Electronic tube audio pre-processing system

Abstract

The invention relates to the technical field of audio processing, and particularly discloses an electronic tube audio pre-processing system which comprises a sound 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 reduction module and an audio output module. According to the invention, the electron tube audio reduction module gives a warm and soft hearing feeling to audio by using the characteristics of even harmonic waves, the problem that a traditional digital audio processor lacks humanized sound performance due to excessive accuracy is obviously improved, the detail performance of the audio is ensured by a multi-band dynamic compression algorithm, distortion is effectively prevented, and the audio processing efficiency is improved. The system is particularly suitable for processing a high dynamic range sound source, such as on-site music recording and high-fidelity sound playing, can dynamically eliminate audio howling and feedback noise through combination of FFT real-time spectrum analysis and a self-adaptive gain control algorithm, and is particularly outstanding in performance in a complex sound environment, and reverberation denoising and balance optimization are realized.

Description

Technical Field

[0001] The invention belongs to the technical field of audio processing, and in particular relates to an electronic tube audio pre-stage processing system. Background Art

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

[0003] Existing audio pre-processing systems often have difficulty in effectively preventing distortion while maintaining audio details when processing sound sources with a high dynamic range. Especially in application scenarios such as live music recording and high-fidelity audio playback, the dynamic range of audio signals is extremely large. Traditional processing methods can easily cause the audio signal to be overly compressed in some parts and too sharp in other parts, thus 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 a certain extent, but cannot completely eliminate it. Especially 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 trouble to users.

[0005] In view of this, the inventor proposes a vacuum tube audio pre-processing system to solve the above problems. Summary of the invention

[0006] The object of the present invention is to provide a tube audio pre-processing system to solve the problems raised in the above background technology.

[0007] To achieve the above object, the present invention provides the following technical solutions:

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

[0009] Audio source input module: used to receive multiple audio source input signals and obtain a unified input audio signal A;

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

[0011] DSP audio processing module: used to refine the audio signal, using a multi-band dynamic range compression algorithm to limit the signal peak while maintaining the audio details to prevent distortion, and obtain a digitally processed audio signal;

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

[0013] The echo and reverberation processing module is used to use a reverberation processor based on a convolution algorithm to perform convolution calculation on the optimized audio signal and a preset room impulse response, and to perform superposition processing of echo and reverberation effects on the optimized audio signal to obtain an audio signal with a rich sense of hierarchy;

[0014] The electron tube audio restoration module is used to perform electron tube processing on the audio signal with rich layering, use a mathematical model that simulates even harmonic generation, and combine the optimization of electron tube equivalent circuit parameters based on SPI CE simulation to improve the sound quality and obtain the final output audio signal;

[0015] Audio output module: used to connect with the tube audio restoration module, output the final processed audio signal, and transmit it to the 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, audio with optimized dynamic range;

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

[0020] T: Compression threshold. Compression will be triggered when the signal strength reaches this value.

[0021] R: Compression ratio, which indicates the degree of compression after the input signal strength exceeds the threshold.

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

[0023]

[0024] X(f): signal amplitude and phase information at frequency f, used to analyze signal spectrum;

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

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

[0027] Ff: The detected feedback frequency, i.e. the frequency that generates the howling;

[0028]

[0029] Where G(f) is the gain at frequency f, which is used to adjust the strength of the frequency signal;

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

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

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

[0033]

[0034] y(t): output signal after reverberation processing, reflecting the sound after reverberation effect;

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

[0036] h(ti): room impulse response, simulating the reflection and attenuation characteristics of sound in space, which is related to the size of the room and the material of the wall;

[0037] N: The sampling length of the signal determines the accuracy and real-time performance of the convolution calculation.

[0038] Preferably, the mathematical model formula for generating even harmonics 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 source input signal includes wired and wireless audio signals received through a microphone 6.35 interface and a Canon interface, a digital audio source signal received through a Bluetooth module, and an external device audio signal of a karaoke machine or a CD player received through a Lotus head or Canon head interface.

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

[0044] Preferably, the system also includes a power supply module: used to receive 110-220 volt AC power to provide stable power support for all module components in the system.

[0045] Compared with the prior art, the present invention has the following beneficial effects:

[0046] (1) The tube audio restoration module of the present invention utilizes the characteristics of even harmonics to give the audio a warm and soft listening experience, significantly improving the traditional digital audio processor's over-precision and lack of humanized sound performance. The multi-band dynamic compression algorithm ensures the detailed performance of the audio while effectively preventing distortion. It is particularly suitable for processing high dynamic range sound sources, such as live music recording and high-fidelity audio playback.

[0047] (2) The present invention combines FFT real-time spectrum analysis with an adaptive gain control algorithm. The system can dynamically eliminate audio howling and feedback noise, and performs particularly well in complex sound environments (such as KTVs and lecture halls). Reverberation denoising and equalization optimization: combined with DSP adjustment and convolution reverberation, it can not only enhance the sound effect, but also reduce background noise through precise frequency distribution control and improve overall clarity. Through the high and low frequency gain adjustment, phase correction and channel delay functions of the DSP components, the system can simulate a variety of sound field effects (such as theaters, churches, and recording studios) to adapt to different usage scenarios.

[0048] (3) The present invention supports multiple preset sound effect modes such as pop, rock, and folk music to meet the personalized needs of users. It is compatible with multiple sound sources such as microphones, Bluetooth devices, and CD players, which enhances the flexibility of the system. Through a carefully designed power supply and heat dissipation system, it ensures that the electronic tube works stably for a long time and extends its service life. BRIEF DESCRIPTION OF THE DRAWINGS

[0049] Figure 1 The present invention is a block diagram of a vacuum tube audio pre-processing system. DETAILED DESCRIPTION

[0050] The following will be combined with the drawings in the embodiments of the present invention to clearly and completely describe the technical solutions in the embodiments of the present invention. Obviously, the described embodiments are only part of the embodiments of the present invention, not all of the embodiments. Based on the embodiments of the present invention, all other embodiments obtained by ordinary technicians in this field without creative work are within the scope of protection of the present invention.

[0051] Embodiment 1:

[0052] See also Figure 1 As shown, a vacuum tube audio pre-stage processing system comprises:

[0053] Audio source input module: used to receive multiple audio source input signals and obtain a unified input audio signal A;

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

[0055] DSP audio processing module: used to refine audio signals and adopt a multi-band dynamic range compression algorithm to limit signal peaks to prevent distortion while maintaining audio details, thereby obtaining digitally processed audio signals. It also provides a variety of sound effect modes such as pop, rock, and folk music according to different needs;

[0056] Feedback suppression module: used for real-time detection and suppression of feedback noise in the digitally processed audio signal, and for dynamically adjusting the signal gain to eliminate howling, improve the audio signal quality, and obtain an optimized audio signal through real-time spectrum analysis and adaptive gain control algorithm based on Fast Fourier Transform (FFT);

[0057] Echo and reverberation processing module: used for using a reverberation processor based on a convolution algorithm, by performing convolution calculation on the optimized audio signal and a preset room impulse response, to achieve simulation of various spatial sound field effects, and to perform superposition processing of echo and reverberation effects on the optimized audio signal, to enhance the spatial sense and expressiveness of the audio, and to obtain an audio signal with a rich sense of hierarchy;

[0058] The electron tube audio restoration module is used to perform electron tube processing on the audio signal with rich layering, use a mathematical model that simulates even harmonic generation, and combine the optimization of electron tube equivalent circuit parameters based on SPI CE simulation to improve the sound quality and obtain the final output audio signal;

[0059] Audio output module: used to connect with the tube audio restoration module, output the final processed audio signal, and transmit it to the power amplifier device or other audio output device to achieve high-quality audio playback.

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

[0061]

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

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

[0064] T: Compression threshold. Compression will be triggered when the signal strength reaches this value.

[0065] R: Compression ratio, which indicates the degree of compression after the input signal strength exceeds the threshold (for example, R = 4:1 means that the excess input signal strength is reduced to 1 / 4 of the original);

[0066] It is used to improve the dynamic range of audio signals, making high notes less harsh and low notes less distorted, thus enhancing the overall audio experience.

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

[0068]

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

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

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

[0072] Ff: The detected feedback frequency, i.e. the frequency that generates the howling;

[0073]

[0074] Where G(f) is the gain at frequency f, which is used to adjust the strength of the frequency signal;

[0075] Gmax: Maximum gain at normal frequency, 0dB, no gain and no 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 and speech scenes.

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

[0079]

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

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

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

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

[0084] It is used to achieve natural reverberation effects, increase the spatial sense of sound, and enable users to feel the real sound field restoration.

[0085] Specifically, the mathematical model formula for generating even harmonics is:

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

[0087] x(t): input signal;

[0088] k1, k2: harmonic weights optimized based on SPICE simulation;

[0089] Used to enhance the warmth, softness and richness of the sound, it is particularly suitable for audio processing of high-quality audio equipment.

[0090] Specifically, the audio source input signal includes wired and wireless audio signals received through the microphone 6.35 interface and the Canon interface, digital audio source signals received through the Bluetooth module, and external device audio signals of the karaoke machine and CD player received through the Lotus head or Canon head interface.

[0091] Specifically, the detailed processing includes channel frequency division, delay adjustment, phase correction, volume adjustment and treble and bass gain adjustment.

[0092] Specifically, the system also includes a power supply module: used to receive 110-220 volt AC power and provide stable power support for all module components in the system.

[0093] As can be seen above, the tube audio restoration module uses the characteristics of even harmonics to give the audio a warm and soft listening experience, significantly improving the traditional digital audio processor's over-precision and lack of humanized sound performance. The multi-band dynamic compression algorithm ensures the audio's detailed performance while effectively preventing distortion. It is particularly suitable for processing high dynamic range sound sources, such as live music recording and high-fidelity audio playback.

[0094] Through FFT real-time spectrum analysis combined with adaptive gain control algorithm, the system can dynamically eliminate audio howling and feedback noise, and performs particularly well in complex sound environments (such as KTV and lecture halls). Reverberation denoising and equalization optimization: combined with DSP adjustment and convolution reverberation, it can not only enhance the sound effect, but also reduce background noise through precise frequency distribution control and improve overall clarity. Through the high and low frequency gain adjustment, phase correction and channel delay functions of the DSP components, the system can simulate a variety of sound field effects (such as theaters, churches, and recording studios) to adapt to different usage scenarios.

[0095] It supports multiple preset sound effect modes such as pop, rock, folk music, etc. to meet the personalized needs of users. It is compatible with multiple sound sources such as microphones, Bluetooth devices, CD players, etc., which enhances the flexibility of the system. Through the carefully designed power supply and heat dissipation system, it ensures that the electronic tube works stably for a long time and extends its service life.

[0096] Embodiment 2:

[0097] Professional studio audio processing

[0098] Background scene

[0099] In a professional recording studio, recording a vocal audio requires the sound quality to be delicate, layered, and have a natural sense of space.

[0100] Specific configuration and parameters

[0101] Input devices

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

[0103] Echo and Reverb Processing Module

[0104] Room impulse response: A medium-sized recording studio was selected to simulate 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), mid-frequency (200Hz–2kHz), and high frequency (>2kHz).

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

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

[0110] Tube Audio Restoration Module

[0111] Harmonic coefficient: even harmonic k1=0.15, odd harmonic k2=0.05.

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

[0113] The output audio frequency response is smooth, the high frequencies are bright but not too sharp, and the mid and low frequencies are mellow and warm.

[0114] Natural reverberation increases the spatial sense of the sound and makes the vocals sound more three-dimensional.

[0115] From the above, we can see that compression and harmonic processing balance the audio dynamics, avoid distortion, and enhance the expressiveness 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] Embodiment three:

[0118] Audio Enhancement for KTV Sound System

[0119] Background scene

[0120] In KTV rooms, microphone audio is prone to howling, and the sound effects need to be enhanced to meet entertainment needs.

[0121] Specific configuration and parameters

[0122] Input devices

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

[0124] Feedback Suppression Module

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

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

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

[0128] DSP audio processing module

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

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

[0131] 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] It can be seen from the above that after the above embodiment is applied, the howling is completely eliminated, and the user's subjective hearing perception shows that the clarity is improved by 20%.

[0135] The combination of reverb and tube processing produces rich audio output without distortion.

[0136] The sound enhancement makes the sound more dynamic, suitable for entertainment environments.

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

[0138] In the description of this specification, the description with reference to the terms "one embodiment", "some embodiments", "examples", "specific examples" or "some examples" etc. means that the specific features, structures, materials or characteristics described in conjunction with the embodiment or example are included in at least one embodiment or example of the present invention. In this specification, the schematic representations of the above terms do not necessarily refer to the same embodiment or example. Moreover, the specific features, structures, materials or characteristics described may be combined in any one or more embodiments or examples in a suitable manner. In addition, those skilled in the art may combine and combine different embodiments or examples described in this specification and the features of different embodiments or examples, unless they are contradictory.

[0139] In the drawings of the embodiments disclosed in the present invention, only the structures related to the embodiments disclosed in the present invention are involved, and other structures can refer to the general design. In the absence of conflict, the same embodiment and different embodiments of the present invention can be combined with each other.

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

Claims

1. A tube audio pre-processing system, characterized in that: include: Audio source input module: used to receive multiple audio source input signals and obtain a unified input audio signal A; MCU intelligent distribution module: used for intelligently processing and distributing the input audio signal to different audio processing modules; DSP audio processing module: used to refine the audio signal, using a multi-band dynamic range compression algorithm to limit the signal peak while maintaining the audio details to prevent distortion, and obtain a digitally processed audio signal; Feedback suppression module: used for real-time detection and suppression of feedback noise in the digitally processed audio signal, and for dynamically adjusting the signal gain to eliminate howling, improve the audio signal quality, and obtain an optimized audio signal through real-time spectrum analysis and adaptive gain control algorithm based on Fast Fourier Transform (FFT); The echo and reverberation processing module is used to use a reverberation processor based on a convolution algorithm to perform convolution calculation on the optimized audio signal and a preset room impulse response, and to perform superposition processing of echo and reverberation effects on the optimized audio signal to obtain an audio signal with a rich sense of hierarchy; The electron tube audio restoration module is used to perform electron tube processing on the audio signal with rich layering, using a mathematical model simulating even harmonic generation, combined with the optimization of electron tube equivalent circuit parameters based on SPICE simulation, to improve the sound quality and obtain the final output audio signal; Audio output module: used to connect with the tube audio restoration module, output the final processed audio signal, and transmit it to the power amplifier device or other audio output device.

2. The electronic tube audio pre-processing system according to claim 1, characterized in that: The formula of the multi-band dynamic compression algorithm is: Where y(t): compressed output signal, audio with optimized dynamic range; x(t): original input signal, containing high peaks with excessive dynamic range; T: Compression threshold. Compression will be triggered when the signal strength reaches this value. R: Compression ratio, which indicates the degree of compression after the input signal strength exceeds the threshold.

3. The electronic tube audio pre-processing system according to claim 1, characterized in that: The spectrum analysis formula of the fast Fourier transform is: X(f): signal amplitude and phase information at frequency f, used to analyze signal spectrum; x(n): time domain input signal, the value at the nth sampling point; N: FFT window length, which determines the resolution of spectrum analysis; Ff: The detected feedback frequency, i.e. the frequency that generates the howling; Where G(f) is the gain at frequency f, which is used to adjust the strength of the frequency signal; Gmax: Maximum gain at normal frequency, 0dB, no gain and no attenuation; Gmin: Minimum gain of feedback frequency, used to suppress howling noise.

4. The electronic tube audio pre-processing system according to claim 1, characterized in that: The formula of the convolution algorithm is: y(t): output signal after reverberation processing; x(i): original input signal, including speech or music signal, representing the sample value at time i; h(ti): room impulse response, simulating the reflection and attenuation characteristics of sound in space, which is related to the room size and wall material; N: sampling length of the signal.

5. The electronic tube audio pre-processing system according to claim 1, characterized in that: The mathematical model formula for even harmonic generation is: y(t)=x(t)+k1x(t) 2 +k2x(t) 3 x(t): input signal; k1, k2: Harmonic weights optimized based on SPICE simulation.

6. The electronic tube audio pre-processing system according to claim 1, characterized in that: The audio source input signal includes wired and wireless audio signals received through the microphone 6.35 interface and the Canon interface, digital audio source signals received through the Bluetooth module, and external device audio signals of the karaoke machine and CD player received through the Lotus head or Canon head interface.

7. The electronic tube audio pre-processing system according to claim 1, characterized in that: The detailed processing includes channel frequency division, delay adjustment, phase correction, volume adjustment and treble and bass gain adjustment.

8. The electronic tube audio pre-processing system according to claim 1, characterized in that: The system also includes a power supply module: used to receive 110-220 volt AC power and provide stable power support for all module components in the system.

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