2987 results about "Sound quality" patented technology

The invention discloses an automatic dubbing method and system based on generative AI. The method comprises the following steps: separating a human voice audio, a background voice audio and a silent video part from an original video, automatically identifying and translating text content in the audio, and combining a translated text with a corresponding time period to generate a translated subtitle text; segmenting the voice audio into segments, marking character attributes for each segmented segment according to voiceprint features, and evaluating the voice quality at the same time; converting the segmented subtitle text into timbre audios of a target language by using a TTS technology, and connecting and merging the audios according to the starting and ending time of the subtitles; and finally, synthesizing the generated target audio with the background sound audio and the silent video to obtain a complete target video. The above steps efficiently complete dubbing work of multiple languages, multiple roles and multiple scenes, and improve dubbing quality and dubbing efficiency.

The invention relates to the technical field of audio signal processing, and discloses a high-sampling-rate audio analysis optimization method and system, a storage medium and equipment, and the method comprises the steps: upgrading an FFTW (Fast Fourier Transform) library, and building an optimized memory management mechanism; carrying out adaptive framing processing on the input audio data, and optimizing the data access efficiency by adopting a multi-stage cache system; the calculation precision of the FFT fast Fourier transform is dynamically adjusted, and the calculation of the FFT fast Fourier transform is optimized by pre-calculating and caching an FFTW plan; performing clock synchronization optimization and jitter elimination on the COAX digital audio signal; establishing a multi-dimensional result cache matrix and an intelligent multiplexing decision tree, and reducing repeated operation through intelligent cache prediction and similarity calculation; system performance indexes are monitored in real time, adaptive parameter adjustment is realized based on an optimization objective function, audio quality and system stability are monitored in combination with subjective and objective evaluation mechanisms, repeated operation is remarkably reduced, and calculation resource allocation is optimized on the premise of ensuring sound quality.

The invention relates to the technical field of audio pushing optimization, and discloses an audio stream pushing crash prevention method and system, a storage medium and equipment, and the method comprises the steps: dynamically generating a format perception pre-detection thread based on a current audio stream format, and adjusting the priority and sampling frequency of the thread according to the audio bit depth and the sampling rate; multi-level intelligent buffer area management with sound quality priority is implemented, and dynamic adjustment of the size of a buffer area and adaptive switching of the buffer area are realized according to different audio formats and input source types; establishing a cross-brand equipment coordination framework, and setting a watchdog timer with preferential tone quality to perform timed monitoring so as to process response timeout; triggering an elegant degradation strategy of sound quality priority according to the resource availability score, and reducing the audio processing quality step by step to maintain the playing continuity; the key characteristics of each transmission protocol are identified and utilized, the transmission protocols are adaptively switched according to the network condition and the processing capability of brand equipment, and real high-quality and zero-interruption audio experience is brought to users.

The invention provides an open earphone audio optimization method based on a self-adaptive compensation algorithm, and aims to improve the audio auditory experience of a user. The method comprises the following steps: firstly, acquiring an environmental noise signal through an earphone microphone, receiving a target audio signal, analyzing auditory perception of a user in a current environment by combining an ISO226 loudness curve and the like, and determining auditory sensitivity of the user to different frequency bands (including high frequency, intermediate frequency and low frequency); and then, enhancing the high-frequency part, deepening the low-frequency part and performing smooth transition compensation on the intermediate-frequency part by using a self-adaptive compensation algorithm. And then, dynamically adjusting the low-frequency part by using a hyperbolic filter, generating an adjusted audio signal, and generating a harmonic signal based on the amplitude and frequency characteristics to further optimize the tone quality. Finally, the optimized audio signal and the generated harmonic signal are provided for a user together, and the effects of automatically optimizing audio output in different environments and enhancing tone quality performance are achieved.

The utility model relates to the technical field of audio, in particular to a loudspeaker and electronic equipment. The embodiment of the utility model provides a loudspeaker, which comprises a shell, a first vibration assembly, a second vibration assembly, a first magnetic circuit assembly, a second magnetic circuit assembly and a magnetic conductive piece, and the magnetic conductive piece is fixed on the inner side of the shell. The loudspeaker comprises two sets of vibration assemblies, the two sets of vibration assemblies are located on the two side faces of the magnetic conductive part, so that the two sets of vibration assemblies can independently produce sound, the first voice coil can drive the first vibrating diaphragm to vibrate, the second voice coil can drive the second vibrating diaphragm to vibrate, and the loudspeaker can have better low-frequency loudness through cooperation of the first vibrating diaphragm and the second vibrating diaphragm; therefore, the sound quality of the loudspeaker is improved. By arranging the magnetic conductive member, the magnetic circuit design is simplified, and the first magnetic circuit assembly and the second magnetic circuit assembly can be supported, so that the design of the support structure of the loudspeaker is simplified, the thin design of the loudspeaker is facilitated, and the adaptability of the loudspeaker to the application environment is improved.

The invention provides a sound quality-based ventilated sound insulation metamaterial design method, which comprises the following steps of: acquiring target noise for spectral analysis, and determining a main sound insulation frequency band of a metamaterial; determining the annoyance degree of a sound sample of the target noise, and calculating psychoacoustic parameters of the sound sample; analyzing the annoyance degree and the psychoacoustic parameters to obtain an annoyance degree correlation model; modeling the acoustic metamaterial structure and calculating transmission loss, and adjusting the sound insulation frequency band of the metamaterial to a specified target frequency band by adjusting the parameters of the acoustic metamaterial structure; the transmission coefficient of the acoustic metamaterial structure is calculated, psychological acoustic parameters are obtained according to the transmission coefficient, an annoyance degree correlation model is established, and the noise annoyance degree is obtained; and adjusting parameters based on the noise annoyance degree under different acoustic metamaterial structures to obtain the acoustic metamaterial structure with the optimal sound quality. According to the invention, the sound quality annoyance degree is used as the basis of the metamaterial design effect, so that the sound quality annoyance degree is greatly reduced under the limited size.

The invention discloses a multi-device high-resolution audio stream intelligent pushing method and device, equipment and a storage medium, which are applied to a multi-device audio system, and the method comprises the following steps: acquiring a current audio stream format and adjusting system working parameters; dynamically adjusting a multi-stage buffer area based on a tone quality priority strategy; operation parameters of multiple devices are synchronously monitored in real time, and data are sent in advance for the device with large delay based on the delay characteristic and the synchronization capability of each device in the system; and dynamically switching system service levels based on the quality of the current audio stream. A protection strategy can be flexibly adjusted according to different audio formats through a format perception pre-detection thread mechanism; the sound quality is kept to the greatest extent while the stability is ensured by the multi-stage buffer region protection with the sound quality priority; a multi-device coordination and timing monitoring mechanism ensures perfect cooperation with a third-party device, and the stability challenge in high-resolution audio transmission is comprehensively solved.

The invention relates to the technical field of sound quality enhancement, in particular to a sound quality improvement method and device of a Bluetooth loudspeaker box, equipment and a storage medium. The method comprises the following steps: collecting sound box environment acoustic signals, and carrying out peripheral audio time-frequency spectrum analysis and three-dimensional sound field fitting to construct a peripheral three-dimensional sound field model; based on the surrounding three-dimensional sound field model, environment noise source semantic deep analysis is carried out, and user real-time scene sound quality demand prediction is carried out, so that real-time scene sound quality demand features are generated; bluetooth sound box loudspeaker monitoring parameters are collected, and adaptive noise frequency suppression adjustment is carried out based on the surrounding three-dimensional sound field model rate, so that a loudspeaker dynamic noise suppression adjustment strategy is generated; and acquiring a Bluetooth audio output signal, and performing band frequency response characteristic optimization according to the real-time scene tone quality demand characteristics to obtain a frequency response tone quality optimized audio signal. According to changes of different environments and requirements, the output tone quality of the Bluetooth loudspeaker box is optimized in real time, and high-fidelity audio output is kept.

The invention relates to the technical field of earphone noise reduction processing, and discloses a motorcycle riding earphone noise reduction method combining different noise reduction algorithms, comprising the following steps: acquiring an original audio signal, and distributing the original audio signal to each processing module; a preprocessing signal is generated, voice activity information is determined, and environmental noise energy information is calculated; according to the environmental noise energy or the auxiliary information, adaptively adjusting a high-low noise energy threshold value; the original signals are input into the traditional and AI noise reduction module in parallel to generate two paths of noise reduction signals; based on the environment noise, the voice information and the threshold value, performing weighted fusion to generate an output signal; and carrying out equalization and compression processing on the output signal, and driving the loudspeaker to output. According to the invention, comprehensive judgment on environmental noise energy information, voice activity information and riding speed is introduced, an output signal of a traditional noise reduction module is set to be zero in a high-speed voice-free environment, and AI noise reduction is combined, so that tone quality distortion of a traditional noise reduction algorithm in a noise environment is effectively avoided.

The invention relates to the technical field of audio coding and decoding, and discloses an audio coding and decoding method and system, and the method comprises the steps: carrying out the time domain analysis and frequency domain analysis of an audio input signal of a Bluetooth earphone, and generating an audio feature matrix; performing feature extraction and fusion processing to generate a target feature vector containing frequency band energy distribution and user preference information; inputting the target feature vector into a preset neural network model for dynamic parameter calculation, and generating an adaptive coding matrix; performing layered division on the audio input signal according to the adaptive coding matrix to obtain basic tone quality layer data and enhanced tone quality layer data; and encoding the basic tone quality layer data and the enhanced tone quality layer data through a heterogeneous computing unit to generate a target encoded data stream, and the method realizes double optimization of audio processing precision and processing time.

The invention discloses an audio data compression method and device, electronic equipment and a storage medium, relates to the technical field of voice processing, can be applied to financial science and technology and medical health business scenarios, and comprises the following steps: obtaining target audio data to be compressed; and inputting the target audio data into an audio coding and decoding quantization compression model to obtain a reconstructed audio signal corresponding to the target audio data, the audio coding and decoding quantization compression model being obtained through audio adversarial training. In the model, feature extraction and compression can be performed on target audio data by using a hierarchical neural network of an encoder to obtain a low-dimensional potential feature vector; a residual vector quantization module performs discrete quantization processing on the low-dimensional potential feature vector through a multi-layer cascaded codebook to obtain a discrete quantization code; and performing audio waveform reduction processing on the discrete quantization code by using a decoder to obtain a reconstructed audio signal corresponding to the target audio data. According to the invention, audio high-fidelity compression can be realized, and audio reconstruction tone quality is improved.

The invention discloses an adjusting method of an intelligent sound box, intelligent equipment and a storage medium. Sound data of an environment where the intelligent sound box is located is collected, acoustic characteristic analysis is performed on the environment, acoustic parameters related to the intelligent sound box are automatically adjusted according to an acoustic analysis result, an acoustic model is formulated, and the intelligent sound box is adjusted according to the sound listening preference of a user. And formulating an acoustic model matched with the listening preference of the user. According to the invention, different listening environments can be automatically identified and adapted, the acoustic characteristics of the environments are analyzed in real time, and the frequency response and sound directivity of the sound box are automatically adjusted by applying a digital signal processing algorithm and a machine learning technology, so that personalized acoustic experience is provided for a user, and the overall sound quality effect is improved.

The invention provides a self-adaptive adjustment method for sound tone of a live broadcast person based on a network model, which relates to the technical field of audio signal processing and comprises the following steps of: determining an accurate starting position and an accurate ending position of packet loss through a sudden change point of a short-time energy function and an abnormal change point of a zero-crossing rate; therefore, the packet loss interval is repaired by pertinently utilizing the fundamental frequency track characteristic and the formant characteristic. The method provided by the invention can solve the problems of fundamental frequency track breakage and unnatural voice caused by the contradiction between acoustic parameter decoupling repair and real-time processing constraint in a high packet loss rate scene in a traditional voice repair method, thereby realizing high-precision and low-delay self-adaptive repair of packet loss voice, and improving the voice recovery efficiency. And the sound quality and the user experience in the live scene are obviously improved.

The invention provides outdoor portable sound equipment, and particularly relates to the technical field of sound equipment, the outdoor portable sound equipment comprises a sound equipment main body, a volume knob is arranged on one side of the top surface of the sound equipment main body, two sounding horns are symmetrically arranged on the surface of one side of the sound equipment main body, and four supporting legs are uniformly and fixedly arranged at the bottom of the sound equipment main body. Through the cooperation of components such as the gear and the gear ring, after the sound equipment main body is used, no matter how many angles a user rotates the volume knob clockwise to increase the volume of the sound equipment main body, after the sound equipment main body is used, the volume knob is driven to rotate anticlockwise through the rotation of the gear ring, so that the volume of the sound equipment main body is increased. Therefore, the sound volume is prevented from being in the maximum state when the sound main body is turned off, and when the sound main body is turned on and used next time, the sound production element in the sound main body is not suddenly in the full-load state, so that the sound production element in the sound main body is protected, and the service life of the sound main body is prolonged.

The invention relates to the field of voice recognition and analysis, in particular to a voice interaction recognition and analysis method and system based on a Bluetooth headset. The method comprises the following steps: identifying environment audio information of a current wearer based on a Bluetooth earphone; carrying out noise space distribution modeling on the environment audio information of the current wearer, and constructing an environment noise real-time distribution field; carrying out noise distribution change trend analysis on the environment noise real-time distribution field, and carrying out dynamic audio receiving adaptation calculation to obtain an optimal audio adaptation receiving parameter; carrying out real-time interactive audio perception on the wearer according to the optimal audio adaptive receiving parameters, carrying out sound quality detail enhancement processing, and constructing a sound quality detail enhanced audio signal; and performing deep audio semantic analysis and instant semantic ambiguity analysis on the sound quality detail enhanced audio signal, and marking an audio segment of semantic ambiguity. According to the invention, accurate and efficient voice interaction recognition analysis is realized.

The invention relates to a self-adaptive frequency optimization system in radio and television microwave transmission, and the system comprises an initial audio collection and processing module which collects an audio signal of a current frequency channel in real time, and generates a first data set; the conversion audio acquisition processing module synchronously acquires a target channel signal according to the frequency switching signal, and calculates the loudness offset between two channels; a feature extraction module extracts frequency domain energy distribution feature vectors of the first data set and the second data set; the model construction module constructs a dynamic gain compensation model and generates a compensation value of a target channel; the tone quality compensation module performs smoothing processing on the compensation value to generate an optimized third data set; and the audio optimization module updates model parameters, dynamically adjusts a frequency domain analysis strategy, and outputs low-delay and high-stability radio and television transmission signals. According to the invention, the problems of high switching delay and degraded tone quality caused by static configuration in a traditional scheme are solved, and the transmission efficiency and user experience in a complex electromagnetic environment are remarkably improved through modular dynamic calibration and compensation.

The invention relates to the technical field of audio processing, in particular to a self-adaptive scene sound effect adjusting method and device and a storage medium, and the method comprises the steps: collecting a mixed audio stream and an original playing audio, and extracting an acoustic fingerprint vector through a pre-trained convolutional neural network; performing acoustic feature deconstruction on the acoustic fingerprint vector to obtain a plurality of acoustic feature components, performing scene recognition judgment according to a preset scene judgment rule based on the plurality of acoustic feature components, and determining an audio playing scene; constructing a time-frequency mask according to the audio playing scene, generating a noise reduction gain matrix based on the time-frequency mask, and performing scene sound field enhancement on the audio playing scene to obtain a sound field optimization matrix; and performing noise reduction and sound effect adjustment on the original playing audio according to the noise reduction gain matrix and the sound field optimization matrix, and outputting the optimized playing audio. According to the invention, sound effect adjustment can be dynamically adapted according to different scenes, and the noise reduction accuracy and the multi-scene sound quality adaptability are improved.

One example method includes performing sound quality operations. Microphone arrays are used to cancel background noise and to enhance speech. With arrays at each environment of each user participating in a call, a first microphone array can cancel or suppress background noise and a second array can generate enhanced speech for transmission to other users. Thus, for user, the audio signal output by the user's device includes an anti-noise signal to cancel background noise present in the user's environment and enhanced speech from other users.

The invention provides a sound field optimization method and device of a multichannel loudspeaker array, equipment and a medium, and relates to the technical field of acoustics, to-be-played audio signals are respectively input to each phase compensation filter for phase compensation filtering, and a plurality of phase-compensated audio signals are obtained; the phase compensation filter is determined according to the phase compensation value of the loudspeaker; respectively inputting each phase-compensated audio signal into a corresponding loudspeaker so as to control the multi-channel loudspeaker array to produce sound; acquiring sound field response data in the target area; and when the difference between the sound field response data and the target sound field data does not meet the set sound effect requirement, dynamically adjusting the phase compensation value of the loudspeaker, and repeating the step of phase compensation filtering until the difference between the obtained sound field response data and the target sound field data meets the set sound effect requirement. According to the invention, through a dynamic adjustment mechanism and a closed-loop optimization process, precise regulation and control of the sound field of the loudspeaker array are realized, and the sound effect quality in the target area is significantly improved.

Methods and apparatus are described herein related to improving the sound quality of a bone conduction speaker. The sound quality of the bone conduction speaker is adjusted in the sound generation, sound transferring, and sound receiving of the bone conduction speaker by designing vibration generation manners and vibration transfer structures.

The invention discloses a meeting window call system based on a voice sensor, and relates to the technical field of communication. Comprising a full-space voice acquisition module, a dynamic confidence evaluation module, a phase coupling enhancement module, a multi-dimensional speech mode recognition module, a dynamic gain adjustment and tone quality enhancement module and a full-link voice integrity verification module, an array type voice sensing unit is deployed based on audio reflection characteristics, a multi-band full-space acquisition matrix is constructed, voice signals of two parties meeting are acquired, and a basic audio data stream with a high signal-to-noise ratio is output. According to the invention, through array sensing acquisition, dynamic confidence evaluation, phase coupling and harmonic enhancement, multi-dimensional speech recognition and dynamic gain adjustment, accurate recognition and fidelity enhancement of low-volume speech are realized, and through combination of full-link integrity verification and mistaken killing backtracking, weak signals are ensured not to be missed and distorted, so that the accuracy of speech recognition is improved. And the privacy, the security and the information integrity of the meeting call are improved.

The invention discloses a call recording method based on dynamic sampling rate adjustment and a Bluetooth earphone, and aims to improve the quality and stability of call recording in a wireless transmission environment. According to the method, uplink audio data and downlink audio data are collected at the same time through the Bluetooth headset, and the transmission rate of a downlink data packet is monitored in real time. When the rate is lower than the preset threshold value, the audio sampling rate is automatically reduced from 16 kHz to 8 kHz to adapt to bandwidth limitation, and recording interruption or tone quality reduction is avoided. The adjusted audio data is cached through a double-layer cache mechanism after being coded and compressed, the first cache temporarily stores the adjusted audio data, and the second cache manages data packets to be sent and retransmitted, so that the continuity and integrity of data transmission are ensured. The method is suitable for wireless audio equipment such as a Bluetooth earphone, and the reliability, adaptability and user experience of call recording are effectively improved.

The invention introduces an unlimited multi-terminal connection Bluetooth audio system based on a broadcast technology, and aims to break through the limitation of a traditional Bluetooth audio system on the equipment connection number and tone quality transmission. By utilizing the technology, the system supports seamless connection with an infinite number of playing terminals, the application range is greatly expanded, and the connection flexibility is improved. The LC3 audio decoding technology integrated in the system significantly improves the quality and efficiency of audio transmission, and ensures excellent auditory experience. Besides, the equipped digital power amplifier and the Bluetooth main control chip allow a user to flexibly process and optimize audio output of each channel through an advanced frequency division control technology, and support independent or combined sound channel output setting. The design of the system not only adapts to household and commercial environments, but also can be widely applied to scenes such as public speech and multi-room music playing, the functionality and operation convenience of the audio system are greatly improved, and the new possibility of the Bluetooth audio technology in multiple scenes is developed.

A loudspeaker assembly and a headphone are provided. The headphone includes a housing assembly, a bone-conduction core module, and an air-conduction core module, and an accommodation space. The bone-conduction core module is provided in the accommodation space and vibrates in a first vibration direction, the air-conduction core module is provided in the accommodation space, and the bone-conduction core module and the air-conduction core module are arranged along the first vibration direction and face each other. In the above manner, the present disclosure reduces the influence of the air-conduction core module on the vibration effect of the bone-conduction core module, increases the sound volume, and improves the bone-conduction effect of the loudspeaker assembly, thereby improving the sound quality effect of the loudspeaker assembly.

The invention relates to the cross technical field of artificial intelligence and multimedia communication, and discloses an intelligent optimization system and method fusing video resource scheduling and voice emotion recognition, and the system comprises a voice emotion recognition module which is used for collecting a user voice stream in real time, extracting acoustic features through a deep learning model, carrying out the emotion classification, and obtaining a voice emotion recognition result; and outputting the emotion label. According to the invention, through voice emotion recognition, the system realizes personalized resource allocation and content adjustment, so that the personalization and intelligence of user experience are greatly improved; the video resource scheduling module dynamically adjusts parameters according to a network, equipment and emotions to ensure smooth playing of a video and excellent image quality and tone quality; a multi-modal decision engine fuses a multivariate data generation strategy, resource utilization is optimized, and the emotion matching degree and the QoS score are improved; the intelligent optimization method is clear in process, video resources are accurately optimized, strategies are continuously updated through user feedback, the system adaptability is enhanced, and high-quality and personalized multimedia service experience is comprehensively provided for users.

The invention discloses a debugging method and system of an audio system, and relates to the technical field of audio debugging, the method comprises the steps of S1, environmental acoustic analysis, S2, basic parameter presetting, S3, real-time monitoring and dynamic adjustment, and S4, adaptive optimizing.The layout of an equalizer or a loudspeaker is adjusted through environmental acoustic analysis; presetting basic parameters of the sound amplifying device and the microphone; real-time multi-band sampling, delay calibration and gain regulation are realized; adaptive optimization is carried out, algorithms such as noise reduction are started, stored parameters are scene presetting, and one-key calling and self-checking updating are supported; the environment is analyzed, the audio system is accurately debugged, the system is dynamically optimized, the tone quality and the debugging efficiency are improved, and diversified scene requirements are met.

Methods and apparatus are described herein related to improving the sound quality of a bone conduction speaker. The sound quality of the bone conduction speaker is adjusted in the sound generation, sound transferring, and sound receiving of the bone conduction speaker by designing vibration generation manners and vibration transfer structures.

The double-loudspeaker OWS Bluetooth earphone comprises a main body, an ear hook or a rear cover is arranged on the main body, a Bluetooth module is arranged in the main body or the ear hook or the rear cover, and a first cavity for placing a first loudspeaker and a second cavity for placing a second loudspeaker are arranged in the main body. The main body is provided with a first sound outlet hole and a second sound outlet hole, the first sound outlet hole is communicated with the first cavity and the second cavity, the periphery of the second cavity is provided with a sound guide tube, one end of the sound guide tube is communicated with the second cavity, and the other end of the sound guide tube is communicated with the second sound outlet hole. According to the structure, the sound of the first loudspeaker and the second loudspeaker can be independently processed and transmitted, the sound interference is avoided, and the sound quality is improved; and the sound of the second loudspeaker can be more directly and effectively transmitted to the second sound outlet hole through the sound guide tube, so that the definition and directivity of the sound are improved, sound leakage is avoided, the loss of low-pitch details is reduced, and high-fidelity sound quality is provided.

The invention relates to an electrostatic vibrating diaphragm automobile audio which is mainly used for an electric vehicle and can compensate vehicle-mounted electromagnetic interference audio and solve the problem of electromagnetic interference of a motor PWM switch and rotating speed harmonics on an audio system. The electrostatic diaphragm automobile sound system comprises an electrostatic diaphragm loudspeaker and an interference feature extraction module, the rotating speed of a motor and the PWM frequency are obtained through a CAN bus, and a dynamic interference feature set is analyzed and extracted through adaptive window length FFT; the EMI quantification module is used for calculating an electromagnetic pollution index at a frequency band of 8kHz-20kHz, and triggering compensation when the electromagnetic pollution index exceeds a threshold value of-40dB; the dynamic notch compensation module is used for generating an inversion signal; according to the double-loop feedback control module, the amplitude is adjusted through PID of an inner loop, and the phase is optimized through gradient descent of an outer loop so as to minimize THD. The method comprises five steps of signal acquisition, feature extraction, interference quantification, dynamic compensation and signal injection. Interference frequency drift can be accurately tracked, a tone quality MOS value is improved after compensation, and an over-compensation risk is avoided.

The invention provides a compensation method and system for a high-precision space sound scene of an earphone, and is applied to the field of data processing. The preset microphone array is used for capturing the environment sound data in real time, dynamic EQ compensation is carried out according to the environment type, the intelligent earphone is helped to adapt to sound scenes in different environments, the tone quality stability and the environment adaptability are improved, meanwhile, the unique space sound scene of a user is constructed according to the music type and the frequency response, and the user experience is improved. In combination with dynamic EQ compensation, sound effect setting can be adjusted in real time according to the level and type of environmental noise, a user can obtain consistent music experience in different environments, immersion is enhanced, dynamic sound source positioning and sound field reconstruction are carried out based on a head related transmission function, and simulation volume is transmitted into ear canals, so that the user experience is improved. The head movement of the user is tracked in real time, audio rendering is adjusted, the omnibearing surround sound effect can be achieved, and more real and immersive music experience is provided for the user.