Reverberation equalization management data processing method and system for music intelligent terminal

Abstract

The application discloses a kind of reverberation equalization management data processing method and system for music intelligent terminal, it is related to data processing technical field, method includes: obtaining the audio signal to be played, obtaining low frequency band, middle frequency band and high frequency band, obtaining standard value;Emit test signal, collect real-time spectrum energy, obtain decay to point in time, obtain the decay duration of low frequency band, middle frequency band and high frequency band;Obtain reverberation requirement, obtain the frequency band to which audio interval belongs, obtain the decay duration of audio interval, judge whether decay duration satisfies reverberation requirement;If the decay duration of audio interval does not satisfy reverberation requirement, obtain the compensation ratio of audio interval according to the decay duration of audio interval, and obtain the target value corresponding to audio interval according to the compensation ratio corresponding to audio interval and standard value, and with target value as the output spectrum energy of audio interval.The application has the advantages of environmental universality, content adaptability and compensation accuracy.

Description

Technical Field

[0001] This invention relates to the field of data processing technology, specifically to a data processing method and system for reverb equalization management in a smart music terminal. Background Technology

[0002] With the popularization of smart music terminals (such as smart music consoles, smart speakers, or smart TVs), music playback in diverse acoustic environments (such as family living rooms, bathrooms, and small conference rooms) has become a daily necessity. However, under different environments and audio content, the environment can affect the reverberation effect, resulting in a significant decrease in music playback quality.

[0003] Specifically, due to the complexity of room size and material reflectivity, reverberation time (i.e., the time characteristic of sound energy decay) varies greatly in the low, mid, and high frequency ranges. For example, in small spaces such as bathrooms, the high-frequency reverberation time may be too short (less than 0.4s), resulting in a dry sound and a lack of spatiality; while in medium to large rooms, the low-frequency reverberation time may be too long (more than 1.5s), causing the sound to be muddy, trailing, and unclear. At the same time, music signals themselves have dynamic characteristics, and the reverberation requirements for different audio ranges (such as bass drum beats, mid-range vocals, and high-frequency instruments) are different. Some require attenuation to maintain clarity (such as speech audio), while others require moderate reverberation enhancement to enhance the sense of stereo (such as symphonic music). However, existing terminals use fixed equalizer presets or simple reverberation modes (such as "conference mode" or "music mode"), which rely on manual selection and cannot perceive and respond to the reverberation characteristics of the actual environment in real time. Finally, existing reverberation compensation is standardized and adjusts the frequency response globally (such as uniformly boosting the energy of all frequency bands), which lacks correlation with the actual playback signal and cannot adapt to changes in the environment, resulting in inaccurate compensation, causing overcompensation or undercompensation. Summary of the Invention

[0004] To address the technical problem in existing technologies that lack an automatic mechanism for real-time measurement of frequency band-specific decay time and binding of audio interval reverberation requirements before playback, resulting in poor sound quality adaptability and unsatisfactory user experience for music terminals, this invention provides a reverberation equalization management data processing method and system for music smart terminals.

[0005] A data processing method for reverb equalization management in a music smart terminal includes: acquiring the audio signal to be played from the music smart terminal, and acquiring the low-frequency, mid-frequency, and high-frequency bands corresponding to the audio signal to be played, and acquiring the standard value of its output spectral energy according to the music smart terminal; before playing the audio signal to be played, transmitting test signals of the low-frequency, mid-frequency, and high-frequency bands respectively, with the spectral energy of each band being the standard value, and acquiring the real-time spectral energy of the test signals in real time, acquiring the time point when the difference between the preset initial value and the real-time spectral energy exceeds the effective energy threshold and using it as the attenuation time point, acquiring the interval between the time point of transmitting the test signals of the low-frequency, mid-frequency, and high-frequency bands and the attenuation time point respectively, and using it as the attenuation duration of the low-frequency, mid-frequency, and high-frequency bands; acquiring the audio signal to be played from the music smart terminal; acquiring the standard value of the output spectral energy of the music smart terminal; before playing the audio signal to be played, transmitting test signals of the low-frequency, mid-frequency, and high-frequency bands respectively, and acquiring the standard value of the output spectral energy of the music smart terminal; ... The system calculates the reverberation requirements for each audio interval in the played audio signal, obtains the frequency band to which each audio interval belongs, and calculates the corresponding decay time for each audio interval based on the decay times of the low-frequency, mid-frequency, and high-frequency bands and the frequency band to which each audio interval belongs. It then determines whether the decay time of each audio interval in the played audio signal meets the reverberation requirements. If the decay time of the i-th audio interval in the played audio signal does not meet the reverberation requirements, the system calculates the compensation ratio for the i-th audio interval based on its decay time, and obtains the target value for the i-th audio interval based on the compensation ratio and the standard value. The target value is then used as the output spectral energy of the i-th audio interval.

[0006] Optionally, obtaining the reverberation requirements corresponding to each audio interval in the audio signal to be played includes: obtaining the minimum reverberation requirement duration and the maximum reverberation requirement duration corresponding to each audio interval in the audio signal to be played, forming a requirement duration interval from the minimum reverberation requirement duration and the maximum reverberation requirement duration, and using the requirement duration interval as the reverberation requirement.

[0007] Optionally, obtaining the attenuation duration corresponding to each audio range based on the attenuation durations of the low-frequency, mid-frequency, and high-frequency bands and the frequency band to which each audio range belongs includes: if the frequency band to which the audio range belongs is the low-frequency band, then the attenuation duration of the low-frequency band is used as the attenuation duration corresponding to that audio range; if the frequency band to which the audio range belongs is the mid-frequency band, then the attenuation duration of the mid-frequency band is used as the attenuation duration corresponding to that audio range; if the frequency band to which the audio range belongs is the high-frequency band, then the attenuation duration of the high-frequency band is used as the attenuation duration corresponding to that audio range.

[0008] Optionally, determining whether the decay duration corresponding to each audio interval in the audio signal to be played meets the reverberation requirements includes: if the decay duration corresponding to the audio interval is not less than the minimum reverberation requirement duration and not greater than the maximum reverberation requirement duration, then the decay duration corresponding to the audio interval meets the reverberation requirements; if the decay duration corresponding to the audio interval is less than the minimum reverberation requirement duration and greater than the maximum reverberation requirement duration, then the decay duration corresponding to the audio interval does not meet the reverberation requirements.

[0009] Optionally, obtaining the compensation ratio corresponding to the i-th audio interval based on the decay duration corresponding to the i-th audio interval includes: if the decay duration corresponding to the i-th audio interval is less than the minimum reverberation requirement duration, then subtract the decay duration corresponding to the i-th audio interval from the minimum reverberation requirement duration to obtain the duration interval corresponding to the i-th audio interval; if the decay duration corresponding to the i-th audio interval is greater than the maximum reverberation requirement duration, then subtract the decay duration corresponding to the i-th audio interval from the maximum reverberation requirement duration to obtain the duration interval corresponding to the i-th audio interval; obtaining the correction coefficient, and obtaining the compensation ratio corresponding to the i-th audio interval based on the correction coefficient, the decay duration corresponding to the i-th audio interval, and the duration interval.

[0010] Optionally, obtaining the target value corresponding to the i-th audio interval based on the compensation ratio and standard value corresponding to the i-th audio interval includes: multiplying the compensation ratio corresponding to the i-th audio interval by the standard value to obtain the compensation value corresponding to the i-th audio interval; and adding the compensation value corresponding to the i-th audio interval to the standard value to obtain the target value corresponding to the i-th audio interval.

[0011] A reverb equalization management data processing system for a music smart terminal is also provided. The system includes: a first data acquisition module, used to acquire the audio signal to be played from the music smart terminal, and to acquire the low-frequency band, mid-frequency band, and high-frequency band corresponding to the audio signal to be played, and to acquire the standard value of its output spectrum energy according to the music smart terminal; a second data acquisition module, used to transmit test signals of low-frequency band, mid-frequency band, and high-frequency band with standard spectral energy respectively before playing the audio to be played, and to collect the real-time spectrum energy of the test signal in real time, to acquire the time point when the difference between the preset initial value and the real-time spectrum energy exceeds the effective energy threshold and to use it as the attenuation time point, and to acquire the interval between the time point of transmitting the test signal of low-frequency band, mid-frequency band, and high-frequency band and the attenuation time point respectively and to use it as the attenuation duration of low-frequency band, mid-frequency band, and high-frequency band; The first data processing module is used to obtain the reverberation requirements corresponding to each audio interval in the audio signal to be played, and to obtain the frequency band to which each audio interval belongs in the audio signal to be played. Based on the decay duration of the low-frequency band, mid-frequency band, and high-frequency band and the frequency band to which each audio interval belongs, the module obtains the decay duration corresponding to each audio interval and determines whether the decay duration corresponding to each audio interval in the audio signal to be played meets the reverberation requirements. The second data processing module is used to obtain the compensation ratio corresponding to the i-th audio interval based on the decay duration corresponding to the i-th audio interval when the decay duration corresponding to the i-th audio interval in the audio signal to be played does not meet the reverberation requirements. Based on the compensation ratio corresponding to the i-th audio interval and the standard value, the module obtains the target value corresponding to the i-th audio interval and uses the target value as the output spectral energy of the i-th audio interval.

[0012] Optionally, the first data processing module is further configured to: obtain the minimum reverberation requirement duration and the maximum reverberation requirement duration corresponding to each audio interval in the audio signal to be played, and form a requirement duration interval by the minimum reverberation requirement duration and the maximum reverberation requirement duration, and use the requirement duration interval as the reverberation requirement.

[0013] Optionally, the second data processing module is further configured to: if the decay duration corresponding to the i-th audio interval is less than the minimum reverberation requirement duration, subtract the decay duration corresponding to the i-th audio interval from the minimum reverberation requirement duration to obtain the duration interval corresponding to the i-th audio interval; if the decay duration corresponding to the i-th audio interval is greater than the maximum reverberation requirement duration, subtract the decay duration corresponding to the i-th audio interval from the maximum reverberation requirement duration to obtain the duration interval corresponding to the i-th audio interval; obtain a correction coefficient, and obtain the compensation ratio corresponding to the i-th audio interval based on the correction coefficient, the decay duration corresponding to the i-th audio interval, and the duration interval.

[0014] Optionally, the second data processing module is further configured to: multiply the compensation ratio corresponding to the i-th audio interval by the standard value to obtain the compensation value corresponding to the i-th audio interval; and add the compensation value corresponding to the i-th audio interval to the standard value to obtain the target value corresponding to the i-th audio interval.

[0015] The beneficial effects of this invention are reflected in:

[0016] The entire music smart terminal uses a reverb equalization management data processing method to achieve closed-loop control of real-time analysis, precise quantification, and predictive adjustment. First, the song is broken down into independent musical phrases based on the natural pauses of the vocals. When the user sings the current phrase (the baseline section), the pitch data of the user and the original singer at multiple key nodes (time / feature points) are collected simultaneously. By calculating the average deviation exceeding the limit, a balance index reflecting the pitch stability of the entire phrase is generated. Furthermore, when a significant pitch deviation is detected (balance exceeds the standard), the volume of the original singer in the next phrase is immediately increased proportionally according to the degree of deviation (not exceeding the safety limit), providing targeted guidance for unfamiliar parts and improving the singing effect. Further, when the user sings accurately (balance meets the standard), the volume of the subsequent original singer is reduced proportionally according to the degree of performance excellence, significantly highlighting the user's vocals and transforming the existing passive response into active guidance. Furthermore, the proportionalization algorithm ensures that the volume change amplitude is precisely matched with the singing quality, eliminating auditory gaps caused by abrupt transitions. At the same time, multi-node aggregation analysis quantifies the overall performance at the micro level, avoiding misjudgment of single-point fluctuations. Attached Figure Description

[0017] To more clearly illustrate the specific embodiments of the present invention or the technical solutions in the prior art, the accompanying drawings used in the description of the specific embodiments or the prior art will be briefly introduced below. In all the drawings, similar elements or parts are generally identified by similar reference numerals. In the drawings, the elements or parts are not necessarily drawn to scale.

[0018] Figure 1 This is a partial flowchart of the reverberation equalization management data processing method for music smart terminals of the present invention;

[0019] Figure 2 This is a schematic diagram of another part of the reverberation equalization management data processing method for music intelligent terminals of the present invention.

[0020] Figure 3 This is a schematic diagram of another part of the reverberation equalization management data processing method for music intelligent terminals of the present invention.

[0021] Figure 4 This is a schematic diagram illustrating the steps of the reverberation equalization management data processing method for a music smart terminal according to the present invention;

[0022] Figure 5This is a schematic diagram of a portion of step S3 in the reverberation equalization management data processing method for music smart terminals of the present invention;

[0023] Figure 6 This is a schematic diagram of a portion of step S4 in the reverberation equalization management data processing method for music smart terminals of the present invention;

[0024] Figure 7 This is a schematic diagram of another part of step S4 in the reverberation equalization management data processing method for music intelligent terminals of the present invention. Detailed Implementation

[0025] To make the objectives, technical solutions, and advantages of the embodiments of the present invention clearer, the technical solutions of the embodiments of the present invention will be clearly and completely described below with reference to the accompanying drawings. Obviously, the described embodiments are only some embodiments of the present invention, and not all embodiments. The components of the embodiments of the present invention described and shown in the accompanying drawings can generally be arranged and designed in various different configurations.

[0026] Therefore, the following detailed description of the embodiments of the invention provided in the accompanying drawings is not intended to limit the scope of the claimed invention, but merely to illustrate selected embodiments of the invention. All other embodiments obtained by those skilled in the art based on the embodiments of the invention without inventive effort are within the scope of protection of the invention.

[0027] It should be noted that similar reference numerals and letters in the following figures indicate similar items; therefore, once an item is defined in one figure, it does not need to be further defined and explained in subsequent figures. Furthermore, the terms "first," "second," etc., are used only to distinguish descriptions and should not be construed as indicating or implying relative importance.

[0028] like Figure 1 , Figure 2 , Figure 3 and Figure 4 As shown, a data processing method for reverb equalization management in a smart music terminal is provided. In one embodiment, the method includes:

[0029] S1. Obtain the audio signal to be played from the music smart terminal, and obtain the low-frequency band, mid-frequency band and high-frequency band corresponding to the audio signal to be played, and obtain the standard value of its output spectrum energy according to the music smart terminal.

[0030] S2. Before playing the audio to be played, transmit test signals in the low-frequency, mid-frequency, and high-frequency bands respectively, with the spectral energy of each band being the standard value. Collect the real-time spectral energy of the test signals in real time, obtain the time point when the difference between the preset initial value and the real-time spectral energy exceeds the effective energy threshold, and use it as the attenuation time point. Obtain the interval between the time point when the test signals in the low-frequency, mid-frequency, and high-frequency bands are transmitted and the attenuation time point, and use it as the attenuation duration of the low-frequency, mid-frequency, and high-frequency bands respectively.

[0031] S3. Obtain the reverberation requirements corresponding to each audio interval in the audio signal to be played, and obtain the frequency band to which each audio interval belongs in the audio signal to be played. Based on the decay time of the low frequency band, mid frequency band and high frequency band and the frequency band to which each audio interval belongs, obtain the decay time corresponding to each audio interval, and determine whether the decay time corresponding to each audio interval in the audio signal to be played meets the reverberation requirements.

[0032] S4. If the decay duration corresponding to the i-th audio interval in the audio signal to be played does not meet the reverberation requirements, then obtain the compensation ratio corresponding to the i-th audio interval according to the decay duration corresponding to the i-th audio interval, and obtain the target value corresponding to the i-th audio interval according to the compensation ratio corresponding to the i-th audio interval and the standard value, and use the target value as the output spectral energy of the i-th audio interval.

[0033] In this embodiment, it should be noted that in S1, when the music smart terminal initiates reverb equalization management, step S1 is the prerequisite and data foundation for environmentally adaptive tuning. The core task includes three aspects: First, it is essential to accurately capture the information of the audio content itself to be played. Whether it's a song from a streaming service, a locally stored recording, or a voice command input by the user in real time, it will be received and parsed by the terminal into a processable audio signal to be played. This means acquiring a complete, unaffected native audio digital stream or file. Second, it is necessary to perform refined and structured spectral analysis on this entire audio signal. It does not treat the audio as a single whole, but rather dynamically and automatically divides the signal spectrum into acoustically significant key parts according to a pre-set proportional rule based on the highest frequency of the content (e.g., the upper limit of the song's spectrum might be 20kHz): namely, the low-frequency band covering deep fundamental sounds (e.g., low-frequency drum beats and bass), the mid-frequency band carrying core clear information (e.g., vocals and mid-frequency instrument melodies), and the high-frequency band shaping spatiality and brightness (e.g., cymbals and violin overtones). This segmentation is crucial because it reflects the fundamental physical differences in the propagation and reflection of different frequency components in space. The segmentation rules can be: low frequency (one-tenth of the highest frequency and below), mid frequency (one-tenth of the highest frequency and above, but less than half the highest frequency), and high frequency (half the highest frequency and above). Third, a standard value for the output spectral energy needs to be obtained. This standard value is the baseline energy level obtained by the smart terminal under the acoustic environment of its factory test. It represents the reliable energy output basis of the device under that acoustic environment. This standard value provides a reliable and quantifiable comparison anchor for subsequent measurements of the actual impact of the environment on sound (such as energy attenuation).

[0034] To further illustrate this, let's take a specific example from a complex home environment: Imagine a user playing a movie soundtrack on a smart TV in their irregularly shaped living room. In step S1, the movie soundtrack to be played is first captured, which contains rich elements—deep and powerful movie bass, clear dialogue and vocals, and highly expressive high-pitched string sections. The system immediately performs intelligent segmentation on the audio signal to be played: it detects the highest effective frequency of the entire piece of music (e.g., 5kHz), and then applies preset rules to classify frequency components below 0.5kHz (one-tenth of the highest frequency) as the crucial low-frequency band, which includes the basic subwoofer effects in movie scores and certain ambient low-frequency sounds; frequency components between 0.5kHz and 2.5kHz (one-tenth to one-half of the highest frequency) are defined as the mid-frequency band, which carries the vocal information that ensures the audience can clearly hear every line of dialogue and lyrics, as well as the core melodies of most instruments; while bright, crisp sound components above 2.5kHz (such as the overtones of the highest register of a violin and the sharp details of certain electronic synthesized sound effects) are classified as the high-frequency band. At the same time, it extracts the standard value of the output spectrum energy stored at the factory from the smart TV firmware.

[0035] In S2, the core measurement phase for environmental adaptation precisely quantifies the reverberation impact of the current specific acoustic environment on different key frequency bands before actual music playback. The execution process consists of two main actions. First, well-controlled test signals are actively emitted. These signals are strictly divided into independent low-frequency, mid-frequency, and high-frequency bands in the frequency domain, covering the main frequency range of the music signal. More importantly, the initial spectral energy of each frequency band test signal is strictly set and emitted according to the standard values ​​obtained in step S1, ensuring that the "source" energy of the test signal is known and consistent. Second, the built-in microphone array captures the sound of these test signals in real time within the playback space. The focus is on capturing the dynamic process of how sound energy decays in space over time. By continuously monitoring the acquired real-time spectral energy and comparing it with a preset effective energy threshold representing significant attenuation of sound energy, the precise time point at which the real-time spectral energy decays to a value exceeding the initial preset value (usually representing the peak value at the time of emission) exceeds this threshold is pinpointed; this point is the attenuation point. By recording the time difference between the start time of transmitting the test signal and the time when the decay reaches its peak, the decay time of the corresponding frequency band (low, mid, and high) in that specific environment is obtained—essentially a direct measurement of the reverberation time characteristics of the environment in that frequency band. This process is performed independently for each frequency band, thus accurately reflecting the differences in the rate of decay of different frequencies of sound in the current room.

[0036] Furthermore, let's illustrate the application of the S2 step using a smart speaker in a home environment. The user prepares to play music in a compact bathroom with hard-tiled walls. Before the smart speaker actually starts playing music, the S2 measurement is automatically triggered. It first emits a low-frequency test signal (e.g., simulating deep bass) with a standard energy level into the bathroom. The small space and hard walls and floor of a bathroom typically prolong the dissipation time of bass signals. Simultaneously, its microphone array closely monitors and captures the decay trajectory of this bass signal. It continues monitoring until the acquired bass energy level steadily decreases, eventually reaching a preset level indicating that the bass has essentially dissipated (effective energy threshold), at which point this point is recorded. Calculating the time interval from the emitted signal to this dissipation point reveals the longer bass decay time in the bathroom environment, reflecting a potential issue of excessively long low-frequency reverberation. Next, this process is repeated: a mid-frequency test signal (simulating human voice or mid-range musical instruments) with a standard energy value is emitted, its decay process is quickly captured, its dissipation point is determined, and the mid-frequency decay time is calculated; then a high-frequency test signal (simulating the sound of a triangle or cymbals) is emitted, its rapid dissipation process is captured (high frequencies dissipate extremely quickly in a small space on a hard surface), and a relatively short high-frequency decay time is calculated. This complete S2 process completes the real-time measurement of the sound energy decay time characteristics (i.e., reverberation characteristics) of the bathroom environment in the three key frequency bands of low, mid, and high frequencies before the music plays, laying the environmental state data foundation for subsequent intelligent compensation.

[0037] In S3, the audio content structure information parsed in step S1 (i.e., the segmented low-frequency, mid-frequency, and high-frequency bands and their respective audio intervals) is intelligently matched and analyzed with the current environmental reverberation characteristics (i.e., the actual decay time of the low, mid, and high frequencies in the environment) measured in real time in step S2. Specifically, S3 needs to know in advance the inherent "reverberation requirements" of each audio interval (such as a pure vocal dialogue, a drum rhythm, or a string melody)—this represents the desired reverberation effect for this type of sound content to achieve the ideal listening experience, usually manifested as an allowable decay time range (from the minimum reverberation requirement to the maximum reverberation requirement). Then, based on the frequency band (low-frequency, mid-frequency, or high-frequency) to which each audio interval belongs, it is directly assigned the decay time of that frequency band measured in S2 in the current actual environment. The final step is to compare the decay time assigned to the audio interval by the current environment with the ideal decay time range required by its content—to determine whether the decay time of the environment falls within the ideal range required by the audio interval. This matching analysis process is key to identifying which specific audio ranges will have "distorted" reverberation effects under the current acoustic environment. For example, human voices may become muddy and unclear due to excessive attenuation, or high notes of the violin may become dry and lack spatial sense due to excessively short attenuation, thus providing a basis for targeted compensation for S4.

[0038] Furthermore, let's take the example of playing a recorded speech with background music using a smart console in a small conference room with hard-tiled floors. In step S3, the different audio ranges contained in the recording are first identified: the speaker's vocals (mainly concentrated in the mid-frequency range), the gentle background strings (containing some high-frequency components), and the intermittent bass cue effects (belonging to the low-frequency range). It is clearly known that the vocals require a relatively short decay time (without trailing) to ensure clear pronunciation; the background strings need a slightly longer decay time to create a moderate sense of spatial surround sound; and the bass cue effects need to avoid an excessively long booming sound. Next, S3 assigned the environmental characteristic data measured by S2 in the conference room—extremely rapid attenuation of high frequencies (due to hard reflections), somewhat prolonged attenuation of low frequencies (due to low-frequency standing waves in a small space), and relatively short attenuation of mid-frequency frequencies—to the corresponding audio ranges: the vocal range inherited the short attenuation duration of the mid-frequency range, the high-frequency components in the background strings inherited the extremely short attenuation duration of the high-frequency range, and the bass cue sound inherited the prolonged attenuation duration of the low-frequency range. At this point, a judgment was made: the short attenuation duration of the vocals met the requirements and required no compensation; however, the extremely short attenuation duration inherited by the high-frequency components of the background strings was far below the relatively long attenuation limit required by the strings themselves, causing the string sound, which should have a sense of space, to sound thin and dry; simultaneously, the prolonged attenuation duration inherited by the bass cue sound exceeded the upper limit required to avoid a booming sound, resulting in a muddy and sluggish sound.

[0039] In S4, the i-th audio interval is any audio interval, where i is a positive integer and less than the number of audio intervals. S4 performs frequency-specific and dynamic energy compensation on audio intervals that S3 determines "does not meet reverberation requirements," thereby correcting the reverberation distortion caused by the current acoustic environment. Its compensation logic is based on the physical characteristics of the audio content and the attenuation characteristics of the environment. First, a compensation direction decision is made. For the i-th audio interval, if its actual attenuation time is shorter than the "minimum reverberation requirement time" needed for the content of that interval (indicating that the environment has consumed sound energy too early, resulting in a dry sound), then the output energy of that interval needs to be positively increased to compensate for the problem of the environment absorbing too quickly by enhancing the initial intensity of the sound source, thus prolonging the effective duration of the sound in that environment. Conversely, if the actual attenuation time is longer than the "maximum reverberation requirement time" (indicating that there is too much residual energy in the environment, resulting in a muddy trailing sound), then the output energy needs to be negatively reduced to weaken the energy input that may cause standing waves or echoes from the source.

[0040] Then, the target energy is calculated. Based on the compensation direction (positive or negative) determined by the above decision, and combined with the reference energy value (standard value) of this frequency band in a standard acoustic environment, the precise spectral energy (target value) that the audio range should output in the current environment to achieve the ideal reverberation effect is calculated. This compensation is non-uniform and content-bound, only acting on specific frequency ranges in the current audio signal that have been identified as needing adjustment (such as vocals and drum beats), rather than a global change, ensuring that the compensation accurately corresponds to the instantaneous characteristics and environmental feedback of the actual playback content.

[0041] To illustrate further, consider this example: a smart control console plays a recording of a speech containing clear vocal narration (dominantly mid-frequency) and soft background strings (containing high-frequency overtones). Analysis by S3 reveals that the actual attenuation time of the vocal section is shorter than the required minimum (the hard conference table causes excessively rapid mid-frequency absorption). According to S4 logic, the energy output of the vocal section needs to be positively increased. S4 will then calculate a target energy value slightly higher than the standard value based on the degree of deficiency and the standard value of the mid-frequency spectrum—a stronger initial vocal energy can counteract the excessively rapid absorption by the environment, ensuring sufficient clarity and continuity of speech even in a room with rapid reflections, preventing slurred pronunciation. The high-frequency overtones in the background strings attenuate much shorter than the required minimum (excessive high-frequency dissipation in a small space). S4 will significantly increase the target energy value of its high-frequency components—a higher initial energy output can slow down the dissipation rate of high frequencies in a hard environment, preventing the string overtones, which should create a sense of space, from sounding thin and lifeless, restoring a certain sense of airiness and surround sound. Sections not identified as distorted (such as low-frequency alerts if they do not exceed the upper limit) will maintain the standard output value to avoid unnecessary interference. The entire process runs dynamically in real time during music playback. If the environment changes suddenly (such as closing a door and changing the reverberation), the terminal will quickly adapt and adjust through a new round of S2-S4 cycles.

[0042] In summary, the data processing method for reverb equalization management in the entire music smart terminal firstly involves real-time acoustic measurements across frequency bands. Before playback, standardized test signals are actively emitted and energy attenuation trajectories are captured using array microphones, precisely quantifying the specific reverberation times (attenuation durations) of the low, mid, and high frequency bands in the current environment. Furthermore, through a content-driven reverberation demand binding mechanism, based on the analysis of the audio signal structure, the measured attenuation duration of each audio band is intelligently matched with the optimal reverberation range required by its specific content type (e.g., short attenuation for clarity in speeches, long attenuation for spatial effect in string music), accurately locating the specific range of distortion. The direction (too short leads to dryness, too long leads to muddiness) allows for precise compensation of the target. Furthermore, through a frequency band adaptive dynamic energy calibration mechanism, the target spectral energy is non-uniformly adjusted according to the distortion type (insufficient or excessive). For intervals requiring extended reverberation, the initial energy is positively increased to counteract excessive environmental absorption (e.g., high-frequency compensation for a bathroom violin), while for intervals requiring suppressed tailing, the energy is negatively reduced to block standing wave formation (e.g., low-frequency energy reduction in a conference room). The compensation value is dynamically calculated based on the device's standard value combined with environmental offset, ensuring that the output energy is always aligned with content requirements and environmental characteristics in real time, thus avoiding over / under-adjustment issues in global compensation to a certain extent. In summary, the entire implementation significantly improves the terminal's sound quality reproduction and user immersion in dynamic scenarios.

[0043] like Figure 5 As shown, in one embodiment, the reverberation requirements corresponding to each audio interval in the audio signal to be played in S3 include:

[0044] S31. Obtain the minimum and maximum reverberation requirement durations corresponding to each audio interval in the audio signal to be played, and form a requirement duration interval from the minimum and maximum reverberation requirement durations, and use the requirement duration interval as the reverberation requirement.

[0045] In this embodiment, it should be noted that in S31, a reverberation effect quantification standard is established for each audio interval. Based on the audio data of the audio signal to be played, the optimal decay time range required is matched. For example, speech dialogue requires a strict interval of short decay (to avoid overlapping of words), while the string section in an orchestra requires a relaxed interval of long decay (to enhance resonance). This required interval is represented as a closed interval of [minimum reverberation required duration, maximum reverberation required duration], serving as an objective benchmark for judging environmental adaptability.

[0046] For example, set quantization metrics for different audio types by presetting an acoustic database. For example, for the human voice dialogue in a conference recording, the upper limit is 0.6 s to prevent the trailing sound from being blurred (such as the trailing sound of the character "lun" in "discussion"), and the required range is [0.3 s, 0.6 s]. For example, for the bass drum in electronic dance music (frequency 80 Hz), the lower limit is 0.8 s to ensure the fullness of the drum sound, and the required range is [0.8 s, 1.2 s]. For another example, for the overtone of the violin in a symphony (frequency 8 kHz), the lower limit is 0.5 s to avoid dryness (such as losing the sense of air in a fast bowing segment), and the required range is [0.5 s, 0.9 s].

[0047] As Figure 5 shown, in one embodiment, obtaining the attenuation duration corresponding to each audio interval according to the attenuation durations of the low-frequency band, the middle-frequency band, and the high-frequency band and the frequency band to which each audio interval belongs in S3 includes:

[0048] S32. If the frequency band to which the audio interval belongs is the low-frequency band, use the attenuation duration of the low-frequency band as the attenuation duration corresponding to this audio interval;

[0049] S33. If the frequency band to which the audio interval belongs is the middle-frequency band, use the attenuation duration of the middle-frequency band as the attenuation duration corresponding to this audio interval;

[0050] S34. If the frequency band to which the audio interval belongs is the high-frequency band, use the attenuation duration of the high-frequency band as the attenuation duration corresponding to this audio interval.

[0051] In this embodiment, it should be noted that in S32, directly assign the full-environment low-frequency attenuation duration measured in S2 to all audio intervals belonging to the low-frequency band. Since the influence of room standing waves on low frequencies is global (such as muddy bass in a living room), this mapping can accurately reflect the distortion risk of all content components in this frequency band. Thus, it avoids repeated measurement of content in the same frequency band and ensures efficiency.

[0052] Further, for example, when the environmental low-frequency attenuation is 1.8 seconds, all bass drum beats and bass lines below 0.5 kHz in the played content inherit this value for subsequent judgment.

[0053] In S33, synchronize the actually measured environmental middle-frequency attenuation duration to all middle-frequency audio intervals. The middle frequency carries core information (such as human voices), and its attenuation characteristics directly affect clarity. Different from low frequencies, the middle-frequency attenuation is more affected by local materials (such as a conference table absorbing middle frequencies), and this mapping accurately captures the distortion of the key frequency band of speech.

[0054] Further, for example, when the actually measured middle-frequency attenuation in a conference room is 0.3 seconds, all human voice dialogues and middle guitar melodies in this environment use this value for evaluation.

[0055] In S34, the environmental high-frequency attenuation duration is bound to all high-frequency audio ranges. High-frequency sound waves are easily absorbed quickly in small spaces (such as bathrooms), and this mapping is directly related to the problem of a lack of spatial awareness. High-frequency attenuation is mainly determined by surface materials (tiles / carpets), and has a consistent impact on content within the same frequency band.

[0056] To illustrate further, for example, if the high-frequency attenuation is measured to be 0.2 seconds in a bathroom, all cymbals and violin overtones in that environment will be tested for compliance based on this value.

[0057] like Figure 5 As shown, in one embodiment, determining whether the decay duration corresponding to each audio interval in the audio signal to be played meets the reverberation requirements in step S3 includes:

[0058] S35. If the decay duration corresponding to the audio range is not less than the minimum reverberation requirement duration and not greater than the maximum reverberation requirement duration, then the decay duration corresponding to the audio range meets the reverberation requirements.

[0059] S36. If the decay duration corresponding to the audio range is less than the minimum reverberation requirement duration but greater than the maximum reverberation requirement duration, then the decay duration corresponding to the audio range does not meet the reverberation requirements.

[0060] In this embodiment, it should be noted that in S35, when the actual attenuation duration of a certain audio range simultaneously meets the following conditions: not lower than the minimum reverberation requirement duration for its content (to prevent dry sound), and not exceeding the maximum reverberation requirement duration (to prevent trailing and muddiness), it is determined that the current environment poses no risk of distortion to that range, and the original output is retained. Essentially, this is a "safety threshold" detection of acoustic effects, requiring no intervention only within the ideal range.

[0061] Furthermore, for example, if the required range of human voices in the conference room is [0.3s, 0.6s], and the ambient mid-frequency attenuation is 0.4 seconds, it is considered compliant, and the original energy output is maintained.

[0062] In S36, when the attenuation duration exceeds the required range (below the lower limit or above the upper limit), two types of distortion flags are triggered: below the lower limit, it indicates that the range is insufficient in sound energy due to excessively rapid environmental absorption (such as the high frequencies of string music in a bathroom being dry and lifeless); above the upper limit, it indicates that the sound is muddy due to energy retention (such as the trailing bass drum beats in a living room). This identifies the specific content components that need compensation and their distortion types, providing targeted instructions for the frequency band compensation in S4.

[0063] Furthermore, for example, the high-frequency requirement of background string music [0.5s, 0.8s], if the actual measurement in the bathroom environment is 0.2 seconds, it is marked as "insufficient high-frequency sound energy", triggering the positive compensation mechanism of S4.

[0064] like Figure 6As shown, in one embodiment, obtaining the compensation ratio corresponding to the i-th audio interval based on the attenuation duration corresponding to the i-th audio interval in S4 includes:

[0065] S41. If the decay duration corresponding to the i-th audio interval is less than the minimum reverberation requirement duration, then subtract the decay duration corresponding to the i-th audio interval from the minimum reverberation requirement duration to obtain the duration interval corresponding to the i-th audio interval.

[0066] S42. If the decay duration corresponding to the i-th audio interval is greater than the maximum reverberation requirement duration, then subtract the decay duration corresponding to the i-th audio interval from the maximum reverberation requirement duration to obtain the duration interval corresponding to the i-th audio interval.

[0067] S43. Obtain the correction coefficient, and obtain the compensation ratio corresponding to the i-th audio interval based on the correction coefficient, the attenuation duration and duration interval corresponding to the i-th audio interval.

[0068] In this embodiment, it should be noted that in S41, when the actual attenuation duration of the audio interval is lower than the lower limit of the content requirement, the "duration of the sound energy to be supplemented" is calculated. A positive duration interval is obtained by subtracting the measured attenuation duration from the minimum reverberation requirement duration. This value quantifies the time gap caused by the premature dissipation of ambient sound energy, such as the missing duration of a broken word ending in a heavily sound-absorbing room. Essentially, it transforms the acoustic defect into a quantifiable time compensation target.

[0069] Furthermore, for example, if the minimum attenuation of the speaker's voice is 0.3 seconds and the measured attenuation of the mid-frequency in the bathroom is 0.2 seconds, then the duration interval is +0.1 seconds (which needs to compensate for the 0.1-second sound energy continuity).

[0070] In S42, when the actual decay time of an audio interval exceeds the upper limit of the content requirement, the "excess dwell time to be reduced" is calculated. This is done by subtracting the measured decay time from the maximum required reverberation time, resulting in a negative duration interval. This value reflects the amount of residual ambient sound energy exceeding the ideal state, such as the redundant duration of a bass drum trailing in a living room. The negative sign indicates the direction of energy reduction to avoid confusion with positive compensation.

[0071] Furthermore, for example, if the maximum attenuation of bass sound effect is 1.0 second, and the actual measured attenuation of low frequency in the living room is 1.8 seconds, then the duration interval is -0.8 seconds (the 0.8-second energy retention needs to be eliminated).

[0072] In S43, based on the acoustic transmission model, the duration interval is converted into an energy correction ratio. First, a correction coefficient is introduced, determined by the device's acoustic characteristics (such as speaker efficiency and frequency response curve). Specifically, the correction coefficient is essentially a quantitative parameter of the device's energy conversion efficiency, reflecting the speaker's ability to convert electrical signals into sound energy. It is calibrated through three key characteristics: frequency response linearity (measured in an anechoic chamber to assess the device's energy conversion efficiency at various frequency bands, such as the low-frequency efficiency decay curve and high-frequency sensitivity decay characteristics); dynamic range upper limit (determining the maximum linear output threshold of the speaker at each frequency band; exceeding this threshold results in distortion); and spectral energy coupling characteristics (measuring the energy interference pattern when multiple frequency bands are output simultaneously, such as the modulation effect of low-frequency resonance on mid-to-high frequencies). For example, when the device needs to compensate for the high-frequency band in the bathroom (S42 detects a -0.2s interval): the high-frequency compensation coefficient 1.19 (factory calibration value) is obtained from a table, and the actual compensation ratio is calculated based on the audio interval duration interval, limiting the output energy to ≤98dB (high-frequency dynamic upper limit) to avoid distortion. Then, a proportional calculation is performed, combining the current decay duration with the duration interval, and generating a proportional value through a non-linear mapping relationship. In summary, this solves the complexity of the "time difference → energy difference" conversion, ensuring that the boost / reduction magnitude conforms to auditory perception characteristics (e.g., +0.1 seconds should correspond to approximately +3dB of energy).

[0073] Furthermore, for example, the +0.1 second interval of human voices in the conference room is compensated by a 0.25 compensation ratio (i.e., 25% energy needs to be increased) after calculation using the equipment correction factor.

[0074] like Figure 7 As shown, in one embodiment, obtaining the target value corresponding to the i-th audio interval in S4 based on the compensation ratio and standard value corresponding to the i-th audio interval includes:

[0075] S44. Multiply the compensation ratio corresponding to the i-th audio interval by the standard value to obtain the compensation value corresponding to the i-th audio interval;

[0076] S45. Add the compensation value corresponding to the i-th audio interval to the standard value to obtain the target value corresponding to the i-th audio interval.

[0077] In this embodiment, it should be noted that in S44, the abstract compensation ratio is transformed into a physically executable energy adjustment. First, a standard value is called to obtain the original output energy reference of this frequency band under standard conditions; then, the compensation amount is calculated, and the compensation ratio is multiplied by the standard value to obtain the specific absolute value of energy to be increased or decreased. The compensation amount dynamically scales with the basic performance of the device to avoid overload on different terminals.

[0078] Furthermore, if the intermediate frequency standard value is 100dB and the compensation ratio is 0.25, then the compensation value is +25dB (in actual execution, it is limited to the linear operating range of the equipment).

[0079] In S45, an output command that meets environmental requirements is generated. The compensation value is algebraically added to the standard value (positive compensation is added / negative compensation is subtracted); with the synthesized value as the target, the terminal speaker is driven to output the signal of that audio range; by changing the initial sound energy intensity, the sound dissipation behavior in the environment is indirectly controlled.

[0080] Furthermore, for example, if the standard value for the high-frequency range of a bathroom violin is 90dB, and the compensation value is +12dB, then the target output value is 102dB—higher initial energy delays the dissipation of high frequencies and restores the sense of space for overtones.

[0081] A reverb equalization management data processing system for music smart terminals is also provided, the system comprising:

[0082] The first data acquisition module is used to acquire the audio signal to be played from the music smart terminal, and to acquire the low-frequency band, mid-frequency band and high-frequency band corresponding to the audio signal to be played, and to acquire the standard value of its output spectrum energy according to the music smart terminal.

[0083] The second data acquisition module is used to transmit test signals with standard spectral energy in low-frequency, mid-frequency, and high-frequency bands respectively before playing the audio to be played, and to collect the real-time spectral energy of the test signals in real time. It obtains the time point when the difference between the preset initial value and the real-time spectral energy exceeds the effective energy threshold and uses it as the attenuation time point. It also obtains the interval between the time point of transmitting the test signals in low-frequency, mid-frequency, and high-frequency bands and the attenuation time point respectively and uses it as the attenuation duration of the low-frequency, mid-frequency, and high-frequency bands.

[0084] The first data processing module is used to obtain the reverberation requirements corresponding to each audio interval in the audio signal to be played, obtain the frequency band to which each audio interval belongs in the audio signal to be played, obtain the decay time corresponding to each audio interval according to the decay time of the low frequency band, mid frequency band and high frequency band and the frequency band to which each audio interval belongs, and determine whether the decay time corresponding to each audio interval in the audio signal to be played meets the reverberation requirements.

[0085] The second data processing module is used to obtain the compensation ratio corresponding to the i-th audio interval based on the attenuation time of the i-th audio interval when the attenuation time of the i-th audio interval in the audio signal to be played does not meet the reverberation requirements, and to obtain the target value corresponding to the i-th audio interval based on the compensation ratio and the standard value, and to use the target value as the output spectral energy of the i-th audio interval.

[0086] Optionally, the first data processing module is further configured to: obtain the minimum reverberation requirement duration and the maximum reverberation requirement duration corresponding to each audio interval in the audio signal to be played, and form a requirement duration interval by the minimum reverberation requirement duration and the maximum reverberation requirement duration, and use the requirement duration interval as the reverberation requirement.

[0087] Optionally, the second data processing module is further configured to: if the decay duration corresponding to the i-th audio interval is less than the minimum reverberation requirement duration, subtract the decay duration corresponding to the i-th audio interval from the minimum reverberation requirement duration to obtain the duration interval corresponding to the i-th audio interval; if the decay duration corresponding to the i-th audio interval is greater than the maximum reverberation requirement duration, subtract the decay duration corresponding to the i-th audio interval from the maximum reverberation requirement duration to obtain the duration interval corresponding to the i-th audio interval; obtain a correction coefficient, and obtain the compensation ratio corresponding to the i-th audio interval based on the correction coefficient, the decay duration corresponding to the i-th audio interval, and the duration interval.

[0088] Optionally, the second data processing module is further configured to: multiply the compensation ratio corresponding to the i-th audio interval by the standard value to obtain the compensation value corresponding to the i-th audio interval; and add the compensation value corresponding to the i-th audio interval to the standard value to obtain the target value corresponding to the i-th audio interval.

[0089] In this embodiment, it should be noted that the specific method of performing the operation of the above-mentioned music smart terminal reverb equalization management data processing system has been described in detail in the embodiment of the music smart terminal reverb equalization management data processing method, and will not be elaborated here.

[0090] The preferred embodiments of this disclosure have been described in detail above with reference to the accompanying drawings. However, this disclosure is not limited to the specific details of the above embodiments. Within the scope of the technical concept of this disclosure, various simple modifications can be made to the technical solutions of this disclosure, and these simple modifications all fall within the protection scope of this disclosure.

[0091] It should also be noted that the various specific technical features described in the above embodiments can be combined in any suitable manner without contradiction. To avoid unnecessary repetition, this disclosure will not describe the various possible combinations separately.

[0092] Furthermore, various different embodiments of this disclosure can be combined in any way, as long as they do not violate the spirit of this disclosure, they should also be regarded as the content disclosed in this disclosure.

[0093] Finally, it should be noted that the above embodiments are only used to illustrate the technical solutions of the present invention, and not to limit them. 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 or all of the technical features. Such modifications or substitutions do not cause the essence of the corresponding technical solutions to deviate from the scope of the technical solutions of the embodiments of the present invention, and they should all be covered within the scope of the claims and specification of the present invention.

Claims

1. A method for processing reverb equalization management data in a smart music terminal, characterized in that, include: Acquire the audio signal to be played from the music smart terminal, and acquire the low-frequency band, mid-frequency band and high-frequency band corresponding to the audio signal to be played, and acquire the standard value of its output spectrum energy according to the music smart terminal; Before playing the audio to be played, test signals with standard spectral energy in low-frequency, mid-frequency, and high-frequency bands are emitted respectively, and the real-time spectral energy of the test signals is collected in real time. The time point when the difference between the preset initial value and the real-time spectral energy exceeds the effective energy threshold is obtained and used as the attenuation time point. The interval between the time point of emitting the test signals in low-frequency, mid-frequency, and high-frequency bands and the attenuation time point is obtained respectively and used as the attenuation duration of low-frequency, mid-frequency, and high-frequency bands. The reverberation requirements for each audio range in the audio signal to be played are obtained, and the frequency band to which each audio range belongs is obtained. Based on the decay duration of the low-frequency band, mid-frequency band, and high-frequency band and the frequency band to which each audio range belongs, the decay duration corresponding to each audio range is obtained, and it is determined whether the decay duration corresponding to each audio range in the audio signal to be played meets the reverberation requirements. Among them, obtaining the reverberation requirements corresponding to each audio interval in the audio signal to be played includes: obtaining the minimum reverberation requirement duration and the maximum reverberation requirement duration corresponding to each audio interval in the audio signal to be played, and forming a requirement duration interval from the minimum reverberation requirement duration and the maximum reverberation requirement duration, and using the requirement duration interval as the reverberation requirement; The determination of whether the decay duration of each audio interval in the audio signal to be played meets the reverberation requirements includes: if the decay duration of the audio interval is not less than the minimum reverberation requirement duration and not greater than the maximum reverberation requirement duration, then the decay duration of the audio interval meets the reverberation requirements; if the decay duration of the audio interval is less than the minimum reverberation requirement duration or greater than the maximum reverberation requirement duration, then the decay duration of the audio interval does not meet the reverberation requirements. If the decay duration of the i-th audio interval in the audio signal to be played does not meet the reverberation requirements, the compensation ratio corresponding to the i-th audio interval is obtained according to the decay duration of the i-th audio interval, and the target value corresponding to the i-th audio interval is obtained according to the compensation ratio and the standard value, and the target value is used as the output spectral energy of the i-th audio interval.

2. The reverb equalization management data processing method for music intelligent terminals according to claim 1, characterized in that, The process of obtaining the attenuation duration corresponding to each audio range based on the attenuation durations of the low-frequency, mid-frequency, and high-frequency bands, as well as the frequency band to which each audio range belongs, includes: If the frequency band to which the audio range belongs is the low frequency band, then the decay time of the low frequency band will be used as the decay time corresponding to the audio range. If the frequency band to which the audio range belongs is the mid-frequency band, then the decay time of the mid-frequency band will be used as the decay time corresponding to the audio range. If the frequency band to which the audio interval belongs is the high frequency band, then the decay time of the high frequency band will be used as the decay time corresponding to the audio interval.

3. The reverb equalization management data processing method for music intelligent terminals according to claim 1, characterized in that, The step of obtaining the compensation ratio corresponding to the i-th audio interval based on the attenuation duration corresponding to the i-th audio interval includes: If the decay duration corresponding to the i-th audio interval is less than the minimum reverberation requirement duration, then the decay duration corresponding to the i-th audio interval is subtracted from the minimum reverberation requirement duration to obtain the duration interval corresponding to the i-th audio interval. If the decay duration corresponding to the i-th audio interval is greater than the maximum reverberation requirement duration, then the decay duration corresponding to the i-th audio interval is subtracted from the maximum reverberation requirement duration to obtain the duration interval corresponding to the i-th audio interval. Obtain the correction coefficient, and based on the correction coefficient, the attenuation duration and duration interval corresponding to the i-th audio interval, obtain the compensation ratio corresponding to the i-th audio interval.

4. The reverb equalization management data processing method for music intelligent terminals according to claim 1, characterized in that, The step of obtaining the target value corresponding to the i-th audio interval based on the compensation ratio and standard value corresponding to the i-th audio interval includes: Multiply the compensation ratio corresponding to the i-th audio interval by the standard value to obtain the compensation value corresponding to the i-th audio interval; The target value for the i-th audio interval is obtained by adding the compensation value corresponding to the standard value.

5. A reverb equalization management data processing system for a music smart terminal, characterized in that, The system includes: The first data acquisition module is used to acquire the audio signal to be played from the music smart terminal, and to acquire the low-frequency band, mid-frequency band and high-frequency band corresponding to the audio signal to be played, and to acquire the standard value of its output spectrum energy according to the music smart terminal. The second data acquisition module is used to transmit test signals with standard spectral energy in low-frequency, mid-frequency, and high-frequency bands respectively before playing the audio to be played, and to collect the real-time spectral energy of the test signals in real time. It obtains the time point when the difference between the preset initial value and the real-time spectral energy exceeds the effective energy threshold and uses it as the attenuation time point. It also obtains the interval between the time point of transmitting the test signals in low-frequency, mid-frequency, and high-frequency bands and the attenuation time point respectively and uses it as the attenuation duration of the low-frequency, mid-frequency, and high-frequency bands. The first data processing module is used to obtain the reverberation requirements corresponding to each audio interval in the audio signal to be played, obtain the frequency band to which each audio interval belongs in the audio signal to be played, obtain the decay time corresponding to each audio interval according to the decay time of the low frequency band, mid frequency band and high frequency band and the frequency band to which each audio interval belongs, and determine whether the decay time corresponding to each audio interval in the audio signal to be played meets the reverberation requirements. The first data processing module is also used to: obtain the minimum reverberation requirement duration and the maximum reverberation requirement duration corresponding to each audio interval in the audio signal to be played, and form a requirement duration interval by the minimum reverberation requirement duration and the maximum reverberation requirement duration, and use the requirement duration interval as the reverberation requirement; The first data processing module is also used to: if the decay time corresponding to the audio interval is not less than the minimum reverberation requirement time and not greater than the maximum reverberation requirement time, then the decay time corresponding to the audio interval meets the reverberation requirements; if the decay time corresponding to the audio interval is less than the minimum reverberation requirement time or greater than the maximum reverberation requirement time, then the decay time corresponding to the audio interval does not meet the reverberation requirements. The second data processing module is used to obtain the compensation ratio corresponding to the i-th audio interval based on the attenuation time of the i-th audio interval when the attenuation time of the i-th audio interval in the audio signal to be played does not meet the reverberation requirements, and to obtain the target value corresponding to the i-th audio interval based on the compensation ratio and the standard value, and to use the target value as the output spectral energy of the i-th audio interval.

6. The reverb equalization management data processing system for music intelligent terminals according to claim 5, characterized in that, The second data processing module is also used for: If the decay duration corresponding to the i-th audio interval is less than the minimum reverberation requirement duration, then the decay duration corresponding to the i-th audio interval is subtracted from the minimum reverberation requirement duration to obtain the duration interval corresponding to the i-th audio interval. If the decay duration corresponding to the i-th audio interval is greater than the maximum reverberation requirement duration, then the decay duration corresponding to the i-th audio interval is subtracted from the maximum reverberation requirement duration to obtain the duration interval corresponding to the i-th audio interval. Obtain the correction coefficient, and based on the correction coefficient, the attenuation duration and duration interval corresponding to the i-th audio interval, obtain the compensation ratio corresponding to the i-th audio interval.

7. The reverb equalization management data processing system for music intelligent terminals according to claim 5, characterized in that, The second data processing module is also used for: Multiply the compensation ratio corresponding to the i-th audio interval by the standard value to obtain the compensation value corresponding to the i-th audio interval; The target value for the i-th audio interval is obtained by adding the compensation value corresponding to the standard value.

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