53results about How to "Good sound experience" patented technology

A circular in-ear headphone or earbud attachment can be attached to in-ear headphones or earbuds to provide traction and outward pressure to the ear of a user in order to secure an earbud in place and increase the quality of the fit. Additionally, the attachment can favorably increase the sound experience of the wearer.

A perceptually motivated spatial decomposition for two-channel stereo audio signals, capturing the information about the virtual sound stage, is proposed. The spatial decomposition allows to re-synthesize audio signals for playback over other sound systems than two-channel stereo. With the use of more front loudspeakers, the width of the virtual sound stage can be increased beyond + / −30° and the sweet spot region is extended. Optionally, lateral independent sound components can be played back separately over loudspeakers on the two sides of a listener to increase listener envelopment. It is also explained how the spatial decomposition can be used with surround sound and wavefield synthesis based audio system. According to the main embodiment of the invention applying to multiple audio signals, it is proposed to generate multiple output audio signals (y1 . . . yM) from multiple input audio signals (x1, . . . , xL), in which the number of output is equal or higher than the number of input signals, this method comprising the steps of: —by means of linear combinations of the input subbands X1(i), . . . , XL(i), computing one or more independent sound subbands representing signal components which are independent between the input subbands, —by means of linear combinations of the input subbands X1(i), . . . , XL(i), computing one or more localized direct sound subbands representing signal components which are contained in more than one of the input subbands and direction factors representing the ratios with which these signal components are contained in two or more input subbands, —generating the output subband signals, Y1(i) . . . YM(i), where each output subband signal is a linear combination of the independent sound subbands and the localized direct sound subbands—converting the output subband signals, Y1(i) . . . YM(i), to time domain audio signals, y1 . . . yM.

The invention discloses a sound effect adjustment method, which comprises an S1 step that a system sets an intelligent sound effect adjustment mode as open or closed based on user selection; an S2 step that the system retrieves the current time; an S3 step that the system retrieves attributes of the current playing content; an S4 step that the system extracts environmental parameters of the environment in which a current user stays; an S5 step that the system comprehensively matches the above three tag attributes and historical data of the user, and automatically selects and sets up a sound effect mode; and an S6 step that the system saves sound effect setting results as historical data. According to the invention, a user is allowed to select an appropriate sound effect adjustment mode according to the actual situation, the user can select conventional manual adjustment and can also select automatic adjustment of the current sound effect mode which requires no user intervention and is based on the current scene of the user, including the physical environment, the playing content, personal preferences and the like, and better sound experience effects can be ultimately provided for the user.

An exemplary embodiment of the invention can generate multiple output audio signals from multiple input audio signals, in which the number of output signals is equal to or higher than the number of input signals. The embodiment includes computing one or more independent sound subbands representing signal components which are independent between the input subbands; computing one or more localized direct sound subbands representing signal components which are contained in more than one of the input subbands and direction factors representing the ratios with which these signal components are contained in two or more input subbands; generating the output subband signals, where each output subband signal is a linear combination of the independent sound subbands and the localized direct sound subbands; and converting the output subband signals to time domain audio signals.

A method, and system, of digital room correction for a device, such as a smart speaker, including a loudspeaker. The method comprises capturing audio from an environment local to the device, for example from one or more microphones of a smart speaker. The captured audio is then processed to recognize one or more categories of sound. A digital room correction procedure may then be controlled dependent upon recognition and / or analysis of at least one of the categories of sound.

An audio enhancement technique comprises an audio source. The audio source is split into at least two branches and each of the branches is processed by a filter bank. Outputs from the filer banks are summed up to create a summed output signal. The summed output signal is splitting into two or more segments to create a split summed output signal. The split summed output signal is processed by one or more audio processing modules. Audio processed by each of the audio processing modules is summed to create a summed audio processed signal. The summed audio processed signal is provided to a balance module to select a balance mix of the audio source and the summed audio processed signal to create a balanced mixed audio signal. The balanced mixed audio signal is fed to a stereo output gain module for gain adjustment to create a gain adjusted stereo signal. The phase of the gain adjusted stereo signal is optionally changed in an amount between zero and 180 degrees to create a final output signal, which is sent to a playback device selected by user.

The invention discloses a method for testing and evaluating the sound quality of an automobile electric device. Aiming at representative noise characteristics including motor sound and friction soundof an electric device, influences of different acoustic parameters such as characteristic loudness, roughness and fluctuation on human ear hearing are analyzed, a development method and an evaluationmethod of the sound of the electric device are established, and a development process of automobile sound quality is perfected. The method comprises the following steps: subjectively evaluating the sound quality of an electric device in a whole vehicle state; carrying out objective test on the sound quality of the electric device in the whole vehicle state; selecting an objective evaluation parameter primary edition; carrying out final determination of sound quality objective evaluation parameters of the electric device; and determining subjective and objective evaluation targets of the soundquality of the electric device.

The invention discloses a sound effect adjustment method, a terminal and a computer readable storage medium. The method comprises the following steps: obtaining a user relative position parameter between an audio output device and a target user, determining a target sound effect parameter of the audio output device in a subsequent playing process based on the user relative position parameter and apre-stored user position-sound effect mapping relationship, and adjusting the output sound effect of the audio output device according to the target sound effect parameter. The problem that the soundeffect parameter output by the audio output device is invariable after the user sets the sound effect parameter, resulting in poor user experience in the prior art is solved. By adoption of the terminal and the computer readable storage medium provided by the invention, by implementation of the scheme, when the relative position of the user and the audio output device changes, the sound effect parameter output by the audio output device is automatically adjusted, so that the user obtains better experience, and the user satisfaction is improved.

The invention discloses a multiple-360-degree-sound-production resonance portable sound system which comprises a shell body, a resonance sound-production device and a control circuit board. The resonance sound-production device and the control circuit board are arranged at the bottom of the shell body. The resonance sound-production device and the control circuit board are electrically connected. The multiple-360-degree-sound-production resonance portable sound system is characterized in that at least one loudspeaker is further included, the loudspeakers are arranged in the shell body and are electrically connected with the control circuit board, a cone-shaped public-address piece which is used for diffusing sound waves emitted out by each loudspeaker around is arranged in the sound-emitting direction of the corresponding loudspeaker, and the cone-shaped public-address pieces are connected with the shell body. The multiple-360-degree-sound-production resonance portable sound system has the advantages that the traditional loudspeakers and the resonance sound-production device are combined, the cone-shaped public-address pieces which are used for diffusing sound waves emitted out by the loudspeakers around are arranged in the sound-emitting direction of the loudspeakers, 360-degree sound production can be achieved, sound effect is greatly improved, perfect sound quality experience is brought to a user, and in addition, the multiple-360-degree-sound-production resonance portable sound system is simple in structure, small in size, convenient to carry, high in practicability and convenient to popularize and use.

The invention provides a sound signal processing method, and a terminal. The processing method comprises the steps of acquiring a three-dimensional head model of a terminal user; calculating a first HRTF referring to the fact that a sound wave emitted by a sound source is transmitted to a left ear of the three-dimensional head model and a second HRTF referring to the fact that the sound wave emitted by the sound source is transmitted to a right ear of the three-dimensional head model; and processing a sound signal emitted by the sound source according to the first HRTF and the second HRTF. Inthe prior art, the virtual auditory technology has the disadvantage of virtual audio space synthesis distortion due to depersonalized HRTF. The method solves the problem in the prior art, and improvesthe sound experience of the user.

The embodiment of the invention provides a sound effect processing method, a device and multimedia equipment; the method specific realizing principle comprises the following steps: firstly obtaining space position parameters, and then processing the space position parameter data, thus obtaining sound effect compensation parameters corresponding to the space position parameters; processing the sound effect outputted by the loudspeaker of the multimedia equipment according to the sound effect compensation parameters. The method and device can greatly reduce volume attenuation or timbre damages caused by the space environment differences of the multimedia equipment.

The invention discloses a real-time audio interaction method of multi-Bluetooth audio equipment. The Bluetooth audio equipment comprises at least one piece of first Bluetooth audio equipment and at least one piece of second Bluetooth audio equipment. The real-time audio interaction method comprises the following steps of respectively arranging LE Audio chips in the first Bluetooth audio equipmentand the second Bluetooth audio equipment, establishing Bluetooth connection between the first Bluetooth audio equipment and the second Bluetooth audio equipment and conducting Bluetooth data packet transmission through the LE Audio chips. The LE Audio chip adopts an LC3 codec, has a very high compression ratio, and can greatly reduce the transmission time of audio data, so that the total transmission delay of the audio data is reduced to meet the requirement of on-site real-time interaction; moreover, Bluetooth broadcasting and multi-stream transmission functions are also achieved, and a purpose that one piece of first Bluetooth audio equipment broadcasts one or more audio streams to a plurality of pieces of second Bluetooth audio equipment can be realized, so as to meet the demands of on-site multi-person real-time entertainment audios.

An audio enhancement technique comprises an audio source. The audio source is split into at least two branches and each of the branches is processed by a filter bank. Outputs from the filer banks are summed up to create a summed output signal. The summed output signal is splitting into two or more segments to create a split summed output signal. The split summed output signal is processed by one or more audio processing modules. Audio processed by each of the audio processing modules is summed to create a summed audio processed signal. The summed audio processed signal is provided to a balance module to select a balance mix of the audio source and the summed audio processed signal to create a balanced mixed audio signal. The balanced mixed audio signal is fed to a stereo output gain module for gain adjustment to create a gain adjusted stereo signal. The phase of the gain adjusted stereo signal is optionally changed in an amount between zero and 180 degrees to create a final output signal, which is sent to a playback device selected by user.

An assembly and method for automatic real time enhancement of audio signal on the basis of listening environment and ambience noise, wherein said assembly comprises one or more audio sensing device such as microphone, a computer implemented system further comprising a non-transitory computer-readable storage medium storing instructions and components to analyze and process audio signals; and a method to capture the sound in listening environment using the audio sensors whereby plurality of audio sensors are located at plurality of locations to capture the proximal sound signals and transferring the captured signal to a system, whereby, the instructions stored in memory on execution by the computer analyze and process the sound signals to provide enhanced sound signal which are transferred to plurality of speakers.

Headphones include a left loudspeaker element; a right loudspeaker element; and a holder for holding the left loudspeaker element and the right loudspeaker element, such that the loudspeaker elements can be attached to the ears, wherein the left loudspeaker element or the right loudspeaker element includes: a first sound converter; a second sound converter, wherein the first sound converter is implemented such that the first sound converter provides directed emission in the direction of an ear in the operating position of the headphones, and the second sound converter is implemented such that the second sound converter provides no or less directed emission than the first sound converter in the direction of the ear in the operating position of the headphones.

A directivity adjustment device that includes a test sound signal generator, a surface detector, and a content processor is described. The test sound signal generator generates test sound signals to scan a listening area (e.g., a room) in a nodding and rotating fashion. The test sounds are emitted by one or more loudspeaker arrays and are sensed by a microphone as they permeate throughout the listening area. The sensed sound is analyzed by the surface detector, to identify reflected sounds in relation to direct sound. Based on the amplitude variance between the peaks that represent the direct and reflected sounds reaching a predefined threshold, an adjusted directivity ratio for the content processor is selected or calculated, which results in a reduction of the level of reflected sounds and corresponding comb filtering effects. Other embodiments are also described.

The invention discloses a wireless multi-channel sound system and a sound channel automatic calibration method thereof. The wireless multi-channel sound system comprises a middle sound box and a plurality of wireless ultrasonic sound boxes with electric pan-tilt bases. A DSP control chip and a microphone array are arranged in the middle sound box; when automatic calibration is performed on channel, a distance measurement command is sent to each wireless sound box in sequence through the DSP control chip; the microphone array feeds back collected ranging commands to the DSP control chip to be processed, the direction of sound transmission is sequentially calculated, the DSP control chip matches sound channels for the wireless ultrasonic sound boxes according to data in all directions, and finally the DSP control chip controls the wireless ultrasonic sound boxes to adjust the deflection angles of the wireless ultrasonic sound boxes. By adopting the scheme, a user does not need to distinguish the originally defined sound channel of each wireless sound box to place, and after the sound system is started, the sound system can automatically identify the position of each wireless sound box and match the corresponding sound channel for the wireless sound box, so that the user can obtain correct surround sound information.

The invention provides a sound effect matching method applied to an electric device. The electric device includes an earphone jack. The method includes a recognition step of recognizing the specification of the current earphone after detecting that an earphone is plugged into the earphone jack; an inquiry step of acquiring the optimal audio parameters corresponding to the current earphone through looking up an earphone and sound parameter table according to the specification of the current earphone; and a processing step of updating the audio parameter in the electric device to be the optimal audio parameter corresponding to the current earphone. By adopting the electric device and the method provided by the invention, the specification of the earphone plugged into the earphone jack currently can be acquired dynamically, so that audio parameters are updated and a user can have better sound effect experience.

The invention belongs to the technical field of automobile headrests, and provides a headrest with turnover sound boxes on two sides.The headrest comprises a headrest framework, wherein external soundboxes are arranged on the two sides of the headrest framework; and rotary arms rotationally connected to the two sides of the headrest framework, wherein microphones are arranged at one ends of the inner sides of the rotary arms, and Bluetooth sound boxes are arranged at the positions, corresponding to the microphones, in the rotary arms. Compared with the prior art, the headrest has the advantages that the position of the rotary arm can be adjusted according to use requirements, and when a call needs to be answered or a user needs to listen to music in the driving process, the sound box in an automobile can be automatically adjusted to the bluetooth sound box in the headrest; meanwhile, a driver or other passengers can turn the rotary arms on the two sides of the headrest to a certain angle in the head direction according to the requirements of the driver or other passengers, so that the sound box and the microphone are closer to ears to achieve the purpose of private conversation ormusic listening to avoid disturbing the other passengers; and the headrest is internally provided with external audio equipment, so that the 3D sound stereo surround field can be formed by the external audio equipment and other vehicle-mounted sound boxes, tone quality experience is optimized, and entertainment and leisure functions are enhanced.

The embodiment of the invention provides an earphone and a tuning method. The earphone comprises a sounding unit, a detection assembly and a processing unit. The detection assembly is arranged at an in-ear part of the earphone, the detection assembly is used for detecting capacitance information between the in-ear part and an ear of a wearer, the sound production unit and the detection assembly are electrically connected with the processing unit, and the processing unit adjusts sound production parameters of the sound production unit according to the capacitance information. According to the embodiment of the invention, the capacitance information between the in-ear part and the ear of the wearer is detected through the detection assembly, the tightness degree of the user wearing the earphone is obtained according to the capacitance information, and the capacitance information under different tightness degrees is transmitted to the processing unit. The processing unit adjusts the soundproduction parameters of the sound production unit according to the capacitance information, so that the sound effect experience when the user uses the earphone is improved.

The invention relates to a multi-channel vehicle-mounted sound system, which comprises an audio processing device, a power amplifier and a plurality of loudspeakers, and is characterized in that the plurality of loudspeakers comprise a central loudspeaker arranged in the center of the front of a vehicle cabin; the left loudspeaker is arranged at the left front part of the vehicle cabin; the right loudspeaker is arranged at the right front part of the vehicle cabin; the left surround loudspeaker is arranged on the left side of the vehicle cabin; the right surround loudspeaker is arranged on the right side of the vehicle cabin; the left front 3D loudspeaker and the right front 3D loudspeaker are arranged on the two sides of the car roof and correspond to the positions of the front row of seats, and for any loudspeaker in the multiple loudspeakers, the loudspeakers located around the loudspeaker are used for conducting azimuth compensation on the loudspeaker. According to the invention, the sound field can be more immersive, better sound experience can be realized, and the tone can be better restored.

The invention provides an earphone and a mobile terminal, and the earphone comprises a housing and a loudspeaker assembly, the interior of the housing comprises a front cavity and a rear cavity which are separated by the loudspeaker assembly, the front cavity is located at one side of the sound production direction of the loudspeaker assembly, and the rear cavity is located at one side opposite to the sound production direction of the loudspeaker assembly; the shell is provided with one or more sound guide channels, each sound guide channel is arranged on the side wall of the shell, and each sound guide channel is used for communicating air in an ear canal of a user with air outside the ear canal under the condition that the air in the front cavity and the air in the rear cavity are not communicated when the earphone is normally worn. When the scheme is adopted, the environment in the ear canal is communicated with the environment outside the ear canal through the sound guide channel arranged in the shell, so that the sound pressure in the ear canal can be exposed or discharged to the surrounding environment outside the earphone. The pressure in the ear canal is reduced; and the sound experience of the user is improved.

The invention belongs to the technical field of automatic driving, and particularly relates to a directional sound production system and method, the system comprises a passenger head position recognition device and a directional sound production device, the passenger head position recognition device is used for positioning the passenger head in real time and sending position information; and the directional sound production device is used for producing sound to the head of the passenger according to the position information. The head position of the passenger in the vehicle is collected in real time and modeled, learned and analyzed, the head position of the passenger in the vehicle is positioned through the ASM algorithm, positioning data is sent to the directional sound production device, sound is always directionally sent according to the head position of the passenger in the vehicle, and the best sound sense experience is provided all the time.

An audio system for an ear mountable playback device comprises a speaker, an error microphone configured to predominantly sense sound being output from the speaker and a further microphone configured to predominantly sense ambient sound. The system further comprises a first noise filter coupling the further microphone to the speaker, a second noise filter coupling the error microphone to the speaker and an adaptation engine. The adaptation engine is configured to adapt a response of the first noise filter depending on error signals from at least the error microphone, estimate a leakage condition from the response of the first noise filter, and adapt a response of the second noise filter depending on the estimated leakage condition.

The invention relates to the technical field of communication equipment, and particularly discloses a sound conduction device. The sound conduction device comprises a mounting part, a metal elastic piece, a ceramic piece and a counterweight block, the mounting part comprises a mounting face and a transmission face which are oppositely arranged, the metal elastic piece is mounted on the mounting face, the ceramic piece and the counterweight block are both installed on the metal elastic piece, the counterweight block is located at one side of the ceramic piece, and the transmission face is used for being fitted with the arm of a user to transmit vibration generated by the ceramic piece to the arm of the user. According to the sound conduction device, a mode that the ceramic piece and the counterweight block are arranged on the metal elastic piece is adopted, the counterweight function of the counterweight block can be utilized, the sound spectral response range is increased, the sound conduction efficiency is improved, and therefore the sound effect is improved, and it is ensured that the user has good sound experience. The invention further discloses wearable equipment with the sound conduction device.

The embodiment of the invention discloses a loudspeaker control method and device, electronic equipment and a storage medium, and relates to the technical field of intelligent control. The method comprises the following steps: acquiring pressure data of a pressure sensor, and determining personnel distribution in a vehicle according to the pressure data; determining a first listening position of audio played by a listening loudspeaker according to the personnel distribution; determining the sitting height of the person in the vehicle according to distance data detected by a distance measuring sensor, and determining a second listening position of the audio played by the listening loudspeaker according to the sitting height of the person in the vehicle; determining an optimal listening position in the vehicle according to the first listening position and the second listening position; and determining the sound channel and the volume of each loudspeaker in the plurality of loudspeakers according to the optimal listening position, and controlling each loudspeaker in the plurality of loudspeakers to play based on the corresponding sound channel and volume. According to the technical scheme, the purpose of intelligently controlling the loudspeaker can be achieved, and better sound experience can be provided for passengers.