A valve audio pre-stage processing system

Abstract

The application relates to the technical field of audio processing, and particularly discloses an electronic tube audio pre-stage processing system, which comprises an audio source input module, an MCU intelligent distribution module, a DSP audio processing module, a feedback suppression module, an echo and reverberation processing module, an electronic tube audio restoration module and an audio output module; in the application, the electronic tube audio restoration module utilizes the characteristics of even harmonics to give the audio a warm and soft listening feeling, obviously improves the sound performance of a traditional digital audio processor which lacks humanization due to excessive precision, a multi-frequency band dynamic compression algorithm guarantees the detail performance of the audio, and effectively prevents distortion, and the application is particularly suitable for processing of high dynamic range audio sources, such as live music recording and high-fidelity sound playing; through real-time spectrum analysis of FFT combined with an adaptive gain control algorithm, the system can dynamically eliminate audio howling and feedback noise, and the performance is particularly outstanding in a complex sound environment, and the system is provided with reverberation denoising and equalization optimization.

Description

TECHNICAL FIELD

[0001] The present application belongs to the technical field of audio processing, and particularly relates to an electronic tube audio pre-stage processing system. BACKGROUND

[0002] In the field of audio processing, traditional pre-stage processing systems mainly rely on the combination of analog circuits and digital signal processing technology (DSP). Although these systems can provide basic functions such as amplification, filtering, and equalization of audio signals to some extent, there are still some deficiencies when processing complex audio signals.

[0003] The existing audio pre-stage processing system often has difficulty in ensuring audio details while effectively preventing distortion when processing high dynamic range sound sources. In particular, in application scenarios such as live music recording and high-fidelity sound playback, the dynamic range of audio signals is extremely large, and traditional processing methods are prone to cause the audio signal to be too compressed in some parts and too sharp in other parts, thereby affecting the overall sound quality.

[0004] In addition, for the processing of feedback noise, most existing systems use simple filters or limiters to suppress howling. However, this method can only reduce the volume of howling to some extent, and cannot completely eliminate it. In particular, in complex sound environments such as KTVs and lecture halls, due to factors such as sound reflection and superposition, the howling problem is particularly prominent, causing users to be disturbed.

[0005] To this end, the present application proposes an electronic tube audio pre-stage processing system to solve the above problems. SUMMARY

[0006] The present application aims to provide an electronic tube audio pre-stage processing system to solve the problems raised in the background art.

[0007] To achieve the above-mentioned purpose, the present application provides the following technical solutions:

[0008] An electronic tube audio pre-stage processing system, comprising:

[0009] a sound source input module for receiving multiple sound source input signals to obtain a unified input audio signal A;

[0010] an MCU intelligent distribution module for intelligently processing and distributing the input audio signal to different audio processing modules;

[0011] a DSP audio processing module for fine processing of audio signals, using a multi-band dynamic range compression algorithm to limit signal peaks to prevent distortion while maintaining audio details, to obtain a digitally processed audio signal;

[0012] Feedback suppression module: for real-time detection and suppression of feedback noise in the digitally processed audio signal, through real-time spectral analysis based on fast Fourier transform (FFT) and adaptive gain control algorithm, for dynamically adjusting signal gain to eliminate howling and improve audio signal quality, to obtain an optimized audio signal;

[0013] Echo and reverberation processing module: for using a reverberation processor based on convolution algorithm, by convolving the optimized audio signal with a preset room impulse response, to perform echo and reverberation effect superposition processing on the optimized audio signal, to obtain an audio signal with rich levels;

[0014] Vacuum tube audio restoration module: for vacuum tube processing of the audio signal with rich levels, using a mathematical model of analog even harmonic generation, combined with SPECTRE simulation-based vacuum tube equivalent circuit parameter optimization, to improve sound quality, to obtain a final output audio signal;

[0015] Audio output module: for connecting with the vacuum tube audio restoration module, outputting the final processed audio signal, and transmitting it to a power amplifier device or other audio output device.

[0016] Preferably, the formula of the multi-band dynamic compression algorithm is:

[0017]

[0018] Where y(t): compressed output signal, dynamic range optimized audio;

[0019] x(t): original input signal, which may contain high peak values with excessive dynamic range;

[0020] T: compression threshold, signal intensity reaching this value will trigger compression;

[0021] R: compression ratio, indicating the degree of compression when the input signal intensity exceeds the threshold.

[0022] Preferably, the formula for spectral analysis of the fast Fourier transform is:

[0023]

[0024] X(f): signal amplitude and phase information at frequency f, for analyzing the signal spectrum;

[0025] x(n): time-domain input signal, value at the nth sampling point;

[0026] N: FFT window length, determines the resolution of spectral analysis;

[0027] Ff: detected feedback frequency, i.e. frequency causing howling;

[0028]

[0029] G(f): Gain at frequency f, used to adjust the intensity of the frequency signal;

[0030] Gmax: Maximum gain at normal frequency, 0dB, no gain or attenuation;

[0031] Gmin: Minimum gain at feedback frequency, used to suppress howling noise.

[0032] Preferably, the formula of the convolution algorithm is:

[0033]

[0034] y(t): Output signal after reverb processing, representing the sound after reverb effect;

[0035] x(i): Original input signal, such as voice or music signal, representing the sample value at time i;

[0036] h(t-i): Room impulse response, simulating the reflection and attenuation characteristics of sound in space, related to room size and wall material;

[0037] N: Sampling length of the signal, determining the accuracy and real-time performance of convolution calculation.

[0038] Preferably, the mathematical model formula of the even harmonic generation is:

[0039] y(t) = x(t) + k1x(t) 2 +k2x(t) 3

[0040] x(t): Input signal;

[0041] k1, k2: Harmonic weights optimized based on SPICE simulation.

[0042] Preferably, the audio input signal includes wired and wireless audio signals received through the microphone 6.35 interface and the canon interface, digital audio signals received through the Bluetooth module, and external device audio signals received from the point machine and CD machine through the lotus head or canon head interface.

[0043] Preferably, the refinement process includes sound channel frequency division, delay adjustment, phase correction, volume adjustment, and bass and treble gain adjustment.

[0044] Preferably, the system further includes a power supply module for receiving 110-220V AC power supply to provide stable power support for all module components in the system.

[0045] Compared with the prior art, the application has the advantages of:

[0046] (1) The electronic tube audio restoration module in the application utilizes the characteristics of even harmonics to give the audio a warm and soft listening experience, significantly improving the lack of humanization in the sound performance of traditional digital audio processors due to excessive precision. The multi-frequency band dynamic compression algorithm ensures the detail performance of the audio while effectively preventing distortion, and is particularly suitable for processing high dynamic range sound sources, such as live music recording and high-fidelity sound playback.

[0047] (2) The application uses FFT real-time spectrum analysis combined with an adaptive gain control algorithm, and the system can dynamically eliminate audio howling and feedback noise. The system is particularly outstanding in complex sound environments (such as KTV and lecture halls). The combination of DSP adjustment and convolution reverberation can enhance the sound effect and reduce background noise through precise frequency distribution control, thereby improving overall clarity. Through the high and low frequency gain adjustment, phase correction, and sound channel delay functions of the DSP component, the system can simulate various sound field effects (such as theaters, churches, and recording studios) to adapt to different use scenarios.

[0048] (3) The application supports various preset sound effect modes such as pop, rock, and folk music, meets the individual needs of users, and is compatible with microphones, Bluetooth devices, CD players, and other sound sources, thereby enhancing the flexibility of the system. Through the carefully designed power supply and heat dissipation system, the electronic tube can work stably for a long time, thereby prolonging the service life. BRIEF DESCRIPTION OF DRAWINGS

[0049] Figure 1 A block diagram of an electronic tube audio pre-processing system according to the application is shown in FIG. DETAILED DESCRIPTION

[0050] The technical solutions in the embodiments of the application will be described in detail below with reference to the accompanying drawings. Obviously, the described embodiments are only a part of the embodiments of the application, rather than all the embodiments. Based on the embodiments in the application, all other embodiments obtained by those skilled in the art without creative work fall within the scope of the application.

[0051] Embodiment 1

[0052] Referring to FIG. Figure 1 An electronic tube audio pre-processing system according to the application is shown in FIG.

[0053] Sound source input module: for receiving various sound source input signals to obtain a unified input audio signal A;

[0054] MCU intelligent distribution module: for intelligently processing and distributing the input audio signal to different audio processing modules.

[0055] DSP audio processing module: for refining the audio signal, using multi-band dynamic range compression algorithm, for limiting signal peak value to prevent distortion while maintaining audio details, obtaining digitally processed audio signal, and providing popular, rock, folk and other sound effect modes according to different needs;

[0056] Feedback suppression module: for real-time detection and suppression of feedback noise in the digitally processed audio signal, through real-time spectral analysis based on fast Fourier transform FFT and adaptive gain control algorithm, for dynamically adjusting signal gain to eliminate howling and improve audio signal quality, obtaining optimized audio signal;

[0057] Echo and reverberation processing module: for using a reverberation processor based on convolution algorithm, by convolving the optimized audio signal with a pre-set room impulse response, realizing simulation of various spatial sound field effects, and performing echo and reverberation effect superposition processing on the optimized audio signal, enhancing the spatial sense and expressiveness of the audio, obtaining an audio signal with rich levels;

[0058] Electronic tube audio restoration module: for processing the audio signal with rich levels through electronic tube processing, using a mathematical model for generating analog even harmonics, and combining with electronic tube equivalent circuit parameter optimization based on SPICE simulation to improve sound quality, obtaining the final output audio signal;

[0059] Audio output module: for connecting with the electronic tube audio restoration module, outputting the final processed audio signal, transmitting it to a power amplifier device or other audio output device, and realizing high-quality audio playback.

[0060] Specifically, the formula of the multi-band dynamic compression algorithm is:

[0061]

[0062] Where y(t): compressed output signal, dynamic range optimized audio;

[0063] x(t): original input signal, which may contain high peak values with excessive dynamic range;

[0064] T: compression threshold, signal intensity exceeding this value will trigger compression;

[0065] R: compression ratio, indicating the compression degree of the input signal intensity exceeding the threshold (for example, R=4:1 means that the input signal intensity exceeding part is reduced to 1 / 4 of the original);

[0066] It is used to improve the dynamic range of the audio signal, so that the high sound is not harsh and the low sound is not distorted, and to enhance the overall audio experience.

[0067] Specifically, the frequency spectrum analysis formula of the fast Fourier transform is:

[0068]

[0069] X(f): signal amplitude and phase information at frequency f, used for analyzing signal spectrum;

[0070] x(n): time-domain input signal, value at the nth sampling point;

[0071] N: FFT window length, determines the resolution of the spectrum analysis;

[0072] Ff: detected feedback frequency, i.e. frequency generating howling;

[0073]

[0074] G(f): gain at frequency f, used to adjust the intensity of the signal at that frequency;

[0075] Gmax: maximum gain of normal frequency, 0dB, no gain or attenuation;

[0076] Gmin: minimum gain of feedback frequency, used to suppress howling noise;

[0077] Eliminate howling noise, improve sound clarity and audibility, suitable for KTV, speech scenes.

[0078] Specifically, the formula of the convolution algorithm is:

[0079]

[0080] y(t): output signal after reverb processing, sound after reverb effect;

[0081] x(i): original input signal, such as speech or music signal, representing the sampling value at time i;

[0082] h(t-i): room impulse response, simulating the reflection and attenuation characteristics of sound in space, related to room size and wall material;

[0083] N: sampling length of the signal, determines the accuracy and real-time performance of convolution calculation;

[0084] Used to achieve natural reverb effect, increase the spatial sense of sound, and make users feel the real sound field restoration.

[0085] Specifically, the mathematical model formula of the even harmonic generation is:

[0086] y(t) = x(t) + k1x(t)2 +k2x(t) 3

[0087] x(t): input signal;

[0088] k1, k2: harmonic weight based on SPICE simulation optimization;

[0089] For enhancing the warmth, softness and fullness of sound, especially suitable for high-quality audio equipment.

[0090] Specifically, the sound source input signal includes wired and wireless audio signals received through the microphone 6.35 interface, the canon interface, digital sound source signals received through the Bluetooth module, and external device audio signals of the song selection machine and CD machine received through the lotus head or canon head interface.

[0091] Specifically, the refinement process includes sound channel frequency division, delay adjustment, phase correction, volume adjustment and high-low sound gain adjustment.

[0092] Specifically, the system further comprises a power supply module: for receiving 110-220 volt alternating current power supply, providing stable power supply support for all module components in the system.

[0093] As can be seen from the above, the electronic tube audio restoration module utilizes the characteristics of even harmonics to give the audio a warm and soft listening experience, significantly improving the lack of humanization in the sound performance of traditional digital audio processors due to excessive precision. The multi-frequency band dynamic compression algorithm ensures the detail performance of the audio, while effectively preventing distortion, and is particularly suitable for processing high dynamic range sound sources, such as live music recording and high-fidelity sound playback.

[0094] Through real-time spectral analysis by FFT combined with adaptive gain control algorithm, the system can dynamically eliminate audio howling and feedback noise, and performs particularly outstanding in complex sound environments (such as KTV, lecture hall), reverberation noise reduction and equalization optimization: combining DSP adjustment and convolution reverberation, it can not only enhance the sound effect, but also reduce background noise through precise frequency distribution control, improve overall clarity, and through the high-low frequency gain adjustment, phase correction and sound channel delay function of the DSP component, the system can simulate multiple sound field effects (such as theater, church, recording studio), adapt to different use scenarios.

[0095] Supporting various preset sound effect modes such as pop, rock and folk music, it meets the individual needs of users, is compatible with microphones, Bluetooth devices, CD machines and other sound sources, enhances the flexibility of the system, and through the carefully designed power supply and heat dissipation system, ensures the stable operation of the electronic tube for a long time, prolongs the service life.

[0096] Embodiment two:

[0097] Professional recording studio audio processing

[0098] Background scenario

[0099] In professional recording studios, when recording a human voice, the sound quality needs to be delicate, layered, and have a natural sense of space.

[0100] Specific configuration and parameters

[0101] Input device

[0102] Microphone type: Neumann U87, connected via a 6.35mm interface, input signal peak value -10dBFS.

[0103] Echo and reverb processing module

[0104] Room impulse response: medium-sized recording studio simulation h(t).

[0105] Convolution calculation parameters: signal length N = 2048, sampling rate fs = 48kHz.

[0106] DSP audio processing module

[0107] Channel frequency division: divided into low frequency (<200Hz), medium frequency (200Hz-2kHz), high frequency (>2kHz).

[0108] Dynamic range compression parameters: compression threshold T = -6dB, compression ratio R = 4:1.

[0109] High and low frequency gain: low frequency gain +3dB, high frequency gain +2dB.

[0110] Electronic tube audio restoration module

[0111] Harmonic coefficients: even harmonics k1 = 0.15, odd harmonics k2 = 0.05.

[0112] Working parameters: input voltage 250V, anode current 10mA.

[0113] Output audio frequency response is smooth, high frequency is bright but not sharp, low and medium frequency is mellow and warm.

[0114] Natural reverb increases the spatial sense of sound, making the vocal sound more stereo.

[0115] As can be seen from the above, compression and harmonic processing balance the audio dynamic, avoid distortion, and at the same time improve the performance of sound details.

[0116] Experimental comparison shows that the system sound quality score is improved by about 15%, and the background noise is reduced by 10dB.

[0117] Example three

[0118] Audio enhancement of KTV sound system

[0119] Background scenario

[0120] In KTV box, microphone audio is prone to howling, and at the same time, the sound effect needs to be enhanced to meet the entertainment needs.

[0121] Specific configuration and parameters

[0122] Input device

[0123] Microphone type: Shure SM58, connected to XLR interface, signal peak -5dBFS

[0124] Feedback suppression module

[0125] FFT spectrum analysis parameters: window length N = 1024, overlap rate 50%.

[0126] Howling detection frequency: Ff = 1.2 kHz.

[0127] Gain adjustment: feedback frequency gain Gfeedback = -15dB

[0128] DSP audio processing module

[0129] Sound effect mode: pop mode, high and low frequency gain +3dB, mid-frequency gain +1dB.

[0130] Surround sound effect enhancement parameters: channel delay Δt = 10ms.

[0131] Electronic tube audio restoration module

[0132] Harmonic coefficient: even harmonic k1 = 0.25, odd harmonic k2 = 0.1.

[0133] Working parameters: input voltage 300V, anode current 15mA.

[0134] As can be seen from the above, after the application of the above embodiment, the howling is completely eliminated, and the user's subjective listening shows that the clarity is improved by 20%.

[0135] Reverb and electronic tube processing are combined, and the output audio is rich in layers and distortion-free.

[0136] Sound effect enhancement makes the sound dynamic stronger, suitable for entertainment environment.

[0137] Experimental comparison shows: compared with traditional KTV sound, the system sound quality score is improved by about 25%, and the peak value of howling frequency is reduced by 20dB.

[0138] In the description of the specification, the description of the terms "one embodiment", "some embodiments", "example", "specific example" or "some examples" etc. means that the specific features, structures, materials or characteristics described in connection with the embodiment or example are contained in at least one embodiment or example of the present application. In the specification, the illustrative description of the above terms does not necessarily refer to the same embodiment or example. Moreover, the specific features, structures, materials or characteristics described can be combined in any appropriate manner in any one or more embodiments or examples. In addition, the person skilled in the art can combine and combine the different embodiments or examples described in the specification and the features of the different embodiments or examples without contradiction.

[0139] In the drawings of the embodiments of the present application, only the structures related to the embodiments of the present disclosure are involved, and other structures can refer to the general design. In the case of no conflict, the same embodiment and different embodiments of the present application can be combined with each other.

[0140] Although the embodiments of the present application have been shown and described, it will be understood by those of ordinary skill in the art that various changes, modifications, substitutions and variations can be made to these embodiments without departing from the principles and spirit of the present application, and the scope of the present application is defined by the appended claims and their equivalents.

Claims

1. An electronic tube audio pre-stage processing system, characterized by, Comprise: Audio source input module: for receiving a variety of audio input signals, obtaining a unified input audio signal A; MCU intelligent distribution module: for intelligent processing and distribution of the input audio signal, and distributing it to different audio processing modules; DSP audio processing module: for detailed processing of audio signals, using a multi-band dynamic range compression algorithm to limit signal peaks to prevent distortion while preserving audio details, to obtain a digitally processed audio signal; Feedback suppression module: for real-time detection and suppression of feedback noise in the digitally processed audio signal, using real-time spectral analysis based on fast Fourier transform FFT and an adaptive gain control algorithm to dynamically adjust signal gain to eliminate howling and improve audio signal quality, to obtain an optimized audio signal; Echo and reverberation processing module: for using a reverberation processor based on a convolution algorithm to perform echo and reverberation effect superposition processing on the optimized audio signal by convolving the optimized audio signal with a pre-set room impulse response, to obtain an audio signal with rich levels; Tube audio restoration module: for tube processing of the audio signal with rich levels, using a mathematical model of analog even harmonic generation combined with SPICE simulation-based tube equivalent circuit parameter optimization to improve sound quality, to obtain a final output audio signal; Audio output module: for connecting with the tube audio restoration module and outputting the final output audio signal to be transmitted to a power amplifier device or other audio output device; The formula of the multi-band dynamic range compression algorithm is: Where y(t): compressed output signal, dynamically range-optimized audio; x(t): original input signal, containing high peak values with excessive dynamic range; T: compression threshold, signal intensity exceeding which triggers compression; R: compression ratio, indicating the degree of compression of input signal intensity exceeding the threshold; The formula of the convolution algorithm is: y(t): reverberation-processed output signal; x(i): original input signal, including voice or music signal, representing the sampling value at time i; h(t-i): room impulse response, simulating the reflection and attenuation characteristics of sound in space, related to room size and wall material; N: signal sampling length; The formula of the mathematical model of even harmonic generation is: y(t) = x(t) + k1x(t) 2 + k2x(t) 3 x(t): input signal; k1, k2: harmonic weights based on SPICE simulation optimization.

2. The electronic tube audio preamplifier system of claim 1, wherein, The spectral analysis formula of the fast Fourier transform is: X(f): signal amplitude and phase information at frequency f, used for spectral analysis of the signal; x(n): time-domain input signal, value at the nth sampling point; N: FFT window length, determines the resolution of spectral analysis; Ff: detected feedback frequency, i.e. the frequency at which howling occurs; Where G(f): gain at frequency f, used to adjust the intensity of the signal at that frequency; Gmax: maximum gain of normal frequencies, 0 dB, no gain or attenuation; Gmin: minimum gain of feedback frequencies, used to suppress howling noise.

3. The electronic tube audio preamplifier system of claim 1, wherein, The audio source input signal includes wired and wireless audio signals received through the microphone 6.35 interface, the canon interface, digital audio source signals received through the Bluetooth module, and external device audio signals of a point machine, a CD machine received through the lotus head or canon head interface.

4. The electronic tube audio preamplifier system of claim 1, wherein, The refinement process includes sound channel frequency division, delay adjustment, phase correction, volume adjustment, and treble and bass gain adjustment.

5. The electronic tube audio preamplifier system of claim 1, wherein, The system further comprises a power supply module for receiving 110-220V AC power supply to provide stable power supply support for all module components in the system.

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