker element, and a right loudspeaker element. A signal processor / renderer can be programmed or configured to process input audio content with a plurality of (e.g., at least five) playback channels (e.g., left, center, right, left elevation, and right elevation channels). The right loudspeaker can be driven by a right main signal and a right elevation-cancellation signal, and / or a combined signal thereof. The left loudspeaker can be driven by a left main signal and a left elevation-cancellation signal, and / or a combined signal thereof. The center / height loudspeaker can be driven by a combination of a center main signal and an elevation / height signal derived or rendered from the input audio signal.

An audio system is provided. The audio system includes a plurality of loudspeakers and a mobile device. The plurality of loudspeakers is capable of being positioned in a listening environment and being arranged to transmit an audio signal in the listening environment. Each loudspeaker is programmed to determine angle information of the audio signal as received at the loudspeaker relative to other loudspeakers of the plurality of loudspeakers and to transmit a first signal indicative of the angle information. The mobile device is programmed to receive the first signal from each of the loudspeakers and to determine a location for each loudspeaker in the listening environment based at least on the angle information.

The present disclosure discloses a vehicle headrest including a main body having a receiving chamber and a first sound hole, and a speaker module including a housing fixed to the main body and including a sound outlet, a speaker unit including a diaphragm, and an acoustic inverting tube. The housing further comprises a back sound hole communicated with the coupling back cavity; the acoustic inverting tube is received in the coupling back cavity. The sound wave output from the back sound hole has a first sound phase, the sound wave output from the sound outlet has a second sound phase, a difference between the first sound phase and the second sound phase is in a range of 2π / 3-4π / 3. The vehicle headrest in the present disclosure has better sound quality and higher acoustic privacy.

The present disclosure relates to a method and system for calibrating a loudspeaker system in an environment (1). The loudspeaker system comprises an audio device (10) with a first and second loudspeaker arranged with a predetermined horizontal separation. The method comprises the steps of emitting (S2a, S2b) a first and second calibration audio signal from the first and second loudspeaker (11) of the audio device (10), wherein the second calibration audio signal is distinguishable from the first calibration audio signal. The method further comprises receiving (S3a, S3b), with a first external microphone (20) the first and second calibration audio signal, determining (S4) the position of the first external microphone (20) relative to the audio device (10) and calibrating (S8) the loudspeaker system based on the determined position of the first external microphone (20).

A computer-implemented method of generating a loudspeaker position profile for sound management of an acoustic system comprising a plurality of loudspeakers is provided. The method comprises receiving measurement data; determining types of loudspeakers; calculating angles of arrival; determining a spatial group for each loudspeaker; grouping loudspeakers into spatial groups; assigning each loudspeaker to a type group; grouping loudspeakers into type groups; generating the loudspeaker position profile; and providing the loudspeaker position profile to a sound field management system.

Equipment including a casing wherein are located: a box; a first loudspeaker disposed inside the box, the first loudspeaker being arranged to broadcast sounds outwards from the casing; a main electronic board placed in the casing outside of the box and on which is mounted, at least one power supply module producing a first power supply voltage; a connecting device passing through the wall of the box; a secondary electronic board inside the box on which is mounted, at least one first amplifier connected to the first loudspeaker, the secondary electronic board being connected to the main electronic board via the connecting device to power the first amplifier with the first voltage.

Various implementations include audio systems and methods for hybrid rendering to enhance audio output. Some implementations include an audio system having: at least one far field speaker configured to output a first portion of an audio signal; and a pair of non-blocking near-field speakers configured to output a second portion of the audio signal in synchronization with the output of the first portion of the audio signal, where the first portion of the audio signal is different from the second portion of the audio signal.

A method is provided for advising placement for a speaker set in a room. A computerized device stores rules of speaker arrangement, acquires interior information of the room based on images of an interior of the room, and determines a seating location and a reference location in the room based on the interior information, so as to generate a speaker placement recommendation with respect to the placement of the speaker set in the room. A wearable display obtains the speaker placement recommendation from the computerized device and displays the speaker placement recommendation in augmented reality or virtual reality.

Example techniques relate to audio generation in a media playback system. Based on one or more first functions and first characteristics of an area, the system may generate first audio that includes a first audio signal and a second audio signal. The system provides the first audio signal to at least one first audio driver and the second audio signal to at least one second audio driver, thereby causing a first playback device and a second playback device to play back the first audio synchronously. The system receives second characteristics of the area and based on one or more second functions and the second characteristics, generates second audio comprising a third audio signal and a fourth audio signal. The system provides the third audio signal to the at least one first audio driver and the fourth audio signal to the at least one second audio driver.

Example techniques relate to audio generation in a media playback system. Based on one or more first functions and first characteristics of an area, the system may generate first audio that includes a first audio signal and a second audio signal. The system provides the first audio signal to at least one first audio driver and the second audio signal to at least one second audio driver, thereby causing a first playback device and a second playback device to play back the first audio synchronously. The system receives second characteristics of the area and based on one or more second functions and the second characteristics, generates second audio comprising a third audio signal and a fourth audio signal. The system provides the third audio signal to the at least one first audio driver and the fourth audio signal to the at least one second audio driver.

To provide an information processing method and an information processing device for achieving the sound image localization taking into account a shape of a physical space.SOLUTION: An information processing method includes the steps of: accepting the setting of the first spatial information and the second spatial information; receiving a first sound image localization information showing a position where the sound image is localized in a first coordinate in the first spatial information; and converting the first sound image localization information into the second sound image localization information corresponding a second coordinate in the second spatial information.SELECTED DRAWING: Figure 2

To provide a soundbar device with which it is possible to realize an optimum surround environment that corresponds to the screen width of a television device. [Solution] This soundbar device 1 comprises: a first housing 10 having a speaker at a front surface and a left side surface; and a second housing 11 having a speaker at a front surface and a right side surface, it being possible to attach / detach the second housing 11 to / from the right side of the first housing 10 using an attachment / detachment mechanism 12. One of the first housing 10 and the second housing 11 is a main housing, and the other of the two housings is a sub-housing. The main housing extracts an audio signal for a channel allocated to the speakers provided to the main housing from a multi-channel audio signal received by a television device 3, plays back the extracted audio signal, and outputs the played-back audio signal from the speakers, and also extracts an audio signal for a channel allocated to the speakers provided to the sub-housing and transmits the extracted audio signal to the sub-housing. The sub-housing receives the audio signal from the main housing, plays back the received audio signal, and outputs the played-back audio signal from the speakers provided to the sub-housing.

Various embodiments set forth techniques for positioning speakers within a venue. The techniques include generating, via a machine learning model, at least one of a two-dimensional (2D) representation or a three-dimensional (3D) representation of a venue based on one or more images of the venue. The techniques further include determining one or more parameters associated with one or more speakers to be placed within the venue based on the at least one of the 2D representation or the 3D representation.

The invention relates to a support system having a connection element (201) for screwing the left side of an acoustic transducer module and a connection element (202) for screwing the right side of the acoustic transducer module, each of which comprises two outer panels (203) between which an adjustment device, in particular for horizontal displaceability, provides a releasable connection of the modules to the elements (301)-(500), which support system is suitable for connecting a number of loudspeaker modules to a flat baffle.

This application provides a method and an apparatus for determining a virtual speaker set. The method for determining a virtual speaker set includes: determining a target virtual speaker from F preset virtual speakers based on a to-be-processed audio signal, where each of the F virtual speakers corresponds to S virtual speakers, F is a positive integer, and S is a positive integer greater than 1; and obtaining, from a preset virtual speaker distribution table, respective position information of S virtual speakers corresponding to the target virtual speaker, where the virtual speaker distribution table includes position information of K virtual speakers, the position information includes an elevation angle index and an azimuth angle index, K is a positive integer greater than 1, F≤K, and F×S≥K. This application can improve audio signal playback effect.

An apparatus and method of rendering audio. A binaural signal is split on an amplitude weighting basis into a front binaural signal and a rear binaural signal, based on perceived position information of the audio. In this manner, the front-back differentiation of the binaural signal is improved.

A loudspeaker includes a housing; one or more transducers within the housing; and a control system for controlling the one or more transducers. The control system includes a user interface supported by the housing, the user interface having one or more input elements for inputting one or more input parameters. The control system further includes a processor within the housing that is operatively coupled to the user interface and the one or more transducers. The processor is configured to perform an optimization routine in which the processor determines one or more control parameters for the one or more transducers based on the one or more input parameters.

An audio system is provided that includes a first loudspeaker and a second loudspeaker. The first loudspeaker transmits an audio signal including audio data and a signature tone into a listening environment. The second loudspeaker includes at least one controller programmed to receive the audio signal including the audio data and the signature tone and to determine a direction of arrival (DOA) of the audio signal as received at the second loudspeaker based at least on the signature tone. The at least one controller is further programmed to perform a time frequency masking operation on the received audio signal prior to determining the DOA of the audio signal.

The invention relates to the technical field of audio acquisition and noise reduction, and discloses a microphone array method for effectively reducing high-speed wind noise, which realizes the elimination of wind noise and echo and the accurate pickup of human voice in a high-speed motion scene through the specific layout design of a five-microphone array in combination with a multi-channel audio data fusion algorithm. The problems that an existing microphone array cannot adapt to a 40km / h high-speed wind noise scene and the sound receiving definition is insufficient are solved. According to the invention, five microphone arrays are deployed according to functions, the MIC1 is specially used for collecting loudspeaker echoes, the MIC2 and the MIC5 symmetrically capture multi-directional wind noise, the MIC3 and the MIC4 accurately pick up human voice in a leeward area, various signals are collected in a classified manner and do not interfere with each other, and a pure data basis is provided for subsequent noise reduction processing; particularly, the MIC3 and the MIC4 form a straight line with the mouth of a wearer and are positioned in a leeward area shielded by the lenses, so that the wind noise interference is greatly reduced, and the voice acquisition definition is improved.

To provide a loudspeaker arrangement disposed closer to listener's ears.SOLUTION: A loudspeaker arrangement includes: a seat configured to support a listener sitting in the seat so that a head 103 of the listener is in a listening position; and loudspeaker arrays 104, 105 which each are secured to the seat and disposed at least partially in a position in front of a backrest of the seat and to one side of the head when the head is in the listening position, the loudspeaker arrays each including one or more loudspeakers. The loudspeaker arrays each have a main broadcasting axes 108, 109 that is representative of a respective main broadcasting direction. The main broadcasting directions of the loudspeaker arrays point to the head.SELECTED DRAWING: Figure 1

To provide rendering of an M-channel input on S speakers (S<M).SOLUTION: An audio renderer for rendering a multi-channel audio signal having M channels to a portable device having S independent speakers, comprises: a first matrix application module for applying a primary rendering matrix to an input audio signal to provide a first pre-rendered signal suitable for playback on the multiple independent speakers; a second matrix application module for applying a secondary rendering matrix to the input audio signal to provide a second pre-rendered signal suitable for playback on the multiple independent speakers; a channel analysis module configured to calculate mixing gain according to a time-varying channel distribution; and a mixing module configured to produce a rendered output signal by mixing the first and second pre-rendered signals based on the mixing gain.SELECTED DRAWING: Figure 1

Audio playback equipment includes microphones, a loudspeaker, emitter means arranged to emit a sound detection signal, and at least one processor component arranged: to acquire detection audio signals produced by the microphones as a result of picking up the detection sound signal; from the audio detection signals, to detect a run of maskings in which at least two distinct microphones are masked in succession; to analyze a detected run of maskings so as to detect a command slide made by a user on the housing via at least two distinct microphones; and to cause at least one predetermined action to take place as a result of detecting said command slide.

1. A method for generating a frame of an encoded bitstream of an audio program including a plurality of audio signals, the frame comprising two or more independent blocks of encoded data, the method comprising: receiving, for one or more of the plurality of audio signals, information indicative of a playback device with which the one or more audio signals are associated; receiving, for the indicated playback device, information indicative of one or more additional associated playback devices; receiving one or more audio signals associated with the indicated one or more additional associated playback devices; encoding the one or more audio signals associated with the playback device; encoding the one or more audio signals associated with the indicated one or more additional associated playback devices; combining the one or more encoded audio signals associated with the playback device and signaling information indicative of the one or more additional associated playback devices into a first independent block; combining the one or more encoded audio signals associated with the one or more additional associated playback devices into one or more additional independent blocks; and combining the first independent block and the one or more additional independent blocks into the frame of the encoded bitstream.

In at least one embodiment, an audio system including a first loudspeaker and a second loudspeaker and at least one controller is provided. The first loudspeaker transmits a first audio signal including a first signature tone into a listening environment. The second loudspeaker transmits a second audio signal including a second signature tone into the listening environment and receive the first audio signal including and the first signature tone. The second loudspeaker receives the second audio signal including the second signature tone after transmitting the second signature tone into the listening environment and determines an estimated distance between the first loudspeaker and the second loudspeaker based at least on the first signature tone and the second signature tone. The second loudspeaker performs a time frequency masking operation to extract a least one of the first signature tone and the second signature tone from the noisy and reverberant mixture.

Various embodiments include an audio system and method for mixed rendering to improve audio output. Certain embodiments include at least one far-field speaker configured to output a first portion of an audio signal and a pair of non-enclosed near-field speakers configured to output a second portion of the audio signal in synchronization with the output of the first portion of the audio signal, wherein the first portion of the audio signal is different from the second portion of the audio signal.

An apparatus and method of rendering audio. A binaural signal is split on an amplitude weighting basis into a front binaural signal and a rear binaural signal, based on perceived position information of the audio. In this manner, the front-back differentiation of the binaural signal is improved.

Disclosed herein are system, apparatus, article of manufacture, method and / or computer program product embodiments, and / or combinations and sub-combinations thereof, for dynamically adjusting sound output of a multi-speaker configuration of a display device. The adjustment may be performed to optimize the sound output based on the physical position of the display device in relation to one or more microphones within a physical environment. An example embodiment operates by the speakers within the display device emitting calibration sound waves and receiving sound data associated with the calibration sound waves. The sound data includes characteristics of the calibration sound waves within the physical environment. The display device may then adjust sound output of speakers within the multi-speaker configuration based on the sound output characteristics.

A multi-driver multi-channel single enclosure Height-Channel (e.g., ATMOS™ or DTS-X®) enabled soundbar loudspeaker system 260 uses a novel signal processing system, driver mounting configuration (310L, 310R) and method to provide a high fidelity home theater listening experience, in a manner which relies on a new method for cancellation of unwanted direct (not ceiling-bounced) radiation of the Height-Channel (or virtual height envelopment) channel's sound 213DS.

Systems and methods for loudspeaker layout mapping in accordance with embodiments of the invention are illustrated. In many embodiments, loudspeakers are able to generate tones which are used by other loudspeakers to generate impulse responses. The impulse responses and the angles at which the tones were received can be used to determine the relative locations and orientations of the loudspeakers. In various embodiments, obstacles which obstruct the path of the tone can be identified and accounted for.