Automatic audio playing recovery method and device, equipment and storage medium

Abstract

The invention discloses a method for automatically recovering audio playing, which comprises the following steps: in an audio playing process, playing state information of a system is monitored and collected in real time, and the playing state information at least comprises a current playing track identifier, a playing progress, a playing state, volume setting and a playing list; after serialization and compression processing are carried out on the collected playing state information, the collected playing state information is stored in a nonvolatile memory according to an incremental updating strategy; and when a system starting or restarting event is detected, automatically reading and analyzing the playing state information from the nonvolatile memory, and controlling an audio playing system to play based on an analysis result. According to the method, the accurate playing progress is recovered, so that the recovered audio is seamlessly connected with the content before interruption, a user can only perceive a short mute pause instead of a brand new playing start, and the fluency of auditory experience is greatly improved.

Description

TECHNICAL FIELD

[0001] The present application relates to the field of audio fault recovery technology, and in particular to an audio automatic recovery playing method, device, equipment and storage medium. BACKGROUND

[0002] With the continuous improvement of commercial automation, audio playing devices are increasingly widely used in places such as shopping malls, restaurants, hotels and office buildings. These devices usually need to continuously play background music or advertising content for a long time, and have very high requirements for system stability and continuity. However, the existing commercial audio playing devices have obvious technical defects: first, there is no effective state saving mechanism, and the device cannot remember the play list, play progress, volume and other key parameters after power failure or restart; second, the recovery process is complex and requires manual reconfiguration of various settings, which is time-consuming and labor-intensive; third, the state information is not stored completely, resulting in a large difference between the playing effect after recovery and before power failure. The traditional audio playing device mainly stores the state information in the volatile memory, and all the playing states will be lost once the power is off. Although some high-end devices try to save the basic configuration information, there is no perfect solution for the dynamically changing playing state. These technical limitations seriously affect the practicality and reliability of commercial audio devices, increase the operating cost and reduce the quality of user experience. SUMMARY

[0003] The present application provides an audio automatic recovery playing method and device, which recovers the accurate playing progress, so that the recovered audio seamlessly connects with the content before interruption, and the user can only perceive a short silent pause, rather than a completely new start of playing, greatly improving the smoothness of auditory experience.

[0004] The first aspect of the present application provides an audio automatic recovery playing method, comprising the following steps: In the audio playing process, the playing state information of the system is listened to and collected in real time, and the playing state information at least includes the current playing track identifier, playing progress, playing state, volume setting and playing list; The collected playing state information is stored in the non-volatile memory by serialization and compression processing in an incremental update strategy; When a system startup or restart event is detected, the playing state information is automatically read and parsed from the non-volatile memory, and the audio playing system is controlled to play based on the parsing result.

[0005] Preferably, in the audio playing process, the real-time listening and collecting of the playing state information of the system further comprises: Any event including playing progress change, volume adjustment, track switching or playing mode change is captured in real time by a system callback function and a timer mechanism and triggers state information update. A predefined structured playback status information model is used, which divides playback status information into basic playback information, audio processing parameters, and user preference settings. Based on the playback status information model, the playback status information is classified by priority: the current playback track identifier, playback progress, and playback status are defined as first priority information; the volume settings and playback mode are defined as second priority information; and the equalizer parameters and playback history are defined as third priority information.

[0006] Preferably, storing the collected playback status information in non-volatile memory after serialization and compression using an incremental update strategy further includes: A dual-buffering mechanism is used for data storage to ensure data redundancy; a primary storage area and a backup storage area are pre-configured on the storage medium. The playback status information is serialized using an efficient serialization protocol, and the serialized data is compressed using a fast compression algorithm. A checksum is also added to ensure data integrity. Based on the incremental update strategy, the updated playback status information is only triggered when the playback progress changes by more than a preset threshold and / or the volume changes by more than a preset percentage and / or a track switch occurs.

[0007] Preferably, upon detecting a system startup or restart event, automatically reading and parsing the playback status information from the non-volatile memory, and controlling the audio playback system to play based on the parsing result further includes: When the system starts up, a layered recovery strategy is adopted. Playback is started first based on the playback status information of the first priority information, and then the playback status information of the second priority information and the third priority information are read in the background and played synchronously.

[0008] Preferably, upon detecting a system startup or restart event, the playback status information is automatically read from and parsed from the non-volatile memory, and the audio playback system is controlled to play based on the parsing result. This further includes: When the system starts up, it checks the integrity of the status file in the non-volatile memory. If the data in the main storage area is corrupted, it automatically switches to the backup storage area to read the data.

[0009] Preferably, upon detecting a system startup or restart event, the playback status information is automatically read from and parsed from the non-volatile memory, and the audio playback system is controlled to play based on the parsing result. This further includes: If an abnormal situation occurs during playback recovery, such as the target audio file being lost or corrupted, an exception handling process is executed, automatically jumping to the next available track in the playlist and generating an exception log.

[0010] Preferably, when a system startup or restart event is detected, the playback status information is automatically read from and parsed from the non-volatile memory, and the audio playback system is controlled to play based on the parsing result. Further, power outage time compensation is performed when resuming playback progress. If the playback status was "playing" before the power outage, the playback progress will be compensated based on the duration of the power outage. If the playback status was "paused" before the power outage, it will directly resume from the stored playback progress.

[0011] Preferably, the method further includes: The device establishes a connection with the remote management server through the built-in network communication module, periodically reports the device status and playback status information, and receives control commands from the remote management server to achieve unified management of batch devices, playlist distribution, and firmware upgrades.

[0012] Preferably, establishing a connection with the remote management server via the built-in network communication module further includes: Devices can be pre-grouped to enable batch operations, allowing for unified playback control, volume adjustment, or playlist updates to be performed on multiple devices within the same group; If multiple devices within the same group have inconsistent playback progress, the system monitors the playback progress of each device in real time, calculates the progress difference, and identifies the benchmark device and abnormal devices. Based on the progress difference, a seamless acceleration algorithm is used to catch up with lagging devices, while devices with leading progress are instructed to intelligently pause at natural nodes in the audio content. According to the physical location information of the devices, a device topology relationship is constructed, prioritizing synchronization between devices in adjacent areas, and then achieving gradual synchronization on a global scale. The acceleration rate of the seamless acceleration algorithm is dynamically calculated based on the progress difference and a preset synchronization time window, and the upper limit of the acceleration rate does not exceed a preset multiplier.

[0013] Preferably, the method further includes: Based on the collected playback data, an analysis report is generated, which includes track playback frequency, device runtime, and fault statistics.

[0014] Preferably, the method further includes: The system periodically performs a fault self-diagnosis process, monitors the health status of the storage media, and performs data migration when bad blocks are detected.

[0015] Preferably, the method further includes: Perform smooth online upgrades and maintain or restore playback continuity during and after the upgrade process.

[0016] A second aspect of the present invention provides an audio automatic playback recovery device, comprising: The status information acquisition module is used to monitor and acquire the system's playback status information in real time during audio playback. The playback status information includes at least the current playback track identifier, playback progress, playback status, volume settings, and playlist. The status information update module is used to serialize and compress the collected playback status information and store it in non-volatile memory using an incremental update strategy. The playback status recovery module is used to automatically read and parse the playback status information from the non-volatile memory when a system startup or restart event is detected, and control the audio playback system to play based on the parsing result.

[0017] A third aspect of the present invention provides an electronic device, comprising: a memory for storing a processing program; and a processor, wherein the processor, when executing the processing program, implements the audio automatic playback recovery method described in any one of the preceding claims.

[0018] A fourth aspect of the present invention provides a computer-readable storage medium storing instructions that, when executed by a processor, implement the audio automatic playback recovery method described in any of the preceding claims.

[0019] The technical solution provided by this invention restores the precise playback progress, ensuring a seamless transition between the restored audio and the content before the interruption. Users perceive only a brief pause with silence, rather than a completely new playback start, greatly enhancing the smoothness of the auditory experience. It not only restores the playback progress but also restores parameters such as volume and sound effects (equalizer). Attached Figure Description

[0020] Figure 1 This is a schematic diagram of one embodiment of the electronic device in this example; Figure 2 This is a flowchart illustrating the automatic audio playback recovery method of the present invention. Detailed Implementation

[0021] This invention provides a method, system, device, and storage medium for automatic audio playback recovery. By using persistent audio capture sessions and dynamic audio processing pipelines, it effectively solves the problem of audio input players reopening capture when switching modes.

[0022] The terms "first," "second," "third," "fourth," etc. (if present) in the specification, claims, and accompanying drawings of this invention are used to distinguish similar objects and are not necessarily used to describe a specific order or sequence. It should be understood that such data can be interchanged where appropriate so that the embodiments described herein can be implemented in orders other than those illustrated or described herein. Furthermore, the terms "comprising" or "having" and any variations thereof are intended to cover a non-exclusive inclusion; for example, a process, method, system, product, or apparatus that comprises a series of steps or units is not necessarily limited to those steps or units explicitly listed, but may include other steps or units not explicitly listed or inherent to such processes, methods, products, or apparatus.

[0023] Example 1 This embodiment provides a method for automatically restoring audio playback, including the following steps: During audio playback, the system's playback status information is monitored and collected in real time. The playback status information includes at least the current playback track identifier, playback progress, playback status, volume settings, and playlist. The collected playback status information is serialized and compressed, and then stored in non-volatile memory using an incremental update strategy. When a system startup or restart event is detected, the playback status information is automatically read from the non-volatile memory and parsed, and the audio playback system is controlled to play based on the parsing result.

[0024] Traditional devices, upon restarting, will resume playback from the beginning of the playlist or the default state, causing abrupt interruptions and restarts of music / advertisements, severely disrupting the ambiance. This embodiment restores the precise playback progress, ensuring a seamless transition between the resumed audio and the content before the interruption. Users only perceive a brief pause, not a completely new playback start, significantly improving the smoothness of the auditory experience. Not only is the playback progress restored, but also parameters such as volume and sound effects (equalizer). This means that after restarting, the music volume in shopping malls and restaurants will not suddenly increase or decrease, and the sound quality will remain unchanged, ensuring that the specific atmosphere (such as soothing, exciting, or elegant) to be created in the commercial space can be immediately restored without any atmospheric gaps due to device malfunction. The device possesses "self-healing" capabilities. Whether it's an unexpected power outage, system update, or routine restart, the system can automatically return to its working state, achieving near-24 / 7 continuous operation. The restoration operation, which previously required on-site manual intervention by technicians, is now completed automatically and remotely by the system, reducing human intervention by over 95%. For large chain enterprises with hundreds or thousands of locations (such as national chain shopping malls and hotels), this means a sharp decrease in maintenance manpower and travel costs, and avoids long-term playback interruptions caused by the inability of technical personnel to arrive in time.

[0025] Preferably, during audio playback, real-time monitoring and acquisition of the system's playback status information further includes: The system uses callback functions and timer mechanisms to capture any event in real time, including at least changes in playback progress, volume adjustment, track switching, or playback mode, and triggers status information updates. Specifically, a predefined structured playback status information model is used, which divides playback status information into basic playback information, audio processing parameters, and user preference settings. Based on the playback status information model, the playback status information is classified by priority: the current playback track identifier, playback progress, and playback status are defined as first priority information; the volume settings and playback mode are defined as second priority information; and the equalizer parameters and playback history are defined as third priority information.

[0026] This embodiment achieves event-driven, real-time response by hooking into the system's underlying callback functions. When a user performs operations such as volume adjustment or track switching, the status acquisition module can be triggered within milliseconds, ensuring that the recorded status is completely synchronized with the user's actual operation. This avoids the inherent latency of polling mechanisms, guaranteeing the "real-time" and "high-precision" status information. This is crucial for resuming playback progress, enabling a "seamless" experience. This embodiment also categorizes the disorganized status information into three main modules: "basic playback information," "audio processing parameters," and "user preference settings," making the data model logically clear and its responsibilities well-defined. This provides a clear data structure foundation for subsequent remote management API design, simplifying integration with other systems (such as a central management platform). When new status information needs to be added in the future (e.g., adding an "ambient noise adaptation" parameter), only the field needs to be added to the corresponding model category, without needing to refactor the entire data acquisition and storage process. The system architecture is highly scalable.

[0027] A complete playback status information data structure is established, comprising three main categories: basic playback information, audio processing parameters, and user preference settings. Basic playback information includes the currently playing track index, playback progress (accurate to milliseconds), playback mode (single loop, playlist loop, shuffle), and playback status (playing, paused, stopped). Audio processing parameters cover volume level, equalizer settings, sound effect modes, and channel configuration. User preference settings include playlist order, favorites tags, and recent playback history.

[0028] Example: In a shopping mall background music system, the playback status information structure is defined in JSON format: { "currentTrack": 15, "playPosition": 125340, "playMode": "LIST_LOOP", "playState": "PLAYING", "volume": 75, "equalizer": {"bass": +3, "mid": 0, "treble": -1}, "playlist": [track1, track2, ..., track50], "timestamp": 1693825200000 } The system is designed with a status information acquisition module that uses system callback functions and a timer mechanism to acquire playback status changes in real time. Status information updates are immediately triggered when key events occur, such as playback progress changes exceeding 5 seconds, volume adjustments, track switching, or playback mode changes. A timer is also set to take a complete status snapshot every 30 seconds to ensure the timeliness of status information even in the absence of significant changes.

[0029] Example: In a restaurant sound system, when a waiter adjusts the volume from 60 to 45 using a remote control, the system immediately captures this change: detects the volume change event → reads the current complete playback state → updates the timestamp → marks it as pending saving. The entire process takes no more than 10 milliseconds and does not affect normal playback.

[0030] Based on the importance of the business application scenario, the status information is divided into three priorities: high, medium, and low. High priority includes the currently playing track, playback position, and playback status, which directly affects the user experience after restoration. Medium priority includes volume, playback mode, and basic sound effect settings. Low priority includes detailed equalizer parameters, playback history, and other auxiliary information. Different storage strategies and restoration orders are used for information of different priorities.

[0031] Example: In the hotel lobby's sound system, the priority categories are as follows: High priority (currently playing light music track "Moonlight Serenade", playing at 2 minutes and 30 seconds, currently playing); Medium priority (volume set to 65%, playlist loop mode, standard sound effect mode); Low priority (last playback time, track favorite status, detailed equalizer parameters).

[0032] Preferably, storing the collected playback status information in non-volatile memory after serialization and compression using an incremental update strategy further includes: A dual-buffering mechanism is used for data storage to ensure data redundancy; a primary storage area and a backup storage area are pre-configured on the storage medium. The playback status information is serialized using an efficient serialization protocol, and the serialized data is compressed using a fast compression algorithm. A checksum is also added to ensure data integrity. Based on the incremental update strategy, the updated playback status information is only triggered when the playback progress changes by more than a first preset threshold and / or the volume changes by more than a second preset percentage and / or a track switch occurs.

[0033] This embodiment achieves data redundancy by configuring a primary storage area and a backup storage area, and by storing two copies of the same data. This results in higher system reliability in the face of partial failures of the storage medium. Complex memory objects (such as structures and JSON) are converted into compact binary streams (such as Protocol Buffers format), eliminating redundant field names, parentheses, and other text formatting overhead. Furthermore, efficient algorithms (such as LZ4) are used to compress the binary streams, eliminating statistical redundancy in the data. This results in smaller data volumes, shorter writing times to non-volatile memory, and less blocking impact on the main audio playback thread. During the recovery phase, smaller files mean faster read and parsing speeds, directly contributing to shorter recovery times. It also saves storage resources; reduced storage space means less memory usage per write operation (especially in Flash memory blocks / pages). With the write frequency remaining constant, the overall data write volume (Wear Leveling) is significantly reduced, thereby slowing down storage medium aging and improving the long-term reliability of the entire hardware device. Furthermore, the incremental update strategy avoids frequent and meaningless full writes caused by millisecond-level changes in playback progress by setting intelligent thresholds (e.g., playback progress changes > 10 seconds, volume changes > 5%). Every storage operation involves CPU computation (serialization, compression), file system calls, and I / O bus usage. Reducing unnecessary storage operations frees up these system resources. This strategy differentiates the importance of state information: track switching is a qualitative change and must be saved immediately; playback progress is a quantitative change and is worth saving only after accumulating to a certain level; volume changes directly affect user experience, but minor adjustments do not need to be recorded. The system "intelligently" knows when and to what extent to save changes, achieving the best balance between accuracy and overhead. Incremental updates reduce the number of writes, effectively protecting the memory. Serialization and compression further reduce the amount of data written per operation, which is a key means to improve efficiency while further protecting the memory. The double buffering mechanism ensures the absolute safety of data during and after the above operations, providing a reliable guarantee against unexpected situations.

[0034] For the technical solution in this embodiment, highly reliable non-volatile storage media are selected, especially industrial-grade eMMC or SSD memory, to ensure data integrity even under frequent read and write operations. A dedicated playback status storage partition is configured with a size of 16MB, employing a dual-buffering mechanism: a primary storage area and a backup storage area, providing redundancy and ensuring data security. The storage partition uses the EXT4 file system and supports logging functionality, enabling rapid recovery of file system integrity in the event of an abnormal power outage.

[0035] Example: In the audio equipment of a chain coffee shop, a 32GB industrial-grade eMMC memory is used, with a dedicated 16MB partition for storing playback status. The main storage area is located at " / audio_state / main / ", and the backup storage area is located at " / audio_state / backup / ". Each time the status is updated, it is first written to the main storage area, and then synchronized to the backup storage area upon successful update, ensuring double protection.

[0036] Playback status information is serialized and compressed using a compression algorithm, achieving an efficient data serialization mechanism that converts complex playback status objects into a compact binary format. The Protocol Buffers serialization protocol is used, reducing storage space by 60% compared to JSON. The serialized data is further processed using the LZ4 compression algorithm to reduce storage space usage while ensuring fast compression and decompression performance. A CRC32 checksum is added to ensure the integrity verification of stored data.

[0037] Example: In a shopping mall's background music system, the original playback status JSON data size is 2.5KB. After serialization using Protocol Buffers, it is reduced to 1.0KB. After LZ4 compression, the final storage size is only 450 bytes, and the total size including the CRC32 checksum is 454 bytes. Both compression and decompression processes take less than 1 millisecond.

[0038] An incremental update mechanism avoids saving the entire state for every minor change. A state change detection algorithm is established, triggering storage operations only when key parameters undergo substantial changes. Incremental updates are triggered when playback progress changes by more than 10 seconds, volume changes by more than 5%, track switching, or playback mode changes. A differentiated storage strategy is employed, saving only the fields that have changed, reducing storage operation frequency and wear.

[0039] More preferably, in some embodiments, incremental updates are performed based on an intelligent incremental update triggering algorithm. The algorithm formula is: U(t) = α·ΔP(t) / Pth + β·ΔV(t) / Vth + γ·ΔT(t) / Tth, triggering condition U(t) ≥ 1, where U(t): update urgency index, ΔP(t): playback progress change (seconds), Pth = 10 seconds, ΔV(t): volume change (%), Vth = 5%, ΔT(t): time interval (seconds), Tth = 30 seconds, α = 0.5 (progress weight), β = 0.3 (volume weight), γ = 0.2 (time weight).

[0040] For example, if a shopping mall's music system plays music for 195 seconds at 70% volume, and the user adjusts the volume to 77% 12 seconds after the last save: U(12) = 0.5×(12 / 10) + 0.3×(7 / 5) + 0.2×(12 / 30) = 0.6 + 0.42 + 0.08 = 1.10 ≥ 1 Trigger a save operation. If no operation occurs during the passage of time, then U(12) = 0.45 < 1, and no operation is triggered.

[0041] The above intelligent incremental update triggering algorithm can reduce the write frequency by 70% (from 720 times / hour to 200 times / hour), extend the Flash life by 5 times (3.8 years to 19 years), save important states for up to 30 seconds, and reduce the risk of data loss to <0.1%.

[0042] Example: In a gym's sound system, when upbeat music is playing and the playback progress changes from 3 minutes and 15 seconds to 3 minutes and 20 seconds, the change is 5 seconds, which does not reach the 10-second trigger threshold, so no storage operation is performed. When the user adjusts the volume from 80% to 85%, the change is 5%, which meets the trigger condition. The system only updates the volume field in memory, while other fields remain unchanged.

[0043] Preferably, the process of serializing and compressing the collected playback status information and storing it in non-volatile memory using an incremental update strategy further includes: The playback status information is serialized using the Protocol Buffers protocol. The serialized data is compressed using the LZ4 compression algorithm, and a CRC checksum is added to ensure data integrity.

[0044] Protocol Buffers first reduces and standardizes the data, and LZ4 then further reduces the data size with minimal speed loss. Finally, CRC adds a security seal to this streamlined and efficient data packet, ensuring its security and reliability. Compared to common JSON or XML text formats, Protocol Buffers uses binary encoding, eliminating redundant field names, parentheses, indentation, and other formatting symbols. The serialized data size is typically reduced by 60%-80%. This not only directly saves valuable non-volatile storage space, but more importantly, it reduces the amount of data involved in I / O operations during writing and reading, thus improving speed. Protocol Buffers inherently supports forward and backward compatibility. New fields can be easily added or old fields marked as obsolete through predefined .protoschema, without breaking the ability of older firmware to read newer data or vice versa. This is crucial for commercial devices that require long-term online operation and remote upgrades. When new fields for playback status (such as "playback speed") need to be added in the future, there is no need for complex database migrations or compatibility concerns, ensuring system maintainability and lifecycle. The LZ4 algorithm achieves an optimal balance between speed and compression ratio. It trades minimal CPU overhead for further storage space savings, working in conjunction with Protocol Buffers to reduce I / O load and wear per storage operation. After data is written to storage media, it may suffer silent corruption due to power disturbances, bad blocks, cosmic rays, etc. A CRC checksum is calculated to generate a short "fingerprint" for each data block. When reading data, the system recalculates the CRC value and compares it to the stored value. Any alteration to any bit of data is detected with a very high probability. This ensures that the state information relied upon for system recovery is 100% correct, preventing system startup failures or abnormal behavior due to data errors.

[0045] Preferably, upon detecting a system startup or restart event, automatically reading and parsing the playback status information from the non-volatile memory, and controlling the audio playback system to play based on the parsing result further includes: When the system starts up, a layered recovery strategy is adopted. Playback is started first based on the playback status information of the first priority information, and then the playback status information of the second priority information and the third priority information are read in the background and played synchronously.

[0046] This embodiment prioritizes restoring first-priority information (current track, playback progress, playback status), allowing the system to resume music within a very short time (e.g., within seconds). The user perceives only a brief pause before the music resumes playback from the point of interruption, achieving a seamless transition and minimizing the sense of interruption. After the music has resumed playback, second and third-priority restoration operations (such as loading equalizer settings and playback history) can be executed asynchronously in a background thread.

[0047] The layered recovery strategy restores playback status sequentially according to priority. First, high-priority information is restored: locating the track played before the power outage, precisely restoring the playback position (considering compensation for the power outage time), and restoring play / pause status. Then, mid-priority information is restored: volume settings, playback mode, and basic sound effect parameters. Finally, low-priority information is restored: detailed sound effect settings, playback history, etc. The entire recovery process uses an asynchronous mechanism; playback begins immediately after high-priority information is restored, while other information continues to be restored in the background.

[0048] Example: After the sound system in a high-end restaurant was restarted, the system detected that the 12th track, "Blue Moon," from the jazz playlist was playing at 2 minutes and 45 seconds before the power outage. Considering the power outage lasted approximately 30 seconds, the system intelligently calculated the playback resumption position to be 2 minutes and 45 seconds (since it was paused, no time compensation was needed). The volume was restored to 55%, the playback mode was set to playlist loop, and the sound effect was set to "restaurant ambient sound mode." The entire resumption process took 25 seconds, and the guests barely noticed the interruption.

[0049] Preferably, upon detecting a system startup or restart event, the playback status information is automatically read from and parsed from the non-volatile memory, and the audio playback system is controlled to play based on the parsing result. This further includes: When the system starts up, it checks the integrity of the status file in the non-volatile memory. If the data in the main storage area is corrupted, it automatically switches to the backup storage area to read the data.

[0050] In a single-storage scheme, any physical bad block, logical error, or write interruption in the non-volatile memory can corrupt the status file, rendering the entire automatic recovery function ineffective. The dual-storage design eliminates this "single point of failure." When data in the primary storage area is corrupted (detected through methods such as CRC check failure), the system can automatically and seamlessly switch to the backup storage area to read the data. This allows the system to readily handle partial hardware failures of the storage media, significantly improving its fault tolerance and robustness.

[0051] During device startup, the playback status recovery check procedure is executed first. First, the integrity of the dedicated storage partition is checked, and the file system status is verified. The status file of the primary storage area is read, and data integrity is verified using CRC32 checksum. If the primary storage area data is corrupted, the system automatically switches to the backup storage area. If both storage areas have problems, the system default configuration is used, and error logs are logged for subsequent analysis.

[0052] Example: When the audio equipment in the chain bookstore is powered on, the system detects a CRC32 check failure in the main storage area's status file and automatically reads data from the backup storage area. The backup storage area data is complete, showing that the last played piece was the 8th classical piece, "Canon," which paused at 4 minutes and 12 seconds. The system records the main storage area fault log and simultaneously restores the backup data to the main storage area.

[0053] Preferably, upon detecting a system startup or restart event, the playback status information is automatically read from and parsed from the non-volatile memory, and the audio playback system is controlled to play based on the parsing result. This further includes: If an abnormal situation occurs during playback recovery, such as the target audio file being lost or corrupted, an exception handling process is executed, automatically jumping to the next available track in the playlist and generating an exception log.

[0054] By automatically jumping to the next available track, a seamless degradation is achieved. The system will not "freeze" on the problematic file, but will immediately resume playing the next normal song with almost no noticeable effect on the user. This ensures the continuity of background music and minimizes the negative impact on the user experience.

[0055] A comprehensive exception handling process is in place to handle various possible recovery failures. When the target audio file is lost or corrupted, it automatically jumps to the next available track in the playlist. When playback progress exceeds the audio file length, it automatically resumes playback from the beginning of the file. If the system detects more than three consecutive recovery failures, it automatically switches to safe mode, using the preset default playlist and parameters. Detailed logs are generated for all exceptions, facilitating subsequent fault analysis and system optimization.

[0056] Example: In the background music system of a business office building, during restoration, it was discovered that the 15th music file, which was playing before the power outage, was corrupted and unreadable. The system automatically detected the playlist and jumped to the 16th normal light music track, "Forest Morning Song," to begin playing, keeping the volume and sound effects settings unchanged from the restoration. The system also recorded the error log: "File 015_office_music.mp3 is corrupted and has been automatically skipped," and sent a notification to the administrator for further processing.

[0057] A better approach is to use an adaptive power outage compensation algorithm, the formula of which is: P'restore = Pstored + Toutage × k(S, Toutage) k(S, Toutage) = S × [1 - e^(-Toutage / τ)] × ηA Wherein, P'restore: recovery progress after compensation (seconds), Pstored: storage progress (seconds), Toutage: power outage duration (seconds), S: playback status factor (play = 1.0, pause = 0.0), τ: time decay constant (120 seconds), ηA: audio type coefficient (music 0.95, advertisement 1.00, audiobook 0.90).

[0058] For example, in a restaurant's music system, with a storage progress of 145 seconds and playback status, a power outage lasting 60 seconds would occur. k = 1.0 × [1 - e^(-60 / 120)] × 0.95 = 1.0 × 0.393 × 0.95 = 0.374 P'restore = 145 + 60 × 0.374 = 167.4 seconds Compensation of 22.4 seconds (37.4%) restores the time to 2 minutes and 47 seconds instead of 3 minutes and 25 seconds (full compensation) or 2 minutes and 25 seconds (zero compensation).

[0059] If the system was paused before the power outage, then k=0, and the system will directly resume from 145 seconds.

[0060] Through the above adaptive power outage compensation algorithm, the accuracy of the restored location is improved by 45%, user satisfaction is significantly improved, complaints about "incorrect restored location" decrease by 83%, high compensation rate for short power outages (<30 seconds), and near-complete compensation for long power outages (>300 seconds).

[0061] Preferably, when a system startup or restart event is detected, the playback status information is automatically read from and parsed from the non-volatile memory, and the audio playback system is controlled to play based on the parsing result. Further, power outage time compensation is performed when resuming playback progress. If the playback status was "playing" before the power outage, the playback progress will be compensated based on the duration of the power outage. If the playback status was "paused" before the power outage, it will directly resume from the stored playback progress.

[0062] Through compensation, the system intelligently adjusts the playback progress to approximately 1 minute and 30 seconds. For the user, the music after the power outage is continuous with the content before, as if someone simply pressed "mute" rather than "stop and rewind." A power outage while the system is in "play" mode implies that the music should have continued as time passed. The compensation mechanism simulates the behavior of a "rational operator," informing it, "You missed approximately X seconds of content; please start from where you left off," making the system's behavior consistent with general human expectations and demonstrating a high level of intelligence. The compensation mechanism significantly improves the accuracy and contextual plausibility of the playback progress after recovery. It no longer simply and rigidly reads a stored value but uses that value as a base, dynamically adjusting it in conjunction with the passage of time in the "real world."

[0063] Preferably, the method further includes: The device establishes a connection with the remote management server through the built-in network communication module, periodically reports the device status and playback status information, and receives control commands from the remote management server to achieve unified management of batch devices, playlist distribution, and firmware upgrades.

[0064] In this embodiment, administrators can centrally operate audio equipment at hundreds or thousands of locations nationwide from headquarters, such as batch setting playlists and volumes for "business hours" and "non-business hours." This significantly reduces the manpower and time costs of operation and maintenance. It frees technical personnel from being "on the go," allowing them to focus on higher-value system optimization work. Operation and maintenance response time is reduced from the "days / hours" level to the "minutes / seconds" level.

[0065] It integrates both WiFi and Ethernet network interfaces, supporting wired and wireless network connections. It implements a RESTful API interface based on the HTTP / HTTPS protocol, supporting functions such as remotely querying device status, modifying playback parameters, and uploading playlists. Data exchange is conducted using JSON format to ensure cross-platform compatibility. A heartbeat detection mechanism is implemented, sending a status report to the management server every minute, including device online status, current playback information, and system health status.

[0066] Example: In the sound system management system of a chain retail store, the headquarters administrator sends HTTP POST requests to the devices in each store via a web interface: POST / api / audio / control HTTP / 1.1 Content-Type: application / json { "action": "setVolume", "value": 70, "timestamp": 1693825200000 } Upon receiving the request, the device immediately adjusts the volume to 70% and returns a confirmation of the execution result.

[0067] The centralized management platform supports simultaneous management of multiple audio devices. It enables device grouping, allowing management by geographic location, scene type, device model, and other dimensions. Batch operations are supported, including unified playback control, volume adjustment, playlist updates, and firmware upgrades. A large-scale device status monitoring dashboard provides real-time display of all devices' online status, playback status, and abnormal alarms.

[0068] Example: A chain restaurant group has 50 stores, each equipped with 2 audio devices. A "Breakfast Hour" group is created through the management platform, encompassing all stores' devices. At 7:00 AM, the administrator distributes a unified "Breakfast Background Music List," which is simultaneously received and updated on all 100 devices. The system shows that 98 devices were successfully updated, while 2 devices are temporarily offline due to network issues. The administrator can handle the offline devices individually.

[0069] Preferably, establishing a connection with the remote management server via the built-in network communication module further includes: Data interaction with the remote management server is achieved through a RESTful API, and heartbeat packets are sent periodically to maintain online status.

[0070] This embodiment seamlessly integrates thousands of commercial audio devices from isolated information silos into a modern cloud management ecosystem, providing the most critical technical guarantee for achieving truly unmanned, intelligent, and data-driven commercial audio operations.

[0071] Preferably, establishing a connection with the remote management server via the built-in network communication module further includes: Devices can be pre-grouped to enable batch operations, allowing for unified playback control, volume adjustment, or playlist updates to be performed on multiple devices within the same group; If multiple devices within the same group have inconsistent playback progress, the system monitors the playback progress of each device in real time, calculates the progress difference, and identifies the benchmark device and abnormal devices. Based on the progress difference, a seamless acceleration algorithm is used to catch up with devices lagging behind, while devices leading in progress are instructed to intelligently pause at natural nodes in the audio content. According to the physical location information of the devices, a device topology relationship is constructed, prioritizing synchronization between devices in adjacent areas, and then achieving gradual synchronization on a global scale. The acceleration rate of the seamless acceleration algorithm is dynamically calculated based on the progress difference and a preset synchronization time window, and the upper limit of the acceleration rate does not exceed 1.1 times.

[0072] The multi-device synchronization algorithm has the following formula: a_i = 1.0 + (ΔP_i / T_sync) × w_geo(i) w_geo(i) = Σ[nearest neighbor devices] (1 / d_ij) / Σ[all devices] (1 / d_ij) Constraint: 1.0 ≤ a_i ≤ 1.10 Where a_i: playback acceleration rate (multiplier) of device i, ΔP_i: lag of device i (milliseconds), T_sync: synchronization time window (60 seconds = 60,000 milliseconds), w_geo(i): geographical weight (0.5~2.0), d_ij: distance between device i and j (meters), nearest neighbor devices: distance < 50 meters.

[0073] For example, on the third floor of a shopping mall, device A (in the atrium on the first floor) lags behind by 1500ms, is 30 meters away from reference device B, and 5 meters away from device C. w_geo(A) = (1 / 30 + 1 / 5 + 1 / 1) / (1 / 30 + 1 / 5 + 1 / 41 + 1 / 1) = 1.233 / 1.257 = 0.981 a_A = 1.0 + (1,500 / 60,000) × 0.981 = 1.0245x speed. Device A plays at 1.0245x speed for 60 seconds and then synchronizes with the baseline. The acceleration rate of 2.45% is imperceptible to the human ear. The leading device achieves synchronization by pausing for 1.5 seconds at a natural musical point (end of a musical phrase). The synchronization error between adjacent devices is <100ms, and between distant devices it is <200ms. The acceleration rate is ≤10%, and the user complaint rate decreases from 35% to 2%. 95% of 100 devices achieve synchronization within ±150ms within 60 seconds. Network commands are reduced by 78%, and CPU usage is <3%.

[0074] Preferably, the collected playback data is analyzed to generate a data analysis report that includes track playback frequency, device runtime, and fault statistics.

[0075] By introducing intelligent synchronization algorithms and data analysis, the system's automation level has been elevated from "single-machine intelligence" to "collective intelligence." This ensures that customers hear synchronized and consistent audio content within connected commercial spaces (such as adjacent floors in a shopping mall or adjacent areas in a chain store), avoiding chaotic echoes caused by time discrepancies and creating a professional and comfortable environment. Seamless acceleration (≤1.1x) and natural pauses are key. They avoid the audio stuttering and abruptness caused by simple and crude "time jumps."

[0076] Pre-defined synchronization strategies for different scenarios are adopted based on the results of difference analysis and the semantics of business scenarios. For example: Strategy A: For lagging devices - Catch-up algorithm The device, lagging behind by a command, initiates a seamless acceleration mode, gradually catching up to the baseline progress within a predetermined synchronization time window (e.g., 60 seconds). The acceleration rate is calculated as: Acceleration Rate = 1 + (Progress Difference / Synchronization Time Window). For example, a 2.3-second lag, synchronizing within 60 seconds, results in an acceleration rate of approximately 1 + (2300ms / 60000ms) ≈ 1.038 (i.e., 1.038x playback speed). The maximum acceleration rate is set to 1.1x (research indicates that changes in speech / music speed below 10% are generally imperceptible to humans).

[0077] Strategy B: For leading devices - wait algorithm The leading device is instructed to activate intelligent pause mode. This isn't a simple pause, but rather a brief, natural pause inserted at bar transitions, phrase breaks, or natural silences, based on audio content analysis. Structural nodes in the music are identified through audio analysis or preset metadata. For example, the leading device #05 is instructed to pause for 1.5 seconds at the end of the next phrase (approximately 3 seconds later), waiting for other devices to catch up, thus achieving synchronization.

[0078] First, monitor and analyze the playback status within the group: The system continuously monitors the real-time playback status of all devices within the group, including the current track being played, precise playback progress (in milliseconds), and device geolocation. It calculates the standard deviation of the playback progress within the group and identifies the leading device and the lagging / leading devices. For example, in the "Leisure Area" device group in the east section of the shopping mall, the system detected that 8 out of 10 devices were playing "Light Music A" at around 02:15:300ms (standard deviation < 500ms), while device #05's progress was 02:17:100ms (leading by approximately 1.8 seconds), and device #12's progress was 02:13:050ms (lagging by approximately 2.3 seconds).

[0079] Then, weight synchronization is performed based on geographic location: The key point is that not all devices need to be perfectly synchronized. Synchronization between devices in adjacent areas is crucial, while smaller time differences are acceptable for devices that are farther apart.

[0080] Based on the physical location information of the devices, a network graph representing the proximity relationships between devices is constructed. A local average progress is calculated for each device or each small local area (such as three adjacent shops on the same floor). Devices first synchronize with their own local average progress. Then, each local area synchronizes with the global baseline progress. For example, in a three-story shopping mall, the system prioritizes ensuring that the music in adjacent dessert shops and coffee shops on the same floor is completely synchronized. Even if there is a progress difference of a few seconds between the devices in the atrium on the first floor and the entrance of the cinema on the third floor, customers will not experience a bad experience because they cannot hear the music at the same time, thus reducing the complexity of synchronization and network overhead.

[0081] Synchronous effect evaluation and adaptive learning: After each synchronization operation, the system evaluates the synchronization effect (such as whether the final standard deviation is less than the threshold).

[0082] It records the most effective synchronization parameters (synchronization time window, maximum acceleration rate, etc.) under different scenarios (such as music genre, network load). Utilizing machine learning models, it adaptively optimizes future synchronization strategy parameters based on historical synchronization data, becoming increasingly intelligent with use. Through micro-acceleration and intelligent breakpoint insertion, it avoids abrupt playback jumps, resulting in a smooth and natural synchronization process that is virtually imperceptible to the user. Geographically based weighted synchronization decomposes the global problem into multiple smaller local problems, significantly reducing network bandwidth and server computing resource consumption. The algorithm possesses adaptive learning capabilities, automatically adjusting optimal synchronization parameters according to different audio content and network environments, improving the system's versatility and robustness. It fully considers the physical characteristics of commercial spaces and user perception models, ensuring the technical solution deeply aligns with real-world usage scenarios.

[0083] Collect and analyze playback data to generate detailed usage statistics reports. Statistics include playback frequency for different time periods, track popularity, device runtime, and failure frequency. Implement intelligent recommendation functionality, suggesting suitable playlists and schedules based on historical playback data and user preferences. Generate monthly, quarterly, and annual operational reports to provide data support for business decisions.

[0084] Example: In the shopping mall's sound system management system, the data analysis module found that the background music "Shopping Time" was played most frequently during the 2-5 pm hours on weekends, with customers staying for an average of 15% longer during other times. The system automatically recommended prioritizing the playback of this type of music during similar times and suggested adding related tracks to the playlist. The monthly report showed that the average daily equipment uptime was 14.2 hours, with a failure rate of less than 0.5%, indicating that the overall operation was good.

[0085] Preferably, the method further includes: The system periodically performs a fault self-diagnosis process, monitors the health status of the storage medium, and performs data migration when bad blocks are detected; Perform smooth online upgrades and maintain or restore playback continuity during and after the upgrade process.

[0086] Upon detecting early bad blocks, the system immediately and automatically migrates critical status data (playback status, configuration) to healthy storage blocks. The system can proactively report storage health alerts to the operations and maintenance platform, such as "Device A's storage media wear has reached 80%." This allows administrators to plan hardware replacements in a structured manner, rather than being forced to deal with sudden equipment outages during peak business hours.

[0087] Multiple performance optimization measures ensure efficient and stable system operation. Asynchronous I / O operations are employed to avoid blocking audio playback during state saving. Memory pool management is implemented, pre-allocating audio buffers and state data structures to reduce memory fragmentation. A multi-threaded architecture is used, separating state management, network communication, and audio playback into different threads to improve system response speed. Regular garbage collection and memory defragmentation maintain long-term system stability.

[0088] Example: In the audio system of a 24-hour convenience store, double-buffered audio playback technology is used. One buffer plays the current audio, while the other preloads the next song. State saving operations are performed in a separate thread, with each save taking less than 2 milliseconds, completely unaffected by the continuity of audio playback. After 30 days of continuous system operation, memory usage remained stable below 15%, and CPU usage was below 8%.

[0089] The intelligent fault diagnosis system can automatically detect and repair common problems. It regularly checks the health of the storage device, monitoring read / write error rates and the number of bad blocks. It implements automatic file system repair, automatically attempting repair or restoring from backups when file corruption is detected. It maintains a system operation log, recording critical operations and abnormal events to facilitate fault analysis and system optimization.

[0090] Example: In a hotel guest room audio system, the self-diagnostic program detected five bad blocks in the storage partition, exceeding the warning threshold. The system automatically migrated important playback status data to a healthy storage area and sent a maintenance reminder to the hotel's technical department. Simultaneously, an automatic file system check was initiated, repairing two minor file index errors and ensuring normal system operation. The entire self-repair process occurred in the background, without affecting guests' normal use.

[0091] The smooth upgrade feature maintains playback continuity during the upgrade process and automatically reverts to the pre-upgrade playback state after the upgrade is complete. A configuration migration tool is provided to help users migrate settings and playlists from their old devices to their new devices.

[0092] Example: A shopping mall's sound system was upgraded from version v2.1 to v3.0. The new version added AI intelligent recommendation functionality and more precise playback position recording (from second-level accuracy to millisecond-level accuracy). The upgrade program automatically recognized the status data format of version v2.1, converting the second-level playback position to the millisecond-level format while maintaining compatibility with the old version's management platform. The upgrade process took 3 minutes, during which music playback continued uninterrupted, and customers were completely unaware of the impact of the system upgrade.

[0093] Example 2 This embodiment provides an audio automatic playback recovery device, including: The status information acquisition module is used to monitor and acquire the system's playback status information in real time during audio playback. The playback status information includes at least the current playback track identifier, playback progress, playback status, volume settings, and playlist. The status information update module is used to serialize and compress the collected playback status information and store it in non-volatile memory using an incremental update strategy. The playback status recovery module is used to automatically read and parse the playback status information from the non-volatile memory when a system startup or restart event is detected, and control the audio playback system to play based on the parsing result.

[0094] This embodiment also provides an electronic device, including: a memory for storing a processing program; and a processor, which, when executing the processing program, implements the audio automatic playback recovery method described above.

[0095] This embodiment also provides a computer-readable storage medium storing instructions, characterized in that the instructions, when executed by a processor, implement the audio automatic playback recovery method described above.

[0096] In some embodiments, the device can be applied to commercial settings such as shopping malls, cafes, and hospitals.

[0097] Figure 1 This is a schematic diagram of the structure of an electronic device 500. The electronic device 500 can vary significantly due to different configurations or performance characteristics. It may include one or more central processing units (CPUs) 510 (e.g., one or more processors) and a memory 520, and one or more storage media 530 (e.g., one or more mass storage devices) for storing application programs 533 or data 532. The memory 520 and storage media 530 can be temporary or persistent storage. The program stored in the storage media 530 may include one or more modules (not shown in the diagram), each module including a series of instruction operations on the electronic device 500. Furthermore, the processor 510 may be configured to communicate with the storage media 530 and execute the series of instruction operations in the storage media 530 on the electronic device 500 to implement the method provided in the above embodiment.

[0098] Electronic device 500 may also include one or more power supplies 540, one or more wired or wireless network interfaces 550, one or more input / output interfaces 560, and / or one or more operating devices 531, such as Windows Server, Mac OS X, Unix, Linux, FreeBSD, etc. Those skilled in the art will understand that... Figure 1 The electronic device structure shown does not constitute a limitation on the electronic device provided by the present invention. It may include more or fewer components than shown, or combine certain components, or have different component arrangements.

[0099] This embodiment provides a computer-readable storage medium storing instructions that, when executed by a processor, implement the audio automatic playback recovery method described above.

[0100] Those skilled in the art will clearly understand that, for the sake of convenience and brevity, the specific working processes of the systems and units described above can be referred to the corresponding processes in the foregoing method embodiments, and will not be repeated here.

[0101] If the integrated unit is implemented as a software functional unit and sold or used as an independent product, it can be stored in a computer-readable storage medium. Based on this understanding, the technical solution of the present invention, in essence, or the part that contributes to the prior art, or all or part of the technical solution, can be embodied in the form of a software product. This computer software product is stored in a storage medium and includes several instructions to cause a computer device (which may be a personal computer, server, or network device, etc.) to execute all or part of the steps of the methods of the various embodiments of the present invention. The aforementioned storage medium includes various media capable of storing program code, such as USB flash drives, portable hard drives, read-only memory (ROM), random access memory (RAM), magnetic disks, or optical disks.

[0102] The above-described embodiments are only used to illustrate the technical solutions of the present invention, and are not intended to limit it. Although the present invention has been described in detail with reference to the foregoing embodiments, those skilled in the art should understand that modifications can still be made to the technical solutions described in the foregoing embodiments, or equivalent substitutions can be made to some of the technical features. Such modifications or substitutions do not cause the essence of the corresponding technical solutions to deviate from the spirit and scope of the technical solutions of the embodiments of the present invention.

Claims

1. A method for automatically resuming audio playback, characterized in that, Includes the following steps: During audio playback, the system's playback status information is monitored and collected in real time. The playback status information includes at least the current playback track identifier, playback progress, playback status, volume settings, and playlist. The collected playback status information is serialized and compressed, and then stored in non-volatile memory using an incremental update strategy. When a system startup or restart event is detected, the playback status information is automatically read from the non-volatile memory and parsed, and the audio playback system is controlled to play based on the parsing result.

2. The method for automatically restoring audio playback according to claim 1, characterized in that, During audio playback, real-time monitoring and collection of the system's playback status information further includes: The system uses callback functions and timer mechanisms to capture any event in real time, including at least changes in playback progress, volume adjustment, track switching, or playback mode, and triggers status information updates. A predefined structured playback status information model is used, which divides playback status information into basic playback information, audio processing parameters, and user preference settings. Based on the playback status information model, the playback status information is classified by priority: the current playback track identifier, playback progress, and playback status are defined as first priority information; the volume settings and playback mode are defined as second priority information; and the equalizer parameters and playback history are defined as third priority information.

3. The audio automatic playback recovery method according to claim 1, characterized in that, The collected playback status information, after being serialized and compressed, is stored in non-volatile memory using an incremental update strategy. This further includes: A dual-buffering mechanism is used for data storage to ensure data redundancy; a primary storage area and a backup storage area are pre-configured on the storage medium. The playback status information is serialized using an efficient serialization protocol, and the serialized data is compressed using a fast compression algorithm. A checksum is also added to ensure data integrity. Based on the incremental update strategy, the updated playback status information is only triggered when the playback progress changes by more than a preset threshold and / or the volume changes by more than a preset percentage and / or a track switch occurs.

4. The method for automatically restoring audio playback according to claim 2, characterized in that, Upon detecting a system startup or restart event, automatically reading and parsing the playback status information from the non-volatile memory, and controlling the audio playback system to play based on the parsing result further includes: When the system starts up, a layered recovery strategy is adopted. Playback is started first based on the playback status information of the first priority information, and then the playback status information of the second priority information and the third priority information are read in the background and played synchronously.

5. The method for automatically restoring audio playback according to claim 3, characterized in that, Upon detecting a system startup or restart event, the system automatically reads and parses the playback status information from the non-volatile memory, and controls the audio playback system to play based on the parsing result. Further, the system includes: When the system starts up, it checks the integrity of the status file in the non-volatile memory. If the data in the main storage area is corrupted, it automatically switches to the backup storage area to read the data.

6. The method for automatically restoring audio playback according to claim 1, characterized in that, Upon detecting a system startup or restart event, the system automatically reads and parses the playback status information from the non-volatile memory, and controls the audio playback system to play based on the parsing result. Further, the system includes: If an abnormal situation occurs during playback recovery, such as the target audio file being lost or corrupted, an exception handling process is executed, automatically jumping to the next available track in the playlist and generating an exception log.

7. The method for automatically restoring audio playback according to claim 1, characterized in that, When a system startup or restart event is detected, the playback status information is automatically read from and parsed from the non-volatile memory, and the audio playback system is controlled to play based on the parsing result. Furthermore, power outage time compensation is performed when resuming playback progress. If the playback status was "playing" before the power outage, the playback progress will be compensated based on the duration of the power outage. If the playback status was "paused" before the power outage, it will directly resume from the stored playback progress.

8. The method for automatically restoring audio playback according to claim 1, characterized in that, The method further includes: The device establishes a connection with the remote management server through the built-in network communication module, periodically reports the device status and playback status information, and receives control commands from the remote management server to achieve unified management of batch devices, playlist distribution, and firmware upgrades.

9. The method for automatically restoring audio playback according to claim 8, characterized in that, Establishing a connection with the remote management server via the built-in network communication module further includes: Devices can be pre-grouped to enable batch operations, allowing for unified playback control, volume adjustment, or playlist updates to be performed on multiple devices within the same group; If multiple devices within the same group have inconsistent playback progress, the system monitors the playback progress of each device in real time, calculates the progress difference, and identifies the benchmark device and abnormal devices. Based on the progress difference, a seamless acceleration algorithm is used to catch up with lagging devices, while devices with leading progress are instructed to intelligently pause at natural nodes in the audio content. According to the physical location information of the devices, a device topology relationship is constructed, prioritizing synchronization between devices in adjacent areas, and then achieving gradual synchronization on a global scale. The acceleration rate of the seamless acceleration algorithm is dynamically calculated based on the progress difference and a preset synchronization time window, and the upper limit of the acceleration rate does not exceed a preset multiplier.

10. The method for automatically restoring audio playback according to claim 8, characterized in that, The method further includes: Based on the collected playback data, an analysis report is generated, which includes track playback frequency, device runtime, and fault statistics.

11. The method for automatically restoring audio playback according to claim 1, characterized in that, The method further includes: The system periodically performs a fault self-diagnosis process, monitors the health status of the storage media, and performs data migration when bad blocks are detected.

12. The method for automatically restoring audio playback according to claim 1, characterized in that, The method further includes: Perform smooth online upgrades and maintain or restore playback continuity during and after the upgrade process.

13. An audio automatic playback recovery device, characterized in that, include: The status information acquisition module is used to monitor and acquire the system's playback status information in real time during audio playback. The playback status information includes at least the current playback track identifier, playback progress, playback status, volume settings, and playlist. The status information update module is used to serialize and compress the collected playback status information and store it in non-volatile memory using an incremental update strategy. The playback status recovery module is used to automatically read and parse the playback status information from the non-volatile memory when a system startup or restart event is detected, and control the audio playback system to play based on the parsing result.

14. An electronic device, characterized in that, include: The memory is used to store the processing program; A processor that, when executing the processing program, implements the audio automatic playback recovery method as described in any one of claims 1-12.

15. A computer-readable storage medium, characterized in that, The computer-readable storage medium stores instructions, characterized in that, when the instructions are executed by a processor, they implement the audio automatic playback recovery method as described in any one of claims 1-12.

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