An electronic microphone audio signal processing system and method

Abstract

This invention relates to the field of electronic microphone audio signal processing technology, and provides an electronic microphone audio signal processing system and method, including a server connected to a signal acquisition module, a digital signal processing module, a digital-to-analog converter module, and an output module. The signal acquisition module includes a sound sensor, an analog front-end, and an analog-to-digital converter. The sound sensor uses a MEMS microphone array to convert changes in sound wave pressure into analog electrical signals. The analog front-end internally includes a preamplifier, a bias circuit, and an anti-aliasing filter. The preamplifier amplifies the electrical signal output by the microphone. The anti-aliasing filter filters out high-frequency components above the Nyquist frequency before analog-to-digital conversion, preventing aliasing distortion. This invention can process digital audio signals in real time and accurately, effectively eliminating echo, suppressing noise, stabilizing volume, and improving speech clarity and sound quality, greatly improving the audio signal processing quality of electronic microphones.

Description

Technical Field

[0001] This invention relates to the field of electronic microphone audio signal processing technology, specifically to an electronic microphone audio signal processing system and method. Background Technology

[0002] An electronic microphone is a device that converts sound waves into electrical signals. It is commonly used in various fields such as audio recording, communication, and sound pickup. The working principle of an electronic microphone is generally based on the change in electrical signals caused by sound pressure. Common types include dynamic microphones, condenser microphones, and electromagnetic microphones.

[0003] Electronic microphone audio signal processing encompasses the entire process from the analog signal output by the microphone to the final signal usable for audio playback or further processing. Every step in electronic microphone audio signal processing is crucial, from signal acquisition, amplification, filtering, and processing to the final output; the quality of any step directly impacts the final audio quality. Precise audio signal processing ensures clarity and accuracy, removes interference, and thus achieves higher quality audio output.

[0004] Chinese Patent Application No. 201910826070.5, Publication No. CN112399304B, discloses a microphone device, an electronic device, and an audio signal processing method thereof. The microphone device includes a microphone array and an integrated circuit. The microphone array includes at least three microphones arranged in a straight line at non-equidistant intervals. The integrated circuit is electrically connected to the microphone array. The integrated circuit is used to process the mixed and added sound signals of different combinations of the microphones through a time-domain filter to generate an output sound signal, wherein the sound gain of the output sound signal at each frequency and each pickup angle is substantially equalized. This invention enables the output sound signal of the microphone device to have substantially equalized sound gain at each frequency and each pickup angle. The microphone device can still provide good sound quality when the user's position changes, thereby enhancing the user experience.

[0005] Chinese Patent Application No. 201610628373.2 and Publication No. CN106060743A discloses a microphone, a microphone assembly, and a microphone signal processing method. The microphone includes at least: a signal processing module and an audio input / output port module connected to the signal processing module. The signal processing module receives a sound signal and processes the received sound signal to generate an audio signal. The audio input / output port module can be selectively connected to an audio playback or recording device with a single-channel audio signal receiving module. When the audio input / output port module is connected to the audio playback or recording device, it transmits the audio signal to the audio playback or recording device. The microphone is suitable for series connection with another microphone having the same structure via the audio input / output port module. This invention provides a microphone with a simple structure, allowing multiple microphones to be used simultaneously by connecting them in series.

[0006] In summary, while existing electronic microphone audio signal processing technologies have improved audio processing effects to some extent, they still have some shortcomings, resulting in insufficient audio quality. Therefore, this application proposes an electronic microphone audio signal processing system and method. Summary of the Invention

[0007] The purpose of this invention is to provide an electronic microphone audio signal processing system and method to solve the problem that although the existing electronic microphone audio signal processing has improved the audio processing effect to a certain extent, it still has some shortcomings, resulting in insufficient audio quality.

[0008] To achieve the above objectives, the present invention provides the following technical solution: The first aspect of the present invention provides an electronic microphone audio signal processing system, including a server, wherein the server is connected to a signal acquisition module, a digital signal processing module, a digital-to-analog conversion module and an output module;

[0009] The signal acquisition module is responsible for converting acoustic signals into digital signals with high fidelity; the signal acquisition module includes a sound sensor, an analog front-end, and an analog-to-digital converter;

[0010] The sound sensor uses a MEMS microphone array to convert changes in sound pressure into analog electrical signals;

[0011] The analog front end internally includes a preamplifier, a bias circuit, and an anti-aliasing filter; the preamplifier amplifies the electrical signal output from the microphone; the bias circuit provides voltage to the microphone; and the anti-aliasing filter is used to filter out high-frequency components above the Nyquist frequency before analog-to-digital conversion to prevent aliasing distortion.

[0012] An analog-to-digital converter converts amplified and filtered clean analog signals into digital samples;

[0013] The digital signal processing module receives digital samples from the signal acquisition module and executes a series of real-time algorithms; the digital signal processing module includes a preprocessing unit, an echo processing unit, a beamforming unit, a noise processing unit, a volume control unit, and an audio processing and encoding unit;

[0014] The preprocessing unit performs framing, windowing, and FFT processing; the echo processing unit eliminates echoes captured by the microphone to ensure clear communication; the beamforming unit utilizes the spatial characteristics of the microphone array to form one or more directional pickup beams by adjusting the delay and weight of each channel signal; the noise processing unit suppresses stable background noise and non-stationary burst noise; the volume control unit dynamically adjusts the signal amplitude to stabilize the output volume at a preset level, avoiding volume fluctuations caused by distance changes; and the audio processing and encoding unit adjusts the frequency response curve to improve speech clarity and performs amplitude limiting and compression encoding on the audio.

[0015] The digital-to-analog conversion module converts the processed digital signal back into an analog signal;

[0016] The output module outputs the final processed audio data in different types, including analog output, digital stream output, audio storage, and AI interface.

[0017] Preferably, the amplification formula of the preamplifier is as follows:

[0018] ;in, This is the amplified analog voltage signal. This is the magnification factor. This is the analog voltage signal output by the MEMS microphone. This is the DC bias voltage.

[0019] Preferably, the analog-to-digital converter performs sampling and quantization, as shown in the formula: ;

[0020] The signal acquisition module acquires digital signal values. This forms set A. ;

[0021] in:

[0022] It is a rounding function; This is the quantization step size; The sampling period is ;

[0023] m = 0, 1, 2, ..., M-1; M is the number of microphones in the MEMS microphone array;

[0024] n = 0, 1, 2, ...; n is a discrete time point, determined by the sampling rate F of the analog-to-digital converter. s It is determined that the time interval between each sample is 1 / F. s Second;

[0025] It is the digitized voltage value of the m-th microphone at sample point n.

[0026] Preferably, the preprocessing unit inputs... Output frequency domain data The solution is as follows:

[0027] ① Divide into short frames that either do not overlap or overlap. k is the frame index, and l is the intra-frame sample index. ;

[0028] N is the frame length, R = N / 2;

[0029] ②Add a window: w is the window function;

[0030] ③ FFT processing: f is the frequency index.

[0031] Preferably, the echo processing unit inputs... An adaptive filter and adaptive filtering algorithm are used to eliminate echoes.

[0032] Preferably, the beamforming unit employs an adaptive beamforming algorithm to suppress interference; the noise processing unit tracks the noise spectrum in real time using MCRA, calculates the real-valued gain using spectral subtraction or Wiener filtering, and multiplies it onto the noisy speech spectrum to suppress noise.

[0033] Preferably, the volume control unit sets a target level a, then calculates the RMS value b of the input signal and compares it with the target level: C = a / (b + ε), where ε is a constant;

[0034] If C > 1, it means that amplification is needed; if C < 1, it means that attenuation is needed; if C = 1, no change is needed.

[0035] A second aspect of the present invention provides a method for processing audio signals from an electronic microphone using the system described in the first aspect, comprising the following steps:

[0036] The signal acquisition module is responsible for converting acoustic signals into digital signals with high fidelity. The digital signal is then processed and transmitted to the server. The server can output the signal directly as a digital stream, audio storage, or AI interface. Alternatively, the digital-to-analog converter module can convert the processed digital signal back into an analog signal for analog output.

[0037] This invention has at least the following beneficial effects:

[0038] This invention provides an electronic microphone audio signal processing system and method. By employing a MEMS microphone array as a sound sensor, combined with a high-precision analog front-end and an analog-to-digital converter, it achieves high-fidelity conversion of acoustic signals to digital signals. Through a digital signal processing module, including multiple units such as preprocessing, echo processing, beamforming, noise reduction, volume control, and audio processing and encoding, this system can process digital audio signals in real time and accurately, effectively eliminating echoes, suppressing noise, stabilizing volume, and improving speech clarity and sound quality, thus greatly improving the audio signal processing quality of the electronic microphone. Attached Figure Description

[0039] Figure 1 This is a schematic diagram of the electronic microphone audio signal processing system module of the present invention. Detailed Implementation

[0040] The technical solutions in the embodiments of the present invention will be clearly and completely described below. Obviously, the described embodiments are only some embodiments of the present invention, and not all embodiments. Based on the embodiments of the present invention, all other embodiments obtained by those skilled in the art without creative effort are within the scope of protection of the present invention.

[0041] Example 1

[0042] This embodiment provides an electronic microphone audio signal processing system, including a server, which is connected to a signal acquisition module, a digital signal processing module, a digital-to-analog conversion module, and an output module.

[0043] The signal acquisition module is responsible for converting acoustic signals into digital signals with high fidelity; the signal acquisition module includes a sound sensor, an analog front-end, and an analog-to-digital converter.

[0044] The sound sensor uses a MEMS microphone array to convert changes in sound wave pressure into analog electrical signals. The MEMS microphone array consists of multiple MEMS microphone units, each of which can accurately capture sound from different directions and provide spatial audio information.

[0045] The analog front end internally includes a preamplifier, a bias circuit, and an anti-aliasing filter; the preamplifier amplifies the electrical signal output from the microphone; the bias circuit provides voltage to the microphone; and the anti-aliasing filter is used to filter out high-frequency components above the Nyquist frequency before analog-to-digital conversion to prevent aliasing distortion.

[0046] The amplification formula for the preamplifier is as follows:

[0047] ;in, This is the amplified analog voltage signal. This is the magnification factor. This is the analog voltage signal output by the MEMS microphone. This is the DC bias voltage.

[0048] An analog-to-digital converter (ADC) converts amplified and filtered clean analog signals into digital samples; the ADC samples and quantizes the signals, as shown in the formula. ;

[0049] The signal acquisition module acquires digital signal values. This forms set A. ;

[0050] in:

[0051] It is a rounding function; This is the quantization step size; The sampling period is ;

[0052] m = 0, 1, 2, ..., M-1; M is the number of microphones in the MEMS microphone array;

[0053] n = 0, 1, 2, ...; n is a discrete time point, determined by the sampling rate F of the analog-to-digital converter. s It is determined that the time interval between each sample is 1 / F. s Second;

[0054] It is the digitized voltage value of the m-th microphone at sample point n.

[0055] The digital signal processing module receives digital samples from the signal acquisition module and executes a series of real-time algorithms; the digital signal processing module includes a preprocessing unit, an echo processing unit, a beamforming unit, a noise processing unit, a volume control unit, and an audio processing and encoding unit.

[0056] The preprocessing unit performs frame segmentation, windowing, and FFT processing; the input to the preprocessing unit is... Output frequency domain data The solution is as follows:

[0057] ① Divide into short frames that either do not overlap or overlap. k is the frame index, and l is the intra-frame sample index. ;

[0058] N is the frame length, R = N / 2;

[0059] ②Add a window: w is the window function;

[0060] ③ FFT processing: f is the frequency index.

[0061] Echo processing unit input It employs adaptive filters and adaptive filtering algorithms to eliminate echoes captured by the microphone, ensuring clear calls.

[0062] The beamforming unit utilizes the spatial characteristics of the microphone array and employs an adaptive beamforming algorithm to form one or more directional pickup beams by adjusting the delay and weight of each channel signal.

[0063] The noise processing unit tracks the noise spectrum in real time using MCRA, calculates the real-valued gain using spectral subtraction or Wiener filtering, and multiplies it onto the noisy speech spectrum to suppress stable background noise and non-stationary burst noise.

[0064] The volume control unit dynamically adjusts the amplitude of the signal to stabilize the output volume at a preset level, avoiding volume fluctuations caused by changes in distance. The volume control unit sets a target level a, then calculates the RMS value b of the input signal and compares it with the target level: C = a / (b + ε), where ε is a constant.

[0065] If C > 1, it means that amplification is needed; if C < 1, it means that attenuation is needed; if C = 1, no change is needed.

[0066] The audio processing and encoding unit adjusts the frequency response curve to improve speech clarity and performs amplitude limiting and compression encoding on the audio. An EQ equalizer enhances sound quality; a limiter prevents the signal amplitude from exceeding its maximum value, avoiding clipping distortion during digital-to-analog conversion or decoding playback; and a suitable audio encoding algorithm, such as AAC or Opus, compresses the processed audio data into a bitstream for easy storage and transmission.

[0067] The digital-to-analog conversion module converts the processed digital signal back into an analog signal via a digital-to-analog converter (DAC). The DAC uses a high-precision converter to ensure signal fidelity during the conversion process and reduce distortion. The converted analog signal is then smoothed and filtered to further remove quantization noise and high-frequency components, resulting in a clean analog audio output.

[0068] The output module outputs the processed audio data in various ways, including analog output, digital stream output, audio storage, and an AI interface. Analog output drives speakers through a power amplifier, providing a high-quality audio playback experience. Digital stream output supports multiple audio formats, such as PCM and I2S, facilitating connection and transmission with other digital audio devices. The audio storage module saves the processed audio data to storage devices, such as SD cards and hard drives, enabling audio recording and playback. The AI ​​interface provides interaction capabilities with other smart devices, using speech recognition and synthesis technology to achieve intelligent control and information exchange, enhancing the system's overall intelligence.

[0069] Example 2

[0070] This embodiment provides a method for processing audio signals from an electronic microphone, the method comprising the following steps:

[0071] S1, through the signal acquisition module, is responsible for converting acoustic signals into digital signals with high fidelity:

[0072] The MEMS microphone array, using a sound sensor, can accurately convert changes in sound pressure into analog electrical signals; the preamplifier in the analog front end amplifies these analog signals; the anti-aliasing filter is responsible for filtering out high-frequency components before analog-to-digital conversion to prevent aliasing distortion; the analog-to-digital converter finally converts these amplified and filtered analog signals into digital samples.

[0073] S2, the digital signal processing module receives these digital samples and executes a series of complex real-time algorithms, including preprocessing, echo processing, beamforming, noise reduction, volume control, and audio processing and encoding, to improve audio quality.

[0074] The preprocessing stage involves framing, windowing, and Fast Fourier Transform (FFT) processing to provide frequency domain data for subsequent processing steps. The echo processing unit uses adaptive filters and algorithms to effectively eliminate echoes captured by the microphone. The beamforming unit utilizes the spatial characteristics of the microphone array to form a directional pickup beam by adjusting signal delay and weights. The noise processing unit tracks the noise spectrum in real time and applies spectral subtraction or Wiener filtering to suppress noise. The volume control unit dynamically adjusts the signal amplitude to ensure stable output volume. Finally, the audio processing and encoding unit adjusts the frequency response curve to improve speech intelligibility and performs amplitude limiting and compression encoding on the audio.

[0075] S3, the processed digital signal, is transmitted to the server. The server then performs the output according to the required type, including but not limited to analog output, digital stream output, audio storage, and AI interface.

[0076] If analog output is to be performed, the digital signal must first be converted back into an analog signal through a digital-to-analog converter module. This step is achieved by a high-precision digital-to-analog converter (DAC).

[0077] If it is not an analog output, the server directly transmits the data to the output module; the digital stream output supports multiple audio formats, facilitating connection and transmission with other digital audio devices; the audio storage module enables the recording and playback of audio data; and the AI ​​interface provides the ability to interact with other smart devices, achieving intelligent control and information interaction through speech recognition and synthesis technology.

[0078] The above embodiments 1-2, by using a MEMS microphone array as a sound sensor, combined with a high-precision analog front-end and analog-to-digital converter, realize the conversion of high-fidelity acoustic signals to digital signals; through the digital signal processing module, including multiple units such as preprocessing, echo processing, beamforming, noise processing, volume control, and audio processing and encoding, this system can process digital audio signals in real time and accurately, effectively eliminate echo, suppress noise, stabilize volume, and improve speech clarity and sound quality.

[0079] In addition, the digital-to-analog conversion module and the output module ensure that the processed audio signal can be output in a high-quality manner. Whether it is analog output, digital stream output, audio storage or AI interface, it can meet the needs of diverse application scenarios. In particular, the application of AI interface enables this system to seamlessly connect with smart devices and realize more intelligent and convenient audio processing and control functions.

[0080] In summary, the electronic microphone audio signal processing system and method provided by this invention significantly improves the processing quality and intelligence level of audio signals through the synergistic interaction of various module structures and processing algorithms, and has broad application prospects and market value.

[0081] The foregoing has shown and described the basic principles, main features, and advantages of the present invention. It will be apparent to those skilled in the art that the present invention is not limited to the details of the exemplary embodiments described above, and that the invention can be implemented in other specific forms without departing from its spirit or basic characteristics. Therefore, the embodiments should be considered exemplary and non-limiting in all respects, and the scope of the invention is defined by the appended claims rather than the foregoing description. Thus, it is intended that all variations falling within the meaning and scope of equivalents of the claims be included within the present invention.

[0082] Although embodiments of the invention have been shown and described, it will be understood by those skilled in the art that various changes, modifications, substitutions and alterations can be made to these embodiments without departing from the principles and spirit of the invention, the scope of which is defined by the appended claims and their equivalents.

Claims

1. An electronic microphone audio signal processing system, characterized by, The system includes a server, which is connected to a signal acquisition module, a digital signal processing module, a digital-to-analog conversion module, and an output module. The signal acquisition module is responsible for converting acoustic signals into digital signals with high fidelity; the signal acquisition module includes a sound sensor, an analog front-end, and an analog-to-digital converter; The sound sensor uses a MEMS microphone array to convert changes in sound pressure into analog electrical signals; The analog front end internally includes a preamplifier, a bias circuit, and an anti-aliasing filter; the preamplifier amplifies the electrical signal output from the microphone; the bias circuit provides voltage to the microphone; and the anti-aliasing filter is used to filter out high-frequency components above the Nyquist frequency before analog-to-digital conversion to prevent aliasing distortion. An analog-to-digital converter converts amplified and filtered clean analog signals into digital samples; The digital signal processing module receives digital samples from the signal acquisition module and executes a series of real-time algorithms; the digital signal processing module includes a preprocessing unit, an echo processing unit, a beamforming unit, a noise processing unit, a volume control unit, and an audio processing and encoding unit; The preprocessing unit performs framing, windowing, and FFT processing; the echo processing unit eliminates echoes captured by the microphone to ensure clear communication; the beamforming unit utilizes the spatial characteristics of the microphone array to form one or more directional pickup beams by adjusting the delay and weight of each channel signal. The noise processing unit suppresses stable background noise and non-stable burst noise; the volume control unit dynamically adjusts the signal amplitude to keep the output volume stable at a preset level and avoids volume fluctuations caused by distance changes; the audio processing and encoding unit adjusts the frequency response curve to improve speech clarity and performs amplitude limiting and compression encoding on the audio. The digital-to-analog conversion module converts the processed digital signal back into an analog signal; The output module outputs the final processed audio data in different types, including analog output, digital stream output, audio storage, and AI interface.

2. An electronic microphone audio signal processing system according to claim 1, wherein, The amplification formula of the preamplifier is as follows: ;in, This is the amplified analog voltage signal. This is the magnification factor. This is the analog voltage signal output by the MEMS microphone. This is the DC bias voltage.

3. The electronic microphone audio signal processing system according to claim 2, characterized in that, The analog-to-digital converter samples and quantizes the data using the following formula: ; The signal acquisition module acquires digital signal values. This forms set A. ; in: It is a rounding function; This is the quantization step size; The sampling period is ; m = 0, 1, 2, ..., M-1; M is the number of microphones in the MEMS microphone array; n = 0, 1, 2,... ; n is a discrete time point, by the sampling rate F of the analog-to-digital converter s It is decided that the time interval between each sample is 1 / F s second; It is the digitized voltage value of the m-th microphone at sample point n.

4. The electronic microphone audio signal processing system according to claim 3, characterized in that, The preprocessing unit input Output frequency domain data The solution is as follows: ① Divide into short frames that either do not overlap or overlap. k is the frame index, and l is the intra-frame sample index. ; N is the frame length, R = N / 2; ② Add a window: w is a window function; ③ FFT processing: f is the frequency index.

5. The electronic microphone audio signal processing system according to claim 4, characterized in that, The echo processing unit input An adaptive filter and adaptive filtering algorithm are used to eliminate echoes.

6. The electronic microphone audio signal processing system according to claim 5, characterized in that, The beamforming unit employs an adaptive beamforming algorithm to suppress interference. The noise processing unit tracks the noise spectrum in real time using MCRA, calculates the real-valued gain using spectral subtraction or Wiener filtering, and multiplies it onto the noisy speech spectrum to suppress noise.

7. The electronic microphone audio signal processing system according to claim 6, characterized in that, The volume control unit sets the target level a, then calculates the RMS value b of the input signal and compares it with the target level: C = a / (b + ε), where ε is a constant; If C > 1, it means that amplification is needed; if C < 1, it means that attenuation is needed; if C = 1, no change is needed.

8. A method for processing audio signals from an electronic microphone using the system described in any one of claims 1 to 7, characterized in that, Includes the following steps: The signal acquisition module is responsible for converting acoustic signals into digital signals with high fidelity. The digital signal is then processed and transmitted to the server. The server can output the signal directly as a digital stream, audio storage, or AI interface. Alternatively, the digital-to-analog converter module can convert the processed digital signal back into an analog signal for analog output.

Images