Self-adaptive frequency compensation system for small multichannel sound equipment and control method of self-adaptive frequency compensation system
Through a small multi-channel audio adaptive frequency compensation system, FFT transformation and filter technology are used to identify and compensate for frequency loss in the audio system, solving sound quality problems and improving the applicability of the audio system and audio experience.
Patent Information
- Application Number
- CN202511002417.6
- Authority / Receiving Office
- CN · China
- Patent Type
- Applications(China)
- Current Assignee / Owner
- Filing Date
- 2025-07-21
- Publication Date
- 2025-10-17
AI Technical Summary
Existing audio systems cannot effectively compensate for the lack of bass and main sound frequencies, which affects the fullness, layering and realism of the sound quality and cannot meet users' demand for high-quality audio.
A small multi-channel audio adaptive frequency compensation system is designed. Through audio signal acquisition, frequency analysis, dynamic filtering and feedback control modules, missing frequencies are identified and compensated. FFT transform, IIR and FIR filters are used for real-time adjustment, and a feedback loop is constructed to optimize the frequency response.
It achieves high-quality audio output in different scenarios and audio content, improves the immersive experience and device applicability, reduces costs, and enhances product competitiveness and brand image.
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Figure CN120812476A_ABST
Abstract
Description
TECHNICAL FIELD
[0001] The present application relates to the technical field of audio signal processing, and in particular to a small multi-channel sound adaptive frequency compensation system and a control method thereof. BACKGROUND
[0002] With the improvement of people's requirements for audio experience, people hope that the sound can restore the sound more accurately and completely. As an important part of music and audio content, the frequency loss of bass and main sound will seriously affect the fullness, level and reality of sound quality. Therefore, technical means need to be taken to compensate for the frequency loss of bass and main sound to meet the needs of users for high-quality audio. SUMMARY
[0003] The present application provides a small multi-channel sound adaptive frequency compensation system, which identifies and compensates for missing frequencies by analyzing, filtering, equalizing and other processing of audio signals.
[0004] A small multi-channel sound adaptive frequency compensation system designed for this purpose includes an audio signal acquisition module, a frequency analysis module and a dynamic filter compensation module in electrical communication with each other; The audio signal acquisition module is used to acquire audio signals through an ADC module at a preset sampling rate; The frequency analysis module is used to perform FFT transform on the acquired audio signals to calculate the energy distribution of the corresponding frequency band and identify the missing or attenuated frequency range; The dynamic filter compensation module is used to select a FIR or IIR filter according to the frequency analysis result, dynamically adjust the filter coefficients and gain value, and perform real-time filter processing on the audio signal to output the compensated audio signal.
[0005] The small multi-channel sound adaptive frequency compensation system further includes a feedback and dynamic control module, and the audio signal acquisition module, the frequency analysis module, the dynamic filter compensation module and the feedback and dynamic control module are in electrical communication with each other; The feedback and dynamic control module is used to acquire the compensated audio signal by the ADC module, perform frequency analysis again, compare the frequency response before and after compensation, and realize dynamic adjustment of filter parameters and gain value through adaptive filter algorithm.
[0006] The feedback and dynamic control module is arranged on the audio output end and constitutes a feedback loop.
[0007] The small multi-channel sound adaptive frequency compensation system further includes a digital audio processor for assisting the audio signal acquisition module; The digital audio processor is provided with an input channel; the input channel is equipped with a first equalizer with adjustable parameters for preliminary frequency response correction; The input channel is provided with 8, each of which is equipped with 5 first equalizers.
[0008] The input channel of the digital audio processor is also equipped with a high-low pass filter that can filter out interference signals of ultra-low and / or ultra-high frequencies; each input channel is equipped with 5 high-low pass filters.
[0009] The input channel of the digital audio processor is also equipped with a noise gate processing module for suppressing ambient noise below a threshold to improve the signal-to-noise ratio; each input channel is equipped with 5 noise gate processing modules.
[0010] The small multi-channel sound self-adaptive frequency compensation system also includes a digital audio processor for assisting the dynamic filter compensation module; The dynamic filter compensation module includes an infinite impulse response or finite impulse response low-pass or band-pass digital filter in a low-frequency part missing state; The digital audio processor is provided with an output channel, and each output channel is equipped with a second equalizer with adjustable parameters to finely adjust the compensated audio signal again.
[0011] The output channel is provided with 8, each of which is equipped with 15 second equalizers.
[0012] A control method of the small multi-channel sound self-adaptive frequency compensation system as described above, comprising the following steps: Acquire the audio signal through the ADC; Convert the audio signal from the time domain to the frequency domain using FFT, and analyze the energy distribution of each frequency band; Design an IIR or FIR digital filter according to the analysis result, and adjust the filter coefficient for gain compensation; Input the audio signal into the filter for real-time processing; Acquire the compensated signal and perform frequency analysis again.
[0013] The FFT transform decomposes the audio signal into multiple frequency components; An IIR or FIR low-pass or band-pass filter is used when the low-frequency part is missing; Quantify the compensation effect by calculating MSE or PSNR; Use the parametric equalizer of the digital audio processor for processing.
[0014] The beneficial technical effects of the present application are as follows: 1. Enhance the immersive experience: The complete frequency response creates a more realistic audio environment for the audience. When watching a movie, the compensated sound system can accurately reproduce low-frequency effects such as explosion sounds, footsteps, and main sound parts such as character dialogues, making the audience feel as if they are in the movie scene and obtaining an immersive experience.
[0015] 2. Adapt to multiple scenarios: This technology can automatically adapt to different playing environments and audio content. Whether enjoying home theater in the living room or holding a party outdoors, the sound system can perform frequency compensation according to the actual situation, ensuring stable and high-quality audio output in various scenarios, improving the applicability and flexibility of the device.
[0016] 3. Improve product competitiveness: Sound products with frequency compensation function have obvious advantages in sound quality, which can attract more consumers and improve the competitiveness of products in the market. This helps sound manufacturers establish a good brand image, expand market share, and achieve higher economic benefits.
[0017] 4. Reduce costs: Compared with developing new sound hardware to improve sound quality, the cost of using frequency compensation technology is lower. Sound manufacturers can integrate this technology into existing products to improve product performance at a lower cost, upgrade products, and improve production efficiency. BRIEF DESCRIPTION OF DRAWINGS
[0018] Figure 1 The figure illustrates the frequency loss mentioned in the background technology.
[0019] Figure 2 The figure illustrates the flow control principle of an embodiment of the present application. DETAILED DESCRIPTION
[0020] The technical solutions in the embodiments of the present application will be described in detail below. Obviously, the described embodiments are only a part of the embodiments of the present application, not all. In order to make the above purposes, features and advantages of the present application more obvious and easy to understand, many specific details are described in the following description to facilitate a full understanding of the present application. However, the present application can be implemented in many other ways different from the description, and those skilled in the art can make similar improvements without departing from the spirit of the present application, so the present application is not limited to the specific embodiments disclosed below.
[0021] Reference Figures 1-2Audio signal processing technology, through the analysis, filtering, equalization and other processing of audio signal, to identify and compensate for missing frequencies. For example, using digital signal processor (DSP) for real-time processing of audio signals, according to the preset frequency response curve, the missing frequency of bass and main sound part is adjusted to restore the balance and integrity of the sound.
[0022] A small multi-channel sound adaptive frequency compensation system, comprising an audio signal acquisition module, a frequency analysis module and a dynamic filter compensation module in electrical communication with each other; The audio signal acquisition module is used to collect audio signals through the ADC module at a preset sampling rate; The frequency analysis module is used to perform FFT transform on the collected audio signals to calculate the energy distribution of the corresponding frequency band and identify the missing or attenuated frequency range; The dynamic filter compensation module is used to select FIR or IIR filter according to the frequency analysis result, dynamically adjust the filter coefficient and gain value, and perform real-time filter processing on the audio signal to output the compensated audio signal.
[0023] The audio signal acquisition module collects audio signals through the ADC module at a preset sampling rate, providing the original data basis for subsequent processing. The frequency analysis module performs FFT transform on the collected audio signals, which can accurately calculate the energy distribution of the corresponding frequency band, so as to quickly and accurately identify the missing or attenuated frequency range, providing a key basis for frequency compensation. The dynamic filter compensation module can flexibly select FIR or IIR filter according to the frequency analysis result, and dynamically adjust the filter coefficient and gain value to perform real-time filter processing on the audio signal to output the compensated audio signal. The coordinated work of this series of modules realizes the complete process from audio signal acquisition, analysis to compensation, which can effectively compensate for the missing bass and main sound frequency in music and audio content, improve the fullness, level and reality of sound quality, and meet the user's demand for high-quality audio. At the same time, the modular design makes the system structure clear, easy to maintain and upgrade, and each module has clear division of labor, which can fully exert its own functional advantages and improve the overall performance and stability of the system.
[0024] The small multi-channel sound adaptive frequency compensation system further comprises a feedback and dynamic control module, and the audio signal acquisition module, the frequency analysis module, the dynamic filter compensation module and the feedback and dynamic control module are in electrical communication with each other; The feedback and dynamic control module is used to collect the compensated audio signal by the ADC module, perform frequency analysis again, compare the frequency response before and after compensation, and realize dynamic adjustment of filter parameters and gain value through adaptive filter algorithm.
[0025] The feedback and dynamic adjustment module collects the compensated audio signal through the ADC module, performs frequency analysis again, compares the frequency response before and after compensation, and realizes dynamic adjustment of filter parameters and gain value by using adaptive filtering algorithm. The introduction of this feedback mechanism enables the system to monitor the effect of frequency compensation in real time, dynamically optimize the filter parameters and gain value according to the actual situation. During audio playback, different audio content and playback environment may affect the frequency compensation effect, while the feedback and dynamic adjustment module can perceive these changes in time and make corresponding adjustments to ensure the accuracy and effectiveness of frequency compensation. Compared with the system without feedback mechanism, this system can continuously adapt to various complex audio playback scenarios, avoid over or under compensation, further improve the restoration degree and stability of sound quality, and provide users with a better and more stable audio experience.
[0026] The feedback and dynamic adjustment module is set on the audio output end and constitutes a feedback loop. This setting method can directly obtain the final output audio signal for analysis, ensuring that the feedback signal is the result of complete processing, making the frequency analysis and parameter adjustment more targeted and accurate. Since the signal at the audio output end is closest to the sound actually heard by the user, feedback adjustment based on this can maximize the accuracy of the compensated audio to meet the user's hearing needs. At the same time, the design of the feedback loop makes the system form a closed-loop control system, and the signal transmission and processing between modules are more closely and continuously, which can quickly respond to changes in audio signals, adjust filter parameters and gain value in time, effectively avoid the problem of inaccurate frequency compensation caused by signal delay or information error, further improve the adaptive ability of the system and the accuracy of frequency compensation, and provide users with better audio quality.
[0027] The small multi-channel sound system adaptive frequency compensation system also includes a digital audio processor for assisting the audio signal acquisition module. The digital audio processor is provided with an input channel; the input channel is equipped with a first equalizer with adjustable parameters for preliminary frequency response correction. The input channel is provided with 8, and each input channel is equipped with 5 first equalizers.
[0028] The digital audio processor is provided with input channels, and the input channels are equipped with first equalizers with adjustable parameters for preliminary frequency response correction. Each input channel is equipped with 8 first equalizers, and each input channel is equipped with 5 first equalizers. The first equalizers can perform preliminary frequency response correction on the audio signals after the audio signals are collected, adjust the gain of different frequency bands in the audio signals, and make the audio signals have a more reasonable frequency distribution before entering the subsequent processing modules. This helps to reduce the processing pressure of the subsequent frequency analysis module and dynamic filter compensation module, and improves the processing efficiency of the entire system. At the same time, the design of adjustable parameters enables the first equalizers to flexibly adjust the degree of frequency response correction according to different audio sources and user needs, enhancing the adaptability of the system to different audio signals. The configuration of multiple first equalizers can finely adjust multiple frequency bands of the audio signals and optimize the frequency response of the audio signals from multiple dimensions, laying a good foundation for subsequent frequency compensation and further improving the quality of the audio signals and the effect of frequency compensation.
[0029] The input channels of the digital audio processor are also equipped with high-low pass filters that can filter out interference signals of ultra-low frequency and / or ultra-high frequency; each input channel is equipped with 5 high-low pass filters.
[0030] During the collection and transmission of audio signals, ultra-low frequency and ultra-high frequency interference signals will inevitably be mixed in, which will affect the purity of the audio signals and the accuracy of the frequency analysis. The high-low pass filter can effectively filter out these interference signals and retain the useful frequency components in the audio signals, making the audio signals input to the subsequent modules more pure and accurate. Each input channel is equipped with multiple high-low pass filters, which can accurately filter interference signals in different frequency ranges and improve the filtering effect. The audio signals processed by the high-low pass filter can reduce the influence of interference on frequency analysis and compensation, so that the frequency analysis module can more accurately identify the missing or attenuated frequency range, and the dynamic filter compensation module can more effectively perform frequency compensation, thereby improving the quality and stability of the entire system for audio signal processing and presenting clearer and higher-quality audio effects to users.
[0031] The input channels of the digital audio processor are also equipped with noise gate processing modules for suppressing environmental noise below a threshold to improve the signal-to-noise ratio; each input channel is equipped with 5 noise gate processing modules.
[0032] In an actual audio playing environment, environmental noise is one of the important factors affecting the quality of audio. The noise gate processing module can automatically detect and suppress environmental noise below the threshold by setting the threshold, and only allow useful audio signals above the threshold to pass. This processing method can effectively improve the signal-to-noise ratio of the audio signal, making the audio content more clear and distinguishable. The configuration of multiple noise gate processing modules can suppress environmental noise of different intensity and frequency characteristics in multiple dimensions, improving the effect of noise suppression. After the audio signal is processed by the noise gate processing module, the interference of environmental noise on the quality of the audio signal can be reduced, the frequency analysis module can more accurately analyze the frequency components of the audio signal, and the dynamic filter compensation module can more accurately compensate the frequency of the audio signal, thereby providing users with a more pure, clear and high-quality audio experience, and effectively improving the applicability and audio playing effect of the audio system in a complex environment.
[0033] The small multi-channel sound adaptive frequency compensation system further comprises a digital audio processor for assisting the dynamic filter compensation module; The dynamic filter compensation module comprises an infinite impulse response or finite impulse response low-pass or band-pass digital filter in a low-frequency part missing state; The digital audio processor is provided with output channels, and each output channel is equipped with a second equalizer with adjustable parameters to finely adjust the compensated audio signal again.
[0034] The dynamic filter compensation module comprises an infinite impulse response or finite impulse response low-pass or band-pass digital filter in a low-frequency part missing state, which can select the appropriate filter type for frequency compensation according to the actual missing situation of the low-frequency part. The digital audio processor is provided with output channels, and each output channel is equipped with a second equalizer with adjustable parameters to finely adjust the compensated audio signal again. The second equalizer can further fine-tune the compensated audio signal, accurately adjust the gain of each frequency band of the audio signal according to different audio content and user needs, and make the frequency response of the audio signal more perfect. This design makes the dynamic filter compensation module complete the preliminary frequency compensation and then fine-tune the signal through the second equalizer, making up for possible insufficient or excessive frequency compensation, and further improving the quality and restoration degree of the audio signal. At the same time, the combination of the second equalizer with adjustable parameters and the multiple types of filters enhances the adaptability and processing capacity of the system to different audio signals and frequency missing situations, and provides users with more personalized and high-quality audio effects.
[0035] The output channel is provided with 8 output channels, and each output channel is equipped with 15 second equalizers.
[0036] Each output channel is equipped with 15 secondary equalizers. The multiple output channels and secondary equalizer configuration significantly enhance the system's ability to fine-tune the compensated audio signal. Eight output channels can simultaneously process multiple audio signal streams, meeting the multi-channel audio signal processing requirements of compact multi-channel speakers, allowing each channel's audio signal to be independently adjusted and optimized. Each output channel is equipped with 15 secondary equalizers, enabling meticulous adjustments across multiple frequency bands of the audio signal, optimizing the frequency response from multiple perspectives. Whether targeting bass, midrange, or treble, the secondary equalizers provide precise gain adjustments for a more balanced and harmonious audio signal across all frequency bands. Compared to systems with fewer output channels and secondary equalizers, this system enables more complex and precise audio adjustments, better adapting to different audio content and the user's personalized sound quality needs. This further enhances the layering and realism of the audio, providing users with an ultimate high-quality audio experience and fully demonstrating the advantages and performance of compact multi-channel speakers in audio processing.
[0037] A control method for the above-mentioned small multi-channel audio adaptive frequency compensation system comprises the following steps: Collect audio signals through ADC; Use FFT to convert the audio signal from the time domain to the frequency domain and analyze the energy distribution of each frequency band; Design IIR or FIR digital filters based on the analysis results and adjust the filter coefficients for gain compensation; Input the audio signal into the filter for real-time processing; Acquire the compensated signal and perform frequency analysis again.
[0038] The audio signal is acquired through an ADC to obtain raw audio data. An FFT is used to convert the audio signal from the time domain to the frequency domain, analyzing the energy distribution of each frequency band to accurately identify missing or attenuated frequency ranges. Based on the analysis results, an IIR or FIR digital filter is designed, and the filter coefficients are adjusted for gain compensation to effectively compensate for the missing frequencies in the audio signal. The audio signal is input into the filter for real-time processing to ensure timely compensation during playback. The compensated signal is collected and frequency analysis is performed again to monitor the effectiveness of the frequency compensation. This series of steps forms a complete audio signal processing and frequency compensation process, with each step working closely together. This control method enables systematic and comprehensive audio signal processing and frequency compensation, ensuring that the missing bass and main sound frequencies are effectively addressed in various audio content and playback scenarios. This improves audio signal quality and sound fidelity, providing users with stable, high-quality audio playback and fully leveraging the capabilities and advantages of a compact multi-channel speaker adaptive frequency compensation system.
[0039] FFT transform decomposes the audio signal into multiple frequency components; IIR or FIR low-pass or band-pass filters are used when the low-frequency part is missing; The compensation effect is quantified by calculating MSE or PSNR; A parametric equalizer using a digital audio processor is used for processing.
[0040] The audio signal is collected by ADC to obtain the original audio data; FFT is used to convert the audio signal from time domain to frequency domain, analyze the energy distribution of each frequency band, and accurately identify the missing or attenuated frequency range; according to the analysis result, IIR or FIR digital filter is designed, filter coefficient is adjusted for gain compensation, and effective compensation is carried out for the frequency missing problem of audio signal; the audio signal is input into the filter for real-time processing to ensure that the audio signal can be compensated in time during the playing process; the compensated signal is collected and frequency analysis is carried out again to monitor the effect of frequency compensation. This series of steps forms a complete audio signal processing and frequency compensation process, each step closely cooperates and is linked together. Through this control method, the audio signal can be processed and frequency compensated systematically and comprehensively, ensuring that the problem of low-frequency and main audio frequency missing can be effectively solved in different audio content and playing scenes, improving the quality and sound quality restoration of the audio signal, providing stable and high-quality audio playing effect for users, and fully exerting the function and advantage of the small multi-channel sound self-adaptive frequency compensation system.
[0041] Specifically: Real-time frequency analysis: The input audio signal is converted from time domain to frequency domain by using Fast Fourier Transform (FFT) algorithm. Taking the common audio sampling rate of 44.1kHz as an example, the audio signal is decomposed into multiple frequency components by FFT, divided into audio frequency bands such as 20-20kHz, the energy distribution of each frequency band is analyzed, and the missing or attenuated frequency range in the bass and main sound frequency band is determined. With a (DSP) digital audio processor, its 8 input channels have 5 segment parametric equalizers, high-low pass filters, and noise gate processing modules, which can preliminarily preprocess the collected audio signal and improve the accuracy of frequency analysis.
[0042] Filter design and gain compensation: Based on the frequency analysis results, a targeted digital filter is designed. If the low-frequency portion is missing, an infinite impulse response (IIR) or finite impulse response (FIR) low-pass or bandpass filter is used. FIR filters have excellent linear phase characteristics and can avoid signal distortion; IIR filters have lower computational complexity while maintaining the same performance. For frequency bands detected to be missing or attenuated, gain compensation is performed by adjusting the filter coefficients. Reasonable gain values are set to prevent audio distortion caused by overcompensation. The digital audio processor (DSP)'s eight output channels each have a 15-band parametric equalization module, which allows for fine-tuning of the compensated audio signal.
[0043] Real-time feedback and dynamic regulation: A feedback loop is established at the audio output, using an ADC to collect the compensated audio signal. Frequency analysis is then performed again, comparing the frequency responses before and after compensation. The compensation effect is quantified by calculating metrics such as mean square error (MSE) and peak signal-to-noise ratio (PSNR). Based on the monitoring results, an adaptive filtering algorithm is employed to dynamically adjust filter parameters and gain values, allowing the system to optimize the compensation strategy in real time based on changes in the audio signal and environment.
[0044] Taking an embedded DSP chip as an example, a system for compensating for the lack of bass and main sound frequencies is constructed: 1. Initialize ADC and configure sampling rate and resolution.
[0045] 2. Write an FFT algorithm program to perform frequency domain analysis on the collected audio signal.
[0046] 3. Based on the frequency analysis results, design the FIR filter and write the filter coefficient update function.
[0047] 4. Input the audio signal into the filter for real-time processing and output the compensated audio signal.
[0048] The professional software tools required are: Audition (graphical process), MATLAB (algorithm verification), and MiniDSP (hardware real-time processing).
[0049] See also Figure 1 , Scenario: Bass is out of sync with the main channel (80Hz-200Hz recessed), Operation logic : 110Hz-160Hz: Core compensation area, boosting +3dB to +4dB to enhance the "fullness" of the sound.
[0050] 60Hz and 210Hz: Moderately boost +2dB to extend the compensation range and smooth the frequency band transition.
[0051] Cooperate with high-pass filter (HPF) to attenuate below 30Hz to avoid invalid low-frequency power occupation.
[0052] The above merely provides the preferred embodiment of the present application and not intended to limit the present application. Accordingly, any modification, equivalent replacement, improvement, etc. made within the spirit and principle of the present application shall fall within the scope of the present application.
Claims
1. A small multi-channel audio adaptive frequency compensation system, characterized by: It includes an audio signal acquisition module, a frequency analysis module and a dynamic filtering compensation module which are electrically connected to each other for communication; The audio signal acquisition module is used to acquire audio signals at a preset sampling rate through the ADC module; The frequency analysis module is used to perform FFT transformation on the collected audio signal to calculate the energy distribution of the corresponding frequency band and identify the missing or attenuated frequency range; The dynamic filtering compensation module is used to select an FIR or IIR filter according to the frequency analysis result, dynamically adjust the filter coefficient and gain value, perform real-time filtering processing on the audio signal, and output the compensated audio signal.
2. The small multi-channel audio adaptive frequency compensation system according to claim 1, characterized in that: It also includes a feedback and dynamic control module, an audio signal acquisition module, a frequency analysis module, a dynamic filtering compensation module and the feedback and dynamic control module, which are electrically connected to each other; The feedback and dynamic control module is used by the ADC module to collect the compensated audio signal, perform frequency analysis again, compare the frequency responses before and after compensation, and dynamically adjust the filter parameters and gain values through the adaptive filtering algorithm.
3. The small multi-channel audio adaptive frequency compensation system according to claim 2, characterized in that: The feedback and dynamic control module is arranged on the audio output end and forms a feedback loop.
4. The small multi-channel audio adaptive frequency compensation system according to claim 1, characterized in that: Also included is a digital audio processor for assisting the audio signal acquisition module; The digital audio processor is provided with an input channel; the input channel is equipped with a first equalizer with adjustable parameters for preliminary frequency response correction; There are 8 input channels, and each input channel is equipped with 5 first equalizers.
5. The small multi-channel audio adaptive frequency compensation system according to claim 4, characterized in that: The input channels of the digital audio processor are further equipped with high-pass and low-pass filters capable of filtering out ultra-low frequency and / or ultra-high frequency interference signals; each input channel is equipped with five high-pass and low-pass filters.
6. The small multi-channel audio adaptive frequency compensation system according to claim 4, characterized in that: The input channel of the digital audio processor is also equipped with a noise gate processing module for suppressing ambient noise below a threshold to improve the signal-to-noise ratio; each input channel is equipped with 5 noise gate processing modules.
7. The small multi-channel audio adaptive frequency compensation system according to claim 1, characterized in that: Also included is a digital audio processor for assisting the dynamic filter compensation module; The dynamic filtering compensation module includes an infinite impulse response or finite impulse response low-pass or band-pass digital filter in a state where the low-frequency part is missing; The digital audio processor is provided with output channels, and each output channel is equipped with a second equalizer with adjustable parameters to finely adjust the compensated audio signal again.
8. The small multi-channel audio adaptive frequency compensation system according to claim 7, characterized in that: There are 8 output channels, and each output channel is equipped with 15 second equalizers.
9. A control method for the small multi-channel audio adaptive frequency compensation system according to claim 1, characterized in that: The following steps are involved: Collect audio signals through ADC; Use FFT to convert the audio signal from the time domain to the frequency domain and analyze the energy distribution of each frequency band; Design IIR or FIR digital filters based on the analysis results and adjust the filter coefficients for gain compensation; Input the audio signal into the filter for real-time processing; Acquire the compensated signal and perform frequency analysis again.
10. The control method of the small multi-channel audio adaptive frequency compensation system according to claim 10, characterized in that: FFT transform decomposes the audio signal into multiple frequency components; Use IIR or FIR low-pass or band-pass filters when the low-frequency part is missing; Quantify the compensation effect by calculating MSE or PSNR; The digital audio processor's parametric equalizer is used for processing.
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