A method and system for real-time audio playback of seismic events

Abstract

The application relates to the technical field of seismology data processing and audio signal processing, and provides a seismic event audio real-time playing method and system, which comprises the following steps: acquiring seismic event data, wherein the seismic event data contains multiple physical dimension parameters; mapping the seismic event data to multi-dimensional audio perception attributes based on the multiple physical dimension parameters, wherein the corresponding tone type is determined according to the range interval to which the intensity parameter in the multiple physical dimension parameters belongs; performing synchronous playback control on the seismic event data based on a playing speed, wherein the synchronous playback control comprises removing the seismic event data based on event identification information, and setting an anti-collision interval to guarantee the synchronous precision of audio triggering and event timing; and outputting corresponding audio signals. The application realizes parallel audio coding communication of seismic multi-dimensional information and guarantees the continuous synchronous precision in the variable-speed and cyclic playback scene.

Description

Technical Field

[0001] This application relates to the fields of seismic data processing and audio signal processing technology, and more specifically, to a method and system for real-time playback of audio from earthquake events. Background Technology

[0002] Currently, earthquake data analysis primarily relies on visual visualization methods, such as map points, time-series charts, and statistical graphs. However, with the continuous improvement of earthquake monitoring networks and the rapid increase in data volume, relying solely on visual analysis faces increasing challenges.

[0003] On the one hand, earthquake events contain multi-dimensional physical parameters such as magnitude, depth, time, and location. Traditional visual methods struggle to effectively express all dimensions of information simultaneously, easily leading to information loss and difficulties in fusion of multi-dimensional information. On the other hand, when it is necessary to play back historical time-series data at varying speeds to analyze the temporal evolution characteristics of earthquake activity, the lack of a precise audio-event synchronization mechanism results in event loss or repeated playback in variable-speed playback scenarios, making it difficult to ensure the accuracy of time-series perception.

[0004] Therefore, there is an urgent need for a technical solution that can effectively convert the multidimensional physical parameters of earthquake events into audio perception attributes and achieve variable-speed synchronous playback, so as to provide a multi-sensory data perception method and ensure playback synchronization accuracy. Summary of the Invention

[0005] To address the difficulties in multidimensional information fusion and the lack of synchronization in variable-speed playback in existing technologies, this application proposes a method and system for real-time audio playback of earthquake events. This method establishes a systematic mapping relationship between multidimensional physical parameters and multidimensional audio perception attributes, and ensures the synchronization accuracy of variable-speed playback through event deduplication and anti-collision mechanisms. This achieves parallel audio encoding transmission of earthquake multidimensional information and guarantees the synchronization accuracy of playback.

[0006] In a first aspect, the present invention provides a method for real-time playback of audio from earthquake events, comprising: Acquire seismic event data, which includes multi-physical dimension parameters; Based on the multi-physical dimension parameters, the earthquake event data is mapped to multi-dimensional audio perception attributes, wherein the corresponding timbre type is determined according to the range of the intensity parameter in the multi-physical dimension parameters. Synchronous playback control is performed on the earthquake event data based on the playback speed. The synchronous playback control includes deduplication of the earthquake event data based on event identification information and setting anti-collision intervals to ensure the synchronization accuracy between audio triggering and event timing. Output the corresponding audio signal.

[0007] In an optional implementation, the mapping relationship between the multi-physical dimension parameters and the multi-dimensional audio perception attributes includes: The intensity parameter is mapped to the volume attribute; Depth parameters are mapped to frequency attributes; Spatial position parameters are mapped to stereo position attributes; The time interval parameter is mapped to the playback interval attribute.

[0008] In an optional implementation, the intensity parameter and the volume attribute satisfy a logarithmic mapping relationship, and the depth parameter and the frequency attribute satisfy an inverse proportional mapping relationship, wherein: The logarithmic mapping relationship between the intensity parameter and the volume attribute is determined based on the ratio of the intensity parameter to the reference intensity value. By performing a logarithmic operation on the ratio and combining it with a coefficient set based on the characteristics of human hearing perception, the dynamic range of audio and the perception sensitivity are balanced to establish the distinguishable volume gradient of the earthquake event data. The inverse proportional mapping relationship between the depth parameter and the frequency attribute is determined based on the ratio of the reference frequency to the depth parameter. By introducing an adjustment parameter to perform an exponential operation on the depth ratio to control the attenuation rate of the frequency with depth, differentiated acoustic characterization of the high-frequency attributes corresponding to shallow-focus earthquake events and the low-frequency attributes corresponding to deep-focus earthquake events is achieved.

[0009] In an optional implementation, the synchronous playback control further includes: The earthquake event data is detected based on a time window, and earthquake event data to be played within the current playback period is filtered. A composite identifier is constructed using the event identifier information and the timestamp, and the deduplication process is performed on the earthquake event data based on the composite identifier information. The anti-collision interval defines the minimum time interval between audio triggering of adjacent seismic event data.

[0010] In an optional implementation, the synchronous playback control further includes a time rollback detection and playback record clearing mechanism to maintain synchronization accuracy in loop playback mode, wherein: The time rollback detection mechanism is configured to monitor the rollback status of the current playback time node relative to the historical playback time node in real time during playback. When it is detected that the current playback time node has rolled back relative to the historical playback time node and the rollback duration exceeds the preset rollback threshold, the playback mode is determined to enter the loop replay stage. The playback record cleanup mechanism is configured to automatically clean up the executed playback records corresponding to the historical playback time nodes and reset the synchronization state in response to the time rollback detection mechanism determining that the loop playback stage has been entered. This eliminates the interference of historical playback records on the event deduplication determination and the anti-collision interval calculation within the current loop cycle, ensuring the continuous synchronization accuracy of audio triggering and event timing in variable speed playback and loop playback scenarios.

[0011] In an optional implementation, before outputting the audio signal, dynamic range compression, spatial reverberation processing, and delay processing are sequentially performed on the synthesized audio signal.

[0012] In an optional implementation, the method further includes an audiovisual synchronization mechanism: The playback of the audio signal triggers corresponding visual highlights and animation effects; The user's interactive operations on the visual interface affect the playback parameters of the audio signal in real time. The playback status remains consistent between audio playback and visual display.

[0013] Secondly, the present invention provides a device for real-time audio playback of earthquake events, comprising: The data acquisition module is configured to acquire seismic event data, which includes multi-physical dimension parameters; The multidimensional mapping module is configured to map the earthquake event data to multidimensional audio perception attributes based on the multi-physical dimension parameters, wherein the corresponding timbre type is determined according to the range of the intensity parameter in the multi-physical dimension parameters. The synchronization control module is configured to perform synchronous playback control on the earthquake event data based on the playback speed. The synchronous playback control includes deduplication of the earthquake event data based on event identification information and setting an anti-collision interval to ensure the synchronization accuracy between audio triggering and event timing. The audio output module is configured to output the corresponding audio signal.

[0014] Thirdly, the present invention provides a real-time audio playback system for earthquake events, including the apparatus described in the foregoing embodiments.

[0015] Beneficial effects: This invention systematically maps the multi-physical-dimensional parameters of earthquake events to multi-dimensional audio-sensory attributes, particularly determining the corresponding timbre type based on the range of intensity parameters. This enables earthquake events of different intensities to achieve differentiated acoustic characterization through non-intuitive timbre selection, thereby transmitting information from multiple physical dimensions such as magnitude, depth, location, and time in parallel through independent audio-sensory channels. This solves the problem of multi-dimensional information fusion difficulties in traditional visual analysis methods and significantly improves the parallel encoding and transmission capabilities of multi-dimensional information and the data perception dimension.

[0016] This invention implements synchronous playback control based on playback speed and introduces a deduplication and anti-collision interval mechanism based on event identifier information. In variable speed playback scenarios, it effectively avoids event loss and repeated playback, ensures the synchronization accuracy of audio triggering and event timing, solves the problem of missing synchronization in existing variable speed playback technologies, and enables the temporal evolution characteristics of seismic activity to be accurately perceived.

[0017] This invention uses a time rollback detection and playback record cleanup mechanism to automatically clean up historical playback records and reset the synchronization state in loop playback mode. This eliminates the interference of historical records on deduplication judgment and anti-collision calculation, ensuring continuous synchronization accuracy in variable speed and loop playback scenarios.

[0018] This invention improves audio quality and spatial perception, and enhances auditory recognition and immersion by sequentially performing dynamic range compression, spatial reverberation and delay effects on the synthesized audio signal.

[0019] This invention uses an audio-visual synchronization mechanism to ensure that audio playback and visual display are consistent in terms of triggering, interaction, and status, thereby achieving integrated audio-visual data perception and providing users with a more comprehensive analytical perspective. Attached Figure Description

[0020] To more clearly illustrate the technical solutions of the embodiments of this application, the accompanying drawings used in the embodiments of this application will be briefly introduced below. It should be understood that the following drawings only show some embodiments of this application and should not be regarded as a limitation of the scope. For those skilled in the art, other related drawings can be obtained based on these drawings without creative effort.

[0021] Figure 1 This is a flowchart of the real-time audio playback method for earthquake events according to an embodiment of the present invention; Figure 2 This is a schematic diagram of the structure of the real-time audio playback device for earthquake events according to an embodiment of the present invention. Detailed Implementation

[0022] The technical solutions in the embodiments of this application will now be described with reference to the accompanying drawings.

[0023] like Figure 1 As shown in the figure, this embodiment provides a method for real-time playback of audio from earthquake events. The method mainly includes the following steps: Step S110: Obtain earthquake event data, which includes multi-physical dimension parameters.

[0024] Specifically, multi-physical-dimensional parameters refer to a set of parameters extracted from earthquake monitoring networks to describe the multiple physical attributes of a single seismic event. For example, multi-physical-dimensional parameters may include, but are not limited to, intensity parameters (such as magnitude) reflecting the scale of earthquake energy release, depth parameters and latitude and longitude coordinates reflecting the spatial location of the seismic source, and time interval parameters reflecting the time characteristics of the event.

[0025] It should be understood that although the specific parameters listed above are listed in this embodiment, in other embodiments, the multi-physical dimension parameters may also include any quantitative indicators that can characterize the physical characteristics of an earthquake, such as energy and duration, as long as these parameters can be systematically encoded into the audio sensing channel.

[0026] Step S120: Based on multi-physical dimension parameters, the earthquake event data is mapped to multi-dimensional audio perception attributes, wherein the corresponding timbre type is determined according to the range of intensity parameters in the multi-physical dimension parameters.

[0027] Specifically, multidimensional audio perception attributes refer to the audio feature dimensions that the human auditory system can perceive independently in parallel, such as timbre type, volume, frequency, and stereo position.

[0028] One of the core innovations of this embodiment is that it breaks through the intuitive linear mapping logic of "the greater the intensity, the greater the volume" in traditional data audio, and adopts a non-intuitive mapping mechanism that determines the corresponding timbre type based on the range of intensity parameters.

[0029] The reason for using this range-timbre mapping instead of simple volume adjustment is that the human ear's sensitivity to volume changes has a severe nonlinear compression effect. When a large number of earthquake events of different intensities occur in a short period of time, relying solely on volume differences can easily lead to small-magnitude events being masked by the volume of large-magnitude events, resulting in the loss of information about low-intensity events.

[0030] Conversely, the human ear is extremely sensitive to the distinction of timbre types and is independent of the volume dimension. By mapping different intensity ranges (such as micro-vibration range, medium-strong vibration range, and large vibration range) to significantly different timbre types (such as string timbre, drum timbre, piano timbre, etc.), even if micro-vibration events and large vibration events are played at the same time, users can still perceive the existence and characteristics of both in parallel through the timbre channel, thereby realizing the anti-masking parallel encoding and transmission of multi-dimensional information.

[0031] Step S130: Perform synchronous playback control on the earthquake event data based on the playback speed. The synchronous playback control includes deduplication of the earthquake event data based on event identification information and setting an anti-collision interval to ensure the synchronization accuracy between audio triggering and event timing.

[0032] Specifically, in variable-speed playback scenarios (such as 5x or 10x speed playback of historical data), the occurrence time of earthquake events is highly compressed, and a large number of events may be triggered densely within a very short physical playback time. Using only deduplication or a single time sorting method cannot effectively guarantee synchronization accuracy: if only deduplication is used without anti-collision intervals, when multiple different events are compressed to trigger at the same millisecond-level time point, the audio engine will generate stacking conflicts, leading to audio signal distortion or event loss; if only anti-collision intervals are used without deduplication, during loop playback or progress bar dragging back, the same event will be repeatedly read and occupy the anti-collision interval queue multiple times, causing the triggering of subsequent new events to be blocked by historical duplicate events, resulting in severe timing lag and event loss.

[0033] Therefore, this embodiment constructs a dual guarantee through a combination of deduplication and anti-collision intervals: deduplication ensures that the same earthquake event is triggered only once under any playback logic based on event identification information, eliminating duplicate occupation; the anti-collision interval limits the minimum time interval between audio triggers of adjacent events, ensuring that even if the event is highly compressed, the audio trigger still follows a strict timing queue and is executed sequentially, thereby maintaining the continuous synchronization accuracy between audio triggers and event timing in complex scenarios such as variable speed playback and loop playback.

[0034] Step S140: Output the corresponding audio signal.

[0035] Specifically, the audio parameters, after being processed by mapping and synchronization control, are sent to the audio synthesis engine to generate the final digital audio signal and output to the playback terminal for users to perform auditory perception and analysis.

[0036] Through the above scheme, this embodiment achieves parallel audio encoding transmission of multidimensional earthquake information and ensures synchronization accuracy for variable speed playback, solving the problems of difficulty in multidimensional information fusion and lack of synchronization in variable speed playback in traditional visual analysis methods.

[0037] We will continue to elaborate on the specific dimensional combinations and algorithm implementation of the multidimensional mapping mechanism.

[0038] The mapping relationship between multi-physical dimension parameters and multi-dimensional audio perception attributes includes: intensity parameters are mapped to volume attributes; depth parameters are mapped to frequency attributes; spatial position parameters are mapped to stereo position attributes; and time interval parameters are mapped to playback interval attributes.

[0039] Specifically, in traditional earthquake data visualization, information from different dimensions is often superimposed on the same visual plane, leading to difficulties in information masking and analysis. This embodiment, however, maps the independent physical dimensions of intensity, depth, spatial location, and time interval to four equally independent and parallel perceptual channels in the human auditory system: volume, frequency, stereo position, and playback interval. This achieves anti-masking parallel transmission of multidimensional information. For example, when two earthquake events overlap on a visual map, users may find it difficult to distinguish them; however, in audio mapping, even if the spatial location parameters of both events are mapped to the same stereo position attribute, as long as their depth parameters differ, resulting in differences in frequency attributes, or their intensity parameters differ, resulting in differences in volume attributes, the human ear can still clearly separate and independently perceive the two events using the frequency or volume channels.

[0040] It should be understood that although this embodiment lists the above four specific dimensional correspondences, in other embodiments, energy parameters can be mapped to overtone quantity attributes, or duration parameters can be mapped to note sustain attributes, etc., according to actual analysis needs. As long as these mapping relationships follow the principle of independent parallelism between physical dimensions and perception channels, they should all be considered to fall within the protection scope of this invention.

[0041] Furthermore, in order to ensure the scientific accuracy and auditory recognition of the above parallel encoding, the intensity parameter and the volume attribute satisfy a logarithmic mapping relationship, and the depth parameter and the frequency attribute satisfy an inverse proportional mapping relationship.

[0042] Regarding the logarithmic mapping relationship from intensity parameters to volume attributes, its specific implementation is based on the ratio of the intensity parameter to the reference intensity value. By performing a logarithmic operation on the ratio and combining it with a coefficient set based on the characteristics of human hearing perception, the dynamic range of audio and the perception sensitivity are balanced to establish the distinguishable volume gradient of the earthquake event data.

[0043] Specifically, earthquake magnitude (a typical example of an intensity parameter) is itself logarithmic, with each increase of one magnitude representing an approximately 31.6-fold increase in energy release. Mapping magnitude to volume using a simple linear mapping would result in severe auditory distortion. To clarify this mechanism, a comparative example is introduced: suppose we linearly map magnitudes from M1.0 to M6.0 to a relative volume range of 0.3 to 1.0, where an M1.0 event has a volume of 0.3 and an M6.0 event has a volume of 1.0. Because the human ear's perception of volume follows the Weber-Fechner law, meaning the perceived increment is proportional to the logarithm of the physical stimulus, under this linear mapping, the volume difference between M1.0 and M2.0 is only 0.14, making it almost impossible for the human ear to distinguish between these two types of micro-vibration events in a complex audio environment. Similarly, the volume difference between M5.0 and M6.0 is also 0.14, but since the overall volume base is close to 1.0, the human ear's sensitivity to changes in this range decreases significantly, causing small-magnitude events to be completely masked by large-magnitude events, making it impossible to achieve a distinguishable volume gradient. In contrast, this embodiment uses a logarithmic mapping relationship, for example, the specific formula can be expressed as V=20*log10(M / M0), where M0 is the reference magnitude value, whose typical range can be set to any value between 0.5 and 2.0 (e.g., M0=1.0), and scaled using a coefficient (e.g., coefficient 20) based on the characteristics of human ear perception.

[0044] Through logarithmic operations, the minute energy differences in the low-magnitude range are nonlinearly amplified into significant volume gradients, while the huge energy differences in the high-magnitude range are compressed into a dynamic range that the human ear can tolerate. This perfectly balances the dynamic range of audio with perceptual sensitivity, establishing a distinguishable volume gradient across the entire frequency range from micro-vibrations to strong vibrations.

[0045] Regarding the inverse proportional mapping relationship between depth parameters and frequency attributes, its specific implementation is based on the ratio of the reference frequency to the depth parameter. By introducing an adjustment parameter to perform an exponential operation on the depth ratio to control the attenuation rate of the frequency as it changes with depth, differentiated acoustic characterization is achieved for shallow-focus earthquake events corresponding to high-frequency attributes and deep-focus earthquake events corresponding to low-frequency attributes.

[0046] Specifically, in seismology, shallow-focus earthquakes typically exhibit higher-frequency seismic wave characteristics and greater destructive power, while deep-focus earthquakes are richer in low-frequency components. To align with this physical intuition and geological understanding in terms of auditory perception, this embodiment deliberately constructs an inverse proportional mapping relationship, where shallow-focus (smaller depth) corresponds to higher frequencies, and deep-focus (larger depth) corresponds to lower frequencies. To demonstrate the non-obviousness of this design, a comparative example is introduced: suppose a depth range of 10km to 600km is linearly mapped to a frequency range of 880Hz to 440Hz. Under this linear mapping, the 90km depth variation from 10km to 100km occupies approximately 660Hz of the frequency bandwidth, while the 500km depth variation from 100km to 600km occupies only approximately 220Hz. This linear allocation leads to overcrowding in the high-frequency band of shallow-focus earthquakes, with minimal frequency differences between different shallow-focus events, making high-frequency distortion and auditory confusion highly likely; conversely, the low-frequency band of deep-focus earthquakes is excessively sparse, wasting frequency resolution.

[0047] This embodiment employs an inverse proportional mapping relationship. For example, the specific formula can be expressed as F=F0 *(D0 / D)^α, where F0 is the reference frequency (typically set to 400Hz to 1000Hz, e.g., F0=880Hz), D0 is the reference depth value (typically set to 5km to 50km, e.g., D0=10km), and α is the adjustment parameter (typically set to any positive number between 0.5 and 2.0, e.g., α=1). By introducing the adjustment parameter α to perform an exponential operation on the depth ratio (D0 / D), the attenuation rate of the frequency with depth can be precisely controlled.

[0048] When α is large, the frequency rapidly decays to the low-frequency region with increasing depth, reserving an extremely wide high-frequency identification space for shallow-source events. When α is small, the decay rate slows down, allowing mid-to-deep-source events to still maintain a certain degree of frequency differentiation. This inverse proportional-exponential adjustment mechanism fundamentally avoids high-frequency distortion and low-frequency waste caused by linear mapping, achieving differentiated acoustic characterization and optimal identification distribution of shallow and deep-source seismic events in the frequency channels.

[0049] The logarithmic and inverse proportional mapping algorithms described above are for illustrative purposes only and not restrictive. In actual testing and verification, the system using the above mapping mechanism calculated audio parameters for earthquake events in the M1.0-M6.0 range. The average calculation time for each event was less than 1ms. It achieved a nonlinear distribution with good auditory discrimination within the 440Hz frequency range, and the change factor of the volume dynamic range conformed to the characteristics of human auditory perception. This fully demonstrates the dual advantages of this multidimensional parallel mapping architecture and specific algorithm in terms of real-time performance and auditory scientific accuracy.

[0050] The specific components of synchronous playback control and the loop playback mechanism are further elaborated in detail. When performing synchronous playback control on earthquake event data based on playback speed, it faces the severe challenge of massive amounts of time-series data being densely triggered on the compressed time axis. It is necessary to maintain the synchronization accuracy between audio triggering and event timing through sophisticated algorithmic logic.

[0051] The synchronous playback control also includes: detecting earthquake event data based on a time window and filtering earthquake event data to be played within the current playback period; constructing composite identifier information using event identifier information and timestamps, and performing deduplication processing on earthquake event data based on the composite identifier information; and limiting the minimum time interval for audio triggering of adjacent earthquake event data to prevent collisions.

[0052] Specifically, the time window detection and filtering mechanism is a pre-filtering step for variable-speed playback. Since historical data accumulated by earthquake monitoring networks often lasts for months or even years, loading all the data into memory at once for time sorting and anti-collision calculations would result in enormous computational overhead and latency.

[0053] This embodiment employs a sliding time window mechanism, extracting only earthquake event data within a preset time range before and after the current playback time point into the playback queue. For example, this preset time range can be set to 1 hour, meaning that only earthquake events within 1 hour before and after the current playback pointer are loaded, ensuring a smooth playback experience while avoiding the risk of memory overflow.

[0054] It should be understood that 1 hour is only a typical example. In other embodiments, the time window can be dynamically adjusted to 30 minutes, 2 hours or any suitable duration depending on the available memory size and data density.

[0055] Furthermore, in terms of deduplication logic, this embodiment abandons the coarse-grained deduplication method that relies solely on a single event identifier. Instead, it utilizes a composite identifier constructed from the event identifier and timestamp for deduplication. In actual earthquake catalog data, due to data revisions or merging, earthquake events occurring at different times may have the same event identifier; or in loop playback and progress bar dragging scenarios, the same event identifier may be read multiple times at different playback cycles or positions. If deduplication is based solely on event identifier, triggering of the same event at different legitimate times may be mistakenly judged as duplicates and removed, resulting in event loss. By introducing a timestamp to construct composite identifier (e.g., concatenating the event ID and the earthquake occurrence timestamp into a hash key), the system can accurately distinguish the unique triggering instance of the same event under different playback logics, ensuring that deduplication judgment neither misses nor falsely reports instances.

[0056] Regarding the anti-collision interval, it defines the minimum time interval between audio triggering of adjacent seismic event data. This embodiment provides a preferred endpoint value example: 50ms.

[0057] The 50ms value is not arbitrarily set, but rather an endpoint based on the minimum temporal resolution of the human ear. Acoustic psychology research indicates that the human ear's perceptual threshold for two consecutive audio events is approximately 40ms to 60ms. Audio events shorter than this interval are fused into a single perceptual object by the auditory system, making it impossible to distinguish their temporal sequence. Therefore, setting the anti-collision interval to 50ms ensures that adjacent events can be clearly separated audibly while maximizing the time compression efficiency of variable-speed playback.

[0058] To demonstrate the non-obviousness and non-substitutability of this endpoint value, an out-of-bounds comparison is introduced: If the anti-collision interval is set to 5ms, although it seems closer to the original timing, at 5x or 10x playback speed, events that were originally spaced several seconds apart are compressed to the millisecond level. The 5ms anti-collision interval cannot prevent a large number of events from stacking and triggering at the same instant, which will cause audio engine overload, signal distortion, and severe stacking chaos. If the anti-collision interval is set to 200ms, although the risk of stacking is completely eliminated, at high-speed playback of dense seismic clusters, the 200ms forced waiting queue will cause a large number of events that should have triggered in a short time to be severely delayed or directly discarded, resulting in catastrophic event loss.

[0059] Therefore, the 50ms endpoint, based on the human ear's perception threshold, achieves an unexpectedly precise balance between preventing stacking and preventing loss.

[0060] Synchronous playback control also includes time rollback detection and playback history cleanup mechanisms to maintain synchronization accuracy in loop playback mode.

[0061] The time rollback detection mechanism is configured to monitor the rollback status of the current playback time node relative to the historical playback time node in real time during playback. When a time rollback is detected between the current playback time node and the rollback duration exceeds a preset rollback threshold, the playback mode is determined to enter the loop replay stage. The playback record cleanup mechanism is configured to automatically clean up the executed playback records corresponding to the historical playback time nodes and reset the synchronization state in response to the time rollback detection mechanism's determination to enter the loop replay stage. This eliminates the interference of historical playback records on the event deduplication determination and anti-collision interval calculation within the current loop cycle, ensuring the continuous synchronization accuracy of audio triggering and event timing in variable speed playback and loop playback scenarios.

[0062] Specifically, timing logic faces unique challenges in loop playback scenarios. When a user sets up a loop to replay a specific historical period (such as yesterday's earthquake swarm activity), the playback pointer automatically jumps back to the starting point after reaching the end. At this point, the current playback time point has significantly regressed compared to historical playback time points. Without a dedicated time regression detection mechanism, this time jump can be misjudged by conventional timing algorithms as data anomalies or illegal user operations.

[0063] This embodiment monitors the displacement direction of the playback pointer in real time. Once it detects that the pointer has moved in the past time direction and the rollback time exceeds the preset rollback threshold (e.g., 1 hour, which is usually matched with the preset duration of the time window mentioned above to identify complete loop jumps rather than minor progress bar adjustments), it determines that the playback mode has entered a new loop playback stage.

[0064] At this point, triggering the playback history cleanup mechanism is crucial. If the historical playback history is not cleaned up, the event deduplication markers and anti-collision interval occupancy records already executed in the previous cycle will remain in memory. When the same event is read again in a new cycle, the deduplication logic will mistakenly determine that the event has already been played based on the remaining composite identifier information and remove it, resulting in no audio output for the entire new cycle. Simultaneously, the remaining anti-collision interval queue will occupy time slots, causing event triggering in the new cycle to be blocked by historical ghost records, resulting in severe timing delays and computational interference.

[0065] Therefore, in response to the determination of the loop playback stage, this embodiment automatically cleans up all the playback records executed in the previous cycle and completely resets the synchronization state of the anti-collision interval, so that each new loop cycle runs as cleanly as the first playback, thereby continuously ensuring the absolute synchronization accuracy of audio triggering and event timing in complex interactive scenarios such as variable speed playback and loop playback.

[0066] To verify the actual effectiveness of the aforementioned synchronous playback control mechanism, this invention conducted real-time synchronous playback performance tests on actual earthquake data. The tests used a 1-hour time window for detection and filtering, deduplication was performed using a composite key constructed based on event ID and timestamp, a 50ms anti-collision interval was set, and playback records were automatically cleared and the status reset when time rollback exceeded 1 hour. Test results showed that at 1x normal speed, the synchronization accuracy reached ±8ms, the event loss rate was only 0.1%, and the repeat playback rate was 0.0%; at 5x fast playback, the synchronization accuracy was ±15ms, the event loss rate was 0.3%, and the repeat playback rate was 0.1%; at 10x high-speed playback, the synchronization accuracy was ±25ms, the event loss rate was 0.8%, and the repeat playback rate was 0.2%; in loop playback mode, thanks to the time rollback detection and record clearing mechanism, the synchronization accuracy recovered to ±12ms, the event loss rate decreased to 0.2%, and the repeat playback rate remained stable at 0.0%.

[0067] The above data fully demonstrates that this embodiment, through a combination of time windows, compound key deduplication, anti-collision intervals, and loop cleaning mechanisms, can maintain reliable synchronization accuracy and an extremely low event error rate under various extreme speed variations and loop playback scenarios, providing a solid technical foundation for the audio-based analysis of seismic data. The specific parameters and test data described above are interpretive rather than restrictive, intended to elucidate the intrinsic principles and defensive depth of the synchronization control mechanism.

[0068] The paper further elaborates on the effects processing mechanism before audio output and the audiovisual synchronization mechanism. The original synthesized audio signal generated by multi-dimensional mapping and synchronization control needs to undergo professional audio effects chain processing before final output, and at the same time, it needs to maintain precise linkage with the visual interface during playback.

[0069] Before outputting the audio signal, the synthesized audio signal is subjected to dynamic range compression, spatial reverberation processing, and delay processing in sequence.

[0070] Specifically, these three elements constitute a strictly sequential and irreversible audio effects processing chain. First is dynamic range compression, whose core function is to compress the vast dynamic range of volume caused by the magnitude differences in earthquake events into a range comfortable for the human ear and safe for speakers to play. This prevents the volume of minor earthquakes from being masked by ambient noise and avoids audio clipping distortion caused by excessively high volume during strong earthquakes. For example, specific parameters for dynamic range compression can be set as follows: threshold -24dB, compression ratio 8:1, start time 3ms, and release time 100ms. When the input volume exceeds the -24dB threshold, the compressor compresses the excess gain at an 8:1 ratio. The fast 3ms start time ensures that transient peaks are not missed, while the 100ms release time avoids the auditory sucking effect during gain recovery.

[0071] Secondly, spatial reverberation processing adds spatial depth and environmental reflection characteristics to the originally dry digitally synthesized sound, giving earthquake events at different depths or locations a more realistic sense of spatial immersion. For example, spatial reverberation parameters can be set as follows: room size 0.7, damping frequency 3000Hz, humidity 0.1, and dryness 0.8. The lower humidity parameter ensures that the reverberation does not obscure the transient clarity of the original event, while the room size of 0.7 provides a moderate spatial attenuation simulation for mid-to-deep source events.

[0072] Finally, there's the delay effect processing. Its function is to add auditory layering and rhythm to dense vibrational sequences by repeating the signal for short periods, preventing a flat, stacked auditory effect when multiple events are triggered in a very short time. For example, the parameters for the delay effect can be set as follows: delay time 125ms, feedback amount 0.2, humidity 0.3, and dryness 0.7. The 125ms delay time is precisely within the critical range where the human ear can perceive independent echoes rather than reverberation blending, and the low feedback amount of 0.2 ensures that the delayed signal decays naturally after 2 to 3 repetitions, without producing endless oscillations.

[0073] The reason for emphasizing the irreversibility of the above processing chain is based on the deep mechanism of acoustic signal processing. If spatial reverberation is performed before dynamic range compression, the long tail reflection signal generated by the reverberator will be treated as a continuous high-level input and indiscriminately compressed by the subsequent compressor, causing the reverberation's decay tail to be abnormally lengthened or truncated, resulting in severe dynamic imbalance and an auditory sucking sensation. If delay is performed before reverberation, the delayed repetitive signal will enter the reverberator and produce secondary spatial reflections, causing the originally clear rhythmic layers to be submerged in an excessively blurred reverberation quagmire.

[0074] Therefore, it is essential to strictly follow the sequential order of compression, reverberation, and delay, completing dynamic control first in the dry signal stage, then imbuing it with spatial characteristics, and finally adding a temporal layer. Only in this way can the dual enhancement of spatial sense and depth be achieved while ensuring dynamic balance.

[0075] To verify the actual performance of this effects chain, the entire effects chain was tested. Test results show that the compressor's processing latency is less than 1ms, with a CPU utilization of 2%, achieving dynamic range balance; the reverb's processing latency is 2-3ms, with a CPU utilization of 5%, enhancing the sense of space; the delay's processing latency is 1-2ms, with a CPU utilization of 3%, improving the sense of depth; the total processing latency of the entire effects chain is controlled within 5ms, and the CPU utilization is controlled within 10%, significantly improving audio quality and professional listening experience while meeting the stringent latency requirements of real-time playback.

[0076] The method also includes an audio-visual synchronization linkage mechanism: the playback of the audio signal triggers corresponding visual highlighting and animation effects; the user's interactive operations on the visual interface affect the playback parameters of the audio signal in real time; and the playback state remains consistent between audio playback and visual display.

[0077] Specifically, the audiovisual synchronization mechanism is not simply a matter of sound and image appearing simultaneously, but rather a deep two-way coupling achieved by binding backend call logic.

[0078] Firstly, the playback of the audio signal triggers corresponding visual highlights and animation effects, a process that is tied to the front-end rendering call logic. When the synchronization control module determines that the audio trigger timing for a certain earthquake event has arrived, it not only sends a playback command to the audio synthesis engine but also sends an event identifier and timestamp to the rendering thread of the visualization interface, triggering the highlighting and flashing of the corresponding map point, the amplitude surge of the waveform graph, or the color transition of the spectrum graph. This allows auditory and visual perception to occur synchronously within a millisecond time difference, enhancing the impression of parallel perception of multi-dimensional information.

[0079] Secondly, user interactions on the visual interface affect the playback parameters of the audio signal in real time, a process linked to event listening and state update logic. For example, when a user changes the playback position by dragging the progress bar with the mouse, changes the playback speed by zooming the timeline with the scroll wheel, or filters the event range by clicking on a specific map area, the front-end event listener immediately captures these interactions and pushes the new playback position, playback speed, or filtering conditions as state update parameters to the synchronization control module and audio synthesis engine in real time. The audio parameters are then dynamically adjusted, providing users with immediate auditory feedback on their data exploration.

[0080] Thirdly, the playback state remains consistent between audio playback and visual display, a process bound to a unified time base. This embodiment establishes a high-precision unified time base clock during system initialization. The triggering sequence of the audio synthesis engine and the animation sequence of the visualization interface both read the current moment from this unified time base, rather than relying on independent system clocks. This unified time base binding mechanism completely eliminates the audio-visual misalignment caused by clock drift between the audio and visual threads, ensuring that auditory and visual perception remain seamlessly integrated even during long-term variable-speed playback and loop playback scenarios, achieving true audiovisual integrated data perception.

[0081] It should be understood that although the above linkage mechanism describes the specific background call logic, in other embodiments, state synchronization can also be achieved through message middleware or shared memory, as long as the core principles of triggering linkage, interactive linkage, and state consistency are met. The specific parameters and call logic described above are for illustrative purposes only and not restrictive.

[0082] This embodiment provides a device for real-time audio playback of earthquake events. The device adopts a purely functional modular architecture, with each step of the method corresponding to a specific step, such as... Figure 2 As shown, it mainly consists of a data acquisition module 201, a multi-dimensional mapping module 202, a synchronization control module 203, and an audio output module 204.

[0083] The data acquisition module 201 is configured to acquire earthquake event data, which includes multi-physical dimension parameters.

[0084] Specifically, the data acquisition module 201 is the input terminal of the entire device, responsible for receiving raw earthquake catalog data in real time or in batches from external earthquake monitoring networks or local historical databases. This module not only performs simple data retrieval operations but also undertakes preliminary data cleaning and formatting tasks, unifying heterogeneous earthquake parameters into structured data objects containing fields such as intensity parameters, depth parameters, spatial location parameters, and time interval parameters. For example, the data acquisition module 201 can maintain a long-term connection with the backend monitoring server via a RESTful API interface, continuously listening for newly occurring events and pushing them to downstream modules in real time; or, in historical playback scenarios, it can read the full set of events for the corresponding time period from local storage based on the user-selected time range.

[0085] It should be understood that although this embodiment describes a specific network connection and local reading method, in other embodiments, the data acquisition module 201 may also acquire data through message queue middleware or shared memory, as long as it can stably output structured seismic event data containing multi-physical dimension parameters.

[0086] The multidimensional mapping module 202 is configured to map the earthquake event data to multidimensional audio perception attributes based on the multi-physical dimension parameters, wherein the corresponding timbre type is determined according to the range of the intensity parameter in the multi-physical dimension parameters.

[0087] Specifically, the multi-dimensional mapping module 202 receives structured data objects from the data acquisition module 201 and serves as the core computing engine of the device, executing non-intuitive interval-timbre mapping and multi-dimensional parallel encoding logic. Regarding signal flow and connectivity, the multi-dimensional mapping module 202 is further subdivided into a parameter parsing unit, an interval determination unit, and an attribute synthesis unit. The parameter parsing unit extracts specific values ​​such as intensity and depth from the multi-physical dimension parameters; the interval determination unit matches the corresponding timbre type identifier (such as strings, drums, etc.) based on the range of intensity values ​​(e.g., micro-seismic interval, moderate-to-strong seismic interval); and the attribute synthesis unit assembles the timbre type identifier with attribute values ​​such as volume, frequency, and stereo position generated by mapping other dimensions into a complete audio-sensory attribute description package, which is then passed to the subsequent synchronization control module 203. This modular mapping architecture makes the updating and expansion of mapping rules extremely flexible. For example, when it is necessary to add a new magnitude interval or replace a timbre sample, only the configuration table of the interval determination unit needs to be modified, without changing the logic of data acquisition or synchronization control, thus decoupling the mapping calculation from the data flow.

[0088] The synchronization control module 203 is configured to perform synchronous playback control on the earthquake event data based on the playback speed. The synchronous playback control includes deduplication of the earthquake event data based on event identification information and setting an anti-collision interval to ensure the synchronization accuracy between audio triggering and event timing.

[0089] Specifically, the synchronization control module 203 is the central scheduling unit that ensures timing accuracy in variable-speed playback and loop playback scenarios. It receives audio perception attribute description packets from the multi-dimensional mapping module 202 and, in conjunction with the currently set playback speed parameters, recalculates and rearranges the event triggering sequence. Internally, the synchronization control module 203 includes a time window filter, a composite key deduplication filter, and an anti-collision interval queuer. The time window filter first extracts valid events based on the current playback pointer position; the composite key deduplication filter uses a composite key constructed from event identifier information and timestamps to eliminate duplicate trigger instances; and the anti-collision interval queuer applies a minimum time interval constraint to the deduplicated events to ensure that adjacent audio triggers do not overlap and conflict. After this series of scheduling steps, the synchronization control module 203 outputs a sequence of audio playback instructions with precise trigger timestamps and sends it to the audio output module 204.

[0090] It should be understood that although this embodiment describes internally subdivided sub-units, such subdivision is only a clear division of logical functions. In actual engineering implementation, the functions of these sub-units can be integrated into the same microprocessor or digital signal processor for execution, as long as they can be combined to achieve dual synchronization protection of deduplication and anti-collision.

[0091] The audio output module 204 is configured to output the corresponding audio signal.

[0092] Specifically, the audio output module 204 is the final execution end of the device. It receives the audio playback command sequence issued by the synchronization control module 203 and calls the underlying audio synthesis engine (such as a synthesizer built on the Web Audio API) to convert the command sequence into actual digital audio waveforms. In some embodiments, the audio output module 204 not only includes a basic waveform generator but also connects a dynamic range compressor, spatial reverb unit, and delay effect unit in series, ultimately outputting the processed audio signal to a playback terminal such as a speaker or headphones for the user to hear. Simultaneously, the audio output module 204 is also responsible for feeding back the playback status to the visual interface, achieving synchronized audio-visual interaction.

[0093] Through the signal flow and connection relationships of the aforementioned data acquisition module 201, multi-dimensional mapping module 202, synchronization control module 203, and audio output module 204, the device in this embodiment materializes the functions of each step of the method, forming a complete closed loop from data input, multi-dimensional encoding, timing scheduling to audio output. This purely functional module division not only strictly corresponds to the method's proprietary rights, facilitating the fight against device-level infringement, but also enables communication between modules through structured data packets and instruction sequences, achieving a highly cohesive and loosely coupled architecture design. This ensures that the system maintains good maintainability and scalability even when facing expansion of mapping rules, addition of playback modes, or adjustment of sound effect chains.

[0094] The module division and connection methods described above are merely illustrative and not restrictive. Those skilled in the art can merge, split, or distribute the modules according to the specific hardware deployment environment or software architecture requirements. As long as they can achieve the above four core functions and maintain the corresponding signal flow logic, they should be considered to fall within the protection scope of this invention.

[0095] This embodiment provides a computer-readable storage medium and a real-time audio playback system for earthquake events, serving as a carrier for the method and device, providing compliance support and product-level fallback.

[0096] On the one hand, this embodiment provides a computer-readable storage medium having a computer program stored thereon, which, when executed by a processor, implements the real-time playback method for audio of earthquake events as described in the foregoing embodiment.

[0097] Specifically, the computer-readable storage medium can be any non-transitory physical medium containing or storing a program that can be used by or in conjunction with an instruction execution system, apparatus, or device. For example, the computer-readable storage medium can include, but is not limited to: portable computer disks, hard disks, random access memory (RAM), read-only memory (ROM), erasable programmable read-only memory (EPROM or flash memory), portable compact disk read-only memory (CD-ROM), optical storage devices, magnetic storage devices, or any suitable combination thereof. In this embodiment, the computer program stored on the medium specifically includes all instruction code implementing steps S100 to S400, as well as the mapping algorithm library, synchronization control logic library, and audio effects chain configuration parameters involved in embodiments 2 to 4. When a general-purpose processor or dedicated audio processing chip loads and executes the program, the raw multi-dimensional physical parameters issued by the earthquake monitoring network can be converted in real time into audio signal outputs with high synchronization accuracy and professional listening quality. It should be understood that although this embodiment lists the specific media types described above, in other embodiments, any physical carrier capable of non-transitoryly storing the above program instructions and allowing the processor to read and execute them should be considered to fall within the protection scope of this invention.

[0098] On the other hand, this embodiment provides a real-time audio playback system for earthquake events, which includes the real-time audio playback device for earthquake events as described in the foregoing embodiments.

[0099] Specifically, to enhance the system's physicality and engineering feasibility, this embodiment further elaborates on the system's hardware deployment architecture and connectivity. In actual operation, the system typically employs a front-end / back-end separation architecture, specifically including a front-end visualization terminal and a back-end audio synthesis server. The front-end visualization terminal can be a personal computer, large-screen workstation, or mobile tablet deployed in the earthquake monitoring center's analysis room. It is primarily responsible for running the visualization interface, receiving user interaction, and rendering visual highlights and animation effects in synchronized audiovisual interaction. The back-end audio synthesis server is a high-performance computing node deployed in a data center. It integrates the core computing logic of the data acquisition module 201, multi-dimensional mapping module 202, synchronization control module 203, and audio output module 204 described in the aforementioned embodiment, and is equipped with a professional digital signal processor (DSP) to handle intensive audio synthesis tasks requiring high-speed playback.

[0100] In terms of hardware connectivity, the front-end visualization terminal and the back-end audio synthesis server establish a communication connection via a high-speed local area network (LAN) or wide area network (WAN). User interaction commands (such as playback speed adjustment and timeline dragging) captured by the front-end terminal through event listeners are encapsulated into status update messages and transmitted to the synchronization control module 203 of the back-end server via the network. After completing audio triggering and synthesis, the back-end server pushes the digital audio stream to the audio playback devices (such as professional monitoring speakers or headphones) connected to the back-end or front-end, and simultaneously sends the trigger timing and event identifier information back to the rendering thread of the front-end terminal to drive the precise triggering of visual animation. This hardware architecture, which separates the front-end and back-end and synchronizes status via network protocols, not only ensures that computationally intensive audio mapping and synthesis tasks do not consume the rendering resources of the front-end terminal, guaranteeing the smoothness of audiovisual interaction, but also allows multiple front-end analysts to access the same back-end audio synthesis service in parallel, realizing multi-user concurrent analysis and data sharing. It should be understood that although this embodiment describes a front-end and back-end separated network deployment architecture, in other embodiments, for single-machine scenarios with sufficient computing resources, all modules of the above-mentioned device can also be integrated and run on the same local workstation. The front-end and back-end can interact with each other through inter-process communication (IPC) or shared memory, as long as they can achieve a complete closed loop of data acquisition, multi-dimensional mapping, synchronization control, and audio output. The media types and system hardware architectures described above are illustrative only and not restrictive.

[0101] To more clearly illustrate the operational effectiveness and application value of the technical solution of this invention in a real-world environment, this embodiment provides a typical application scenario for earthquake monitoring network data analysis. It should be understood that this scenario is merely illustrative and not restrictive; those skilled in the art can certainly extend it to other types of time-series physical event data analysis.

[0102] The application scenario is as follows: An earthquake monitoring center needs to perform audio playback analysis on 1,000 earthquake events recorded in the past week to quickly capture the spatiotemporal evolution characteristics of seismic activity in the region. The magnitudes of these 1,000 events range from M1.0 to M6.0, the depths range from 10 km to 600 km, and their spatial distribution spans multiple latitude and longitude grids.

[0103] In the specific parameter input and mapping stage, after acquiring the data of these 1,000 earthquake events, the system maps them to multi-dimensional audio perception attributes based on multi-physical dimension parameters. Specifically, the corresponding timbre type is determined according to the range of the intensity parameter: micro-earthquake events with magnitudes of M1.0 to M2.5 fall into the first range and are mapped to piano timbre, whose crisp, short notes can indicate the release of small amounts of energy; moderate-to-strong earthquake events with magnitudes of M2.5 to M4.5 fall into the second range and are mapped to string timbre, whose long, sustained notes can indicate the release of continuous energy; strong earthquake events with magnitudes of M4.5 to M6.0 fall into the third range and are mapped to drum timbre, whose deep and impactful striking sounds can instantly capture the analyst's attention. Meanwhile, the depth parameter is mapped to the frequency attribute, with extremely shallow events at 10km mapped to a high frequency of 880Hz, while deep events at 600km are mapped to a low frequency of 440Hz. The sharp high frequencies of shallow events and the deep low frequencies of deep events create distinct acoustic characteristics. The spatial location parameter is mapped to the stereo location attribute, with latitude and longitude coordinates being linearly or nonlinearly compressed into a stereo sound image range of 0.0 to 1.0. This allows events located on the west side of the monitoring network to be emitted in the left channel, and events located on the east side to be emitted in the right channel, achieving a sense of spatial audio immersion.

[0104] In the synchronized playback and interactive analysis phase, analysts first browsed the data at a normal playback speed of 1x, at which point the system controlled the synchronized playback of earthquake event data based on the playback speed. Subsequently, to focus on analyzing the dense earthquake swarm activity that occurred over the weekend, analysts increased the playback speed from 1x to 5x for rapid playback. At 5x high-speed playback, earthquake events that were originally spaced several seconds apart were compressed into a millisecond-level rapid triggering. The system ensured the synchronization accuracy of audio triggering and event timing by deduplicating the earthquake event data based on event identifier information and setting anti-collision intervals. This ensured that the dense events would not cause audio distortion due to stacking, nor would key timing information be lost due to omissions. Specifically, the system used event identifier information and timestamps to construct composite identifier information for deduplication and limited the anti-collision interval to 50ms, allowing these 100 dense earthquake swarm events to still trigger sequentially with a clear rhythm even at 5x speed. When analysts set up a loop playback of the active period for repeated verification, the system's time rollback detection mechanism determines that the playback mode has entered the loop playback stage. The playback record cleanup mechanism automatically cleans up the historical playback records and resets the synchronization state, ensuring that each loop playback is as clean and undisturbed as the first playback.

[0105] In the multi-dimensional information parallel perception and audiovisual linkage verification stage, analysts captured the temporal evolution characteristics of the earthquake swarm sequence through hearing: in 5x fast playback, dense small-magnitude events that were originally difficult to distinguish on the visual chart appeared as a crisp sequence of notes jumping rapidly from left to right, mapped to piano timbres and possessing different frequencies and stereo positions; when a strong earthquake of M5.5 suddenly appeared, the heavy impact of the drum timbre instantly penetrated the background of the piano timbre, indicating a sharp jump in the activity of the strong earthquake swarm. At this time, the audiovisual synchronization linkage mechanism played a key role: the playback of the drum timbre was not only output through the speaker, but also simultaneously triggered the highlighting and flashing of the corresponding map points on the visualization interface and the amplitude surge of the waveform graph. The analysts' vision and hearing received this abnormal signal synchronously within a millisecond time difference. Subsequently, the analysts clicked on the highlighted area with the mouse, and the interactive operation affected the playback parameters of the audio signal in real time. The system immediately reduced the playback speed from 5x back to 1x and performed a separate audio preview and waveform analysis of the strong earthquake event, achieving a smooth transition from macroscopic rapid browsing to microscopic detailed analysis.

[0106] As can be clearly seen from the above application scenarios, this invention, through the systematic mapping of multi-dimensional physical parameters to multi-dimensional audio perception attributes, especially the non-intuitive mapping from intensity range to timbre type, enables the human ear to process information in multiple dimensions such as magnitude, depth, and location simultaneously and in parallel, breaking through the bottleneck of information masking and overlap in traditional visual analysis. At the same time, through the combination mechanism of deduplication processing and anti-collision interval, as well as the recording cleanup mechanism of loop playback, the absolute temporal synchronization accuracy in variable speed and loop playback scenarios is guaranteed, enabling analysts to capture the temporal evolution characteristics and anomaly patterns of seismic activity with extremely high efficiency, fully demonstrating the practical value of multi-dimensional information parallel perception and variable speed synchronous playback.

[0107] The above description is merely a specific embodiment of the present invention, but the scope of protection of the present invention is not limited thereto. Any variations or substitutions that can be easily conceived by those skilled in the art within the technical scope disclosed in the present invention, such as adaptively adjusting the mapping combination of multi-dimensional physical dimensions and audio perception attributes, replacing logarithmic or inverse proportional mapping relationships with other equivalent mathematical mapping models, dynamically adjusting the endpoint values ​​of anti-collision intervals or the preset duration of time windows according to actual hardware performance, or implementing the same data flow and synchronization control logic in other functional module division methods, should all be covered within the scope of protection of the present invention. Therefore, the scope of protection of the present invention should be determined by the scope of the claims.

Claims

1. A method for real-time playback of earthquake event audio, characterized in that, include: Acquire seismic event data, which includes multi-physical dimension parameters; Based on the multi-physical dimension parameters, the earthquake event data is mapped to multi-dimensional audio perception attributes, wherein the corresponding timbre type is determined according to the range of the intensity parameter in the multi-physical dimension parameters. Synchronous playback control is performed on the earthquake event data based on the playback speed. The synchronous playback control includes deduplication of the earthquake event data based on event identification information and setting anti-collision intervals to ensure the synchronization accuracy between audio triggering and event timing. Output the corresponding audio signal.

2. The method according to claim 1, characterized in that, The mapping relationship between the multi-physical dimension parameters and the multi-dimensional audio perception attributes includes: The intensity parameter is mapped to the volume attribute; Depth parameters are mapped to frequency attributes; Spatial position parameters are mapped to stereo position attributes; The time interval parameter is mapped to the playback interval attribute.

3. The method according to claim 2, characterized in that, The intensity parameter and the volume attribute satisfy a logarithmic mapping relationship, and the depth parameter and the frequency attribute satisfy an inverse proportional mapping relationship; The logarithmic mapping relationship between the intensity parameter and the volume attribute is determined based on the ratio of the intensity parameter to the reference intensity value. By performing a logarithmic operation on the ratio and combining it with a coefficient set based on the characteristics of human hearing perception, the dynamic range of audio and the perception sensitivity are balanced to establish the distinguishable volume gradient of the earthquake event data. The inverse proportional mapping relationship between the depth parameter and the frequency attribute is determined based on the ratio of the reference frequency to the depth parameter. By introducing an adjustment parameter to perform an exponential operation on the depth ratio to control the attenuation rate of the frequency with depth, differentiated acoustic characterization of the high-frequency attributes corresponding to shallow-focus earthquake events and the low-frequency attributes corresponding to deep-focus earthquake events is achieved.

4. The method according to claim 1, characterized in that, Also includes: The earthquake event data is detected based on a time window, and earthquake event data to be played within the current playback period is filtered. A composite identifier is constructed using the event identifier information and the timestamp, and the deduplication process is performed on the earthquake event data based on the composite identifier information. The anti-collision interval defines the minimum time interval between audio triggering of adjacent seismic event data.

5. The method according to claim 4, characterized in that, It also includes time rollback detection and playback history cleanup mechanisms to maintain synchronization accuracy in loop playback mode.

6. The method according to claim 5, characterized in that, The time rollback detection mechanism is configured to monitor the rollback status of the current playback time node relative to the historical playback time node in real time during playback. When it is detected that the current playback time node has rolled back relative to the historical playback time node and the rollback duration exceeds the preset rollback threshold, the playback mode is determined to enter the loop replay stage. The playback record cleanup mechanism is configured to automatically clean up the executed playback records corresponding to the historical playback time nodes and reset the synchronization state in response to the time rollback detection mechanism determining that the loop playback stage has been entered. This eliminates the interference of historical playback records on the event deduplication determination and the anti-collision interval calculation within the current loop cycle, ensuring the continuous synchronization accuracy of audio triggering and event timing in variable speed playback and loop playback scenarios.

7. The method according to claim 1, characterized in that, Before outputting the audio signal, the process further includes sequentially performing dynamic range compression, spatial reverberation processing, and delay processing on the synthesized audio signal.

8. The method according to claim 7, characterized in that, The method further includes: The playback of the audio signal triggers corresponding visual highlights and animation effects; The user's interactive operations on the visual interface affect the playback parameters of the audio signal in real time. The playback status remains consistent between audio playback and visual display.

9. A device for real-time audio playback of earthquake events, characterized in that, include: The data acquisition module is configured to acquire seismic event data, which includes multi-physical dimension parameters; The multidimensional mapping module is configured to map the earthquake event data to multidimensional audio perception attributes based on the multi-physical dimension parameters, wherein the corresponding timbre type is determined according to the range of the intensity parameter in the multi-physical dimension parameters. The synchronization control module is configured to perform synchronous playback control on the earthquake event data based on the playback speed. The synchronous playback control includes deduplication of the earthquake event data based on event identification information and setting an anti-collision interval to ensure the synchronization accuracy between audio triggering and event timing. The audio output module is configured to output the corresponding audio signal.

10. A real-time audio playback system for earthquake events, characterized in that, Includes the apparatus according to claim 9.

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