Debugging method and system of audio system

Abstract

The invention discloses a debugging method and system of an audio system, and relates to the technical field of audio debugging, the method comprises the steps of S1, environmental acoustic analysis, S2, basic parameter presetting, S3, real-time monitoring and dynamic adjustment, and S4, adaptive optimizing.The layout of an equalizer or a loudspeaker is adjusted through environmental acoustic analysis; presetting basic parameters of the sound amplifying device and the microphone; real-time multi-band sampling, delay calibration and gain regulation are realized; adaptive optimization is carried out, algorithms such as noise reduction are started, stored parameters are scene presetting, and one-key calling and self-checking updating are supported; the environment is analyzed, the audio system is accurately debugged, the system is dynamically optimized, the tone quality and the debugging efficiency are improved, and diversified scene requirements are met.

Description

Technical Field

[0001] The present invention relates to the technical field of audio debugging, and in particular to a debugging method and system for an audio system. Background Art

[0002] With the rapid iterative evolution of information technology, the market demand for audio quality and personalization is increasing, and audio system debugging is in urgent need of innovation; audio application scenarios are becoming increasingly diverse, and places such as indoor environments have increasingly stringent requirements for sound quality; traditional debugging methods rely too much on manual experience, are inefficient and easily affected by environmental interference, and are difficult to accurately deal with complex factors such as reverberation and noise. In order to meet the demand for high-quality audio in indoor environments and other scenarios, it is necessary to accurately analyze audio characteristics, quickly generate optimized debugging solutions, and improve debugging efficiency and accuracy; in order to promote the advancement of audio system debugging towards intelligence and precision, provide accurate and effective debugging solutions for various audio scenarios, and promote the development of audio system debugging towards precision and intelligence to meet the diverse and high-quality needs of modern audio applications and enhance the core competitiveness of the industry.

[0003] Existing technology, such as the invention application patent with announcement number: CN119342395A, discloses a debugging method and device for an audio system. This invention is directed to an audio system debugging method, which uses an array microphone to collect the spatial position information of a second test audio played by a sound amplification device, calculates the energy difference at each test point, and adjusts the playback gain so that the energy difference reaches a set threshold to achieve uniform calibration of the sound field; the frequency response can also be calibrated through the first test audio, and the array microphone can be calibrated based on parameters such as environmental noise, delay, and reverberation obtained based on the third test audio, to optimize noise reduction and echo cancellation, etc., thereby improving the performance of the audio system, ensuring a uniform sound field and accurate frequency response, and adapting to complex audio scenes.

[0004] Combining the above solutions, it can be found that the current audio system debugging methods have limitations. The existing technology lacks environmental adaptability and is easily disturbed by sudden environmental noise, which affects data accuracy. The test point setting is mechanical and does not take into account the actual scene layout. It lacks real-time performance and is prone to delays when processing real-time audio streams, affecting the immediate feedback and effect of debugging. Parameter adjustment is rigid, and calibration parameters are determined in a fixed order. If the frequency response and other changes occur later, it is difficult to adapt in real time. The level of intelligence is low, and the audio adjustment parameters cannot be quickly adjusted according to scene changes. Summary of the Invention

[0005] The purpose of the present invention is to provide a debugging method and system for an audio system, which solves the problems existing in the background technology.

[0006] To solve the above technical problems, the present invention adopts the following technical solution: The present invention provides a debugging method and system for an audio system, including: environmental acoustic analysis, basic parameter pre-adjustment, real-time monitoring and dynamic adjustment, and adaptive optimization.

[0007] S1. Environmental acoustics analysis: measuring the reverberation time of the indoor audio system, modeling background noise, and locating sound field defects. Based on this data, the speaker layout is adjusted to analyze the potential impact of the environment on sound quality.

[0008] S2. Pre-adjust basic parameters to obtain the power of the sound amplification device and the sensitivity parameters of the microphone in the indoor environment, preliminarily adjust the volume of the sound amplification device, microphone gain, adjust the microphone input sensitivity, and adjust the equalizer EQ.

[0009] S3: Real-time monitoring and dynamic adjustment, multi-band sampling, marking of abnormal audio and characteristic frequency bands, and delay calibration and gain control between microphones and amplification devices.

[0010] S4, adaptive optimization, automatically enables the noise reduction algorithm or adjusts the gain compression threshold to achieve dynamic adaptation, and stores the optimized parameters as scene presets, supporting one-click call or periodic self-check updates.

[0011] The beneficial effects of the present invention are: 1. The present invention performs reverberation duration measurement, background noise modeling and sound field defect location in environmental acoustic analysis, accurately analyzes environmental acoustic characteristics, provides data support for adjusting EQ and speaker layout, optimizes sound quality, and improves sound clarity and naturalness.

[0012] 2. The present invention uses multi-band sampling to mark abnormal audio during real-time monitoring and dynamic adjustment, performs delay calibration and gain control, solves audio problems in real time, and ensures stable audio output.

[0013] 3. The present invention automatically enables the noise reduction algorithm and adjusts the gain compression threshold in adaptive optimization to achieve dynamic adaptation, and stores the optimized parameters as scene presets. It supports one-click call and periodic self-check updates, making it convenient to quickly call the optimal parameters for different scenarios, maintain system high performance, and enhance the listening experience. BRIEF DESCRIPTION OF THE DRAWINGS

[0014] In order to more clearly illustrate the embodiments of the present invention or the technical solutions in the prior art, the following briefly introduces the drawings required for use in the embodiments or the description of the prior art. Obviously, the drawings described below are only some embodiments of the present invention. For ordinary technicians in this field, other drawings can be obtained based on these drawings without paying any creative work.

[0015] Figure 1 This is a connection diagram for executing the method of the present invention.

[0016] Figure 2 This is a schematic diagram of the system structure connection of the present invention. DETAILED DESCRIPTION

[0017] The technical solutions in the embodiments of the present invention will be clearly and completely described below in conjunction with the drawings in the embodiments of the present invention. Obviously, the described embodiments are only part of the embodiments of the present invention, rather than all the embodiments; based on the embodiments of the present invention, all other embodiments obtained by ordinary technicians in this field without making creative work are within the scope of protection of the present invention.

[0018] Reference Figure 1 As shown, the present invention provides a debugging method and system for an audio system, comprising: S1. Environmental acoustics analysis: measuring the reverberation time of the indoor audio system, modeling background noise, and locating sound field defects. Based on this data, the speaker layout is adjusted to analyze the potential impact of the environment on sound quality.

[0019] In a specific embodiment of the present invention, the reverberation duration is measured by the following specific steps: A1. Place the sound source, such as a speaker, at a typical location in the room, such as the center of the stage. A2. Place the measurement microphone at key locations within the audience area. For example, divide the room into six equal zones and place a microphone in the center of each zone, approximately 1.2 meters high to simulate human ear height. A3. Using the noise interruption method, play pink noise continuously until the sound field stabilizes, then suddenly turn it off. Measure the time it takes for the sound energy to decay to -60dB. A4. Multi-band measurement, divided into low frequency (125Hz), mid-frequency (500Hz), and high frequency (2kHz) measurements, because the reverberation characteristics of different frequency bands vary significantly; A5. Based on the obtained audio signal, use software to obtain the curve of sound pressure level attenuation over time, find the interval of the attenuation curve from -5dB to -35dB, calculate the slope and extrapolate it to -60dB, and get ,in are two time points on the sound pressure level attenuation curve, The time point when the starting decibel value is -5dB, The end point of decibel value is -35dB. For arrive The decibel attenuation is 60dB, and 60 is the target attenuation.

[0020] It should be noted that the reverberation time RT60 refers to the time required for the sound energy to decay by 60 decibels after the sound source stops making sound. It is a key parameter for measuring the acoustic characteristics of a space and directly affects the clarity, naturalness and listening experience of the sound. If the RT60 is too long, such as >1 second, the speech will produce reverberation superposition due to multiple reflections, resulting in blurred words. If the RT60 is too short, such as <0.3 seconds, the sound will appear dry and lack fullness, and the listener will easily fatigue. The RT60 is controlled at 0.4-0.8 seconds to balance clarity and naturalness. For example, through measurement, it was found that the RT60 of a certain indoor environment is 1.2 seconds, and sound-absorbing materials such as sound-absorbing cotton and curtains need to be installed to reduce it to 0.7 seconds. Pink noise is a noise signal with the same energy in each octave bandwidth over a wide frequency range.

[0021] In a specific embodiment of the present invention, the background noise modeling quantifies the continuous noise in the environment, such as air conditioning, ventilation system, and external traffic noise, when there is no signal input. A noise characteristic model is established through spectrum analysis and sound pressure level measurement to provide a basis for EQ adjustment or sound insulation design of the audio system.

[0022] B1. Use a sound level meter to continuously measure noise levels at key locations within the indoor environment, such as the front and middle rows of the auditorium, and in corners, for a period of time. Record the noise spectrum at each location. Import this data into acoustic simulation software to generate a noise distribution heat map showing the differences in noise intensity in different areas. Use an acoustic camera or mobile microphone array to identify major noise sources, such as air conditioning vents and door and window gaps, based on the sound pressure level distribution map. B2. Build a steady-state noise model: This involves taking multiple measurements, such as 10 samplings, and averaging the sound pressure levels in each frequency band to create a smooth noise spectrum. This allows the identification of the frequency bands of the primary noise source within the spectrum, specifically peak frequencies significantly above the background level, such as 120Hz air conditioning harmonics and 63Hz low-frequency resonances. If the air conditioning noise level at 120Hz is consistently 55dB, the model will label that frequency as the "primary noise source." B3. Build a transient noise model: Calculate peak sound pressure level, duration, and frequency; B4. Describe the noise energy distribution through the power spectral density (PSD) function. , Sort the collected discrete frequencies, , is any integer greater than 2, where is the noise intensity at the i-th frequency point, is the Dirac function, which represents the frequency There is a discrete noise peak at It is the energy distribution of broadband noise (such as airflow sound and environmental noise), usually represented by a smooth curve.

[0023] It should be noted that the statistical peak sound pressure level is the maximum instantaneous sound pressure level of the noise event; the duration specifically refers to the length of time from the appearance to the disappearance of the noise; and the frequency of occurrence is the number of times the noise event occurs per unit time, such as 5 times per hour.

[0024] In a specific embodiment of the present invention, the sound field defect positioning uses a sound field scanner or a mobile microphone array to detect the sound pressure level distribution and frequency response differences in the indoor environment; and mark areas with obvious sound focus, blind spots or standing waves.

[0025] C1. Play the test signal source through the main speakers, covering the full frequency range of 20 Hz-20 kHz. Use a mobile microphone array to collect data at multiple points in the audience area, such as the center of each row, corners, and the edge of the stage. C2. Record key parameters at each point, including sound pressure level, frequency response curve (1 / 3 octave analysis), and impulse response, for use in detecting the reflected sound path. C3. Import the data into acoustic software to generate a sound pressure level heat map, visually displaying high sound pressure areas (red) and low sound pressure areas (blue). Compare the frequency response curves at different locations and mark abnormal frequency bands. For example, if the low-frequency (100Hz) sound pressure level in one area is significantly higher than that in other locations, C4. Identify and mark defect types. Position and mark several different types of defects and adjust the speaker position and pointing angle based on the defect type.

[0026] It should be noted that several different types of defects and their adjustment measures include (1) sound focusing, where the sound energy is concentrated in a specific area due to concave reflective structures, such as domes and curved walls, resulting in an abnormal increase in the sound pressure level; by observing the local highlighted areas in the sound pressure level thermal map, impulse response analysis shows dense reflected sound, such as the superposition of multiple reflections; adjustment measures: physical adjustment, adding diffusers (such as QRD diffusers) to the reflective surface (such as curved walls) to break up the sound energy concentration; electroacoustic compensation, reducing the mid- and high-frequency gain in the speaker channel corresponding to the focus area; (2) Sound blind spots: The sound pressure level is significantly lower than that of the surrounding areas due to obstruction or excessive sound absorption; the sound pressure level heat map shows a continuous low-level area or the frequency response curve shows obvious attenuation of the entire frequency band; adjustment measures: optimize the speaker layout, add auxiliary speakers, such as ceiling speakers or supplementary speakers, to cover the blind spots; adjust the sound absorption material, reduce the sound absorption material near the blind spots, such as removing overly thick carpets; (3) Standing waves: due to the repeated reflection of low-frequency sound waves between parallel walls, fixed nodes and antinodes are formed, resulting in sharp peaks or valleys in the frequency response curve; measure the low-frequency response at different positions, such as 50-200Hz, and mark the frequency band with severe fluctuations; or use the moving microphone method, slowly move the microphone, and observe the periodic change of the sound pressure level with position; adjustment measures: low-frequency trap, install porous sound absorbers or Helmholtz resonators in the corners of the wall to absorb specific low frequencies; EQ adjustment, narrow-band attenuation of the standing wave frequency band; (4) Early reflected sound interference: The time difference between the reflected sound from the wall or ceiling and the direct sound is too small (<50ms), resulting in unclear speech. In the marked impulse response analysis, the reflected sound peak appears within 20ms after the direct sound. Adjustment measures: sound absorption treatment, install sound-absorbing panels (such as 25mm thick porous fiberboard) at the reflection point (such as the first reflection area on the side wall); electronic delay, add delay to the auxiliary speaker through DSP, so that the time difference between the reflected sound and the direct sound is >50ms.

[0027] S2. Pre-adjust basic parameters to obtain the power of the sound amplification device and the sensitivity parameters of the microphone in the indoor environment, preliminarily adjust the volume of the sound amplification device, microphone gain, adjust the microphone input sensitivity, and adjust the equalizer EQ.

[0028] In a specific embodiment of the present invention, the power of the sound amplifying device and the sensitivity parameters of the microphone in the indoor environment are obtained as follows: Use a power meter to measure the actual power of the sound amplification device; use a sound level meter and signal generator to measure the sensitivity of the microphone according to the standard test process.

[0029] Play a 1kHz sine wave through a signal generator and adjust the power amplifier so that the sound pressure level at the reference microphone is 94dB SPL (corresponding to 1Pa sound pressure). Connect the microphone to be tested to an audio interface or analyzer, record its output voltage, and calculate the microphone sensitivity. ,in is the output voltage, is the input sound pressure, which is the effective value of the pressure change caused by the sound wave passing through the medium.

[0030] In a specific embodiment of the present invention, the method of preliminarily adjusting the volume of the sound amplification device, the microphone gain, adjusting the microphone input sensitivity, and adjusting the equalizer EQ is as follows: D1. Calculate the required sound pressure level based on the indoor volume and purpose, such as a speech or music performance, using the speaker power formula: ,in is the target sound pressure level, V is the space volume, and Q is the loudspeaker directivity factor; Calculating Total Gain Requirements ,in is the target sound pressure level, is the microphone sensitivity, d is the average distance between the listener and the loudspeaker, R is the room constant, which is related to the sound absorption, and , S is the total surface area of the room, is the average sound absorption coefficient; D2. Sound amplification equipment adjustment: Adjust the channel gain knob to match the microphone output level. For dynamic microphones, slowly increase the input level to -20dBFS to -12dBFS to avoid excessive noise floor. For condenser microphones, after turning on 48V phantom power, set the gain to 30-40dB. Fine-tune the gain based on the pickup distance. For close-range pickup, such as a podium mic, reduce the gain by 10dB. For longer-range pickup, such as a handheld mic, increase the gain to ensure a stable vocal signal. D3. Microphone Adjustment: Switch the polar pattern. Cardioid or supercardioid polar patterns reduce noise from the podium and audience coughs. Omnidirectional is only used for close-range pickup in quiet environments, such as desktop gooseneck microphones. Supercardioid or sharp cardioid polar patterns are preferred for indoor environments to suppress side and rear reflections. D4. Adjust the equalizer EQ, optimize the frequency band and reverberation time based on the acquired data If the reverberation time is too long, add an attenuation filter in the corresponding frequency band, set Q=1-2, attenuate -2~-4dB, and suppress the superposition of sound wave reflections; if the reverberation time is too short, slightly increase the corresponding frequency band, such as +1~+2dB, to compensate for the lack of direct sound energy; Make different equalizer adjustments based on the main noise frequency band. For low-frequency noise: enable the high-pass filter (HPF) to cut off frequencies below 80Hz to prevent the microphone from picking up low-frequency vibration noise from air conditioners and projectors. For high-frequency noise: in the noise-concentrated frequency band, such as 6-16kHz, slightly attenuate by -1 to -2dB, reducing the gain at the corresponding frequency point to reduce the overlap of noise and useful signals, balance noise and voice clarity, and avoid excessive attenuation that may cause muffled sound. For narrowband noise, such as 50Hz power hum, use a notch filter for precise attenuation, setting a Q value ≥ 5 and attenuating by at least -10dB. To suppress howling, set Q=10 and attenuation of -6dB in the feedback-prone frequency band such as 3.15 kHz.

[0031] It should be noted that Q is the directivity factor of the loudspeaker, omnidirectional is 1, and horn loudspeaker is 10. The directivity factor is an important parameter to measure the degree of sound energy concentration of the loudspeaker in a specific direction. It reflects the distribution characteristics of its sound radiation energy in space. It means the ratio of the sound intensity of the loudspeaker in a specific direction, usually axial, that is, directly in front, to the average sound intensity of the omnidirectional (omnidirectional) sound source of the same power at the same distance. Howling refers to the sound signal received by the microphone in the audio system. After being amplified, it is played back through the loudspeaker, picked up by the microphone again and continuously amplified, resulting in a sharp, harsh high-frequency sound, which needs to be adjusted and suppressed.

[0032] S3: Real-time monitoring and dynamic adjustment, multi-band sampling, marking of abnormal audio and characteristic frequency bands, and delay calibration and gain control between microphones and amplification devices.

[0033] In a specific embodiment of the present invention, the multi-band sampling and marking of abnormal audio and characteristic frequency bands are performed in the following specific methods: E1. Signal crossover: Split the audio signal into multiple frequency bands, such as low frequency 20-200 Hz, mid-frequency 200-2 kHz, and high frequency 2-20 kHz. Use DSP filter modules or third-party crossover plug-ins, such as FabFilter Pro-Q3. E2. Multi-band real-time monitoring: Open a multi-window view in the spectrum analyzer to display the sound pressure level and frequency response curve of each frequency band, marking key parameters such as average energy (RMS level) and peak level (transient maximum), THD (total harmonic distortion), in order to detect distortion in each frequency band, such as low-frequency amplifier overload; E3. Abnormal audio marking: Enable the "peak hold" function of the spectrum analyzer to mark frequency bands that continuously exceed the standard, such as the howling frequency of 3.15kHz, and set a threshold alarm. Since howling is often caused by positive feedback in a specific frequency band, real-time filtering can quickly eliminate the problem of excessive loop gain.

[0034] In a specific embodiment of the present invention, the delay calibration is synchronized with the gain, and the specific method is as follows: Play test signals, such as warble tones, and use time alignment tools to measure the transmission delay between microphones and speakers. Adjust delay parameters in the digital signal processor to ensure audio-visual synchronization, such as lip synchronization in conferencing systems. Maintain a clear and stable sound field through automatic gain control, multi-band dynamic equalization, and feedback suppressor linkage adjustment, effectively responding to complex environmental changes in indoor environments, ensuring clear speech, natural music, and no howling interference.

[0035] S4, adaptive optimization, automatically enables the noise reduction algorithm or adjusts the gain compression threshold to achieve dynamic adaptation, and stores the optimized parameters as scene presets, supporting one-click call or periodic self-check updates.

[0036] In a specific embodiment of the present invention, the specific method for achieving dynamic adaptation is as follows: F1. Real-time noise reduction algorithm identifies ambient noise through spectrum analysis and dynamically suppresses it using an adaptive filter. It collects the ambient noise spectrum when there is no input signal and generates a noise model. It compares the input signal with the noise model in real time and inserts a notch filter to attenuate the noise by -6 to -12 dB in the dominant noise frequency band, such as 100-200 Hz. F2. Gain compression threshold adjustment: Automatically adjusts the compression threshold and compression ratio based on the dynamic range of the input signal to prevent clipping. Calculates the signal's RMS and peak levels in real time. If the signal is consistently below -20dBFS, lowers the threshold to increase sensitivity. If the signal frequently triggers clipping, raises the threshold and increases the compression ratio. F3. Adjust the noise reduction intensity, gain compression parameters, EQ curve, etc., and save them as preset modes, such as "Report Mode" and "Question and Answer Mode", which can be called up with one click through the mixer preset button or the central control panel.

[0037] It should be noted that clipping is a distortion phenomenon that occurs when the audio signal exceeds the maximum processing capacity of the system, which manifests as the waveform being "truncated", resulting in harsh sound or noise; dBFS is the unit for expressing signal level in digital audio systems, with 0dBFS as the maximum non-clipping level, and all other levels are expressed negatively based on this.

[0038] Reference Figure 2 As shown, the second aspect of the present invention provides a method for executing the mall rights and interests platform application system of the present invention, including: an environmental acoustic analysis module, which measures the reverberation time of the audio system of the indoor environment, models the background noise, locates the sound field defects, and adjusts the speaker layout to improve the impact of the environment on the sound quality; The basic parameter pre-adjustment module obtains the power and microphone sensitivity parameters of the indoor environment, preliminarily adjusts the volume of the loudspeaker, microphone gain, adjusts the microphone input sensitivity, and adjusts the equalizer EQ; Real-time monitoring and dynamic adjustment module performs multi-band sampling, marks abnormal audio and characteristic frequency bands, and performs delay calibration and gain control between the microphone and the sound amplification device; The adaptive optimization module automatically enables the noise reduction algorithm or adjusts the gain compression threshold to achieve dynamic adaptation, and stores the optimized parameters as scene presets, supporting one-click call or periodic self-check updates.

Claims

1. A method and system for debugging an audio system, characterized in that: include: S1. Environmental acoustics analysis: measuring the reverberation time of the audio system in the indoor environment, modeling background noise, locating sound field defects, adjusting speaker layout based on this data, and analyzing the potential impact of the environment on sound quality; S2. Pre-adjust basic parameters to obtain the power and microphone sensitivity parameters of the indoor environment, preliminarily adjust the volume of the loudspeaker, microphone gain, adjust the microphone input sensitivity, and adjust the equalizer EQ; S3: Real-time monitoring and dynamic adjustment, multi-band sampling, marking of abnormal audio and characteristic frequency bands, and delay calibration and gain control between microphones and amplification devices; S4, adaptive optimization, automatically enables the noise reduction algorithm or adjusts the gain compression threshold to achieve dynamic adaptation, and stores the optimized parameters as scene presets, supporting one-click call or periodic self-check updates.

2. The audio system debugging method and system according to claim 1, characterized in that: The reverberation duration measurement specifically comprises the following steps: A1. Place the sound source at a typical location in the indoor environment. A2. Place the measurement microphones at strategic locations in the audience area. A3. Using the noise interruption method, play pink noise continuously until the sound field stabilizes, then suddenly turn it off. Measure the time it takes for the sound energy to decay to -60dB. A4. Multi-band measurement, divided into low frequency, mid frequency and high frequency, because the reverberation characteristics of different frequency bands vary significantly; A5. Based on the obtained audio signal, use software to obtain the curve of sound pressure level attenuation over time, find the interval of the attenuation curve from -5dB to -35dB, calculate the slope and extrapolate it to -60dB, and get ,in are two time points on the sound pressure level attenuation curve, The time point when the starting decibel value is -5dB, The end point of decibel value is -35dB. For arrive The decibel attenuation is 60dB, and 60 is the target attenuation.

3. The audio system debugging method and system according to claim 1, characterized in that: The background noise modeling includes: B1. Use a sound level meter to continuously measure noise levels at multiple key locations within the indoor environment for a period of time, recording the noise spectrum at each point. Import this data into acoustic simulation software to generate a noise distribution heat map showing the differences in noise intensity in different areas. Use an acoustic camera or mobile microphone array to identify the main noise sources based on the sound pressure level distribution map. B2. Build a steady-state noise model. Through multiple measurements, average the sound pressure level in each frequency band to generate a smooth noise spectrum curve. Identify the frequency bands of the main noise sources in the spectrum, specifically the peak frequencies that are significantly higher than the background. B3. Build a transient noise model and calculate peak sound pressure level, duration, and frequency; B4. The noise energy distribution is described by the power spectral density (PSD) function. The specific formula is: , Sort the collected discrete frequencies, , is any integer greater than 2, where is the noise intensity at the i-th frequency point, is the Dirac function, which represents the frequency There is a discrete noise peak at is the energy distribution of broadband noise, usually represented by a smooth curve.

4. The audio system debugging method and system according to claim 1, characterized in that: The specific contents of the sound field defect positioning include: C1. Play the test signal source through the main speakers, covering the full frequency band, and use a mobile microphone array to collect data at multiple points in the audience area; C2. Record the key parameters of each point, including the sound pressure level, frequency response curve, and impulse response; C3. Import the data into acoustic software to generate a sound pressure level heat map, visually displaying high and low sound pressure areas. Compare the frequency response curves at different locations and mark abnormal frequency bands. C4. Identify and mark defect types. Position and mark several different types of defects and adjust the speaker position and pointing angle based on the defect type.

5. The audio system debugging method and system according to claim 1, characterized in that: The power of the sound amplification device and the microphone sensitivity parameters of the indoor environment are obtained as follows: Use a power meter to measure the actual power of the sound amplification device; use a sound level meter and signal generator to measure the sensitivity of the microphone according to the standard test process; Play a 1kHz sine wave through a signal generator and adjust the power amplifier so that the sound pressure level at the reference microphone is 94dBSPL. Connect the microphone to the audio interface or analyzer, record its output voltage, and calculate the microphone sensitivity. ,in is the output voltage, is the input sound pressure.

6. The audio system debugging method and system according to claim 1, characterized in that: The specific method for preliminarily adjusting the volume of the sound amplification device, the microphone gain, adjusting the microphone input sensitivity, and adjusting the equalizer EQ is as follows: D1. Calculate the required sound pressure level based on the indoor volume and usage, and combine it with the speaker power formula: ,in is the target sound pressure level, V is the space volume, and Q is the loudspeaker directivity factor; Calculating Total Gain Requirements ,in is the target sound pressure level, is the microphone sensitivity, d is the average distance between the listener and the loudspeaker, R is the room constant, which is related to the sound absorption, and , S is the total surface area of the room, is the average sound absorption coefficient; D2. Sound amplification equipment adjustment: Adjust the channel gain knob to match the microphone output level: For dynamic microphones, slowly increase the input level to -20dBFS to -12dBFS. For condenser microphones, after turning on 48V phantom power, set the gain to 30-40dB. For close-range miking, reduce the gain by 10dB, and increase the gain for long-range miking to ensure a stable vocal signal. D3. Microphone adjustment: switch the polar pattern to cardioid or supercardioid: reduce the noise from the podium and the coughing of the audience; Omnidirectional: Only used for close-range pickup in quiet environments. In indoor environments, supercardioid or sharp cardioid polar patterns are preferred to suppress reflected sound from the sides and rear. D4. Adjust the equalizer EQ, optimize the frequency band and reverberation time based on the acquired data If the reverberation time is too long, add an attenuation filter in the corresponding frequency band, set the Q value, attenuate dB, and suppress the superposition of sound wave reflections; if the reverberation time is too short, slightly increase the corresponding frequency band to compensate for the lack of direct sound energy; Different equalizer adjustments are made based on the main noise frequency bands. For low-frequency noise, the high-pass filter (HPF) is enabled to cut off frequencies below 80Hz to prevent the microphone from picking up low-frequency vibration noise from air conditioners and projectors. For high-frequency noise, the gain of the corresponding frequency points in the noise-concentrated frequency band is reduced to reduce the overlap of noise and useful signals, slightly attenuating the dB level to balance noise and voice clarity, and avoid excessive attenuation that can cause muffled sound. For narrowband noise, use a notch filter to precisely attenuate it, set the Q value, and attenuate in dB; To suppress howling, set the Q value and attenuate dB in the frequency band prone to feedback.

7. The audio system debugging method and system according to claim 1, characterized in that: The specific method of performing multi-band sampling and marking abnormal audio and characteristic frequency bands is as follows: E1. Signal frequency division: Split the audio signal into multiple frequency bands using DSP filter modules or third-party frequency division plug-ins; E2. Real-time monitoring of multiple frequency bands. Open a multi-window view in the spectrum analyzer to display the sound pressure level and frequency response curve of each frequency band, and mark key parameters. E3. To mark abnormal audio, enable the spectrum analyzer's "peak hold" function, mark the frequency band that continuously exceeds the standard, and set a threshold alarm.

8. The audio system debugging method and system according to claim 1, characterized in that: The delay calibration is synchronized with the gain, and the specific method is as follows: Play the test signal and use the time alignment tool to measure the transmission delay between the microphone and the speaker. Adjust the delay parameters in the digital signal processor to ensure sound and image synchronization. Maintain a clear and stable sound field through automatic gain control, multi-band dynamic equalization, and feedback suppressor linkage adjustment.

9. The audio system debugging method and system according to claim 1, characterized in that: The specific method for achieving dynamic adaptation is as follows: F1. Real-time noise reduction algorithm identifies ambient noise through spectrum analysis and dynamically suppresses it using an adaptive filter. It collects the ambient noise spectrum when there is no input signal and generates a noise model. It compares the input signal with the noise model in real time and inserts a notch filter in the noise-dominated frequency band. F2. Gain compression threshold adjustment: Automatically adjusts the compression threshold and compression ratio based on the dynamic range of the input signal to prevent clipping. Calculates the signal's RMS and peak levels in real time. If the signal is consistently below the minimum level, lowers the threshold to increase sensitivity. If the signal frequently triggers clipping, raises the threshold and increases the compression ratio. F3. Adjust noise reduction intensity, gain compression parameters, EQ curve, etc., save them as preset modes, and call them up with one click through the mixer preset button or the central control panel.

10. A system for executing the audio system debugging method according to any one of claims 1 to 9: The environmental acoustics analysis module measures the reverberation time of the indoor audio system, models background noise, locates sound field defects, and adjusts the speaker layout to improve the impact of the environment on sound quality. The basic parameter pre-adjustment module obtains the power and microphone sensitivity parameters of the indoor environment, preliminarily adjusts the volume of the loudspeaker, microphone gain, adjusts the microphone input sensitivity, and adjusts the equalizer EQ; Real-time monitoring and dynamic adjustment module performs multi-band sampling, marks abnormal audio and characteristic frequency bands, and performs delay calibration and gain control between the microphone and the sound amplification device; The adaptive optimization module automatically enables the noise reduction algorithm or adjusts the gain compression threshold to achieve dynamic adaptation, and stores the optimized parameters as scene presets, supporting one-click call or periodic self-check updates.

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