Sound box control method, system and equipment based on built-in sound source and storage medium

Abstract

The invention relates to the technical field of audio processing and loudspeaker box control technologies, in particular to a loudspeaker box control method, system and device based on a built-in sound source and a storage medium, and the method comprises the steps: obtaining a selection instruction of a user, and calling a corresponding sound source file based on the selection instruction; receiving the called sound source file by using a built-in preset audio processing algorithm, and adjusting audio parameters of the sound source file; performing deep optimization combination on the processed audio parameters corresponding to the plurality of audio source files subjected to audio parameter adjustment by using an audio mixing algorithm so as to obtain a playing audio signal; and transmitting the playing audio signal to an integrated amplification circuit for power amplification, and enhancing the signal strength to drive a loudspeaker to work. The application has the effect of improving the audio quality and the user experience.

Description

Technical Field

[0001] The present application relates to the technical field of audio processing and speaker control technology, and in particular to a speaker control method, system, device and storage medium based on a built-in sound source. Background Art

[0002] At present, most professional-level speakers on the market have the function of adjusting the sound quality, high, medium and low frequencies or volume, which greatly improves the user experience. However, as consumers' demand for personalized sound quality and timbre continues to increase, traditional speakers are no longer able to meet the diverse needs. Especially in the fields of music performance, teaching and entertainment, users hope to achieve flexible changes in timbre through simple operations, rather than just adjusting the volume and sound quality.

[0003] In order to meet the above needs, the existing technical solutions usually connect a dedicated controller or musical instrument interface to the speaker, for example, using musical instruments such as electric blowpipes, electronic keyboards, and electronic drums as intermediate media to adjust the sound characteristics of the speaker. In this way, users can achieve richer timbre changes, thereby improving the musical expression. However, although this method of relying on external devices can enrich the expressiveness of the sound to a certain extent, it also brings many inconveniences.

[0004] The method of relying on external devices not only increases the complexity and cost of the system, but also greatly limits the portability and ease of use of the speaker. Especially for beginners, this complicated setting process is often daunting. Therefore, how to realize the selection and switching of multiple preset tones inside the speaker without relying on external devices has become a technical problem that needs to be solved urgently. Summary of the invention

[0005] The purpose of this application is to provide a speaker control method, system, device and storage medium based on a built-in sound source.

[0006] The first object of this application is to provide a speaker control method based on a built-in sound source, including: Obtaining a user's selection instruction, and calling a corresponding audio source file based on the selection instruction; Using a built-in preset audio processing algorithm, receiving the retrieved audio source file, and adjusting the audio parameters of the audio source file; Using a mixing algorithm, the processed audio parameters corresponding to the multiple sound source files after the audio parameters have been adjusted are deeply optimized and combined to obtain a playback audio signal; The played audio signal is transmitted to the integrated amplifier circuit for power amplification to enhance the signal strength so as to drive the speaker to work.

[0007] By adopting the above technical solution, the built-in audio source files are directly called based on the user's selection instructions, getting rid of the dependence on external devices, and greatly improving the convenience and autonomy of the speaker. The built-in preset audio processing algorithm can accurately adjust the audio parameters to meet the user's diverse sound quality requirements. The use of the mixing algorithm realizes the deep optimization combination of multiple sound sources, enriches the expressiveness and layering of the audio, and creates a more outstanding sound effect. Finally, the speaker is driven by power amplification to ensure the high quality of the audio output, providing users with an efficient, flexible, personalized and high-quality audio experience as a whole.

[0008] In a possible implementation of the present application, the step of using a mixing algorithm to deeply optimize and combine processed audio parameters corresponding to multiple sound source files after the audio parameters are adjusted to obtain a playing audio signal includes: Analyzing and comparing style features of the multiple audio source files selected by the user; Determining whether there are significant differences in the styles of the multiple audio source files selected by the user; If there are significant differences in the style features, obtaining the user's preference setting information for the mixing effect and the current environment information of the speaker; By using audio analysis technology, multi-dimensional audio feature information of each of the audio source files is extracted, including timbre features, rhythm features, pitch features, and audio energy distribution features; According to the preference setting information, the environmental information and the multi-dimensional audio feature information, a mixing strategy is formulated using an intelligent mixing algorithm to optimize the distribution of the timbre features of the sound source files on the frequency spectrum so that different timbres complement each other in the frequency band and avoid interference; rhythm synchronization is performed based on rhythm features, and sound sources of different rhythm types are integrated on the time axis by stretching, compressing or filling rhythm segments; a suitable harmony framework is constructed based on pitch features and melody features, and the pitch and melody direction of the sound source corresponding to each sound source file are adjusted to achieve harmony; the dynamic range of the sound source corresponding to each sound source file is compressed or expanded according to the environmental information and user preferences to balance the relationship between the volume and dynamic performance of each sound source; During the mixing process, the machine learning model and preset music aesthetic rules are continuously used to evaluate and optimize the mixing effect, and the mixing parameters are constantly adjusted to adapt to different music styles and user needs; Output the optimized mixed audio signal.

[0009] By adopting the above technical solutions, the style characteristics and difference judgments of the audio source files selected by the user are analyzed, and the mixing strategy is made more targeted and adaptable by combining user preferences and environmental information; multi-dimensional audio features are extracted with the help of audio analysis technology, and intelligent mixing algorithms are used to optimize the processing from multiple aspects such as timbre, rhythm, pitch and dynamic range, which greatly improves the professionalism and coordination of the mixing, effectively avoids audio conflicts, achieves the harmonious integration of various sound sources, and significantly enhances the richness and three-dimensionality of the audio effects. In the mixing process, machine learning models and music aesthetic rules are introduced for continuous evaluation and optimization, and parameters can be adjusted in real time according to different music styles and user needs to ensure that the mixing effect always meets user expectations, providing users with a more high-quality, personalized and creative audio experience, and effectively promoting the development of speaker mixing technology towards intelligence and precision.

[0010] In a possible implementation of the present application, during the mixing process, the steps of continuously using a machine learning model and preset music aesthetic rules to evaluate and optimize the mixing effect, and continuously adjusting the mixing parameters to adapt to different music styles and user needs include: Use real-time audio analysis technology to continuously monitor the current mixing effect to obtain real-time mixing effect evaluation results; According to the real-time mixing effect evaluation result, an intelligent recommendation algorithm is used to screen out recommended sound source files that improve the mixing effect in the sound source library; The recommended audio source files are added to the current mix, and the mixing algorithm is re-applied to deeply optimize and combine all the audio source files.

[0011] By adopting the above technical solutions and using real-time audio analysis technology to continuously monitor the mixing effect, the current mixing status can be obtained in a timely and accurate manner, providing an accurate basis for subsequent optimization. Based on this evaluation result, an intelligent recommendation algorithm is used to filter the recommended audio source files, which greatly enriches the selection range of mixing materials and introduces more elements that may improve the mixing effect. Adding and remixing the recommended audio source files allows the mixing results to be continuously iterated and optimized to better adapt to changes in different music styles and user needs, effectively improving the innovation and flexibility of the mixing. This mechanism of dynamic evaluation, intelligent recommendation and remixing creates a high-quality audio experience for users that is more in line with personal preferences and meets professional music standards.

[0012] In a possible implementation manner of the present application, the method further includes: Creating multiple intelligent premixed sound source templates covering different music styles, emotional atmospheres and application scenarios, each of the intelligent premixed sound source templates is composed of multiple pre-processed sound source files, and each of the intelligent premixed sound source templates is attached with corresponding audio feature tags and metadata descriptions; Acquire user demand information, and based on the acquired user demand information, perform a quick matching search in a premixed sound source template library to find one or more templates that best match the user demand as a mixing basis; Determine a matching premixed sound source template, load the sound source file in the determined premixed sound source template into a mixing processing unit, and perform fast mixing according to preset parameters of the premixed sound source template to generate an initial mixed audio signal; During playback, the system continuously monitors users’ real-time feedback and environmental change information, and uses adaptive algorithms to dynamically adjust mixing parameters.

[0013] By adopting the above technical solutions, a variety of intelligent pre-mixed sound source templates are created and detailed information is attached, providing users with rich and targeted mixing starting point options, saving a lot of creative time and energy. The fast matching retrieval mechanism based on user needs can accurately locate templates that meet user expectations, improving the efficiency and accuracy of mixing creation. Rapid mixing based on template preset parameters generates initial signals, laying a good foundation for subsequent adjustments. During playback, feedback and environmental changes are continuously monitored and parameters are dynamically adjusted to ensure that the mix always adapts to different scenarios and user preferences, providing users with a more personalized, high-quality and stable audio experience, allowing the speaker to maintain excellent mixing effects in various complex usage environments, enhancing the practicality of the product and user satisfaction, and strongly promoting the development of speaker mixing technology in intelligence and adaptability.

[0014] In a possible implementation manner of the present application, the steps before obtaining the playing audio signal include: Pre-design and generate multiple excellent mixing templates covering different music styles, emotional atmospheres and application scenarios; When the user selects multiple audio source files for mixing creation, the user's operation process is monitored in real time, and when the user completes the preliminary mixing settings or adjusts the mixing parameters in real time, the audio analysis technology is used to extract the audio features of the mixing project created by the user; Compare and analyze the audio features of the mixing project described by the user with the audio features of each excellent mixing template in the template library one by one to obtain a contrast ratio; Based on the priority order of the contrast, an excellent mixing template with a high contrast has a high corresponding priority, and a recommended excellent mixing template list and a corresponding comparison report are displayed to the user.

[0015] By adopting the above technical solutions, a rich variety of excellent mixing templates are pre-designed to provide users with a wide range of reference examples and sources of creative inspiration. Real-time monitoring of user operations and extraction of audio features of mixing projects can achieve accurate grasp of the user's creative process. By comparing and analyzing with the template library to obtain contrast and priority, users can effectively guide them to find templates that are similar to their own creative style or needs but more distinctive. Displaying recommendation lists and comparison reports can help users deeply understand the characteristics and differences of different templates, making it easier for users to quickly select suitable templates for optimization and improvement, improve mixing quality and efficiency, and promote users to continuously explore and innovate, greatly enhancing the user's autonomy, convenience and professionalism in the mixing creation process.

[0016] In a possible implementation manner of the present application, the steps after transmitting the played audio signal to the integrated amplifier circuit for power amplification and enhancing the signal strength to drive the speaker to work include: Real-time monitoring of the actual sound quality and effect of the speaker output; Using audio analysis technology to perform multi-dimensional testing on the actual played sound, including but not limited to volume stability, timbre accuracy, frequency response flatness, distortion, and spatial sound positioning accuracy; Obtain current environment information and user feedback information; Based on the current environment information and the user feedback information, an adaptive control algorithm is used to adjust the played audio signal in real time.

[0017] By adopting the above technical solutions, real-time monitoring of the sound quality and effect of the speaker output and multi-dimensional audio analysis can accurately grasp the actual audio playback situation and promptly discover possible problems. Combining the current environmental information with user feedback information, the adaptive control algorithm is used to adjust the playback audio signal in real time, so that the speaker can automatically optimize the audio output according to different environments (such as automatically improving the volume clarity in noisy environments and optimizing the sound quality in quiet environments) and user personalized needs (such as users' real-time preferences for timbre and volume), ensuring that users always get the best listening experience. This not only improves the intelligence and adaptability of the speaker, but also effectively improves the stability and quality of audio playback.

[0018] In a possible implementation manner of the present application, the method further includes: Obtaining registration information of a smart home device, assigning a unique device identifier to the corresponding smart home device, and obtaining basic function information of the smart home device; Classify and mark the sound source files in the built-in sound source library, and establish a mapping relationship between each category and the smart home device and operation action; When the speaker is playing, feature recognition is performed on the sound source file corresponding to the played audio signal in real time to obtain a feature recognition result; Automatically generate corresponding smart home control instructions according to the feature recognition results and the mapping relationship; The generated smart home control instructions are sent to the corresponding smart home devices via a wireless communication protocol.

[0019] By adopting the above technical solutions, the smart home device information is obtained and identifiers are assigned, so that the speaker can accurately identify and manage smart home devices. The audio source files are classified and marked and a mapping relationship is established, laying the foundation for the intelligent linkage of audio and smart device operations. Real-time feature recognition and automatic generation of control instructions during playback achieve innovative functions of real-time control of smart home devices based on audio content, such as adjusting the lighting atmosphere according to the rhythm of the music, starting the corresponding electrical appliances according to the audio scene, etc., creating a more intelligent and immersive home environment experience for users. The wireless communication protocol ensures stable transmission of instructions, enhances the reliability and timely response of the system, effectively improves the convenience, fun and automation of home life, promotes the development of smart home control technology in a more intelligent, efficient and personalized direction, and further expands the application scenarios and practical value of speakers in the smart home ecosystem.

[0020] The second purpose of this application is to provide a speaker control system based on a built-in sound source, the system comprising: Audio source file retrieval module: obtains the user's selection instruction and retrieves the corresponding audio source file based on the selection instruction; Audio parameter adjustment module: using a built-in preset audio processing algorithm, receiving the retrieved audio source file, and adjusting the audio parameters of the audio source file; Audio signal generation module: uses a mixing algorithm to deeply optimize and combine the processed audio parameters corresponding to multiple audio source files after the audio parameters have been adjusted to obtain the audio signal for playback; Audio signal transmission module: transmits the audio signal to the integrated amplifier circuit for power amplification, thereby increasing the signal strength to drive the speaker to work.

[0021] By adopting the above technical solution, the built-in audio source files are directly called based on the user's selection instructions, getting rid of the dependence on external devices, and greatly improving the convenience and autonomy of the speaker. The built-in preset audio processing algorithm can accurately adjust the audio parameters to meet the user's diverse sound quality requirements. The use of the mixing algorithm realizes the deep optimization combination of multiple sound sources, enriches the expressiveness and layering of the audio, and creates a more outstanding sound effect. Finally, the speaker is driven by power amplification to ensure the high quality of the audio output, providing users with an efficient, flexible, personalized and high-quality audio experience as a whole.

[0022] The third object of this application is to provide a speaker control device based on a built-in sound source, the device comprising: A memory and a processor, wherein the memory stores a computer program that can be loaded by the processor and execute the above-mentioned speaker control method based on the built-in sound source.

[0023] The fourth objective of this application is to provide a storage medium.

[0024] The fourth application objective of this application is achieved through the following technical solutions: A storage medium stores a computer program that can be loaded by a processor and execute the above-mentioned speaker control method based on a built-in sound source.

[0025] In summary, the present application includes at least one of the following beneficial technical effects: 1. Get rid of the dependence on external devices. Through the built-in sound source and related algorithms, users can easily select, process and mix a variety of sound source files to achieve rich and diverse timbre changes to meet the needs of different music styles and personal preferences.

[0026] 2. The mixing algorithm deeply optimizes and combines audio parameters from multiple dimensions, including timbre, rhythm, pitch, and dynamic range, thereby improving the expressiveness, layering, and coordination of the audio, avoiding audio conflicts, and creating more outstanding sound effects.

[0027] 3. Introducing machine learning models and music aesthetic rules into the mixing process, combined with real-time audio analysis technology, can continuously evaluate and optimize the mixing effect, dynamically adjust mixing parameters according to different music styles and user needs, ensure that the audio always meets user expectations, and provide a more high-quality, personalized and creative audio experience. BRIEF DESCRIPTION OF THE DRAWINGS

[0028] Figure 1 It is a flowchart of a method for controlling a speaker based on a built-in sound source provided in an embodiment of the present application; Figure 2 It is a virtual structural diagram of a speaker control system based on a built-in sound source provided in an embodiment of the present application. DETAILED DESCRIPTION

[0029] In order to make the purpose, technical solution and advantages of the embodiments of the present application clearer, the technical solution in the embodiments of the present application will be clearly and completely described below in conjunction with the drawings in the embodiments of the present application. Obviously, the described embodiments are part of the embodiments of the present application, not all of the embodiments. Based on the embodiments in the present application, all other embodiments obtained by ordinary technicians in this field without creative work are within the scope of protection of this application.

[0030] In addition, the term "and / or" in this article is only a description of the association relationship of associated objects, indicating that there can be three relationships. For example, A and / or B can represent: A exists alone, A and B exist at the same time, and B exists alone. In addition, the character " / " in this article, unless otherwise specified, generally means that the associated objects before and after are in an "or" relationship.

[0031] The embodiments of the present application are further described in detail below in conjunction with the drawings in the specification.

[0032] The present application embodiment provides a speaker control method based on a built-in sound source, referring to Figure 1 , the main process of the method is described as follows: S1: Obtaining a user's selection instruction, and calling a corresponding audio source file based on the selection instruction; The speaker is equipped with a variety of user interaction methods, such as touch screen, physical buttons or mobile phone applications, to facilitate users to input selection commands. Users can quickly locate and select the required sound source files by browsing the directory structure of the built-in sound source library of the speaker, or using the search function. When the user completes the selection, the speaker system receives the command and accurately retrieves the corresponding sound source file from the built-in memory to the audio processing unit to prepare for subsequent audio parameter adjustment.

[0033] S2: using a built-in preset audio processing algorithm to receive the retrieved audio source file and adjust the audio parameters of the audio source file; Among them, the built-in preset audio processing algorithm includes a variety of audio adjustment function modules, such as equalizer (EQ), dynamic range compressor, noise reducer, etc. After receiving the retrieved sound source file, the audio processing unit first analyzes the audio signal to determine its basic audio characteristics, such as frequency distribution, volume dynamic range, noise level, etc. Then, according to the preset algorithm parameters or user-defined settings, the gain of different frequency bands is adjusted through the equalizer to enhance or weaken specific frequency components to improve the timbre; the dynamic range compressor is used to control the maximum and minimum volume of the audio signal to make the sound more balanced during playback and avoid excessive or low volume; the noise reducer is used to remove background noise in the audio and improve the purity of the audio. After these processes, the sound source file with adjusted audio parameters is obtained, providing optimized audio material for the mixing algorithm.

[0034] S3: using a mixing algorithm to deeply optimize and combine the processed audio parameters corresponding to the multiple sound source files after the audio parameters have been adjusted, so as to obtain a playback audio signal; Among them, after the mixing algorithm is started, the style characteristics of multiple audio source files selected by the user after the audio parameters are adjusted are analyzed to determine whether there are significant differences in their music styles (such as classical, pop, rock, jazz, etc.), rhythm types (such as slow, medium, fast, complex rhythm types, etc.), melody trends (upward, downward, stable, fluctuating, etc.) and harmony characteristics. If there are significant differences, the user's preference setting information for the mixing effect (such as the user's preferred timbre balance, rhythm intensity, harmony richness, etc.) and the current environment information of the speaker (such as environmental noise level, space size, acoustic characteristics, etc.) are further obtained.

[0035] Through audio analysis technology, we deeply extract multi-dimensional audio feature information of each sound source file, including timbre characteristics (such as timbre brightness, warmth, thickness, etc.), rhythm characteristics (such as beat speed, rhythm stability, rhythm pattern complexity, etc.), pitch characteristics (such as pitch range, melody contour, interval relationship, etc.) and audio energy distribution characteristics (such as the energy proportion of each frequency band, dynamic energy changes of audio signals, etc.).

[0036] According to the obtained preference information, environmental information and multi-dimensional audio feature information, the intelligent mixing algorithm is used to formulate a mixing strategy. The timbre characteristics of the sound source files are optimized and distributed on the spectrum to ensure that different timbres complement each other in the frequency band and avoid interference. For example, the low-frequency rich timbre is reasonably matched with the high-frequency bright timbre to make the entire mix fuller and clearer on the spectrum; the rhythm synchronization is performed according to the rhythm characteristics, and the sound sources of different rhythm types are integrated on the time axis by stretching, compressing or filling the rhythm fragments to ensure the continuity and coordination of the rhythm; based on the pitch characteristics and melody characteristics, a suitable harmony framework is constructed, and the pitch and melody direction of the sound source corresponding to each sound source file are adjusted to achieve harmony and create a harmonious music effect; according to the environmental information and user preferences, the dynamic range of the sound source corresponding to each sound source file is compressed or expanded, and the relationship between the volume and dynamic performance of each sound source is balanced, so that the mix can maintain a good auditory effect in different environments.

[0037] During the mixing process, the machine learning model and preset music aesthetic rules are continuously used to evaluate and optimize the mixing effect. By learning from a large number of excellent mixing works, the machine learning model can predict the effects under different mixing parameter settings and compare and analyze the current mixing effects. The preset music aesthetic rules evaluate the balance, harmony, and layering of the mixing from the perspective of music theory and auditory perception. Based on the evaluation results, the mixing parameters are continuously adjusted to adapt to different music styles and user needs until the playback audio signal that meets the preset standards is obtained.

[0038] S4: transmitting the played audio signal to the integrated amplifier circuit for power amplification to enhance the signal strength so as to drive the speaker to work.

[0039] Among them, the playback audio signal obtained by the optimized combination of the mixing algorithm is transmitted to the integrated amplifier circuit. The integrated amplifier circuit uses a high-performance power amplifier chip to amplify the voltage and current of the audio signal and enhance the signal strength to make it sufficient to drive the speaker to work normally. During the power amplification process, the circuit will perform appropriate impedance matching and distortion control on the audio signal to ensure that the amplified audio signal can accurately restore the sound effect after mixing while maintaining a low degree of distortion. Finally, high-quality audio is played through the speaker, providing users with a clear, full, and infectious sound experience.

[0040] Specifically, in some possible embodiments, the step of using a mixing algorithm to deeply optimize and combine processed audio parameters corresponding to a plurality of audio source files after the audio parameters have been adjusted to obtain a playing audio signal includes: Analyzing and comparing style features of the multiple audio source files selected by the user; Determining whether there are significant differences in the styles of the multiple audio source files selected by the user; If there are significant differences in the style features, obtaining the user's preference setting information for the mixing effect and the current environment information of the speaker; By using audio analysis technology, multi-dimensional audio feature information of each of the audio source files is extracted, including timbre features, rhythm features, pitch features, and audio energy distribution features; According to the preference setting information, the environmental information and the multi-dimensional audio feature information, a mixing strategy is formulated using an intelligent mixing algorithm to optimize the distribution of the timbre features of the sound source files on the frequency spectrum so that different timbres complement each other in the frequency band and avoid interference; rhythm synchronization is performed based on rhythm features, and sound sources of different rhythm types are integrated on the time axis by stretching, compressing or filling rhythm segments; a suitable harmony framework is constructed based on pitch features and melody features, and the pitch and melody direction of the sound source corresponding to each sound source file are adjusted to achieve harmony; the dynamic range of the sound source corresponding to each sound source file is compressed or expanded according to the environmental information and user preferences to balance the relationship between the volume and dynamic performance of each sound source; During the mixing process, the machine learning model and preset music aesthetic rules are continuously used to evaluate and optimize the mixing effect, and the mixing parameters are constantly adjusted to adapt to different music styles and user needs; Output the optimized mixed audio signal.

[0041] Among them, when the user selects multiple sound source files, the system first calls the style feature analysis module. This module uses a deep learning algorithm to analyze the musical elements of each sound source file. For example, the music style of each sound source file is determined by pattern recognition in terms of melody patterns, harmonic progressions, rhythmic patterns, and instrument timbre combinations. For pop music, its typical rhythmic patterns (such as 4 / 4 beats, strong-weak-secondary strong-weak beat patterns), common chord progressions (such as I-IV-V, etc.) and a rich variety of electronic instrument timbres may be identified. For classical music, its complex harmonic structure, rich melodic lines, and the performance characteristics of specific instruments (such as violins, pianos, etc.) are analyzed. By comparing these analysis results, it is determined whether there are significant differences in the styles of the selected sound source files. The judgment standard can be based on a pre-set style similarity threshold. If the distance between the style feature vectors exceeds the threshold, it is considered that there is a significant difference.

[0042] If it is judged that there is a significant difference, the system will ask the user for the preference information of the mixing effect through the user interface of the speaker (such as the settings menu on the display, the preference options in the mobile phone application). The user can adjust parameters such as timbre balance (such as wanting heavier bass or brighter treble), rhythm intensity (preferring strong rhythm impact or soft rhythm rhythm), harmony complexity (simple and harmonious harmony or complex and varied harmony). At the same time, the built-in environmental sensors of the speaker (such as microphones for detecting ambient noise and light sensors for assisting in judging the size and layout of the indoor space) obtain information about the current environment of the speaker. For example, in a noisy environment, the system will tend to increase the overall volume of the mix and enhance the penetration of the sound; in a small space environment, the low-frequency components may be appropriately reduced to avoid the sound being too turbid.

[0043] For each sound source file, audio analysis technology uses algorithms such as fast Fourier transform (FFT) to perform spectrum analysis on the audio signal and extract timbre features, such as the spectral center of gravity of the timbre (determines the brightness or warmth of the timbre) and the harmonic structure (reflects the richness and characteristics of the timbre). The rhythm detection algorithm calculates the beat speed (the number of beats per minute), rhythm stability (the degree of change in the time interval between beats), and rhythm complexity (such as a simple uniform rhythm or a complex syncopated rhythm). The pitch detection algorithm is used to determine the pitch characteristics, including the pitch range (the span between the highest and lowest notes), the melody contour (the rise, fall, or fluctuation of the pitch over time), and the interval relationship (the pitch distance between adjacent notes). At the same time, the audio energy distribution characteristics are calculated to determine the energy proportion of each frequency segment (such as low frequency, medium frequency, and high frequency) and the dynamic energy changes of the audio signal over time.

[0044] Based on the obtained preference information, environmental information, and multi-dimensional audio feature information, the intelligent mixing algorithm begins to formulate a mixing strategy. In terms of spectrum optimization allocation, if the user prefers bright timbre and the current environmental noise is low, the system will appropriately increase the gain of the bright timbre sound source in the high frequency band, and reasonably allocate other complementary timbre (such as warm low-frequency timbre) to the corresponding frequency band to avoid frequency conflicts. For rhythm synchronization processing, if one sound source has a stable 4 / 4 beat rhythm and the other has a complex syncopated rhythm, the algorithm calculates the rhythm phase difference between the two and appropriately stretches or compresses the syncopated rhythm so that it can be smoothly integrated with the stable rhythm on the timeline. In terms of harmony construction, according to the pitch characteristics and melody direction, choose appropriate chords to match, adjust the pitch and melody direction of each sound source, and ensure harmony. For example, when multiple melody lines exist at the same time, adjust their pitch relationship to avoid discordant intervals. For dynamic range processing, if the user wants to highlight the dynamic changes of a main sound source in the preference settings and the environment is relatively quiet, the system will expand the dynamic range of the sound source and appropriately compress the dynamic range of other auxiliary sound sources to balance the relationship between the volume and dynamic performance of each sound source.

[0045] During the mixing process, the machine learning model continuously evaluates the mixing effect. The model is trained on a large number of audio samples of different styles and different mixing parameter settings to learn the relationship between different mixing parameters and the final auditory effect. It predicts the quality score of the mixing effect based on the audio features and parameter settings of the current mix. At the same time, the preset music aesthetic rules are evaluated from the perspective of music theory, such as checking the functionality of harmony, the fluency of melody, the rationality of rhythm, etc. If the evaluation result does not reach the preset satisfaction threshold, the system adjusts the mixing parameters according to the feedback of the machine learning model and the guidance of the music aesthetic rules, such as changing the spectrum distribution ratio of the timbre, fine-tuning the synchronization parameters of the rhythm, and adjusting the chord composition of the harmony. This process will continue to iterate until the mixing effect meets the requirements of different music styles and the personalized needs of users.

[0046] After a series of analysis, processing, evaluation and optimization processes, the mixing effect that meets the requirements is finally obtained, and the system outputs the optimized mixed audio signal to the subsequent power amplifier circuit. The audio signal has achieved an optimized balance in terms of spectrum, rhythm, harmony, dynamics, etc., and can provide users with high-quality, creative and personalized audio experience.

[0047] Specifically, in some possible embodiments, during the mixing process, the steps of continuously using the machine learning model and preset music aesthetic rules to evaluate and optimize the mixing effect, and continuously adjusting the mixing parameters to adapt to different music styles and user needs include: Use real-time audio analysis technology to continuously monitor the current mixing effect to obtain real-time mixing effect evaluation results; According to the real-time mixing effect evaluation result, an intelligent recommendation algorithm is used to screen out recommended sound source files that improve the mixing effect in the sound source library; The recommended audio source files are added to the current mix, and the mixing algorithm is re-applied to deeply optimize and combine all the audio source files.

[0048] Among them, the system uses real-time audio analysis technology to continuously monitor the audio signals in the mixing process. This technology collects audio signals at a high sampling frequency (for example, thousands of times per second), and uses algorithms such as fast Fourier transform (FFT) to perform spectrum analysis on audio signals to obtain information such as energy distribution and harmonic structure of audio signals in different frequency bands, so as to evaluate the timbre characteristics; the beat tracking algorithm is used to detect the stability of the rhythm, the beat speed and the accuracy of the rhythm type; the pitch detection algorithm is used to determine the accuracy of the pitch, the pitch range and the fluency of the melody. At the same time, the multi-dimensional information such as the volume balance relationship and phase relationship between the sound sources is analyzed, and this information is combined with the preset music aesthetic standards and the target mixing effect model to obtain the real-time mixing effect evaluation results. For example, if it is found that the energy of a certain frequency band is too high, resulting in a harsh timbre, or the rhythm is unstable and affects the overall sense of rhythm, these problems will be reflected in the evaluation results.

[0049] Based on the real-time mixing effect evaluation results, the intelligent recommendation algorithm is activated. The algorithm first analyzes the problems or directions for improvement in the current mix. For example, if the evaluation results show that the current mix lacks the power of the bass part, the algorithm will focus on screening audio source files with rich low-frequency components. The algorithm will conduct a comprehensive search in the sound source library, and based on audio feature tags, metadata descriptions, and previous mixing effect data, it will screen out recommended audio source files that match the current mixing style and can make up for the shortcomings of the existing mixing. For example, if the current mix is ​​in pop music style and needs to enhance the sense of rhythm, the algorithm will give priority to recommending audio source files with strong rhythmic patterns and rich percussion elements; if the mix is ​​in classical music style and the harmony is slightly thin, the algorithm will look for classical music clips with complex harmonic structures and beautiful melody lines as recommended sound sources.

[0050] Add the selected recommended audio source files to the current mix. At this time, the mixing processing unit will treat the newly added audio source files and the original multiple audio source files as a whole. Re-apply the mixing algorithm to deeply optimize and combine all audio source files. This process is similar to the previous mixing process, but it will pay more attention to the integration of the newly added audio source files with the original mix. In terms of spectrum allocation, the distribution of each audio source in the frequency band will be readjusted to ensure that the timbre characteristics of the new audio source complement the original timbre and do not conflict with each other; for rhythm processing, the rhythm synchronization analysis will be performed again to make the rhythm of the new audio source perfectly integrated with the original rhythm to maintain the overall rhythm continuity; in terms of harmony construction, the pitch and melody direction of the new audio source will be reconsidered to build a more harmonious and rich harmony framework; at the same time, according to the new mixing situation, the dynamic range of each audio source will be rebalanced to achieve a new balance in volume and dynamic performance of the entire mix. After this series of remixing operations, an optimized and improved mixing effect is obtained, and real-time monitoring and evaluation are continued, and the cycle is repeated until the mixing effect reaches the optimal state to meet different music styles and user needs.

[0051] Specifically, in some possible embodiments, the method further includes: Creating multiple intelligent premixed sound source templates covering different music styles, emotional atmospheres and application scenarios, each of the intelligent premixed sound source templates is composed of multiple pre-processed sound source files, and each of the intelligent premixed sound source templates is attached with corresponding audio feature tags and metadata descriptions; Acquire user demand information, and based on the acquired user demand information, perform a quick matching search in a premixed sound source template library to find one or more templates that best match the user demand as a mixing basis; Determine a matching premixed sound source template, load the sound source file in the determined premixed sound source template into a mixing processing unit, and perform fast mixing according to preset parameters of the premixed sound source template to generate an initial mixed audio signal; During playback, the system continuously monitors users’ real-time feedback and environmental change information, and uses adaptive algorithms to dynamically adjust mixing parameters.

[0052] Among them, the music professional team and audio engineers cooperated to carefully select and edit multiple sound source files according to different music styles (such as classical, pop, rock, jazz, electronic, etc.), emotional atmosphere (cheerful, sad, passionate, soothing, quiet, etc.) and application scenarios (such as parties, study, meditation, sports, movie soundtracks, etc.) to form intelligent pre-mixed sound source templates. For each template, professional audio processing software is used to pre-process the sound source files it contains, including audio format unification, volume standardization, noise removal and other operations to ensure the audio quality and compatibility of the template. At the same time, detailed audio feature tags are attached to each template, such as music style tags (such as "classical-baroque style" and "pop-folk style"), emotional atmosphere tags (such as "cheerful-positive" and "sad-lyrical"), application scenario tags (such as "party-lively" and "study-quiet"), and key audio feature descriptions (such as "mainly string music, light rhythm, rich harmony"), etc. In addition, metadata descriptions are added, including information such as the creation background of the template, the applicable population, and recommended matching equipment, etc., to facilitate user understanding and selection.

[0053] The speaker system obtains user demand information in a variety of ways. Users can input their needs through the speaker's touch screen interface, mobile phone applications, or voice interaction, such as selecting a specific music style (such as "I want a rock-style mix"), the desired emotional atmosphere (such as "create a quiet and relaxing atmosphere"), or applicable scenarios (such as "for yoga practice"). The system converts the information entered by the user into recognizable demand parameters, and then performs a quick matching search in the pre-mixed sound source template library. The matching algorithm calculates the similarity between the user's demand parameters and the audio feature tags and metadata descriptions of the template, and gives priority to recommending templates with high similarity. For example, if the user's demand is "for running, with a strong rhythm and vitality", the system will search for templates with "sports" scene tags, "strong rhythm" and "vitality" and other emotional atmosphere tags, as well as "electronic music" and "rock rhythm" and other music style features in the template library, and sort them from high to low according to the matching degree and display them to the user.

[0054] After the user determines the matching pre-mixed sound source template, the speaker system loads the sound source files in the template into the mixing processing unit. The mixing processing unit performs fast mixing operations according to the preset parameters of the template. These preset parameters are set by professionals according to the characteristics of the template and the target effect during the template creation process, including the initial volume ratio, sound and image position, basic equalizer settings, etc. of each sound source file. The mixing processing unit quickly combines and processes the loaded sound source files according to these preset parameters to generate an initial mixed audio signal. In this process, efficient mixing algorithms and optimized computing resources are used to ensure that the mixing operation is completed in a shorter time, reducing the user's waiting time, while ensuring that the mixing quality reaches a certain benchmark level, providing a good foundation for further adjustments.

[0055] During the playback of the initial mixed audio signal, the speaker system continuously monitors the user's real-time feedback information and environmental change information through a variety of sensors and interactive methods. For example, the built-in microphone collects the user's voice feedback (such as the user saying "the sound is too loud" or "the bass is not enough"), and monitors the user's behavior (such as the user's movement increases as the music rhythm speeds up) and environmental changes (such as the dimming of the ambient light may indicate the night rest scene) through acceleration sensors, light sensors, etc. Adaptive algorithms are used to analyze and process these feedback and change information, and the mixing parameters are dynamically adjusted according to preset rules. If the user feedback is too loud, the system will automatically reduce the overall volume and adjust the dynamic range appropriately; if the user's movement increases and the ambient light does not change significantly, it may indicate that the user is active, and the system will enhance the rhythm intensity and bass effect. At the same time, the algorithm will continue to learn the user's preferences for adjusting mixing parameters in different scenarios, and continuously optimize the adjustment strategy to provide a mixing effect that is more in line with user needs and environmental changes, ensuring that users always get the best audio experience.

[0056] Specifically, in some possible embodiments, the steps before obtaining the playing audio signal include: Pre-design and generate multiple excellent mixing templates covering different music styles, emotional atmospheres and application scenarios; When the user selects multiple audio source files for mixing creation, the user's operation process is monitored in real time, and when the user completes the preliminary mixing settings or adjusts the mixing parameters in real time, the audio analysis technology is used to extract the audio features of the mixing project created by the user; Compare and analyze the audio features of the mixing project described by the user with the audio features of each excellent mixing template in the template library one by one to obtain a contrast ratio; Based on the priority order of the contrast, an excellent mixing template with a high contrast has a high corresponding priority, and a recommended excellent mixing template list and a corresponding comparison report are displayed to the user.

[0057] Among them, the professional audio designer team, with rich experience in music creation and mixing, uses professional audio production software to carefully design and generate excellent mixing templates covering a wide range of music styles (such as the elegance and solemnity of classical music, the diverse styles of pop music, the innovative rhythm of electronic music, etc.), diverse emotional atmospheres (from passionate and enthusiastic to quiet and soothing), and various application scenarios (including stage performances, family gatherings, personal creations, etc.). During the design process, the team deeply analyzed the combination effects of different music elements, and finely processed and optimized the configuration of the sound source files in each template. For example, when building a rock-style mixing template for stage performances, carefully select drum sound sources with a strong sense of rhythm, penetrating electric guitar sounds, and infectious vocal tracks, and create a passionate and energetic stage sound effect by accurately adjusting the volume balance, sound image positioning, frequency balance, and dynamic range of each sound source. At the same time, each template is assigned detailed audio feature labels, such as rhythm characteristics (such as "fast and intense 4 / 4 beat"), harmonic structure (such as "a combination of powerful chords and melodic harmonies"), timbre characteristics (such as "bright and granular electric guitar tone"), etc., to facilitate subsequent accurate comparative analysis.

[0058] When the user selects multiple sound source files on the speaker system for mixing creation, the system monitors the user's operation process in real time. Whether adjusting the mixing parameters (such as volume sliders, pan knobs, effect switches, etc.) through the touch interface or controlling the mixing process through voice commands, the system can capture the user's operation behavior in time. At the key node where the user completes the preliminary mixing settings or adjusts the mixing parameters in real time, the system uses audio analysis technology to conduct a comprehensive analysis of the mixing project created by the user. The energy distribution and timbre characteristics (such as spectral center of gravity, harmonic richness, etc.) of the mixing project in different frequency bands are obtained through the spectrum analysis algorithm; the rhythm detection algorithm is used to determine the rhythm speed, rhythm stability and rhythm pattern complexity of the mixing; the pitch range, melody direction and harmonic relationship are identified with the help of the pitch analysis tool. These extracted audio features will serve as the basic data for comparison with excellent mixing templates to accurately reflect the unique nature of the user's mixing project.

[0059] The audio features of the user's mixing project are compared and analyzed with the audio features of each excellent mixing template in the template library one by one. For each audio feature dimension, the system calculates the degree of difference between the two. For example, in the comparison of rhythm features, the beat speed difference and rhythm type similarity between the mixing project and the template are calculated; in terms of timbre feature comparison, the difference in spectrum distribution and the similarity of harmonic structure are analyzed; for harmony features, the consistency of chord progressions and the matching degree of pitch relationships are evaluated. By comprehensively considering the differences in various audio feature dimensions, the preset weighting algorithm is used to calculate the contrast between the mixing project and each template. The weighting algorithm assigns corresponding weights according to the degree of influence of different audio features on the mixing effect. For example, rhythm and harmony features may have higher weights in the overall comparison because they play a key role in the overall structure and emotional expression of music.

[0060] According to the priority order of contrast, the system sorts the excellent mixing templates, with the templates with high contrast ranked at the top. A list of recommended excellent mixing templates is shown to the user. In addition to the display name and thumbnail, each template in the list also provides a brief description to highlight its similarities with the user's mixing project and possible improvement directions. At the same time, a detailed comparison report is generated for the user, which is presented in the form of a combination of charts and text descriptions. The chart part intuitively shows the difference between the mixing project and the recommended template in key audio features (such as rhythm, timbre, and harmony), such as the difference in frequency band energy distribution shown by a line chart and the difference in beat speed compared by a bar chart. The text description part explains in depth the impact of these differences on the mixing effect and provides professional suggestions to help users understand how the recommended template can improve the quality of the mixing. Users can choose the excellent mixing template that suits them as a reference based on the recommended list and comparison report to further optimize the mixing creation.

[0061] Specifically, in some possible embodiments, the steps after transmitting the played audio signal to the integrated amplifier circuit for power amplification and enhancing the signal strength to drive the speaker to work include: Real-time monitoring of the actual sound quality and effect of the speaker output; Using audio analysis technology to perform multi-dimensional testing on the actual played sound, including but not limited to volume stability, timbre accuracy, frequency response flatness, distortion, and spatial sound positioning accuracy; Obtain current environment information and user feedback information; Based on the current environment information and the user feedback information, an adaptive control algorithm is used to adjust the played audio signal in real time.

[0062] Among them, while the playing audio signal is driven by the power amplifier to drive the speaker to work, the system starts the real-time monitoring module. This module uses a high-precision audio acquisition device connected to the output end of the speaker to collect the actual playing sound in real time at an extremely high sampling frequency (for example, hundreds of thousands of times per second). Using advanced audio analysis technology, the collected audio signal is tested in multiple dimensions. In terms of volume stability, the fluctuation range of volume per unit time is calculated to determine whether the volume fluctuates; in terms of timbre accuracy, the frequency spectrum is compared with the original mixed audio signal to analyze the changes in timbre in terms of frequency components, harmonic structure, etc., to ensure that the actual playback timbre is consistent with expectations; in the frequency response flatness test, the sound intensity at different frequency points is measured, and the frequency response curve is drawn to evaluate whether it remains flat within a reasonable range to ensure balanced playback of sounds at all frequencies; in terms of distortion, a distortion analysis algorithm is used to detect indicators such as harmonic distortion and intermodulation distortion in audio signals to ensure that the sound is not distorted; in terms of spatial sound positioning accuracy, multi-channel audio analysis technology is used, combined with the placement of the speakers and the acoustic environment, to determine whether the sound is accurately positioned in space, such as whether the left and right channel separation, the front and rear sound field layering, etc. have achieved the expected results.

[0063] In order to have a more comprehensive understanding of the playback environment and user experience, the system obtains current environmental information and user feedback information. Environmental information is obtained through a variety of sensors, such as ambient light sensors used to detect ambient brightness and determine whether it is in a bright or dim environment, so as to adjust the brightness or warmth of the sound to match it; noise sensors monitor the ambient noise level in real time to provide a basis for adaptive volume adjustment; temperature sensors can consider the impact of temperature on the speed of sound propagation and speaker performance (although the impact is relatively small, it can still be used as a reference factor in high-precision audio processing). At the same time, the system collects user feedback information in a variety of ways, such as setting a convenient feedback button on the speaker device, so that users can feedback problems such as too loud, too soft, and poor timbre with one click; or connecting to the speaker through a mobile phone application, users can enter detailed feedback in the application; in addition, voice recognition technology can be used to recognize users' verbal evaluation of sound effects in real time, such as "the sound is a bit dull" and "the bass is not shocking enough".

[0064] Based on the current environmental information and user feedback information obtained, the adaptive control algorithm begins to make real-time adjustments to the playback audio signal. When the ambient noise is large, the algorithm automatically increases the volume of the audio signal to ensure that the user can hear the sound clearly. At the same time, according to the frequency characteristics of the noise, the frequency distribution of the audio signal is appropriately adjusted to highlight the mid- and low-frequency parts and enhance the penetration of the sound; if the user feedback is that the sound is too bright, the algorithm reduces the gain of the treble part and adjusts the timbre to make it softer; if the user wants to enhance the spatial sound effect, the algorithm optimizes the phase relationship and volume balance of the multi-channel audio signal according to the layout of the speakers and the acoustic characteristics of the room to create a more three-dimensional and immersive spatial sound effect. During the adjustment process, the algorithm will comprehensively consider the weight of environmental information and user feedback, give priority to meeting the direct needs of users, and take into account the impact of environmental factors on the sound effect. Through continuous iterative adjustments, the playback audio signal always adapts to environmental changes and user preferences, providing users with the best audio experience. In addition, the algorithm will also learn the user's preference settings in different environments and scenarios, gradually forming a personalized audio adjustment strategy to further improve user satisfaction.

[0065] Specifically, in some possible embodiments, the method further includes: Obtaining registration information of a smart home device, assigning a unique device identifier to the corresponding smart home device, and obtaining basic function information of the smart home device; Classify and mark the sound source files in the built-in sound source library, and establish a mapping relationship between each category and the smart home device and operation action; When the speaker is playing, feature recognition is performed on the sound source file corresponding to the played audio signal in real time to obtain a feature recognition result; Automatically generate corresponding smart home control instructions according to the feature recognition results and the mapping relationship; The generated smart home control instructions are sent to the corresponding smart home devices via a wireless communication protocol.

[0066] Among them, the speaker system starts the smart home device discovery and connection function, and detects and identifies the connected smart home devices around by searching the local network or specific communication frequency band. For devices that support Bluetooth connection, the speaker uses the Bluetooth scanning function to search for nearby devices in pairing mode; for devices that support Wi-Fi connection, it communicates with the home network router to obtain a list of smart home devices connected to the same network. When a new smart home device is discovered, the speaker system sends a request to the device to obtain its registration information such as device model, manufacturer, device type (such as lights, curtains, air conditioners, etc.). Based on this information, a unique device identifier is assigned to the device, which is stored in the device management database of the speaker system for subsequent accurate identification and control of the device. At the same time, the basic functional information of smart home devices is deeply obtained, such as the brightness adjustment range and color mode of lighting devices, the opening and closing degree control accuracy of curtain devices, the temperature adjustment range and wind speed gear of air conditioning devices, etc. These functional information will serve as an important basis for building mapping relationships.

[0067] Professional music teams and audio editors comprehensively classify and label the audio files in the built-in audio library. They classify the files in detail based on multiple dimensions, such as music style (such as classical, pop, rock, jazz, etc.), emotional atmosphere (cheerful, sad, passionate, soothing, etc.), rhythm characteristics (fast, medium, slow, strong rhythm, soft rhythm, etc.), and audio scenes (such as morning wake-up, night sleep aid, party entertainment, study and work, etc.). For example, pop music with a brisk rhythm and lively melody is marked as "cheerful-popular-fast rhythm-party entertainment". For each category, a mapping relationship is established between smart home devices and operation actions. For example, for audio source files in the "Cheerful - Party Entertainment" category, mapping relationships are established with the "turn on bright lights, switch to dynamic lighting effects" operation of lighting equipment, the "fully open curtains" operation of curtain equipment, and the "increase volume, enhance bass effect" operation of the smart speaker itself; for audio source files in the "Relax - Nighttime Sleeping Aid" category, mapping is done to the "dim lights, switch to warm tones" operation of lighting equipment, the "close curtains" operation of curtain equipment, and the "lower volume, play soft sound effects" operation of the speaker. These mapping relationships are stored in the mapping relationship database of the speaker system and can be flexibly adjusted and expanded according to user needs and actual usage scenarios.

[0068] When the speaker plays audio, the real-time audio analysis module starts to identify the features of the sound source file corresponding to the played audio signal. Using deep learning algorithms and audio signal processing technology, analyze the rhythm, melody, harmony, timbre and other characteristics of the audio. The rhythm detection algorithm determines the beat speed and rhythm type of the music, such as identifying whether it is a stable 4 / 4 beat rhythm or a complex syncopated rhythm; the melody analysis tool is used to extract the direction of the melody line and the range of pitch changes; the harmony recognition technology is used to analyze the chord progression and harmony function; the timbre feature analysis algorithm is used to determine the brightness, warmth, richness, etc. of the timbre. These feature recognition results are matched and queried with the previously established mapping relationship database to find the corresponding smart home devices and operation action combinations, and automatically generate corresponding smart home control instructions. For example, if it is recognized that the currently playing music is a rock style with a strong rhythm, and the melody is passionate and the timbre is heavy, the system will generate control instructions based on the mapping relationship, turn up the lights and switch to the flashing effect, and turn on the air purifier to create a vibrant environment.

[0069] The generated smart home control instructions are sent to the corresponding smart home devices through the wireless communication protocol. The speaker system supports a variety of mainstream wireless communication protocols, such as Wi-Fi, Bluetooth, Zigbee, etc., and automatically adapts and communicates according to the protocols supported by the smart home devices. Before sending the instructions, the system encrypts and verifies the instructions to ensure the security and integrity of the instructions. The encrypted instructions are transmitted to the target smart home device through the wireless network. After the device receives the instructions, it first decrypts and verifies the instructions. If the instructions are correct, the device performs the corresponding operation. For example, the lighting equipment adjusts the brightness, color or lighting effects according to the received instructions; the curtain equipment controls the motor to drive the curtains to open and close to the specified position; the air conditioning equipment adjusts the operating parameters such as temperature and wind speed. After the device completes the operation, some smart devices will send operation feedback information to the speaker system. After receiving the feedback, the speaker system confirms that the command has been executed successfully, and can display the operation result prompt to the user as needed, such as displaying "The lights have been dimmed to 80%" on the speaker display or using voice prompts "The curtains are fully open", etc., to achieve two-way communication and interaction between the speaker and smart home devices, providing users with a more intelligent and convenient integrated home control experience.

[0070] Another embodiment of the present application provides a speaker control system based on a built-in sound source, wherein Figure 2 , a speaker control system based on built-in sound source, including: The audio source file retrieving module 100 is used to obtain the user's selection instruction and retrieve the corresponding audio source file based on the selection instruction; The audio parameter adjustment module 200 uses a built-in preset audio processing algorithm to receive the retrieved audio source file and adjust the audio parameters of the audio source file; The playing audio signal generating module 300 uses a mixing algorithm to deeply optimize and combine the processed audio parameters corresponding to the multiple sound source files after the audio parameters are adjusted, so as to obtain a playing audio signal; The audio signal transmission module 400 transmits the audio signal to the integrated amplifier circuit for power amplification, thereby increasing the signal strength to drive the speaker to work.

[0071] The speaker control system based on the built-in sound source provided in this embodiment can implement the various steps of the aforementioned embodiments due to the functions of its modules themselves and the logical connections between each other, and thus can achieve the same technical effects as the aforementioned embodiments. The principle analysis can be found in the relevant description of the steps of the aforementioned speaker control method based on the built-in sound source, which will not be repeated here.

[0072] An embodiment of the present application also provides a speaker control device based on a built-in sound source, including a memory and a processor, wherein the memory stores a computer program that can be loaded by the processor and execute the above-mentioned speaker control method based on the built-in sound source.

[0073] An embodiment of the present application further provides a storage medium, in which a computer program that can be loaded by a processor and execute the above-mentioned speaker control method based on a built-in sound source is stored.

[0074] The storage medium provided in this embodiment can achieve the same technical effects as the aforementioned embodiments because the computer program therein implements the various steps of the aforementioned embodiments after being loaded and run on the processor. The principle analysis can be found in the relevant description of the aforementioned method steps, which will not be repeated here.

[0075] The storage medium includes, for example, various media that can store program codes, such as a USB flash drive, a mobile hard disk, a read-only memory (ROM), a random access memory (RAM), a magnetic disk or an optical disk.

[0076] The steps of the method or algorithm described in conjunction with the embodiments disclosed herein may be implemented directly using hardware, a software module executed by a processor, or a combination of the two. The software module may be placed in a random access memory (RAM), a memory, a read-only memory (ROM), an electrically programmable ROM, an electrically erasable programmable ROM, a register, a hard disk, a removable disk, a CD-ROM, or any other form of storage medium known in the art.

[0077] In the description of this specification, the description with reference to the terms "one embodiment", "some embodiments", "example", "specific example", or "some examples" etc. means that the specific features, structures, materials or characteristics described in conjunction with the embodiment or example are included in at least one embodiment or example of the present invention. In this specification, the schematic representations of the above terms do not necessarily refer to the same embodiment or example. Moreover, the specific features, structures, materials or characteristics described may be combined in any one or more embodiments or examples in a suitable manner. In addition, those skilled in the art may combine and combine the different embodiments or examples described in this specification and the features of the different embodiments or examples, without contradiction.

[0078] In addition, the term "first" or "second" may explicitly or implicitly include at least one of the features. In the description of the present invention, "plurality" means at least two, such as two, three, etc., unless otherwise clearly and specifically defined, and is only used for descriptive purposes, and cannot be understood as indicating or implying relative importance or implicitly indicating the number of the indicated technical features.

[0079] Thus, any process or method description in a flowchart or otherwise described herein may be understood to represent a module, segment or portion of code comprising one or more executable instructions for implementing the steps of a custom logical function or process, and the scope of the preferred embodiments of the present invention includes alternative implementations in which functions may not be performed in the order shown or discussed, including performing functions in a substantially simultaneous manner or in reverse order depending on the functions involved, which should be understood by technicians in the technical field to which the embodiments of the present invention belong.

[0080] The embodiments of this specific implementation method are all preferred embodiments of the present application, and are not intended to limit the protection scope of the present application. Therefore, all equivalent changes made based on the structure, shape, and principle of the present application should be included in the protection scope of the present application.

Claims

1. A speaker control method based on a built-in sound source, characterized in that: include: Obtaining a user's selection instruction, and calling a corresponding audio source file based on the selection instruction; Using a built-in preset audio processing algorithm, receiving the retrieved audio source file, and adjusting the audio parameters of the audio source file; Using a mixing algorithm, the processed audio parameters corresponding to the multiple sound source files after the audio parameters have been adjusted are deeply optimized and combined to obtain a playback audio signal; The played audio signal is transmitted to the integrated amplifier circuit for power amplification to enhance the signal strength so as to drive the speaker to work.

2. The method for controlling a speaker based on a built-in sound source according to claim 1, characterized in that: The steps of using a mixing algorithm to deeply optimize and combine the processed audio parameters corresponding to the multiple sound source files after the audio parameters are adjusted to obtain the playing audio signal include: Analyzing and comparing style features of the multiple audio source files selected by the user; Determining whether there are significant differences in the styles of the multiple audio source files selected by the user; If there are significant differences in the style features, obtaining the user's preference setting information for the mixing effect and the current environment information of the speaker; By using audio analysis technology, multi-dimensional audio feature information of each of the audio source files is extracted, including timbre features, rhythm features, pitch features, and audio energy distribution features; According to the preference setting information, the environmental information and the multi-dimensional audio feature information, a mixing strategy is formulated using an intelligent mixing algorithm to optimize the distribution of the timbre features of the sound source files on the frequency spectrum so that different timbres complement each other in the frequency band and avoid interference; rhythm synchronization is performed based on rhythm features, and sound sources of different rhythm types are integrated on the time axis by stretching, compressing or filling rhythm segments; a suitable harmony framework is constructed based on pitch features and melody features, and the pitch and melody direction of the sound source corresponding to each sound source file are adjusted to achieve harmony; the dynamic range of the sound source corresponding to each sound source file is compressed or expanded according to the environmental information and user preferences to balance the relationship between the volume and dynamic performance of each sound source; During the mixing process, the machine learning model and preset music aesthetic rules are continuously used to evaluate and optimize the mixing effect, and the mixing parameters are constantly adjusted to adapt to different music styles and user needs; Output the optimized mixed audio signal.

3. The method for controlling a speaker based on a built-in sound source according to claim 2, characterized in that: During the mixing process, the steps of continuously using machine learning models and preset music aesthetic rules to evaluate and optimize the mixing effect and continuously adjust the mixing parameters to adapt to different music styles and user needs include: Use real-time audio analysis technology to continuously monitor the current mixing effect to obtain real-time mixing effect evaluation results; According to the real-time mixing effect evaluation result, an intelligent recommendation algorithm is used to screen out recommended sound source files that improve the mixing effect in the sound source library; The recommended audio source files are added to the current mix, and the mixing algorithm is re-applied to deeply optimize and combine all the audio source files.

4. The method for controlling a speaker based on a built-in sound source according to claim 1, characterized in that: The method further comprises: Creating multiple intelligent premixed sound source templates covering different music styles, emotional atmospheres and application scenarios, each of the intelligent premixed sound source templates is composed of multiple pre-processed sound source files, and each of the intelligent premixed sound source templates is attached with corresponding audio feature tags and metadata descriptions; Acquire user demand information, and based on the acquired user demand information, perform a quick matching search in a premixed sound source template library to find one or more templates that best match the user demand as a mixing basis; Determine a matching premixed sound source template, load the sound source file in the determined premixed sound source template into a mixing processing unit, and perform fast mixing according to preset parameters of the premixed sound source template to generate an initial mixed audio signal; During playback, the system continuously monitors users’ real-time feedback and environmental change information, and uses adaptive algorithms to dynamically adjust mixing parameters.

5. The method for controlling a speaker based on a built-in sound source according to claim 1, characterized in that: The steps before obtaining the playing audio signal include: Pre-design and generate multiple excellent mixing templates covering different music styles, emotional atmospheres and application scenarios; When the user selects multiple audio source files for mixing creation, the user's operation process is monitored in real time, and when the user completes the preliminary mixing settings or adjusts the mixing parameters in real time, the audio analysis technology is used to extract the audio features of the mixing project created by the user; Compare and analyze the audio features of the mixing project described by the user with the audio features of each excellent mixing template in the template library one by one to obtain a contrast ratio; Based on the priority order of the contrast, an excellent mixing template with a high contrast has a high corresponding priority, and a recommended excellent mixing template list and a corresponding comparison report are displayed to the user.

6. The method for controlling a speaker based on a built-in sound source according to claim 1, characterized in that: The steps after transmitting the playing audio signal to the integrated amplifier circuit for power amplification and enhancing the signal strength to drive the speaker to work include: Real-time monitoring of the actual sound quality and effect of the speaker output; Using audio analysis technology to perform multi-dimensional testing on the actual played sound, including but not limited to volume stability, timbre accuracy, frequency response flatness, distortion, and spatial sound positioning accuracy; Obtain current environment information and user feedback information; Based on the current environment information and the user feedback information, an adaptive control algorithm is used to adjust the played audio signal in real time.

7. The method for controlling a speaker based on a built-in sound source according to claim 1, characterized in that: The method further comprises: Obtaining registration information of a smart home device, assigning a unique device identifier to the corresponding smart home device, and obtaining basic function information of the smart home device; Classify and mark the sound source files in the built-in sound source library, and establish a mapping relationship between each category and the smart home device and operation action; When the speaker is playing, feature recognition is performed on the sound source file corresponding to the played audio signal in real time to obtain a feature recognition result; Automatically generate corresponding smart home control instructions according to the feature recognition results and the mapping relationship; The generated smart home control instructions are sent to the corresponding smart home devices via a wireless communication protocol.

8. A speaker control system based on a built-in sound source, characterized in that: include: Audio source file retrieval module: obtains the user's selection instruction and retrieves the corresponding audio source file based on the selection instruction; Audio parameter adjustment module: using a built-in preset audio processing algorithm, receiving the retrieved audio source file, and adjusting the audio parameters of the audio source file; Audio signal generation module: uses a mixing algorithm to deeply optimize and combine the processed audio parameters corresponding to multiple audio source files after the audio parameters have been adjusted to obtain the audio signal for playback; Audio signal transmission module: transmits the audio signal to the integrated amplifier circuit for power amplification, thereby increasing the signal strength to drive the speaker to work.

9. A speaker control device based on a built-in sound source, characterized in that: include: A memory and a processor, wherein the memory stores a computer program that can be loaded by the processor and execute any one of the speaker control methods based on a built-in sound source according to claims 1-7.

10. A storage medium, characterized in that: A computer program is stored which can be loaded by a processor and execute any one of the speaker control methods based on a built-in sound source according to claims 1-7.

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