Environmental acoustic adaptive sound effect control method and sound based thereon

Abstract

This invention discloses an environmental acoustic adaptive sound effect control method, relating to the field of sound effect control. By integrating three modes—manual calibration, intelligent inspection, and event triggering—it can not only respond to user commands but also automatically trigger calibration when the speaker moves, there are sudden changes in environmental noise, playback mode switches, or a preset cycle is reached. This ensures that the speaker can proactively optimize the sound effect in a timely manner after changes in the usage environment, maintaining the speaker's sound effect at a suitable listening level. By quickly judging the suitability of the environment, it first assesses the background noise level, and only performs subsequent acoustic feature analysis when the environment is suitable for measurement. This avoids ineffective or even harmful calibration in high-noise environments, improving the reliability of system decisions and the accuracy of the final sound effect adjustment.

Description

Technical Field

[0001] This invention relates to the field of sound effect control, and in particular to a sound effect control method based on environmental acoustics and its audio system. Background Technology

[0002] The actual listening experience of a speaker is significantly affected by the acoustic characteristics of its placement environment and real-time ambient noise. The same speaker may produce muddy low frequencies, harsh high frequencies, or indistinct human voices in different environments. Traditional fixed sound effect presets cannot adapt to such dynamic changes.

[0003] There are two main types of adaptive sound technology commonly found on the market: one type relies on users to manually select preset modes, but it cannot perceive specific changes in the environment, and the adjustment is rough and inaccurate. Another type has automatic room correction, but it is usually only manually triggered during the initial setup or relies on a dedicated test microphone array, making it impossible to perform real-time and convenient recalibration when the speaker is moved, when there are sudden changes in ambient noise, or when the acoustic characteristics drift after long-term use.

[0004] Existing technologies lack adaptability, cannot automatically trigger optimization in a timely manner after environmental changes, and lack multimodal intelligent judgment of triggering conditions. Summary of the Invention

[0005] The purpose of this invention is to provide an environmental acoustic adaptive sound effect control method to solve the problems mentioned in the background art.

[0006] To achieve the above objectives, the present invention provides the following technical solution: an environmental acoustic adaptive sound effect control method, comprising the following steps: S1: In response to the trigger condition, collect the sound signal in the current environment; S2: Collect sound signals in the current environment and perform acoustic feature analysis on the sound signals to generate an acoustic feature analysis report; S3: Based on the acoustic feature analysis report and the preset audio playback mode target, a set of audio processing parameter instructions is generated; S4: Controls the speaker to adjust sound effects according to the audio processing parameter instruction set.

[0007] Preferably, based on step S1, the triggering condition includes at least one of the following modes: Manual calibration mode: Responds to active trigger commands issued by the user via physical buttons, touch interface, or application programming interface; Intelligent inspection mode: Automatic adjustment commands triggered by a timer based on a preset time period; Event-triggered mode: Responds to speaker displacement events detected by the accelerometer, events triggered by playback mode switching operations, or / and ambient noise spectrum abrupt events detected by the noise monitor.

[0008] Preferably, before step S1, a system initialization step is also included: Perform a power-on self-test on the speaker system. The self-test should include at least a check of the status of the pickup module, audio processing module, speaker unit, and main control unit. If the self-test passes, the system enters a low-power monitoring state and starts monitoring the trigger conditions. If the self-test fails, a fault alarm will be triggered and the sound effect adjustment process will be prevented from starting.

[0009] Preferably, step S1 specifically includes: Record the ambient audio for the current time period and calculate the background noise frequency of the current environment; Determine whether the peak background noise exceeds the masking threshold; If the noise level is not exceeded, a specific test signal is played, and the test sound after environmental reflection is recorded simultaneously. The background noise and the test sound are used together as the sound signal for acoustic feature analysis. If the limit is exceeded, the current adjustment process will be terminated or suspended.

[0010] Preferably, step S2 includes: Noise characteristic analysis: Spectral analysis of the background noise is performed to identify peak noise frequency bands with significantly higher energy than the background. Frequency response characteristic analysis: By comparing the spectrum of the test signal with that of the recorded test sound, identify the frequency response dips where the attenuation caused by the environment exceeds the threshold and / or the frequency response peaks where the gain exceeds the threshold. Reverberation characteristic analysis: The room impulse response is calculated based on the test signal and test sound, and then the reverberation time of the frequency band is analyzed.

[0011] Preferably, in step S3, the audio processing parameter instruction set includes adjustment instructions for at least one of the following parameters: gain parameter, equalizer parameter, and delay parameter.

[0012] Preferably, based on step S3, it includes priority decision logic: Determine if the ambient noise level is too high; If the ambient noise level is determined to be too high, the priority decision is to generate an instruction to increase the total output gain of the speaker. If the ambient noise level is determined to be not too high, then a decision is made based on the acoustic feature analysis report to attenuate or compensate the gain of a specific frequency band.

[0013] Preferably, the preset audio processing targets include music mode, movie mode, and voice mode; When in music mode, the audio processing focuses on high fidelity and a wide soundstage. When in movie mode, audio processing focuses on a wide dynamic range and immersive surround sound effects. When in voice mode, the audio processing focuses on improving mid-frequency clarity and voice intelligibility.

[0014] Preferably, the audio processing parameter instruction set also includes delay compensation parameters for multi-channel speaker systems; The decision generation steps include: analyzing the relative delay of sound signals arriving at microphones at different locations or between different channels; and when the delay exceeds the tolerance value, generating instructions to compensate for the delay of each channel.

[0015] A type of speaker that uses an environmental acoustic adaptive sound effect control method.

[0016] The technical effects and advantages of this invention are as follows: 1. By integrating three modes—manual calibration, intelligent inspection, and event triggering—it can not only respond to user commands but also automatically trigger calibration when the speaker is moved, there are sudden changes in environmental noise, playback mode is switched, or a preset cycle is reached. This ensures that the speaker can proactively optimize the sound effect in a timely manner after changes in the usage environment, maintaining the speaker's sound effect at a suitable listening level.

[0017] 2. By quickly assessing environmental suitability and evaluating the background noise level, subsequent acoustic feature analysis is only performed when the environment is suitable for measurement. This avoids ineffective or even harmful calibration in high-noise environments, improving the reliability of system decisions and the accuracy of final sound effect adjustment.

[0018] 3. The system operates in low-power monitoring mode by default, and the complete adjustment process is only triggered by specific conditions, rather than continuously performing high-power acoustic analysis. While ensuring intelligent adaptive capabilities, it significantly reduces the overall power consumption of the speaker.

[0019] 4. By combining the acoustic feature analysis results with specific playback mode goals for decision-making, sound effect adjustment can be targeted to solve specific problems, serve the fine-tuning of the current playback content, and meet the diverse scenario needs of users. Attached Figure Description

[0020] Figure 1 This is a schematic diagram of the overall process of the present invention; Figure 2 This is a flowchart of the sound signal acquisition and analysis process of the present invention; Figure 3 This is a flowchart of the system initialization steps of the present invention; Figure 4This is a flowchart of the acoustic feature analysis process of the present invention; Figure 5 This is a flowchart illustrating the decision-making process of the audio processing parameter instruction set of the present invention. Figure 6 This is a flowchart of the system control process of the present invention; Figure 7 This is a schematic diagram of the main structure of the speaker of the present invention. Detailed Implementation

[0021] The technical solutions of the embodiments of the present invention will be clearly and completely described below with reference to the accompanying drawings. Obviously, the described embodiments are only some embodiments of the present invention, and not all embodiments. Based on the embodiments of the present invention, all other embodiments obtained by those skilled in the art without creative effort are within the scope of protection of the present invention.

[0022] This invention provides, for example Figures 1-6 The environmental acoustic adaptive sound effect control method shown includes the following steps: S1: In response to the trigger condition, collect the sound signal in the current environment; S2: Collect sound signals in the current environment and perform acoustic feature analysis on the sound signals to generate an acoustic feature analysis report; S3: Based on the acoustic feature analysis report and the preset audio playback mode target, a set of audio processing parameter instructions is generated; S4: Controls the speaker to adjust sound effects according to the audio processing parameter instruction set.

[0023] Furthermore, the response of the trigger condition monitoring and judgment module can be divided into different working modes. Based on step S1, the trigger condition includes at least one of the following modes: Manual calibration mode: Responds to active trigger commands issued by the user via physical buttons, touch interface, or application programming interface; Intelligent inspection mode: Automatic adjustment commands triggered by a timer based on a preset time period; Event-triggered mode: Responds to speaker displacement events detected by the accelerometer, events triggered by playback mode switching operations, or / and ambient noise spectrum abrupt events detected by the noise monitor; The first manual calibration mode, which is specifically disclosed, is triggered by the user through the APP or a designated physical button via a user interface. It receives instructions from the mobile APP or physical button, and restarts the adjustment upon receiving the instructions. Under the action of the main control unit module, it detects the external environment and noise, and makes corresponding adjustments based on the decisions. The second intelligent inspection mode, which is specifically disclosed, is triggered by a set timer to deal with slow changes in the environment, such as acoustic characteristic drift caused by temperature and humidity, and to perform periodic maintenance fine-tuning. It records the last adjustment time and the adjustment interval through the timer. When the adjustment time is reached, the adjustment starts again. Under the action of the main control unit module, the external environment and noise are detected and decisions are made to make corresponding adjustments. The third event-triggered mode, which is specifically disclosed, is triggered by specific events, including displacement events, playback mode switching, and noise change events. When the built-in accelerometer detects that the speaker has been moved, a displacement event is detected, that is, the accelerometer detects that power is on. At this time, it is determined that displacement has occurred, which may be due to a change in the usage environment, and the adjustment will start again.

[0024] When users actively adjust the speaker's usage mode, such as switching from music mode to movie mode, this is a playback mode switch, and corresponding adjustments need to be made according to the environment.

[0025] With the help of the noise monitor, the background noise spectrum collected by the microphone is analyzed in real time. When the rate of change of the ambient noise spectrum continuously exceeds the threshold, the adjustment is restarted. In actual use, continuous monitoring will detect drastic changes in the ambient background noise spectrum, which can trigger a noise mutation event. In any of the three working modes mentioned above, when any condition is triggered, a trigger signal is sent to the main control unit module, and adjustment and testing are performed under the action of the main control unit module.

[0026] Before step S1, a system initialization step is also included: Perform a power-on self-test on the speaker system. The self-test includes at least checking the status of the pickup module, audio processing module, speaker unit, and main control unit. If the self-test passes, the system enters a low-power monitoring state and starts monitoring the triggering conditions. If the self-test fails, a fault alarm will be triggered and the sound effect adjustment process will be prevented from starting.

[0027] Specifically, when using the speaker, power it on, and a system initialization self-test is performed to determine whether the self-test passes. After the self-test passes, it enters the low power consumption monitoring state. In the low power consumption monitoring state, it reduces its power consumption. Under the action of the main control unit module, it controls the trigger condition monitoring and judgment module to start and perform corresponding monitoring to meet the low power consumption monitoring state. When the self-test fails, the speaker will issue a fault alarm and display an indicator light that flashes to indicate the fault. At this time, the speaker needs to be stopped for repair. The corresponding test should be performed according to the indication of the fault light to prevent incorrect adjustments in the faulty state.

[0028] Self-testing is performed to ensure that all critical hardware is in a working state, which is the foundation for reliable system operation. It checks the microphone circuit connection through the pickup module, performs a short recording and playback test to ensure normal sound output, checks the memory, storage and system clock through the main control unit, checks the communication status and initialization status of the audio processing module to ensure that they are normal, and performs transient current test through the speaker unit to ensure that the coil is normal.

[0029] After the self-test is completed, in the low power monitoring state, the main control unit issues a command to control the trigger condition monitoring and judgment module to monitor. When an abnormality is detected, the speaker sound effect is controlled. The speaker is equipped with an accelerometer, a timer, a noise monitor, and a user interface. Step S1 specifically includes: Record the ambient audio for the current time period and calculate the background noise frequency of the current environment; Determine whether the peak background noise exceeds the masking threshold; If the noise level is not exceeded, a specific test signal is played, and the test sound after environmental reflection is recorded simultaneously. The background noise and the test sound are used together as the sound signal for acoustic feature analysis. If the limit is exceeded, the current adjustment process will be terminated or suspended.

[0030] Specifically disclosed, the main control unit sends adjustment test commands to the sound pickup module. Under the action of the sound pickup module, feature acquisition and feature analysis are performed. This is divided into two steps. The first step is to record the ambient audio of the current time period, calculate the ambient background noise frequency, and determine whether the background noise peak exceeds the masking threshold. The masking threshold is a settable threshold. That is, if any value in the currently recorded audio segment exceeds the masking threshold, adjustment is required. If it does not exceed the masking threshold, no adjustment is required. The determination result is fed back to the main control unit module. The main control unit module performs the next control step based on its feature analysis. Furthermore, the second step is to play a specific test signal, which can be a swept tone or white noise, and simultaneously record the sound reflected by the room with high precision, and collect a pure ambient noise. The background noise and the test sound are used together as the sound signal for acoustic feature analysis. Based on step S2, including: Noise characteristic analysis: Spectral analysis of the background noise is performed to identify peak noise frequency bands with energy significantly higher than the background. Frequency response characteristic analysis: By comparing the spectrum of the test signal with that of the recorded test sound, identify the frequency response dips where attenuation due to the environment exceeds a threshold and / or the frequency response peaks where gain exceeds a threshold; Reverberation characteristic analysis: Based on the test signal and the test sound, the room impulse response is calculated, and then the reverberation time of the key frequency band is analyzed.

[0031] Specifically, the reflected sound is compared with pure ambient noise, the room impulse response is calculated, the reverberation time of each frequency band is analyzed, the noise spectrum is analyzed, the interference frequency band is identified, the frequency response is compared, the frequency response depression or peak caused by the environment is found, and the sound energy attenuation curve is generated. Specifically disclosed is the method for judging reverberation characteristics. This method involves performing Schroeder integral and other calculations on the impulse response, calculating the room impulse response based on the test signal and test sound, and calculating the reverberation time of key frequency bands such as low frequency, mid frequency, and high frequency. If the audio value of a certain frequency band exceeds the preset comfort threshold, it is marked as excessive reverberation in that frequency band, which needs to be adjusted. Noise characteristic judgment involves performing spectral analysis on the background noise to obtain its spectrum diagram, identifying peak noise frequency bands that are significantly higher than the background in the spectrum, and marking these frequency bands as having noise masking. Frequency response characteristic judgment: By comparing the spectrum of the test signal with that of the recorded test sound, frequency response dips and / or gain peaks that exceed the threshold due to environmental factors are identified. Sound field characteristic judgment: For multi-microphone or multi-speaker systems, analyze the time difference of the signal arriving at different locations to determine the sound field asymmetry.

[0032] The aforementioned feature analysis report is output to the main control unit module, which then outputs decision instructions.

[0033] It is particularly important to emphasize that a quick test can be performed first to determine whether the current environment is suitable for measurement. This can be done by recording the audio of the current time period and calculating whether the local noise frequency of the current environment exceeds the maximum threshold. If it exceeds the maximum threshold, it can be determined that the external environment is poor and cannot be controlled. If it is within the maximum threshold, it can be determined that the external noise is within the controllable range and can then be controlled.

[0034] The main control unit module will output decision instructions. By combining the sound pickup diagnostic report, user preferences, and task objectives, it will output the quality control instructions to the audio processing module, make corresponding judgments and decisions, and generate a set of audio processing parameter instructions. This will map acoustic problems into specific executable device operations, such as reducing the gain by 2dB at 100Hz, setting the audio equalizer value to 1.2, and ensuring the control between multiple adjustment items, ultimately achieving overall control and optimization of the speaker's sound effects. The preset audio processing targets include music mode, movie mode and voice mode. Corresponding settings are made in the corresponding modes. When in music mode, the audio processing targets focus on high fidelity and wide sound field performance. When in movie mode, audio processing focuses on a wide dynamic range and immersive surround sound effects. When in voice mode, the audio processing focuses on improving mid-frequency clarity and voice intelligibility.

[0035] Based on step S3, the audio processing parameter instruction set includes adjustment instructions for at least one of the following parameters: gain parameter, equalizer parameter, and delay parameter; Based on step S3, including priority decision logic: Determine if the ambient noise level is too high; If the ambient noise level is determined to be too high, the priority decision is to generate an instruction to increase the total output gain of the speaker. If the ambient noise level is determined to be not too high, then a decision is made based on the acoustic feature analysis report to attenuate or compensate the gain of a specific frequency band.

[0036] The audio processing parameter instruction set also includes delay compensation parameters for multi-channel speaker systems; The decision generation steps include: analyzing the relative delay of sound signals arriving at microphones at different locations or between different channels; and when the delay exceeds the tolerance value, generating instructions to compensate for the delay of each channel.

[0037] In the audio processing module, delay parameter decisions are made to determine whether there is a delay in the multi-channel audio. When a delay exists, the delay value of each channel is compensated, and the decision command is output to the main control unit module to control the sound effect adjustment. When there is no multi-channel delay, no adjustment is made. Under the gain parameter decision, it is determined whether the ambient noise is too high. If it is too high, the total gain is increased, and a decision command is output to the main control unit module to control the sound effect adjustment. If there is no excessive ambient noise, the equalization parameter decision is made to determine whether the frequency band reverberation is too heavy. If the frequency band reverberation is too heavy, the gain of that frequency band can be attenuated, and a decision command is output to the main control unit module to control the sound effect adjustment. If the frequency band reverberation is not too heavy, no adjustment is made. Based on step S3, including priority decision logic: Determine if the ambient noise level is too high; If the ambient noise level is determined to be too high, the priority decision is to generate an instruction to increase the total output gain of the speaker. If the ambient noise level is determined to be not too high, then based on the results of the frequency response characteristic analysis and / or the reverberation characteristic analysis, a decision is made to generate equalizer parameter values ​​to adjust the gain of a specific frequency band.

[0038] It is particularly important to emphasize that the gain parameter decision takes precedence over the equalization parameter decision. When the ambient noise is too high, the overall gain should be increased first. Only when the ambient noise is not too high should the gain of that frequency band be attenuated.

[0039] like Figure 7 Specifically, this is an embodiment of a sound system, which is controlled using an environmental acoustic adaptive sound effect control method; Finally, it should be noted that the above description is only a preferred embodiment of the present invention and is not intended to limit the present invention. Although the present invention has been described in detail with reference to the foregoing embodiments, those skilled in the art can still modify the technical solutions described in the foregoing embodiments or make equivalent substitutions for some of the technical features. Any modifications, equivalent substitutions, improvements, etc., made within the spirit and principles of the present invention should be included within the protection scope of the present invention.

Claims

1. A method for adaptive sound effect control based on environmental acoustics, characterized in that, Includes the following steps: S1: In response to the trigger condition, collect the sound signal in the current environment; S2: Collect sound signals in the current environment and perform acoustic feature analysis on the sound signals to generate an acoustic feature analysis report; S3: Based on the acoustic feature analysis report and the preset audio playback mode target, a set of audio processing parameter instructions is generated; S4: Controls the speaker to adjust sound effects according to the audio processing parameter instruction set.

2. The environmental acoustic adaptive sound effect control method according to claim 1, characterized in that, Based on step S1, the triggering conditions include at least one of the following modes: Manual calibration mode: Responds to active trigger commands issued by the user via physical buttons, touch interface, or application programming interface; Intelligent inspection mode: Automatic adjustment commands triggered by a timer based on a preset time period; Event-triggered mode: Responds to speaker displacement events detected by the accelerometer, events triggered by playback mode switching operations, or / and ambient noise spectrum abrupt events detected by the noise monitor.

3. The environmental acoustic adaptive sound effect control method according to claim 1, characterized in that, Before step S1, there is also a system initialization step: Perform a power-on self-test on the speaker system. The self-test should include at least a check of the status of the pickup module, audio processing module, speaker unit, and main control unit. If the self-test passes, the system enters a low-power monitoring state and starts monitoring the trigger conditions. If the self-test fails, a fault alarm will be triggered and the sound effect adjustment process will be prevented from starting.

4. The environmental acoustic adaptive sound effect control method according to claim 1, characterized in that, Step S1 specifically includes: Record the ambient audio for the current time period and calculate the background noise frequency of the current environment; Determine whether the peak background noise exceeds the masking threshold; If the noise level is not exceeded, a specific test signal is played, and the test sound after environmental reflection is recorded simultaneously. The background noise and the test sound are used together as the sound signal for acoustic feature analysis. If the limit is exceeded, the current adjustment process will be terminated or suspended.

5. The environmental acoustic adaptive sound effect control method according to claim 4, characterized in that, Based on step S2, including: Noise characteristic analysis: Spectral analysis of the background noise is performed to identify peak noise frequency bands with significantly higher energy than the background. Frequency response characteristic analysis: By comparing the spectrum of the test signal with that of the recorded test sound, identify the frequency response dips where the attenuation caused by the environment exceeds the threshold and / or the frequency response peaks where the gain exceeds the threshold. Reverberation characteristic analysis: The room impulse response is calculated based on the test signal and test sound, and then the reverberation time of the frequency band is analyzed.

6. The environmental acoustic adaptive sound effect control method according to claim 1, characterized in that, Based on step S3, the audio processing parameter instruction set includes adjustment instructions for at least one of the following parameters: gain parameter, equalizer parameter, and delay parameter.

7. The environmental acoustic adaptive sound effect control method according to claim 6, characterized in that, Based on step S3, including priority decision logic: Determine if the ambient noise level is too high; If the ambient noise level is determined to be too high, the priority decision is to generate an instruction to increase the total output gain of the speaker. If the ambient noise level is determined to be not too high, then a decision is made based on the acoustic feature analysis report to attenuate or compensate the gain of a specific frequency band.

8. The environmental acoustic adaptive sound effect control method according to claim 1, characterized in that, The preset audio processing targets include music mode, movie mode, and voice mode; When in music mode, the audio processing focuses on high fidelity and a wide soundstage. When in movie mode, audio processing focuses on a wide dynamic range and immersive surround sound effects. When in voice mode, the audio processing focuses on improving mid-frequency clarity and voice intelligibility.

9. The environmental acoustic adaptive sound effect control method according to claim 1, characterized in that, The audio processing parameter instruction set also includes delay compensation parameters for multi-channel speaker systems; The decision generation steps include: analyzing the relative delay of sound signals arriving at microphones at different locations or between different channels; and when the delay exceeds the tolerance value, generating instructions to compensate for the delay of each channel.

10. An audio system using the environmental acoustic adaptive sound effect control method according to any one of claims 1-9.

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