Audio adaptive adjustment control method and related device

Abstract

The invention discloses an audio adaptive adjustment control method and a related device, and relates to the technical field of audio control, and the method comprises the steps: carrying out the performance evaluation based on the processor operation load, the power supply state, the data transmission error rate and the wireless link quality of each sound box, so as to determine a main sound box and a plurality of slave sound boxes; the main sound box receives the audio stream and distributes the audio stream to each slave sound box for playing; in the playing process, analyzing environmental interference characteristics to determine signal emission information, and emitting a detection signal; analyzing the pulse response based on the detection signal to determine an audio adjustment parameter to adjust the audio stream; and in the playing process, switching control of the main loudspeaker box is carried out based on the initial adaptive switching threshold, the accumulated auditory cost and the initial role viscosity weighted component in combination with the performance evaluation result of each slave loudspeaker box. According to the invention, the stability of an interconnected sound box network, the adaptability of audio playing and the fluency of user experience can be obviously improved.

Description

Technical Field

[0001] This invention relates to the field of audio control technology, and in particular to an audio adaptive adjustment control method and related apparatus. Background Technology

[0002] In modern life, Bluetooth speakers have become an indispensable part of home entertainment and office meetings. People increasingly prefer to connect multiple Bluetooth speakers to form a network, allowing sound to be heard in every corner of the home or simultaneously playing music in different rooms, thus enhancing the auditory experience. Furthermore, thanks to advancements in Bluetooth technology, a single audio source can now transmit sound to many compatible devices simultaneously, breaking the previous limitation of one-to-one Bluetooth connections and further driving the demand for multi-speaker interconnection. Such interconnected systems use a smart device as the sound source, achieving synchronized sound playback through the cooperation of master and slave speakers.

[0003] However, existing Bluetooth speaker interconnection technology still faces several challenges. When assembling a speaker team, the primary speaker is often simply chosen based on signal strength or randomness. This leads to poor system stability, high audio transmission latency, and even connection interruptions if the primary speaker's performance is inadequate. Secondly, during sound playback, the system doesn't effectively adjust to changes in the surrounding environment, easily causing sound quality deviations. Traditional audio adjustments rely on rough distance estimations based on signal strength, lacking accuracy and failing to precisely adapt to the environment. Furthermore, the system doesn't correlate the energy characteristics of each sound segment with environmental parameters in real time, resulting in fixed sound adjustment parameters that struggle to handle dynamically changing scenarios, ultimately leading to a significant discrepancy between the played sound and the user's expectations. Summary of the Invention

[0004] The purpose of this invention is to overcome the shortcomings of the prior art. This invention provides an audio adaptive adjustment control method and related device, which can significantly improve the stability of interconnected speaker networks, the adaptability of audio playback, and the smoothness of user experience, and realize intelligent management and optimized control of multi-speaker interconnection systems.

[0005] To address the aforementioned technical problems, this invention provides an audio adaptive adjustment control method, the method comprising: Performance evaluations are conducted based on the processor load, power supply status, data transmission error rate, and wireless link quality of each speaker to obtain the performance evaluation results of each speaker. Based on the performance evaluation results of each speaker, the main speaker and several slave speakers are determined. The main speaker and several slave speakers form an interconnected speaker network. The main speaker in the interconnected speaker network receives the audio stream and distributes the audio stream to each slave speaker for playback based on one-to-many broadcast technology. During the playback of the audio stream, the main speaker and each of the slave speakers analyze the environmental interference characteristics, determine the signal transmission information based on the environmental interference characteristics, and transmit detection signals based on the signal transmission information; Based on the detection signal, the impulse response is analyzed, and based on the impulse response, audio adjustment parameters are determined. Based on the audio adjustment parameters, the audio stream is adjusted. During the playback of the audio stream, the initial adaptive switching threshold, cumulative auditory cost, and initial role stickiness weighted score of the main speaker are obtained. Based on the initial adaptive switching threshold, cumulative auditory cost, and initial role stickiness weighted score, combined with the performance evaluation results of each slave speaker, the switching control of the main speaker is performed using interactive behavior.

[0006] Optionally, the analysis of environmental interference characteristics and the determination of signal transmission information based on the environmental interference characteristics include: Acquire environmental acoustic data and perform spectral analysis on the environmental acoustic data based on fast Fourier transform to obtain spectral data; Based on a preset spectrum energy threshold table and a periodic detector, the spectrum data is used to determine the characteristics of environmental interference. The frequency band range and time window for signal transmission are determined based on the environmental interference characteristics, and the signal transmission information is determined based on the frequency band range and time window.

[0007] Optionally, the step of analyzing the impulse response based on the detection signal and determining the audio adjustment parameters based on the impulse response includes: The impulse response is calculated by combining the transmitted and received probe signal information with the Fast Fourier Transform and Inverse Fast Fourier Transform. The reverberation time is determined based on the impulse response, and the audio adjustment parameters are determined based on the reverberation time and environmental interference characteristics.

[0008] Optionally, obtaining the initial adaptive switching threshold, cumulative auditory cost, and initial role stickiness weighted score of the main speaker includes: The fluctuation index of the current radio environment is calculated based on the Bluetooth packet error rate and received signal strength indication of each speaker. The initial adaptive switching threshold is determined based on the volatility index; Obtain the speaker switching frequency count of the main speaker, and determine the cumulative auditory cost based on the speaker switching frequency count; The initial role stickiness weighting score is determined based on the duration of the main speaker's presence.

[0009] Optionally, the step of controlling the switching of the main speaker based on the initial adaptive switching threshold, cumulative auditory cost, and initial role stickiness weighted score combined with the performance evaluation results of each slave speaker, using interactive behavior, includes: The system acquires user interaction behavior with the interconnected speaker network, and adjusts the initial role stickiness weighted score and the initial adaptive switching threshold based on the interaction behavior to obtain the target adaptive switching threshold and the target role stickiness weighted score. If the performance evaluation result of a secondary speaker is greater than the sum of the cumulative auditory cost of the primary speaker, the target adaptive switching threshold, and the target role stickiness weighted score, then the secondary speaker with a performance evaluation result greater than the sum will be switched to the new primary speaker.

[0010] Optionally, switching the speaker whose performance evaluation result is greater than the sum of the values ​​to the new main speaker includes: Identify the audio content type of the audio stream; Sensitivity analysis of main speaker switching is performed based on the audio content type to obtain target sensitivity information, and speaker switching time period is determined based on the target sensitivity information; The audio content characteristics of the audio stream are obtained, and the audio buffer information and audio synchronization information for main speaker switching are determined based on the audio content characteristics. Based on the speaker switching period, audio buffer information, and audio synchronization information, the speaker with a performance evaluation result greater than the sum of the values ​​will be switched to the new main speaker.

[0011] Optionally, adjusting the initial character stickiness weighted score and the initial adaptive switching threshold based on the interaction behavior to obtain the target adaptive switching threshold and the target character stickiness weighted score includes: The interaction behavior is correlated with the speaker system operation log to obtain correlation behavior data; Based on the associated behavior data, user behavior pattern matching is performed to obtain user behavior pattern matching results; Based on the user behavior pattern matching results, user intent is inferred to obtain user intent information. Based on the user intent information, the initial role stickiness weighted score and the initial adaptive switching threshold are adjusted to obtain the target adaptive switching threshold and the target role stickiness weighted score.

[0012] In addition, the present invention also provides an audio adaptive adjustment control device, the device comprising: Master-slave speaker determination module: used to perform performance evaluation based on the processor operating load, power supply status, data transmission error rate and wireless link quality of each speaker, obtain the performance evaluation results of each speaker, and determine the master speaker and several slave speakers based on the performance evaluation results of each speaker; Audio playback module: used to form an interconnected speaker network between the main speaker and several slave speakers. The main speaker in the interconnected speaker network receives the audio stream and distributes the audio stream to each slave speaker for playback based on one-to-many broadcast technology. Detection signal transmission module: used to analyze environmental interference characteristics of the main speaker and each slave speaker during the playback of audio stream, determine signal transmission information based on the environmental interference characteristics, and transmit detection signals based on the signal transmission information; Audio adjustment module: used to analyze the impulse response based on the detection signal, determine the audio adjustment parameters based on the impulse response, and adjust the audio stream based on the audio adjustment parameters; Speaker switching module: During the playback of an audio stream, it acquires the initial adaptive switching threshold, cumulative auditory cost, and initial role stickiness weighted score of the main speaker, and controls the switching of the main speaker using interactive behavior based on the initial adaptive switching threshold, cumulative auditory cost, and initial role stickiness weighted score, combined with the performance evaluation results of each slave speaker.

[0013] In addition, the present invention also provides an electronic device, which includes a processor and a memory, wherein the memory is used to store instructions, and the processor is used to call the instructions in the memory to cause the electronic device to execute the above-described audio adaptive adjustment control method.

[0014] In addition, the present invention provides a computer-readable storage medium that stores computer instructions that, when executed on an electronic device, cause the electronic device to perform the above-described audio adaptive adjustment control method.

[0015] In this embodiment of the invention, performance evaluation is performed based on the processor load, power supply status, data transmission error rate, and wireless link quality of each speaker to determine the master speaker and several slave speakers. This solves the problem of arbitrary master speaker selection and system instability in traditional solutions. The master speaker and several slave speakers form an interconnected speaker network. The master speaker in the interconnected speaker network receives the audio stream and distributes it to each slave speaker for playback using one-to-many broadcast technology, ensuring audio synchronization in a multi-speaker environment. During audio stream playback, the master speaker and each slave speaker analyze environmental interference to determine signal transmission information. Based on this information, they transmit detection signals to avoid conflicts with environmental noise and ensure the effectiveness of the detection signals. The impulse response is analyzed based on the detection signals, and audio adjustment parameters are determined accordingly. The audio stream is then adjusted based on these parameters, overcoming the limitation of traditional speakers that cannot adaptively adjust sound quality according to environmental changes. During audio stream playback, the switching control of the main speaker is based on the initial adaptive switching threshold of the main speaker, the cumulative auditory cost, and the initial role stickiness weighted score, combined with the performance evaluation results of each slave speaker. This effectively solves the problems of inflexible speaker switching control and easy connection interruption or delay in existing technologies. It can significantly improve the stability of interconnected speaker networks, the adaptability of audio playback, and the smoothness of user experience, realizing intelligent management and optimized control of multi-speaker interconnection systems. Attached Figure Description

[0016] To more clearly illustrate the technical solutions in the embodiments of the present invention or the prior art, the drawings used in the description of the embodiments or the prior art will be briefly introduced below. Obviously, the drawings described below are only some embodiments of the present invention. For those skilled in the art, other drawings can be obtained based on these drawings without creative effort.

[0017] Figure 1 This is a flowchart illustrating the audio adaptive adjustment control method in an embodiment of the present invention; Figure 2 This is a flowchart illustrating an audio adaptive adjustment control method according to another embodiment of the present invention; Figure 3 This is a schematic diagram of the structural composition of the audio adaptive adjustment control device in an embodiment of the present invention; Figure 4 This is a schematic diagram of the structural composition of the electronic device in an embodiment of the present invention. Detailed Implementation

[0018] The technical solutions of the embodiments of the present invention will be clearly and completely described below with reference to the accompanying drawings. Obviously, the described embodiments are only some embodiments of the present invention, and not all embodiments. Based on the embodiments of the present invention, all other embodiments obtained by those skilled in the art without creative effort are within the scope of protection of the present invention.

[0019] Example 1 Please see Figure 1 , Figure 1 This is a flowchart illustrating the audio adaptive adjustment control method in an embodiment of the present invention, the method comprising: S11: Based on the processor operating load, power supply status, data transmission error rate and wireless link quality of each speaker, a performance evaluation is performed to obtain the performance evaluation results of each speaker, and the main speaker and several slave speakers are determined based on the performance evaluation results of each speaker. In the specific implementation of this invention, the processor load, power supply status, data transmission error rate, and wireless link quality of each speaker are obtained through the speaker parameter table and database. Performance evaluation is then performed based on these parameters to obtain the performance evaluation results for each speaker. Based on these results, a master speaker and several slave speakers are determined. The system intelligently selects the speaker with the best performance as the master speaker and assigns the remaining speakers as slave speakers. This solves the problem of arbitrary master speaker selection and system instability in traditional solutions, laying the foundation for the stable operation of the entire interconnected speaker network.

[0020] S12: The main speaker and several slave speakers form an interconnected speaker network. The main speaker in the interconnected speaker network receives the audio stream and distributes the audio stream to each slave speaker for playback based on one-to-many broadcast technology. In the specific implementation of this invention, the main speaker and several slave speakers form an interconnected speaker network. The main speaker in the interconnected speaker network receives the audio stream and distributes the audio stream to each slave speaker for playback based on one-to-many broadcast technology, ensuring audio synchronization in a multi-speaker environment and improving the overall listening experience.

[0021] S13: During the playback of the audio stream, the main speaker and each slave speaker analyze the environmental interference characteristics, determine the signal transmission information based on the environmental interference characteristics, and transmit detection signals based on the signal transmission information; In the specific implementation of this invention, during the playback of the audio stream, environmental acoustic data is acquired, and spectral analysis is performed on the environmental acoustic data based on Fast Fourier Transform to obtain spectral data. Based on a preset spectral energy threshold table and a periodic detector, the spectral data is used to determine environmental interference characteristics. Based on the environmental interference characteristics, the frequency band range and time window for signal transmission are determined to determine signal transmission information. Based on the signal transmission information, a detection signal is transmitted, which can intelligently determine the signal transmission information, avoid conflict with environmental noise, and ensure the effectiveness of the detection signal.

[0022] S14: Analyze the pulse response based on the detection signal, determine the audio adjustment parameters based on the pulse response, and adjust the audio stream based on the audio adjustment parameters; In the specific implementation of this invention, the impulse response is calculated based on the transmitted and received detection signal information using Fast Fourier Transform and Inverse Fast Fourier Transform. The reverberation time is determined based on the impulse response, and audio adjustment parameters are determined based on the reverberation time and environmental interference characteristics. The audio stream is then adjusted according to these parameters, compensating for acoustic defects in the environment and optimizing sound quality. This allows audio playback to adapt to different acoustic environments in real time, solving the problem of fixed audio adjustment parameters and difficulty in handling dynamic scenarios in traditional solutions.

[0023] S15: During the playback of the audio stream, obtain the initial adaptive switching threshold, cumulative auditory cost, and initial role stickiness weighted score of the main speaker, and use interactive behavior to control the switching of the main speaker based on the initial adaptive switching threshold, cumulative auditory cost, and initial role stickiness weighted score combined with the performance evaluation results of each slave speaker.

[0024] In the specific implementation of this invention, the fluctuation index of the current radio environment is calculated based on the Bluetooth data packet error rate and received signal strength indication of each slave speaker; the initial adaptive switching threshold is determined based on the fluctuation index; the cumulative auditory cost is determined based on the speaker switching frequency count of the master speaker; and the initial role stickiness weighting score is determined based on the master speaker's on-premises duration. The initial role stickiness weighting score and the initial adaptive switching threshold are adjusted based on the user's interaction behavior with the interconnected speaker network to obtain the target adaptive switching threshold and the target role stickiness weighting score. If the performance evaluation result of a slave speaker is greater than the sum of the master speaker's cumulative auditory cost, the target adaptive switching threshold, and the target role stickiness weighting score, then the slave speaker with the performance evaluation result greater than the sum is switched to the new master speaker. This means that the master speaker will only be switched when the performance of a slave speaker is significantly better than the current master speaker, and this advantage is sufficient to overcome the switching threshold adjusted by the user's interaction behavior (i.e., the sum of the cumulative auditory cost, the target adaptive switching threshold, and the target role stickiness weighting score), thus avoiding frequent or inappropriate switching due to slight performance fluctuations. This switching mechanism ensures that the best-performing speaker always serves as the main speaker throughout the entire playback process, thereby maximizing the stability, low latency, and high sound quality of audio transmission. It effectively solves the problem of overall system performance degradation caused by the performance degradation of the main speaker in traditional solutions.

[0025] In this embodiment of the invention, performance evaluation is performed based on the processor load, power supply status, data transmission error rate, and wireless link quality of each speaker to determine the master speaker and several slave speakers. This solves the problem of arbitrary master speaker selection and system instability in traditional solutions. The master speaker and several slave speakers form an interconnected speaker network. The master speaker in the interconnected speaker network receives the audio stream and distributes it to each slave speaker for playback using one-to-many broadcast technology, ensuring audio synchronization in a multi-speaker environment. During audio stream playback, the master speaker and each slave speaker analyze environmental interference to determine signal transmission information. Based on this information, they transmit detection signals to avoid conflicts with environmental noise and ensure the effectiveness of the detection signals. The impulse response is analyzed based on the detection signals, and audio adjustment parameters are determined accordingly. The audio stream is then adjusted based on these parameters, overcoming the limitation of traditional speakers that cannot adaptively adjust sound quality according to environmental changes. During audio stream playback, the switching control of the main speaker is based on the initial adaptive switching threshold of the main speaker, the cumulative auditory cost, and the initial role stickiness weighted score, combined with the performance evaluation results of each slave speaker. This effectively solves the problems of inflexible speaker switching control and easy connection interruption or delay in existing technologies. It can significantly improve the stability of interconnected speaker networks, the adaptability of audio playback, and the smoothness of user experience, realizing intelligent management and optimized control of multi-speaker interconnection systems.

[0026] Example 2 Please see Figure 2 , Figure 2 This is a flowchart illustrating an audio adaptive adjustment control method according to another embodiment of the present invention, the method comprising: S201: Based on the processor operating load, power supply status, data transmission error rate and wireless link quality of each speaker, a performance evaluation is performed to obtain the performance evaluation results of each speaker, and the main speaker and several slave speakers are determined based on the performance evaluation results of each speaker. In the specific implementation of this invention, processor load refers to the proportion of computing resources occupied by the speaker's internal processor when performing various tasks, reflecting the processor's workload. Power supply status refers to the speaker's battery level or external power connection status, affecting the speaker's continuous working capability. Data transmission error rate refers to the proportion of data packet transmission failures during wireless communication, reflecting the reliability of the wireless link. Wireless link quality refers to a comprehensive indicator of the stability, bandwidth, and signal strength of the wireless connection between speakers. The evaluation results of these performance indicators will directly affect the selection of the main speaker and the overall performance of the system.

[0027] The processor load can be obtained by querying the CPU utilization of the speaker's operating system; the power supply status can be obtained by reading battery level or power connection status sensor data; the data transmission error rate can be obtained by counting the number of error packets reported by the wireless communication module; and the wireless link quality can be obtained by measuring indicators such as received signal strength and signal-to-noise ratio. Based on this raw data, methods such as weighted averaging, fuzzy logic, or machine learning models can be used to comprehensively calculate the overall performance score of each speaker, which is the performance evaluation result. The speaker with the highest performance evaluation result is selected as the master speaker, and the rest are slave speakers. This step solves the problem of arbitrary master speaker selection in traditional solutions, which can easily lead to system instability, and lays the foundation for the stable operation of the entire interconnected speaker network.

[0028] S202: The main speaker and several slave speakers form an interconnected speaker network. The main speaker in the interconnected speaker network receives the audio stream and distributes the audio stream to each slave speaker for playback based on one-to-many broadcast technology. In the specific implementation of this invention, a main speaker and several slave speakers form an interconnected speaker network. This interconnected speaker network refers to a wireless audio transmission and playback system consisting of a main speaker and several slave speakers. The main speaker is responsible for receiving the original audio stream and distributing it to the slave speakers, while the slave speakers are responsible for synchronously playing the received audio stream. One-to-many broadcast technology means that the main speaker can simultaneously send the same audio stream data to multiple slave speakers, ensuring synchronized audio playback. For example, the main speaker can encapsulate the audio stream into data packets and send them simultaneously to all slave speakers via a wireless network. The slave speakers then decode and play the data packets. This ensures audio synchronization in a multi-speaker environment and improves the overall listening experience.

[0029] S203: During the playback of the audio stream, the main speaker and each slave speaker analyze the environmental interference characteristics, determine the signal transmission information based on the environmental interference characteristics, and transmit a detection signal based on the signal transmission information; In the specific implementation of this invention, the step of analyzing environmental interference characteristics and determining signal transmission information based on the environmental interference characteristics includes: acquiring environmental acoustic data and performing spectral analysis on the environmental acoustic data based on Fast Fourier Transform to obtain spectral data; determining environmental interference characteristics using the spectral data based on a preset spectral energy threshold table and a periodic detector; determining the frequency band range and time window for signal transmission based on the environmental interference characteristics, and determining signal transmission information based on the frequency band range and time window.

[0030] Specifically, during audio stream playback, ambient acoustic data is acquired. Ambient acoustic data refers to sound information collected from the environment in which the speaker is located, which may include background noise, human voices, music, or other ambient sounds. Ambient acoustic data can be acquired in real time through the speaker's built-in microphone or other acoustic sensors. Spectral analysis is then performed on the ambient acoustic data using Fast Fourier Transform (FFT), an algorithm that converts time-domain signals into frequency-domain signals. By performing spectral analysis on the ambient acoustic data, the energy distribution of different frequency components can be obtained, i.e., spectral data. Spectral data can intuitively reflect the frequency characteristics and intensity of various sound interferences present in the environment.

[0031] Environmental interference characteristics are determined using the spectrum data based on a preset spectrum energy threshold table and a periodic detector. The preset spectrum energy threshold table can store energy thresholds for different frequency bands in advance to distinguish between normal environmental noise and potential interference signals. The periodic detector is used to identify whether there are interference signals with periodic characteristics in the spectrum data, such as periodic noise generated by electrical equipment or mechanical vibration. By combining the preset spectrum energy threshold table and the periodic detector to analyze the spectrum data, environmental interference characteristics, such as the frequency range, intensity, and periodicity of the interference, can be accurately identified and determined.

[0032] The frequency band range and time window for signal transmission are determined based on the environmental interference characteristics, and the signal transmission information is determined based on the frequency band range and time window. For example, if there is strong interference in a certain frequency band in the environment, the detection signal can be avoided in that frequency band, or the transmission can be carried out in a time window with weaker interference. The signal transmission information may also include parameters such as the transmission frequency, transmission power, transmission duration, and transmission timing of the detection signal. The determination of these parameters aims to ensure that the detection signal can be effectively propagated and received in a complex environment, while minimizing its interference with the audio stream playback.

[0033] A probe signal is an audio signal used to measure the characteristics of an acoustic environment. By transmitting the probe signal according to the signal transmission information, it is possible to avoid transmitting the probe signal in a strong interference frequency band or time period, effectively reducing the risk of the probe signal being masked by environmental interference. This ensures the effectiveness and accuracy of the probe signal, laying the foundation for subsequent impulse response analysis and determination of audio adjustment parameters.

[0034] S204: Analyze the impulse response based on the detection signal, determine the audio adjustment parameters based on the impulse response, and adjust the audio stream based on the audio adjustment parameters; In a specific implementation of the present invention, the step of analyzing the impulse response based on the detection signal and determining the audio adjustment parameters based on the impulse response includes: calculating the impulse response based on the transmitted and received detection signal information by combining Fast Fourier Transform and Inverse Fast Fourier Transform; determining the reverberation time based on the impulse response; and determining the audio adjustment parameters based on the reverberation time and environmental interference characteristics.

[0035] Specifically, the impulse response is calculated based on the transmitted and received probe signal information, using Fast Fourier Transform (FFT) and Inverse Fast Fourier Transform (IFT). The transmitted probe signal information refers to the raw data or characteristic parameters of the probe signals emitted by the main speaker and each slave speaker according to the signal transmission information, such as the signal waveform, frequency, amplitude, and transmission time. The received probe signal information refers to the probe signal data received by the microphones inside or outside the speakers, after propagating through the environmental acoustic path. These received signals include the influence of the environment on the probe signals, such as reflection, absorption, and diffraction. The IFT is the inverse operation of the FFT, used to convert the frequency domain signal back to the time domain signal. By processing the transmitted and received probe signal information in the frequency domain and then using the IFT, the impulse response can be calculated. The impulse response describes the acoustic space's response to an instantaneous sound source, including the room's acoustic characteristics such as reflection and reverberation.

[0036] The reverberation time is determined based on the impulse response. Reverberation time refers to the time required for the sound energy density in a space to decrease by 60 dB after the sound source stops emitting sound; it is an important parameter for measuring the reverberation level of a room. Reverberation time can be calculated by analyzing the decay curve of the impulse response, such as calculating the time required for the energy of the impulse response to decay to one millionth of its initial energy. Audio adjustment parameters are determined based on the reverberation time and environmental interference characteristics. These parameters are used to adjust the audio stream, and are various parameters used to optimize the playback effect of the audio stream, such as equalizer settings, volume gain, delay dynamic range compression, etc. By combining reverberation time (reflecting the inherent acoustic characteristics of the room) and environmental interference characteristics (reflecting real-time external interference), the current acoustic environment can be assessed more comprehensively and accurately, thereby determining audio adjustment parameters that can effectively counteract interference, optimize sound quality, and improve listening comfort. This achieves refined and adaptive adjustment of the audio stream.

[0037] S205: During the playback of the audio stream, obtain the initial adaptive switching threshold, cumulative auditory cost, and initial role stickiness weighted score of the main speaker; In a specific implementation of this invention, obtaining the initial adaptive switching threshold, cumulative auditory cost, and initial role stickiness weighting score of the main speaker includes: calculating the fluctuation index of the current radio environment based on the Bluetooth data packet error rate and received signal strength indication of each slave speaker; determining the initial adaptive switching threshold based on the fluctuation index; obtaining the speaker switching frequency count of the main speaker and determining the cumulative auditory cost based on the speaker switching frequency count; and determining the initial role stickiness weighting score based on the duration of the main speaker's presence.

[0038] Specifically, during audio stream playback, a fluctuation index for the current radio environment is calculated based on the Bluetooth packet error rate and received signal strength indicator (RSS) of each speaker. The Bluetooth packet error rate reflects the reliability of data transmission, while the RSS indicates the signal strength. By comprehensively analyzing these indicators, a fluctuation index can be calculated; the higher the index, the more unstable the radio environment.

[0039] Determining the initial adaptive switching threshold based on the aforementioned fluctuation index refers to dynamically setting an initial judgment criterion for main speaker switching based on the calculated radio environment fluctuation index. For example, when the radio environment fluctuation index is high, it indicates that the communication link may be unstable. In this case, the initial adaptive switching threshold can be set lower so that switching is triggered when performance degrades slightly, thereby improving the system's response speed and stability. Conversely, when the fluctuation index is low, the threshold can be appropriately increased to avoid unnecessary frequent switching.

[0040] Obtain the speaker switching frequency count of the main speaker. The speaker switching frequency count refers to the number of times the main speaker switches roles within a certain period of time. This is used to assess the potential impact on the user's auditory experience. Based on the speaker switching frequency count, determine the cumulative auditory cost. This can be done by quantifying a coefficient that affects the cumulative auditory discomfort of each switch and calculating this coefficient with the speaker switching frequency count. The cumulative auditory cost aims to quantify the negative impact of frequent switching on the user experience so that it can be taken into account in the switching decision.

[0041] The initial role stickiness weighting score, determined by the duration of the main speaker's presence, assigns an initial stickiness weight based on the length of time the main speaker has continuously assumed its main speaker role. A longer presence generally indicates more stable operation of the main speaker, resulting in a higher initial role stickiness weighting score. This weighting score aims to increase the likelihood of a currently stable main speaker maintaining its role, thereby reducing system instability and preventing role switching without sufficient justification.

[0042] S206: Obtain the user's interaction behavior with the interconnected speaker network, and adjust the initial role stickiness weighted score and the initial adaptive switching threshold based on the interaction behavior to obtain the target adaptive switching threshold and the target role stickiness weighted score. In a specific implementation of this invention, adjusting the initial role stickiness weighted score and the initial adaptive switching threshold based on the interaction behavior to obtain the target adaptive switching threshold and the target role stickiness weighted score includes: performing correlation analysis between the interaction behavior and the speaker system operation log to obtain correlated behavior data; performing user behavior pattern matching based on the correlated behavior data to obtain user behavior pattern matching results; performing user intent inference based on the user behavior pattern matching results to obtain user intent information; and adjusting the initial role stickiness weighted score and the initial adaptive switching threshold based on the user intent information to obtain the target adaptive switching threshold and the target role stickiness weighted score.

[0043] Specifically, the system acquires user interaction behavior with the interconnected speaker network. Interaction behavior refers to the various operations and feedback performed by the user and the speaker network, collected and analyzed by the system. These interactions can include, but are not limited to, user operations on the speakers via voice commands, touch panels, mobile applications, etc., such as volume adjustment, speaker grouping / disgrouping, and manual switching of master / slave speaker roles. Furthermore, physical user actions, such as moving or approaching a speaker, can also be considered interaction behavior. The interaction behavior is then correlated with the speaker system's operational logs to obtain correlated behavior data. The speaker system's operational logs record various operational states, events, performance data, and historical operation records of the speaker network. The purpose of this correlation analysis is to integrate the user's specific interaction behavior with the system's internal state, environmental parameters, and subsequent system responses at the time of the interaction, thereby forming more comprehensive and contextualized correlated behavior data, providing a rich data foundation for subsequent user behavior pattern matching.

[0044] User behavior pattern matching is performed based on the associated behavioral data to obtain user behavior pattern matching results. This involves comparing the associated behavioral data with predefined or machine learning-trained typical user behavior patterns. These patterns may include user preferences for speaker switching, sound quality adjustment habits, etc. For example, if a user frequently switches the main speaker manually in a specific scenario, a scenario switching preference pattern may form. The matching results can indicate which known pattern the current user behavior is most similar to, or identify new, unpredictable patterns.

[0045] User intent inference is performed based on the user behavior pattern matching results to obtain user intent information. User intent inference further analyzes the underlying motivations and expectations behind user behavior, building upon the behavior pattern matching results. For example, if the matching results show that the user frequently manually switches the main speaker when speaker performance degrades, it can be inferred that the user's intent is to hope that the system can switch the main speaker more promptly and intelligently to ensure a good listening experience. User intent information may include the user's expectations for switching sensitivity and preferences for speaker role stickiness. Based on this user intent information, the initial role stickiness weighted score and the initial adaptive switching threshold are adjusted to obtain the target adaptive switching threshold and the target role stickiness weighted score. The initial role stickiness weighted score reflects the main speaker's position or inertia in the switching decision; higher stickiness means the main speaker is less likely to be switched. The initial adaptive switching threshold determines the tolerance for performance differences that trigger main speaker switching. Based on the inferred user intent information, these two initial parameters can be dynamically adjusted. For example, if a user intends to show high sensitivity to switching and wants the system to more actively switch the main speaker to optimize the experience, the target adaptive switching threshold can be lowered, and the target role stickiness weighting score can be reduced, making it easier for the system to trigger a switch. Conversely, if the user prefers to maintain the stability of the main speaker, these parameters can be increased. This incorporates the user's personalized needs, achieving a more intelligent and human-centered switching decision.

[0046] S207: If the performance evaluation result of a secondary speaker is greater than the sum of the cumulative auditory cost of the primary speaker, the target adaptive switching threshold, and the target role stickiness weighted score, then the secondary speaker with a performance evaluation result greater than the sum will be switched to the new primary speaker.

[0047] In a specific implementation of this invention, the step of switching a secondary speaker with a performance evaluation result greater than the sum of its values ​​to a new primary speaker includes: identifying the audio content type of the audio stream; performing sensitivity analysis on the primary speaker switching based on the audio content type to obtain target sensitivity information, and determining the speaker switching time period based on the target sensitivity information; acquiring the audio content characteristics of the audio stream, and determining the audio buffer information and audio synchronization information for primary speaker switching based on the audio content characteristics; and switching the secondary speaker with a performance evaluation result greater than the sum of its values ​​to a new primary speaker based on the speaker switching time period, audio buffer information, and audio synchronization information.

[0048] Specifically, identifying the audio content type of the audio stream involves analyzing the currently playing audio stream to determine its category, such as music, speech, ambient sound, or silent segments. This step aims to provide contextual information for subsequent switching decisions, as different types of audio content have varying sensitivities to switching; for example, switching while playing music may be more noticeable to the user than switching while playing speech.

[0049] Sensitivity analysis of main speaker switching is performed based on the audio content type to obtain target sensitivity information. This target sensitivity information assesses the user's perception of main speaker switching based on the audio content type, reflecting the appropriateness of switching under the current audio content. For example, switching during a musical climax or a key moment in a voice conversation elicits higher user sensitivity; while switching during a musical interlude, a quiet segment, or background sound effects elicits lower user sensitivity. Based on this target sensitivity information, speaker switching time periods are determined. This allows for the selection of one or more speaker switching time periods within the audio stream, meaning switching at the point in the audio stream with the least impact on the user's hearing. For example, switching can be performed during quiet segments, low-volume sections, or segments with minimal rhythmic changes.

[0050] The audio content characteristics of the audio stream are obtained. These characteristics refer to analyzing the technical parameters of the audio stream, such as its encoding format, sampling rate, bit rate, and number of channels, as well as its real-time latency and jitter during playback. Based on these audio content characteristics, audio buffering information and audio synchronization information for switching the main speaker are determined. The audio buffering information ensures sufficient audio data reserves during the switching process to avoid playback stuttering caused by data transmission delays or interruptions. The audio synchronization information ensures that the new main speaker and the slave speaker can play in precise synchronization after switching, avoiding echoes or desynchronization.

[0051] When the performance evaluation result of a secondary speaker exceeds the sum of the cumulative auditory cost of the primary speaker, the target adaptive switching threshold, and the target role stickiness weighted score, the secondary speaker with the performance evaluation result exceeding the sum is switched to the new primary speaker based on the speaker switching period, audio buffer information, and audio synchronization information. Switching during periods of silence or low volume in the audio stream effectively avoids abrupt transitions. By acquiring the audio content characteristics of the audio stream and determining the audio buffer and synchronization information accordingly, the continuity of audio data and precise synchronization of playback are ensured during switching. Therefore, even when the primary speaker role changes, the interconnected speaker network can maintain high-quality audio output, avoiding audio interruptions or synchronization issues that may occur with traditional switching methods.

[0052] It should be noted that the identification of the audio content type of the audio stream includes: performing acoustic feature analysis on the audio stream to obtain time-domain features and frequency-domain features; calculating the degree of matching between the time-domain features and frequency-domain features and feature information in a preset audio content feature template; and determining the audio content type of the audio stream based on the degree of matching and the dynamic change information of the audio stream.

[0053] Specifically, acoustic feature analysis is performed on the audio stream to obtain time-domain and frequency-domain features. Time-domain features may include, but are not limited to, short-time energy, zero-crossing rate, fundamental frequency, loudness, etc., which mainly reflect the variation law of the audio signal in the time dimension. Frequency-domain features may include, but are not limited to, Mel-frequency cepstral coefficients, spectral centroid, spectral bandwidth, spectral roll-off, chromaticity features, etc., which mainly reflect the energy distribution and timbre characteristics of the audio signal in the frequency dimension. The extraction of these features can be achieved through digital signal processing techniques, such as Fourier transform, wavelet transform, etc.

[0054] The degree of matching between the time-domain and frequency-domain features and the feature information in the preset audio content feature templates is calculated. The preset audio content feature templates are pre-stored sets of typical acoustic features corresponding to different audio content types. These templates can be obtained by training and learning on a large amount of audio data of known types. The degree of matching can be calculated using various similarity measurement methods, such as Euclidean distance and cosine similarity, to quantify the similarity between the features of the audio stream to be analyzed and the features of each preset template.

[0055] The audio content type of the audio stream is determined based on the matching degree combined with the dynamic change information of the audio stream. Dynamic change information refers to the trend, rhythm, and volume fluctuations of the audio stream's acoustic characteristics over time during playback. Combining the matching degree with the dynamic change information of the audio stream allows for a more comprehensive and accurate assessment of the true content type. For example, even if the static features of an audio segment have a high matching degree with a speech template, if its dynamic change information shows a strong musical rhythm, it may be more accurately identified as music with lyrics rather than pure speech. This ensures the accuracy and robustness of audio content type identification, providing reliable basic data for subsequent main speaker switching sensitivity analysis.

[0056] It should be noted that the sensitivity analysis of main speaker switching based on the audio content type to obtain target sensitivity information includes: determining initial sensitivity information for main speaker switching based on the audio content type; acquiring user identity information, environment information, historical interaction behavior, preset user preferences, and environmental acoustic data; determining sensitivity weights based on the identity information, environment information, historical interaction behavior, preset user preferences, and environmental acoustic data; and adjusting the initial sensitivity information based on the sensitivity weights to obtain target sensitivity information.

[0057] Specifically, the initial sensitivity information for switching the main speaker is determined based on the type of audio content. The initial sensitivity information refers to a basic switching sensitivity value preset according to the type of currently playing audio. For example, for music that requires a high degree of immersion, the initial sensitivity may be set to a lower value to avoid frequent switching that would interrupt the user experience. For background music, the initial sensitivity may be set to a higher value to allow for more flexible switching.

[0058] The system acquires user identity information, environmental information, historical interaction behavior, preset user preferences, and environmental acoustic data. User identity information may include user ID, login status, etc., used to identify the user; environmental information may include the room type, space size, current time period, etc., where the speaker is located; historical interaction behavior refers to the record of the user's past operations such as switching speakers, adjusting volume, and selecting playback content; preset user preferences refer to the user's explicit settings in the system settings regarding speaker switching tendencies or audio type preferences; and environmental acoustic data can be collected in real time through the speaker's built-in microphone to analyze acoustic characteristics such as environmental noise and reverberation.

[0059] Sensitivity weights are determined based on the aforementioned identity information, environmental information, historical interaction behavior, preset user preferences, and environmental acoustic data. Specifically, machine learning models or preset rules can be used to comprehensively analyze the above multi-dimensional data and calculate a weight value for adjusting the initial sensitivity. For example, if it is identified that the current user has extremely high requirements for sound quality and is in a quiet listening environment, the sensitivity weight may tend to decrease the switching sensitivity to maintain sound quality stability; if the user frequently listens to podcasts while moving, the sensitivity weight may tend to increase the switching sensitivity to adapt to environmental changes.

[0060] The initial sensitivity information is adjusted based on the aforementioned sensitivity weights to obtain target sensitivity information. The initial sensitivity information is then combined with the calculated sensitivity weights, such as through multiplication or weighted averaging, to arrive at a final sensitivity value that better reflects the current user, environment, and preferences. The aim is to make the main speaker's switching decisions more intelligent and personalized. Correcting the initial sensitivity information determined based on audio content type avoids the problem of overly rigid switching decisions or decisions that don't meet actual user needs, which might result from relying solely on audio content type.

[0061] In this embodiment of the invention, performance evaluation is performed based on the processor load, power supply status, data transmission error rate, and wireless link quality of each speaker to determine the master speaker and several slave speakers. This solves the problem of arbitrary master speaker selection and system instability in traditional solutions. The master speaker and several slave speakers form an interconnected speaker network. The master speaker in the interconnected speaker network receives the audio stream and distributes it to each slave speaker for playback using one-to-many broadcast technology, ensuring audio synchronization in a multi-speaker environment. During audio stream playback, the master speaker and each slave speaker analyze environmental interference to determine signal transmission information. Based on this information, they transmit detection signals to avoid conflicts with environmental noise and ensure the effectiveness of the detection signals. The impulse response is analyzed based on the detection signals, and audio adjustment parameters are determined accordingly. The audio stream is then adjusted based on these parameters, overcoming the limitation of traditional speakers that cannot adaptively adjust sound quality according to environmental changes. During audio stream playback, the switching control of the main speaker is based on the initial adaptive switching threshold of the main speaker, the cumulative auditory cost, and the initial role stickiness weighted score, combined with the performance evaluation results of each slave speaker. This effectively solves the problems of inflexible speaker switching control and easy connection interruption or delay in existing technologies. It can significantly improve the stability of interconnected speaker networks, the adaptability of audio playback, and the smoothness of user experience, realizing intelligent management and optimized control of multi-speaker interconnection systems.

[0062] Example 3 Please see Figure 3 , Figure 3 This is a schematic diagram of the structural composition of the audio adaptive adjustment control device in an embodiment of the present invention. The device includes: Master-slave speaker determination module 31: used to perform performance evaluation based on the processor operating load, power supply status, data transmission error rate and wireless link quality of each speaker, obtain the performance evaluation results of each speaker, and determine the master speaker and several slave speakers based on the performance evaluation results of each speaker. Audio playback module 32: used to form an interconnected speaker network between the main speaker and several slave speakers, wherein the main speaker in the interconnected speaker network receives the audio stream and distributes the audio stream to each slave speaker for playback based on one-to-many broadcast technology; Detection signal transmission module 33: During the playback of an audio stream, the main speaker and each of the slave speakers analyze environmental interference characteristics, determine signal transmission information based on the environmental interference characteristics, and transmit detection signals based on the signal transmission information; Audio adjustment module 34: used to analyze the pulse response based on the detection signal, determine the audio adjustment parameters based on the pulse response, and adjust the audio stream based on the audio adjustment parameters; Speaker switching module 35: During the playback of an audio stream, it acquires the initial adaptive switching threshold, cumulative auditory cost, and initial role stickiness weighted score of the main speaker, and controls the switching of the main speaker using interactive behavior based on the initial adaptive switching threshold, cumulative auditory cost, and initial role stickiness weighted score combined with the performance evaluation results of each slave speaker.

[0063] In the specific implementation of this invention, the specific implementation of the device item can be referred to the implementation of the method item above, and will not be repeated here.

[0064] In this embodiment of the invention, performance evaluation is performed based on the processor load, power supply status, data transmission error rate, and wireless link quality of each speaker to determine the master speaker and several slave speakers. This solves the problem of arbitrary master speaker selection and system instability in traditional solutions. The master speaker and several slave speakers form an interconnected speaker network. The master speaker in the interconnected speaker network receives the audio stream and distributes it to each slave speaker for playback using one-to-many broadcast technology, ensuring audio synchronization in a multi-speaker environment. During audio stream playback, the master speaker and each slave speaker analyze environmental interference to determine signal transmission information. Based on this information, they transmit detection signals to avoid conflicts with environmental noise and ensure the effectiveness of the detection signals. The impulse response is analyzed based on the detection signals, and audio adjustment parameters are determined accordingly. The audio stream is then adjusted based on these parameters, overcoming the limitation of traditional speakers that cannot adaptively adjust sound quality according to environmental changes. During audio stream playback, the switching control of the main speaker is based on the initial adaptive switching threshold of the main speaker, the cumulative auditory cost, and the initial role stickiness weighted score, combined with the performance evaluation results of each slave speaker. This effectively solves the problems of inflexible speaker switching control and easy connection interruption or delay in existing technologies. It can significantly improve the stability of interconnected speaker networks, the adaptability of audio playback, and the smoothness of user experience, realizing intelligent management and optimized control of multi-speaker interconnection systems.

[0065] This invention provides a computer-readable storage medium storing a computer program. When executed by a processor, this program implements the audio adaptive adjustment control method of any of the above embodiments. The computer-readable storage medium includes, but is not limited to, any type of disk (including floppy disk, hard disk, optical disk, CD-ROM, and magneto-optical disk), ROM (Read-Only Memory), RAM (Random Access Memory), EPROM (Erasable Programmable Read-Only Memory), EEPROM (Electrically Erasable Programmable Read-Only Memory), flash memory, magnetic cards, or optical cards. In other words, the storage device includes any medium that stores or transmits information in a readable form by a device (e.g., a computer, a mobile phone), and can be a read-only memory, a disk, or an optical disk, etc.

[0066] Example 4 Please see Figure 4 , Figure 4 This is a schematic diagram of the structural composition of the electronic device in an embodiment of the present invention.

[0067] This invention also provides an electronic device, such as... Figure 4 As shown, the electronic device includes a memory 41, a processor 43, and a computer program 42 stored in the memory 41 and executable on the processor 43. Those skilled in the art will understand that... Figure 4The illustrated electronic device does not constitute a limitation on all devices and may include more or fewer components than illustrated, or combine certain components. Memory 41 can be used to store computer program 42 and various functional modules. Processor 43 runs the computer program 42 stored in memory 41, thereby performing various functional applications and data processing of the device. Memory can be internal memory or external memory, or both. Internal memory may include read-only memory (ROM), programmable ROM (PROM), electrically programmable ROM (EPROM), electrically erasable programmable ROM (EEPROM), flash memory, or random access memory. External memory may include hard disks, floppy disks, ZIP disks, USB flash drives, magnetic tapes, etc. Processor 43 may be a central processing unit (CPU), or other general-purpose processors, digital signal processors (DSPs), application-specific integrated circuits (ASICs), field-programmable gate arrays (FPGAs), or other programmable logic devices, discrete gate or transistor logic devices, discrete hardware components, etc. The general-purpose processor can be a microprocessor, a single-chip microcomputer, or a processor 43, or any conventional processor, etc. The processors and memories disclosed in this invention include, but are not limited to, these types of processors and memories. The processors and memories disclosed in this invention are merely examples and not intended to be limiting.

[0068] As one embodiment, the electronic device includes: one or more processors 43, a memory 41, and one or more computer programs 42, wherein the one or more computer programs 42 are stored in the memory 41 and configured to be executed by the one or more processors 43, and the one or more computer programs 42 are configured to perform the audio adaptive adjustment control method in any of the above embodiments. For specific implementation details, please refer to the above embodiments, which will not be repeated here.

[0069] In this embodiment of the invention, performance evaluation is performed based on the processor load, power supply status, data transmission error rate, and wireless link quality of each speaker to determine the master speaker and several slave speakers. This solves the problem of arbitrary master speaker selection and system instability in traditional solutions. The master speaker and several slave speakers form an interconnected speaker network. The master speaker in the interconnected speaker network receives the audio stream and distributes it to each slave speaker for playback using one-to-many broadcast technology, ensuring audio synchronization in a multi-speaker environment. During audio stream playback, the master speaker and each slave speaker analyze environmental interference to determine signal transmission information. Based on this information, they transmit detection signals to avoid conflicts with environmental noise and ensure the effectiveness of the detection signals. The impulse response is analyzed based on the detection signals, and audio adjustment parameters are determined accordingly. The audio stream is then adjusted based on these parameters, overcoming the limitation of traditional speakers that cannot adaptively adjust sound quality according to environmental changes. During audio stream playback, the switching control of the main speaker is based on the initial adaptive switching threshold of the main speaker, the cumulative auditory cost, and the initial role stickiness weighted score, combined with the performance evaluation results of each slave speaker. This effectively solves the problems of inflexible speaker switching control and easy connection interruption or delay in existing technologies. It can significantly improve the stability of interconnected speaker networks, the adaptability of audio playback, and the smoothness of user experience, realizing intelligent management and optimized control of multi-speaker interconnection systems.

[0070] Furthermore, the above provides a detailed description of the audio adaptive adjustment control method and related apparatus provided by the embodiments of the present invention. Specific examples have been used to illustrate the principles and implementation methods of the present invention. The description of the above embodiments is only for the purpose of helping to understand the method and core ideas of the present invention. At the same time, for those skilled in the art, there will be changes in the specific implementation methods and application scope based on the ideas of the present invention. Therefore, the content of this specification should not be construed as a limitation of the present invention.

Claims

1. An audio adaptive adjustment control method, characterized in that, The method includes: Performance evaluations are conducted based on the processor load, power supply status, data transmission error rate, and wireless link quality of each speaker to obtain the performance evaluation results of each speaker. Based on the performance evaluation results of each speaker, the main speaker and several slave speakers are determined. The main speaker and several slave speakers form an interconnected speaker network. The main speaker in the interconnected speaker network receives the audio stream and distributes the audio stream to each slave speaker for playback based on one-to-many broadcast technology. During the playback of the audio stream, the main speaker and each of the slave speakers analyze the environmental interference characteristics, determine the signal transmission information based on the environmental interference characteristics, and transmit detection signals based on the signal transmission information; Based on the detection signal, the impulse response is analyzed, and based on the impulse response, audio adjustment parameters are determined. Based on the audio adjustment parameters, the audio stream is adjusted. During the playback of the audio stream, the initial adaptive switching threshold, cumulative auditory cost, and initial role stickiness weighted score of the main speaker are obtained. Based on the initial adaptive switching threshold, cumulative auditory cost, and initial role stickiness weighted score, combined with the performance evaluation results of each slave speaker, the switching control of the main speaker is performed using interactive behavior.

2. The audio adaptive adjustment control method according to claim 1, characterized in that, The analysis of environmental interference characteristics and the determination of signal transmission information based on these characteristics include: Acquire environmental acoustic data and perform spectral analysis on the environmental acoustic data based on fast Fourier transform to obtain spectral data; Based on a preset spectrum energy threshold table and a periodic detector, the spectrum data is used to determine the characteristics of environmental interference. The frequency band range and time window for signal transmission are determined based on the environmental interference characteristics, and the signal transmission information is determined based on the frequency band range and time window.

3. The audio adaptive adjustment control method according to claim 1, characterized in that, The step of analyzing the impulse response based on the detected signal and determining the audio adjustment parameters based on the impulse response includes: The impulse response is calculated by combining the transmitted and received probe signal information with the Fast Fourier Transform and Inverse Fast Fourier Transform. The reverberation time is determined based on the impulse response, and the audio adjustment parameters are determined based on the reverberation time and environmental interference characteristics.

4. The audio adaptive adjustment control method according to claim 1, characterized in that, The acquisition of the initial adaptive switching threshold, cumulative auditory cost, and initial role stickiness weighted score of the main speaker includes: The fluctuation index of the current radio environment is calculated based on the Bluetooth packet error rate and received signal strength indication of each speaker. The initial adaptive switching threshold is determined based on the volatility index; Obtain the speaker switching frequency count of the main speaker, and determine the cumulative auditory cost based on the speaker switching frequency count; The initial role stickiness weighting score is determined based on the duration of the main speaker's presence.

5. The audio adaptive adjustment control method according to claim 1, characterized in that, The method of controlling the switching of the main speaker based on the initial adaptive switching threshold, cumulative auditory cost, and initial role stickiness weighted score, combined with the performance evaluation results of each slave speaker, and utilizing interactive behavior includes: The system acquires user interaction behavior with the interconnected speaker network, and adjusts the initial role stickiness weighted score and the initial adaptive switching threshold based on the interaction behavior to obtain the target adaptive switching threshold and the target role stickiness weighted score. If the performance evaluation result of a secondary speaker is greater than the sum of the cumulative auditory cost of the primary speaker, the target adaptive switching threshold, and the target role stickiness weighted score, then the secondary speaker with a performance evaluation result greater than the sum will be switched to the new primary speaker.

6. The audio adaptive adjustment control method according to claim 5, characterized in that, The step of switching the secondary speaker whose performance evaluation result is greater than the sum value to the new primary speaker includes: Identify the audio content type of the audio stream; Sensitivity analysis of main speaker switching is performed based on the audio content type to obtain target sensitivity information, and speaker switching time period is determined based on the target sensitivity information; The audio content characteristics of the audio stream are obtained, and the audio buffer information and audio synchronization information for main speaker switching are determined based on the audio content characteristics. Based on the speaker switching period, audio buffer information, and audio synchronization information, the speaker with a performance evaluation result greater than the sum of the values ​​will be switched to the new main speaker.

7. The audio adaptive adjustment control method according to claim 5, characterized in that, The step of adjusting the initial character stickiness weighted score and the initial adaptive switching threshold based on the interaction behavior to obtain the target adaptive switching threshold and the target character stickiness weighted score includes: The interaction behavior is correlated with the speaker system operation log to obtain correlation behavior data; Based on the associated behavior data, user behavior pattern matching is performed to obtain user behavior pattern matching results; Based on the user behavior pattern matching results, user intent is inferred to obtain user intent information. Based on the user intent information, the initial role stickiness weighted score and the initial adaptive switching threshold are adjusted to obtain the target adaptive switching threshold and the target role stickiness weighted score.

8. An audio adaptive adjustment control device, characterized in that, The device includes: Master-slave speaker determination module: used to perform performance evaluation based on the processor operating load, power supply status, data transmission error rate and wireless link quality of each speaker, obtain the performance evaluation results of each speaker, and determine the master speaker and several slave speakers based on the performance evaluation results of each speaker; Audio playback module: used to form an interconnected speaker network between the main speaker and several slave speakers. The main speaker in the interconnected speaker network receives the audio stream and distributes the audio stream to each slave speaker for playback based on one-to-many broadcast technology. Detection signal transmission module: used to analyze environmental interference characteristics of the main speaker and each slave speaker during the playback of audio stream, determine signal transmission information based on the environmental interference characteristics, and transmit detection signals based on the signal transmission information; Audio adjustment module: used to analyze the impulse response based on the detection signal, determine the audio adjustment parameters based on the impulse response, and adjust the audio stream based on the audio adjustment parameters; Speaker switching module: During the playback of an audio stream, it acquires the initial adaptive switching threshold, cumulative auditory cost, and initial role stickiness weighted score of the main speaker, and controls the switching of the main speaker using interactive behavior based on the initial adaptive switching threshold, cumulative auditory cost, and initial role stickiness weighted score, combined with the performance evaluation results of each slave speaker.

9. An electronic device, the electronic device comprising a processor and a memory, characterized in that, The memory is used to store instructions, and the processor is used to call the instructions in the memory to cause the electronic device to execute the audio adaptive adjustment control method as described in any one of claims 1 to 7.

10. A computer-readable storage medium, characterized in that, The computer-readable storage medium stores computer instructions that, when executed on an electronic device, cause the electronic device to perform the audio adaptive adjustment control method as described in any one of claims 1 to 7.

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