201 results about "Surround sound" patented technology

The invention provides a control method and device of a vehicle-mounted external loudspeaker and a vehicle-mounted external loudspeaker system. The control method comprises the following steps that in response to starting of a vehicle-mounted loudspeaker box mode, the relative position relation between a user and a vehicle is obtained at a preset time interval, and the vehicle is provided with a plurality of vehicle-mounted external loudspeakers making sounds outwards; determining a target loudspeaker and a corresponding sound channel from the plurality of vehicle-mounted external loudspeakers at least based on the relative position relationship; and controlling each target loudspeaker to play the audio data of the corresponding sound channel. The invention provides a mode of taking the vehicle as a large mobile sound box capable of providing a surround sound field, and the provided surround sound field can be adjusted according to the relative position relationship between the user and the vehicle, so that the hearing effect of the user when using the vehicle-mounted sound box is improved, the application scene of the vehicle is expanded, and the user experience is improved.

Example techniques relate to primary-ambient playback of surround audio by audio playback devices. Example playback devices described herein may include multiple speakers, such as a forward-firing audio transducer and side-firing audio transducers. Using example techniques, such playback devices may perform primary-ambient decomposition to separate surround channels into primary′ and ambient channels. The playback devices play back the ambient channel(s) via the side-firing transducers and the primary channel via the forward-firing transducer.

The invention relates to a main sound device and a sound device control method, the main sound device and at least one auxiliary sound device form a surround sound system, and the main sound device comprises a controller configured to send a volume acquisition request used for requesting volume information of the auxiliary sound device; receiving first volume information, wherein the first volume information is used for indicating an output sound pressure level supported by the target auxiliary sound equipment; determining a target sound pressure difference, wherein the target sound pressure difference is a sound pressure level corresponding to the difference between the distance from the main sound box to the user and the distance from the target auxiliary sound box to the user; based on a first audio signal corresponding to the main sound box, the current output sound pressure level of the main sound box, a second audio signal corresponding to the target auxiliary sound box, the first volume information and the target sound pressure difference, determining a third audio signal played by the main sound box based on the current output sound pressure level and a fourth audio signal played by the target auxiliary sound box based on the target output sound pressure level, the sound pressure level when the third audio signal reaches the user and the sound pressure level when the fourth audio signal reaches the user meet the consistency condition.

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).

Surround sound systems can dramatically expand the size of a user's sound field. Much surround sound content is mixed in a simplistic way where the front audio is copied to the rear, at a lower volume. It can be difficult for users to appreciate the value proposition of a surround sound system without more compelling spatially complex content. Quantifying surround sound complexity of various content items based on an audio spatial complexity scoring system can address this issue. Algorithms can be implemented to determine an audio spatial complexity score based on audio channels of a content item. Large catalog of content items can be analyzed, and audio spatial complexity scores can be associated with various content items. If a user has a surround sound system, content items with a high audio spatial complexity can be retrieved or recommended to the user to demonstrate the surround sound system's value better.

An experience sharing system for sharing an experience with others using VR and a method thereof includes a VR recorder that records a surrounding environment image, a user response image, and a surrounding sound of a first user to generate experience content, a VR player that plays and provides the experience content to a second user, and a data processing device connected to the VR recorder and the VR player over a communication network. The data processing device pre-analyzes the experience content received from the VR recorder to classify a category, adds an image matched with the classified category to the experience content, stores the experience content to which the image is added in a DB, and accesses the experience content from the DB depending on a request of the VR player and transmits the accessed experience content to the VR player.

Methods and apparatus for acquiring and encoding omnidirectional surround sound are described herein. An exemplary omnidirectional surround microphone may include a first microphone capsule oriented substantially toward a first vertex of a conceptual tetrahedron, a second microphone capsule oriented substantially toward a second vertex of the conceptual tetrahedron, a third microphone capsule oriented substantially toward a third vertex of the conceptual tetrahedron and a fourth microphone capsule oriented substantially toward a fourth vertex of the conceptual tetrahedron.

In one embodiment, an A / V interconnection architecture is provided that includes a TX A / V endpoint that supports native audio and stereo down-mixed audio on separate networks. The TX A / V endpoint outputs native audio over a video network to an RX A / V endpoint coupled to an A / V sink component that is capable of processing or outputting the native audio, and produces a stereo down-mixed version of the native audio which is output over an audio network to an audio system coupled to an audio sink component that is incapable of handling the native audio. In another embodiment, an A / V interconnection architecture is provide that enables expandable surround sound. If the number of audio channels for a type of surround sound is less than or equal to a number of local amplified output channels, surround sound is provided using only local unpowered speakers. If the number exceeds the number of the local amplified output channels, surround sound is provided using both the local unpowered speakers and one or more add on devices accessible over the audio network.

The application discloses an implementation method and system of an intelligent earphone interrupt function based on a positioning technology. The method comprises the following steps: establishing a connection between an intelligent earphone and a mobile host terminal and synchronizing a clock, and determining a relative position of multiple parties in space; calculating a distance difference by receiving sound data, and solving three-dimensional coordinates and distance information of a sound source based on a hyperbolic positioning principle; determining an interrupt priority value in combination with a semantic analysis result of the sound data, and sending an intelligent reminder to a user according to the interrupt priority value. The application realizes accurate spatial perception and intelligent importance judgment of surrounding sound events, enables the intelligent earphone to actively provide an interrupt type reminder at a critical moment, and improves the intelligence and practicability of human-computer interaction.

This product is a cartilage conduction earphone that is used by placing ear hooks over the ears and abutting the cartilage conduction speakers at both ends against the ear cartilage at the entrance to the ear canal. The product can also be used as a sound collector that amplifies surrounding sounds to make them easier to hear. By abutting the convex parts on the inner surface of the cartilage conduction speaker against the user's ear cartilage, the product prevents sound from diffusing and efficiently transmits vibrations emitted by the vibrator to the ear cartilage.

The present invention relates to a method for managing the low frequency content obtained by a loudspeaker system comprising a plurality of loudspeaker devices, such as a surround sound loudspeaker system, wherein each individual loudspeaker device has a known response as a function of frequency under anechoic conditions comprising a phase response, the method comprising the steps of providing a plurality of loudspeaker devices (1, 2, 3, 4) and for each of said plurality of loudspeaker devices (1, 2, 3, 4) providing the corresponding phase response as a function of frequency obtained under anechoic conditions and for each of the individual loudspeaker devices (1, 2, 3, 4) inserting a filter device (5, 6, 7, 8) in the signal chain to the corresponding loudspeaker device (1, 2, 3, 4), where the individual filter device (5, 6, 7, 8) is configured such that the resulting phase response of each individual loudspeaker device (1, 2, 3, 4) under anechoic conditions is substantially the same for all of the loudspeaker devices (1, 2, 3, 4). The invention further relates to a loudspeaker system implementing the method of the invention.

1. The name of the design product: sound box support (adapt to KEF, Bowers & Wilkins). 2. The use of the design product: used for floor stand of surround sound box and sound. 3. The design points of the design product: in shape. 4. The picture or photo that can best show the design points: design 1 perspective view. 5. Design 1 is specified as the basic design.

Embodiments of the present invention disclose a method, apparatus, electronic device, and storage medium for audio signal mixing and playback. In these embodiments, correlation analysis is performed on the left and right channel signals in a two-channel signal, and a specific signal processing strategy is determined based on the analysis results. By employing targeted processing methods for the two-channel signal in different states, the accuracy of signal processing is improved. Finally, when played back in a 5.1-channel surround sound system, the sound and image are naturally coherent, achieving a good sense of surround and envelopment.

The invention discloses an audio virtual surround processing method, a computer program product and electronic equipment. The audio virtual surround processing method comprises the following steps: acquiring a stereo signal containing two channels as an original audio, and separating an initial middle sound and an initial surround sound corresponding to the stereo signal; in the initial middle sound, first azimuth information corresponding to the two channels is extracted, and target middle sound corresponding to the two channels is generated based on the first azimuth information; extracting second azimuth information respectively corresponding to the two channels from the stereo signal, and generating target surround sounds respectively corresponding to the two channels based on the second azimuth information; the second azimuth information is related to the initial surround sound; and performing virtual surround processing on the original audio based on the initial middle sound, the initial surround sound, the target middle sound and the target surround sound to obtain a multi-channel audio signal serving as a target audio. The target audio can keep the sound image position of the original audio as much as possible.

The application provides a workshop machine noise adsorption shielding room, which comprises a base, side plate assemblies are arranged on the top of the base, noise reduction assemblies are arranged in the side plate assemblies, drying assemblies are fixedly installed on the top of the side plate assemblies and extend into the noise reduction assemblies, a top plate is arranged on the top of the side plate assemblies, a hanging plate is fixedly installed on the bottom of the top plate, two groups of surrounding sound insulation structures are arranged on the hanging plate, sound insulation curtains are rolled on the surrounding sound insulation structures, and a supporting assembly is slidably installed on the top of the base; the surrounding sound insulation structure comprises a folding assembly and a transmission assembly, and the drying assembly comprises a cross arm, a sleeve, a gas guide port and an electric heating rod. The shielding room realizes the double-layer sound insulation and noise reduction effect through the surrounding sound insulation structure, the sound insulation structure can be automatically carried out and rolled up, the noise reduction of the equipment machine is met, and the actual regulation and control of the personnel on the machine are not hindered.

To provide an "audio signal processor" for down-mixing a surround audio signal so as to obtain an excellent sound image feeling.SOLUTION: This device is provided with a pseudo surround audio signal generating part 1 for generating a surround audio signal SigA of 7.1. 2ch from an inputted audio stereo signal SigIN of 4ch, a signal processing part 2 for processing the surround audio signal SigA and outputting it as a signal SigB for down mix, and a down mix part 3 for synthesizing the signal SigB for down mix and outputting a stereo audio signal SigOUT of 2ch. The signal processing unit 2 combines the input audio signal of each channel of the SigA with the audio signal obtained by convolving the head-related transfer function from the speaker corresponding to the audio signal to the listener at a ratio corresponding to the position of the speaker, and outputs the combined signal as the audio signal of the SigB.SELECTED DRAWING: Figure 1

A method for generating a spatialized audio output, comprising: obtaining at least two high-order panoramic surround sound sources, where the at least two high-order panoramic surround sound sources are associated with corresponding audio source locations within an audio environment; obtaining a listener location within the audio environment, where the listener location is free to move within the audio environment; determining at least one source as an active source; determining at least one current higher order panoramic surround sound audio source from the at least one determined activity source based on the current listener location; determining at least one previous higher order panoramic surround sound source associated with the signal interpolation determination; processing all channel signals of the at least one current high-order panoramic surround sound frequency source; performing processing of at least one channel signal of other active sources of the at least one determined active source; determining a difference between at least one current high-order panoramic surround sound source and at least one previous high-order panoramic surround sound source associated with the signal interpolation determination; continuing to process all channel signals of the at least one previous active high-order panoramic surround sound source; processing at least one channel signal of at least one previous higher order panoramic surround sound source associated with the signal interpolation determination; continuing the processing of at least one channel signal of the other one of the at least one determined active source; processing all channel signals of at least one current high-order panoramic surround sound frequency source in the current high-order panoramic surround sound frequency sources; the signal interpolation is performed based on: during a first period of time after determining the difference, continuing processing of a channel signal of at least one of the at least one previous higher order panoramic surround sound source associated with the signal interpolation; after a first time period after the difference is determined; continuous processing of all channel signals of at least one previous higher-order panoramic surround sound source of at least one previous higher-order panoramic surround sound source associated with signal interpolation and at least one of at least one current higher-order panoramic surround sound source determined from at least two higher-order panoramic surround sound sources cross fade-in and fade-out between processing of all channel signals of the current high-order panoramic surround sound frequency source; and after the end of the cross fade-in and fade-out, stopping processing of all channel signals of at least one previous active high-order panoramic surround sound source; and generating a spatialized audio output based on the determined at least one signal interpolation.

The invention relates to the technical field of white noise healing, in particular to a white noise healing wall adjusting system fused with psychoacoustics, which comprises a wall device, an audio generation module, a parametric modeling module, a sensor module, a control module and an interactive feedback module. The wall body device is composed of a surface patch made of a light elastic material and an aluminum alloy frame, and a natural sound frequency curve is simulated through mechanical motion; the audio generation module extracts a natural white noise sample and accurately adjusts sound parameters in combination with a mathematical function; the parametric modeling module is used for realizing dynamic adjustment of a wall body by utilizing Grasshopper; the sensor module monitors environment and user behaviors; the interaction feedback module enhances immersion experience through multichannel surround sound and audio visualization. According to the application, by simulating natural auditory experience, dynamically changing sound elements and designing multi-sensory interaction, a user can be helped to recover attention, relieve pressure and improve sleep quality, meanwhile, personalized customization and intelligent environment adaptation are supported, and an innovative solution is provided for healing application in a building environment.

This invention relates to the field of vehicle-mounted microphone technology and provides a microphone module and a method for maintaining the microphone module. The microphone module includes a housing, a circuit board disposed within a receiving cavity, an acoustic module connected to the circuit board and communicating with a sound hole, a sound-transparent protective component covering the sound hole, a vibration assembly arranged around the sound hole on the outside of the housing, and a control module. The control module enables the vibration assembly to transmit vibration to the sound-transparent protective component during operation, causing it to vibrate to clean adhering dust, particles, or liquid residue. The sound-transparent protective component, as the only passage between the sound hole and the outside, ensures sound wave transmission while blocking foreign objects from entering the receiving cavity and confining them to its surface or pores. The vibration assembly, arranged around the sound hole, concentrates vibration on the key area at the sound hole entrance, causing inertial shedding and removal of foreign objects, thereby actively restoring transparency, reducing acoustic channel impedance, improving sound pickup attenuation, and enhancing the reliability of the vehicle-mounted waterproof microphone.

The invention relates to a control method and device for stereo surround sound of intelligent equipment and the intelligent equipment, and the method comprises the steps: determining the audio position information of a first wearing position relative to an ear position based on the first wearing position of a first intelligent wearing component and the ear position of a current user, the first intelligent wearing component is the intelligent wearing component currently worn on the human body in the at least one intelligent wearing component; based on the audio position information, performing sound channel distribution on the audio signal, and determining an audio sound channel corresponding to each first intelligent wearable component; splitting the audio signal to obtain audio sub-signals correspondingly played by each audio channel; and playing the corresponding audio sub-signal through the first intelligent wearable component to obtain the three-dimensional surround sound of the intelligent wearable device relative to the human ear. According to the control method for the stereo surround sound of the intelligent device, the technical problem that multiple intelligent wearable devices cannot give out the stereo surround sound in a linkage mode can be solved.

The invention provides acoustic equipment and ear-hanging type hearing aid equipment. The acoustic equipment is used for assisting a user in acquiring sound transmitted from a neighborhood along a target direction, and the acoustic sensor module collects external sound from the outside of the acoustic equipment and internal sound emitted from the position of the loudspeaker module to generate a sensor signal; the signal processing circuit performs target operation based on the sensor signal to generate a driving signal; the loudspeaker module receives the driving signal and outputs loudspeaker sound meeting target sensitivity to surrounding sound, the target sensitivity is within a preset auditory frequency range, the acoustic equipment has directivity in distribution of sensitivity to external sound and has the highest value in the target direction, and the loudspeaker sensitivity to internal sound is lower than 3dB. The acoustic equipment not only has a good spatial pointing effect, but also can realize effective suppression of a squeal phenomenon, and provides better acoustic experience for a user.

Processes, methods, systems, and devices are disclosed for pairing a target device with a source device and pairing the target device with a partner device. A user may choose between using the target device and the partner device without actively connecting the partner device to the source device. The target device and the partner device may each be a specialized device providing certain functions. For example, the source device may be a computing device, such as a smart phone or a tablet computer. The target device may be a sound bar dedicated for playing high definition surround sound that outperforms internal speakers of the source device. And the partner device may be a noise-canceling headset. The user may want to seamlessly switch between playing sounds on the sound bard and the headset from time to time under different circumstances, without needing to manually pair the headset to the computing device.

The utility model provides a rotary screen, a combined screen and a spinning, relates to the technical field of sports equipment, and solves the problems that an existing spinning is mainly a mechanical part, and a user is boring and lacks sports experience feeling in the using process. The rotary screen comprises a rotary screen body capable of rotating by 360 degrees, a screen connecting assembly and a screen supporting pipe, the screen connecting assembly comprises a screen connecting plate detachably connected to the back face of the rotary screen body and a transverse rotary sleeve, and a vertical rotary sleeve is integrally formed on the outer wall of the transverse rotary sleeve. The vertical rotating sleeve is rotationally connected to the screen connecting plate through a rotating shaft; and the transverse rotating sleeve is rotationally arranged on the screen supporting tube between the lower limiting convex ring and the upper limiting cap in a sleeving manner. The spinning has the advantages that when the spinning is used, a user can watch fitness video courses, meanwhile, the spinning is matched with front and back sound boxes for 3D surrounding, and the experience feeling is better; the front and rear sound boxes are detachably mounted and are convenient to mount and dismount.

The invention discloses a pipe die centrifugal chamber automatic noise reduction system which comprises a plurality of centrifugal chambers distributed adjacently, the centrifugal chambers are defined by surrounding sound insulation walls, the centrifugal chambers are arranged to be first-class centrifugal chambers and second-class centrifugal chambers which are in pairs and distributed alternately, and sound insulation cabin covers which are opened in a staggered sliding mode are arranged above the first-class centrifugal chambers and the second-class centrifugal chambers. According to the automatic noise reduction system for the pipe die centrifugal chamber, the sound insulation cabin cover of the centrifugal chamber slides to the adjacent centrifugal chamber to achieve opening of the centrifugal chamber by means of the high position and the low position of the sound insulation cabin cover of the adjacent centrifugal chamber, hoisting, taking and placing of a pipe pile die are conducted, extra space does not need to be reserved for opening of the sound insulation cabin cover of the centrifugal chamber, and operation is convenient. Therefore, the occupied area of the centrifugal chamber is reduced, and more pipe die centrifugal chambers with automatic noise reduction can be arranged in a plant space with the same size.

To adjust the sound quality depending on how the sound is transmitted.SOLUTION: An ear device 100 includes a control unit 110 that performs acoustic processing on the basis of humidity data acquired by a humidity acquisition unit 150 when performing acoustic processing on the surrounding sound acquired by a sound acquisition unit 130 and outputting it from a sound output unit 140.SELECTED DRAWING: Figure 5

This product is a charger for a hearing aid, which amplifies surrounding sounds and stores the hearing aid in a recess inside the device for charging.

The system according to the embodiment aims to ensure the safety of a user by analyzing surrounding sounds to detect danger and providing route guidance to a destination. [Solution] A system according to an embodiment includes a collection unit, an analysis unit, a warning unit, a reception unit, a calculation unit, and a guidance unit. The collection unit collects ambient sounds. The analysis unit analyzes the sounds collected by the collection unit. The warning unit detects danger and issues a warning based on the results of the analysis by the analysis unit. The reception unit accepts input of a destination. The calculation unit calculates the most suitable route based on the information accepted by the reception unit. The guidance unit provides route guidance based on the route calculated by the calculation unit.

The present invention belongs to the technical field of electroacoustics in electronics, and relates to an audio system formed by a speaker-enclosure planar array, and a preparation method for the audio system. Different from the prior art, the present invention uses a plurality of small / -micro-aperture speakers to form a special planar array audio device, such that by using the plurality of small- / micro-aperture speakers, audio signals with frequencies of 20-120 Hz or even higher, which are typically cut off in small-aperture speaker systems, are caused, by means of synchronous superposition operation, to output powerful, clear and pure bass sound waves. Moreover, an algorithm for a linear digital filter is used to perform frequency division and surround sound processing on digital audio signals of the speakers without generating a phase shift. Therefore, a series of functional defects in conventional large-aperture subwoofers are overcome, thereby enabling the planar-array audio system to achieve a surround sound effect in a three-dimensional or two-dimensional space.

Disclosed in the present invention is a near-field surround sound system, comprising: a headband main body which is provided with connecting parts at two ends; auxiliary clamping frames arranged adjacent to the upper ends of the connecting parts, wherein the auxiliary clamping frames are connected to the two side surfaces of the headband main body, and the curvature of the auxiliary clamping frames corresponds to the curvature of the two sides of the head; soft cushions arranged on the end surfaces of the headband main body and the auxiliary clamping frames in contact with the head; and near-ear loudspeaker modules, wherein two near-ear loudspeaker modules are provided and are respectively connected to the two connecting parts, each near-ear loudspeaker module comprises a plurality of loudspeaker units, and when a user wears the headband main body, the connecting parts are used for preventing the near-ear loudspeaker modules from coming into contact with the ears of the user. The near-field surround sound system comprises a top loudspeaker module, wherein the top loudspeaker module is connected to the headband main body and is approximately arranged at the top end of the headband main body, and a sound emitting surface faces the head. At least one of the two near-ear loudspeaker modules is provided with microphones.

The invention relates to a surround sound generation method and device, equipment and a medium, and the method comprises the steps: recognizing a first dry sound signal and a first wet sound signal contained in a first stereo audio signal based on a preset neural network model after the first stereo audio signal is obtained; the first dry sound signal and the first wet sound signal are extracted from the first stereo audio signal, and the first dry sound signal and the first wet sound signal are fused with the first stereo audio signal to obtain the second dry sound signal and the second wet sound signal, so that the accuracy of extracting the dry sound signal and the wet sound signal can be improved. On the basis, the second dry sound signal and the second wet sound signal are signals originally carried in the first stereo audio signal, and no extra reverberation effect is added, so that the surround sound is generated based on the second dry sound signal and the second wet sound signal, distortion of the surround sound can be reduced, and the tone quality effect is improved.