A voice call noise reduction circuit

CN114974290BActive Publication Date: 2025-03-25SHIJIAZHUANG KEHENG ELECTRONICS CO LTD
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Patent Information

Application Number
CN202210515450.9
Authority / Receiving Office
CN · China
Patent Type
Patents(China)
Current Assignee / Owner
Filing Date
2022-05-12
Publication Date
2025-03-25
Estimated Expiration
2042-05-12

AI Technical Summary

Technical Problem

The prior art has poor results when dealing with stable noise in machining workshops or armored vehicles, resulting in poor call quality.

Method used

A voice call noise reduction circuit is adopted, including a first-level filtering module and an audio processing module. The first-level filtering module filters signals above the normal frequency range of voice, while the audio processing module identifies and filters the stable noise components, and uses these steps to generate a clear audio signal.

Benefits of technology

It effectively reduces the high-frequency component and the stable noise component, improves the communication clarity of both parties, and significantly improves the call quality.

✦ Generated by Eureka AI based on patent content.

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Abstract

The present application relates to the technical field of call equipment, and in particular to a voice call noise reduction circuit, comprising a first-level filtering module, connected to an audio input device, for receiving an audio signal, filtering out high-frequency components in the audio signal, and generating a first audio signal; wherein the high-frequency component is a signal in the audio signal whose frequency is higher than the upper limit of the normal frequency range of human voice; an audio processing module, for receiving the first audio signal, filtering out the stationary noise component in the first audio signal, and generating a second audio signal; a power amplifier module, connected to the audio processing module, for receiving the second audio signal, amplifying the second audio signal according to a preset amplification ratio, and generating an audio power amplifier signal; the power amplifier module is connected to an audio output device, and the audio output device receives the audio power amplifier signal and plays audio. The present application has the effect of improving the quality of voice calls.
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Description

Technical Field

[0001] The present application relates to the technical field of call equipment, and in particular to a voice call noise reduction circuit. Background Art

[0002] Telephone voice communication plays an extremely important role in people's current social activities, economic activities and production life. At present, in social life, mobile phones or circuits are often used for communication, relying on base stations or network communication methods to achieve communication between two parties.

[0003] In some special occasions, such as machine processing workshops and armored vehicles, calls often use internal lines to achieve information transmission between two parties. Noise reduction is often required during communication to remove noise so that the human voice is clearer and more convenient for signal transmission during communication. In related technologies, dual MIC noise reduction is often used at the microphone to filter the voice signal collected from the microphone and other voice input devices.

[0004] Dual MIC noise reduction uses two microphones, one for calls and the other for user XXX. It separates noise and voice signals based on the differences in signal strength, phase and time generated by the two microphones, and then filters out the noise.

[0005] However, dual MIC noise reduction can only eliminate non-stationary noise, while in machining workshops, armored vehicles and other places, a large amount of mechanical noise with a long duration is often generated, such as the roaring noise of motors or engines when they are running. These mechanical noises greatly affect the call quality, and dual MIC noise reduction cannot filter these stationary noises well, and still generate large noise, resulting in poor call quality. Summary of the invention

[0006] In order to improve the quality of voice calls, the present application provides a voice call noise reduction circuit, comprising:

[0007] A voice call noise reduction circuit, comprising:

[0008] A primary filtering module, connected to an audio input device, for receiving an audio signal, filtering out high-frequency components in the audio signal, and generating a first audio signal;

[0009] The high-frequency component is a signal in the audio signal whose frequency is higher than the upper limit of the normal frequency range of human voice;

[0010] an audio processing module, configured to receive the first audio signal, filter out a stationary noise component in the first audio signal, and generate a second audio signal;

[0011] The stationary noise component is a component whose frequency remains unchanged within a preset reference time length;

[0012] a power amplification module, connected to the audio processing module, and configured to receive the second audio signal, amplify the second audio signal according to a preset amplification ratio, and generate an audio power amplifier signal;

[0013] The power amplification module is connected to an audio output device, and the audio output device receives an audio power amplifier signal to play audio.

[0014] By adopting the above technical solution, a first-level filtering module is used to filter signals that are higher than the upper limit of the normal frequency range of human voice, so that the high-frequency components are greatly attenuated, thereby improving the clarity of human voice. Secondly, the audio processing module is used to filter out the steady noise component. The steady noise component is a component of the signal whose frequency remains unchanged and remains unchanged within a preset reference time length, such as the noise component generated by continuous vibration when a machine is working. The audio processing module is used to filter out the steady noise component, and then the power amplifier module amplifies the second audio signal with the steady noise component filtered out into an audio power amplifier signal, thereby retaining the human voice to the greatest extent while reducing various continuous steady noises generated in complex environments, thereby improving the clarity of communication between the two parties on the call.

[0015] In a possible implementation, the audio processing module includes a first processing module, a secondary filtering module and a second processing module;

[0016] The first processing module is connected to the primary filtering module, and is used to receive the first audio signal, filter the stationary noise component in the first audio signal, and generate a first audio processing signal;

[0017] The secondary filtering module is connected to the first processing module and is used to receive the first audio processing signal to attenuate the high frequency components in the first audio processing signal and then generate a second audio processing signal;

[0018] The second processing module is connected to the secondary filtering module, and is used to receive a second audio processing signal, perform secondary filtering on a stationary noise component in the second audio processing signal, and generate a second audio signal.

[0019] By adopting the above technical solution, the first processing module is first used to perform preliminary filtering on the stationary noise component in the first audio signal, and then the secondary filtering module is used to attenuate the high-frequency component, and then the second processing module performs secondary processing, so that the effectiveness of attenuation and filtering of the stationary noise component and the high-frequency component is improved, thereby better improving the call quality.

[0020] In a possible implementation manner, when filtering the stationary noise component, the first processing module is specifically configured to:

[0021] identifying each signal component in the first audio signal according to differences in frequency and / or duration in the first audio signal;

[0022] According to each of the duration periods, the duration corresponding to each signal component is determined respectively. If the duration exceeds a preset reference duration, the signal component corresponding to the duration is determined to be a stationary noise component, and the stationary noise component is filtered.

[0023] By adopting the above technical solution, when the communicating parties are talking normally, the tone of the human voice keeps changing, causing the frequency of the generated signal component to keep changing. However, the stationary noise component persists in the surrounding environment and its frequency does not change for a long time. Therefore, based on the difference between the two, the stationary noise component is identified and filtered out, thereby improving the accuracy of filtering the stationary noise component.

[0024] In a possible implementation manner, before the first processing module identifies each signal component in the first audio signal according to differences in frequency and / or duration in the first audio signal, it is further configured to:

[0025] Each signal component in the first audio signal that is within a normal frequency range of human voice is amplified according to a preset first ratio.

[0026] By adopting the above technical solution, the signal components in the first audio signal that are within the normal frequency range of human voice are first amplified, and then the duration of each signal component is identified to determine whether each signal component is a stationary noise component; the amplitude of each signal component within the normal frequency range of human voice after amplification will be increased, which makes it easier to identify the stationary noise components within the normal frequency range of human voice, thereby improving the accuracy of identifying the stationary noise components.

[0027] In a possible implementation, after the first processing module amplifies each signal component in the first audio signal within a normal frequency range of human voice according to a preset first ratio, it is further configured to:

[0028] Detecting the amplitude of each signal component;

[0029] If the amplitude of any one of the signal components is smaller than the preset first reference voltage value, each signal component with an amplitude smaller than the preset first reference voltage value is filtered out.

[0030] By adopting the above technical solution, after amplifying each signal within the normal frequency range of human voice, the signal component whose amplitude is less than the preset first reference voltage value is filtered, wherein the signal component whose amplitude is less than the preset first reference voltage value represents a sound with extremely low volume. Most of the sounds with extremely low volume are tiny sounds emitted by other sound sources in the surrounding environment, which will affect the human voices of the communicating parties. Filtering out the signal component corresponding to the sound with smaller amplitude is beneficial to improving the quality of the signal.

[0031] In a possible implementation, after the first processing module amplifies each signal component in the first audio signal within a normal frequency range of human voice according to a preset first ratio, it is further configured to:

[0032] Identify the amplitude of each signal component;

[0033] All signal components with amplitudes greater than a preset second reference voltage value are filtered out.

[0034] By adopting the above technical solution, the signal component with an amplitude greater than the preset second reference voltage value is a component with an abnormally high amplitude and a very high volume, which will affect the call quality of both parties. Filtering out the component with an abnormally high amplitude is beneficial to improving the clarity of the human voice and further improving the call quality.

[0035] In a possible implementation, the power amplifier circuit includes an operational amplifier, a resistor R15, a resistor R16, a resistor R17, a resistor R18, a resistor R19, a resistor R20, a resistor R21, a resistor R22, a resistor R23, a resistor R25, a resistor RX, a capacitor C23, a capacitor C24, a polar capacitor C25, a polar capacitor C26, and a polar capacitor C27;

[0036] One end of the resistor R15 is used to receive the second audio signal, and the other end is connected to the negative electrode of the polarity capacitor C25, the positive electrode of the polarity capacitor C25 is connected to the positive input terminal of the operational amplifier, one end of the capacitor C23 is connected to the negative electrode of the polarity capacitor C25, and the other end is grounded; the resistor RX is connected in parallel to both ends of the capacitor C23; one end of the resistor R17 is connected to the positive input terminal of the operational amplifier, and the other end is grounded; one end of the resistor R16 is connected to the positive input terminal of the operational amplifier, and the other end is connected to the 26V power supply; the negative input terminal of the operational amplifier is connected to one end of the resistor R18, and the positive input terminal of the resistor R18 is connected to the positive input terminal of the operational amplifier. The other end is connected to the positive electrode of capacitor C27, and the negative electrode of capacitor C27 is grounded; one end of resistor R19 is connected to the negative input end, and the other end is connected to one end of resistor R20, the other end of resistor R20 is connected to the output end of the operational amplifier, and capacitor C24 is connected in parallel to both ends of resistor R19; the output end of the operational amplifier is connected to the positive electrode of polarity capacitor C26, the negative electrode of polarity capacitor C26 is connected to one end of resistor R21, and the other end of resistor R21 outputs the audio power amplifier signal; one end of resistor R22 is connected to one end of resistor R21 outputting the audio power amplifier signal, and the other end is grounded.

[0037] By adopting the above technical solution, the second audio signal is amplified by the power amplifier module, and the signal component in the second audio signal is shaped, so that the audio power amplifier signal can be directly used for playback in the audio output device, thereby improving the loading capacity.

[0038] In a possible implementation manner, the resistor RX is a variable resistor.

[0039] By adopting the above technical solution, the resistor RX is a variable resistor. By changing the resistance value of the resistor RX, the amplification ratio of the second audio signal can be changed, thereby improving applicability.

[0040] In a possible implementation, the primary filtering module includes a resistor R1, a resistor R2, a capacitor C1, and a capacitor C2;

[0041] Among them, one end of the resistor R1 is connected to the audio input device, and the other end is grounded; one end of the capacitor C1 is connected to the end of the resistor R1 that is not grounded, and the other end is connected to one end of the resistor R2, and the other end of the resistor R2 is connected to the audio processing module for outputting the first audio signal; one end of the capacitor C2 is connected to the end of the capacitor C1 connected to the resistor R2, and the other end is grounded.

[0042] By adopting the above technical solution, the high-frequency component and the low-frequency component in the audio signal are filtered using a bandpass filter composed of a primary filtering module to generate a first audio signal which is output from one end of the resistor R2 connected to the audio processing module.

[0043] In summary, the present application includes at least one of the following beneficial technical effects:

[0044] 1. Use the first-level filtering module to filter the signal that is higher than the upper limit of the normal frequency range of human voice, so that the high-frequency component is greatly attenuated, improving the clarity of human voice. Secondly, use the audio processing module to filter out the steady noise component. The steady noise component is the component of the signal whose frequency remains unchanged and remains unchanged within a preset reference time. For example, the noise component generated by the continuous vibration of the machine when it is working. The audio processing module is used to filter out the steady noise component, and then the power amplifier module amplifies the second audio signal with the steady noise component filtered out into an audio power amplifier signal, which retains the human voice to the greatest extent while reducing various continuous steady noises generated in complex environments, thereby improving the clarity of communication between the two parties on the call.

[0045] 2. First, the signal components in the first audio signal that are within the normal frequency range of human voice are amplified, and then the duration of each signal component is identified to determine whether each signal component is a stationary noise component; the amplitude of each signal component within the normal frequency range of human voice after amplification will increase, which makes it easier to identify the stationary noise components within the normal frequency range of human voice, thereby improving the accuracy of identifying the stationary noise components. BRIEF DESCRIPTION OF THE DRAWINGS

[0046] Figure 1 It is a module schematic diagram of a voice call noise reduction circuit according to an embodiment of the present application;

[0047] Figure 2 is a circuit diagram of a voice call noise reduction circuit according to an embodiment of the present application;

[0048] In the figure, 1, primary filtering module; 2, audio processing module; 21, first processing module; 22, secondary filtering module; 23, second processing module; 3, power amplification module. DETAILED DESCRIPTION

[0049] The following is combined with Figure 1 -Attached Figure 2 , further details of this application are given.

[0050] A voice call noise reduction circuit, referring to Figure 1 , including: a primary filter module 1, an audio processing module 2 and a power amplifier module 3, wherein the primary filter module 1 is connected to an audio input device, the primary filter module 1 is also connected to the audio processing module 2, the audio processing module 2 is connected to the power amplifier module 3, and the power amplifier module 3 is connected to an audio output device. The audio input device can be a microphone or a microphone for collecting audio signals; the audio output device can be a speaker or a speaker for playing audio.

[0051] Specifically, the first-level filtering module 1 is connected to the audio input device, and is used to receive an audio signal, filter out the high-frequency component in the audio signal, and generate a first audio signal; wherein the high-frequency component is a signal in the audio signal whose frequency is higher than the upper limit of the normal frequency range of human voice.

[0052] The high-frequency component is a signal with a frequency higher than the normal frequency of human voice, usually a short-wave high-frequency signal, such as car horns, firecrackers, or metal cutting, which are very penetrating and harsh sounds. They not only affect the clarity of human voices, but also cause negative emotions and harm people's hearing. The high-frequency component in the audio signal is filtered out by the first-level filtering module 1 to improve the clarity of speech.

[0053] The normal frequency range of human voice is between 20HZ and 20KHZ, that is, the frequency range that human ears can hear is basically between 20HZ and 20KHZ. The possibility of human voice in signals outside the normal frequency range of human voice is extremely small, and they are basically all noise; the signals within the normal frequency range of human voice may be human voice or noise; therefore, the high-frequency components exceeding the upper limit of the normal frequency range of human voice are first filtered out through the first-level filtering module 1, that is, the high-frequency noise signals mixed in the audio signal are filtered out, and the signals within the normal frequency range of human voice are retained.

[0054] Furthermore, the sound with a frequency lower than the lower limit of the normal frequency range of human voice is basically low-frequency noise, which will cause harm to the human body and affect the clarity of the human voice. Therefore, it is also necessary to filter out the low-frequency components with a frequency lower than the lower limit of the normal frequency range of human voice. In a feasible method, the high-frequency components and the low-frequency components can be filtered out by the first-level filtering module 1.

[0055] Reference Figure 2 The primary filter module 1 includes a resistor R1, a resistor R2, a capacitor C1 and a capacitor C2, wherein one end of the resistor R1 is connected to the audio input device and the other end is grounded, one end of the capacitor C1 is connected to the ungrounded end of the resistor R1 and the other end is connected to one end of the resistor R2, and the other end of the resistor R2 is connected to the audio processing module 2; one end of the capacitor C2 is connected to the end of the capacitor C1 connected to the resistor R2 and the other end is grounded. The primary filter module 1 is essentially a bandpass filter, and the bandwidth allowed by the bandpass filter is set to the normal frequency range of human voice, that is, the bandwidth is set to between 20HZ and 20KHZ, that is, the signal with a frequency in the normal frequency range of human voice will pass through the primary filter module 1, and the signal with a frequency outside the normal frequency range of human voice will be greatly attenuated or directly filtered out.

[0056] Among them, the resistor R1 and the capacitor C1 form a high-pass filter to limit the passage of signals below the lower frequency limit. The size of the lower frequency limit is determined by the values ​​of the resistor R1 and the capacitor C1. In the embodiment of the present application, it is set to limit the passage of signals below 20HZ; the resistor R2 and the capacitor C2 form a low-pass filter to limit the passage of signals above the upper frequency limit. Among them, the value of the upper frequency limit is determined by the values ​​of the resistor R2 and the capacitor C2. In the embodiment of the present application, it is set to limit the passage of signals with a frequency higher than 20KHZ.

[0057] By using a first-stage filtering circuit, the signals in the audio signal that are within the normal sound range of human voice are retained, and the signals whose frequencies are outside the normal sound range of human voice are filtered out, thereby reducing the harsh high-frequency components and improving the clarity of the voice.

[0058] Engine noise components and vibration components that exist for a long time and greatly interfere with the communication between the two parties during their existence are collectively referred to as stationary noise components. Based on the effect of the communication between the two parties, the voice component should be retained as much as possible and other noises should be eliminated, especially the vibration component generated by mechanical vibration, the engine noise component generated by the roar of the engine, and other stationary noise components that exist for a long time and greatly interfere with the transmission of the human voice.

[0059] Furthermore, the audio processing module 2 is used to receive the first audio signal, identify the stationary noise component in the first audio signal, filter out the stationary noise component, and generate a second audio signal, wherein the stationary noise component is a component whose frequency maintains a constant value for a period exceeding a preset duration, and represents the component of noise generated by mechanical vibration and engine operation; the audio processing module 2 is used to purify and filter the signal within the normal range of human voice again. The power amplifier module 3 is connected to the audio processing module 2, and is used to receive the second audio signal, and after amplifying the second audio signal according to a preset amplification ratio, generate an audio power amplifier signal to the audio output device, and the audio output device is used to play the audio power amplifier signal.

[0060] The audio processing module 2 includes a first processing module 21, a secondary filtering circuit and a second processing module 23. The first processing module 21 is connected to the primary filtering module 1, receives the first audio signal output by the primary filtering module 1, and preliminarily identifies and filters the stationary noise component in the first audio signal.

[0061] Reference Figure 2The first processing module 21 includes a signal processing chip, a resistor R3, a resistor R4, a resistor R5, a resistor R6, a resistor R7, a capacitor C3, a capacitor C4, a polar capacitor C5, a polar capacitor C6, a polar capacitor C7, a polar capacitor C8, a polar capacitor C9, a polar capacitor C10, and a polar capacitor C11; the signal processing chip adopts a chip model MC33219, and the MC33219 chip includes pin MCI, pin MCO, pin TAI, pin TAO, pin XDI, pin CPT, pin CP2, pin TLI, pin TLO, pin RLO, pin VB, pin CT, pin CF and pin MUTE.

[0062] Reference Figure 2Pin MCI is the inverting input terminal of the microphone amplifier configured inside the chip MC33219, which is used as the input terminal of the first processing module 21, connected to the output of the first-stage filtering module 1, and receives the first audio signal. One end of the resistor R3 is connected to the pin MCI, and the other end is connected to the pin MCO; Pin MCO is the output terminal of the microphone amplifier, and the resistor R3 is connected between the pin MCI and the pin MCO to form a negative feedback gain. The resistance value of the resistor R3 determines the amplification factor of the amplifier. Pin TAI is the input end of the attenuator configured inside the chip MC33219, and is connected to pin MCO. The signal amplified by the microphone amplifier is attenuated by the attenuator and output from the output end of the attenuator, namely pin TAO; Pin TAI is also connected to one end of the resistor R4, the other end of the resistor R4 is connected to one end of the capacitor C3, the other end of the capacitor C3 is connected to pin XDI, and pin XDI is the detection end of the background noise detector configured inside the chip MC33219, which is used to detect the background noise in the signal amplified by the microphone amplifier; Pin CPT is the setting end of the background noise, and is connected to one end of the resistor R5, the other end of the resistor R5 is connected to a 6.5V power supply, the positive pole of the polarity capacitor C5 is connected to the 6.5V power supply, and the negative pole is connected to pin CPT. The background noise is set by the values ​​of the resistor R5 and the polarity capacitor C5. The time constant of the scene noise monitor; one end of the capacitor C4 is connected to the pin TLI, and the other end is connected to one end of the resistor R6, the other end of the resistor R6 is connected to the pin TA0, and the pin TAO is also connected to the positive electrode of the polarity capacitor C6, and the negative electrode of the polarity capacitor C6 outputs the first audio processing signal; one end of the pin CP2 is connected to the positive electrode of the polarity capacitor C7, and the negative electrode of the polarity capacitor C7 is grounded; the pin TLO is connected to the positive electrode of the polarity capacitor C8, and the negative electrode of the polarity capacitor C8 is grounded; the pin VB is connected to the positive electrode of the polarity capacitor C9, and the negative electrode of the polarity capacitor C9 is grounded; the pin CT is connected to the positive electrode of the polarity capacitor C10, and the negative electrode is grounded; the pin CD is grounded; the pin RLO is connected to the positive electrode of the polarity capacitor C11, and the negative electrode of the polarity capacitor C11 is grounded; the pin MUTE is grounded; one end of the resistor R7 is connected to the pin VB and the other end is connected to the pin CT.

[0063] The first audio signal is received by the pin MCI, the frequency and amplitude of the signal component in the first audio signal are detected, and then the stationary noise component in the first audio signal is preliminarily filtered using the internal control logic of the chip to generate a first audio processing signal.

[0064] Among them, the first audio signal often includes multiple signal components, for example, the human voice component generated by human speech, the vibration component generated by mechanical vibration, the engine noise component generated by the roar of the engine, and components of various frequencies and amplitudes generated in the surrounding environment. Each signal component corresponds to a sound amplitude, a sound frequency and a duration period according to the sound source. The duration period represents the time period corresponding to when the frequency is maintained at the sound frequency. For example: in the time period of 10:00-10:01, the frequency of the vibration component generated by mechanical vibration is maintained at 10KHZ, then the sound frequency is 10KHZ, and the duration period is 10:00-10:01. At the same time, the first audio signal may be accompanied by multiple components, there may be only one signal component, or there may be no signal component.

[0065] Therefore, in order to weaken or filter out the stationary noise component, when the first processing module 21 receives the first audio signal, in order to eliminate the stationary noise component, it is first necessary to determine which signal components in the first audio signal are vocal components and which signal components are stationary noise components, and then filter out the stationary noise components after determining them.

[0066] In a possible implementation of the embodiment of the present application, the method of identifying and filtering the stationary noise component is:

[0067] According to the difference in frequency and / or duration of each component in the first audio signal, each signal component in the first audio signal is identified.

[0068] Specifically, at the same moment, the first audio signal may be mixed with components of multiple frequencies, and each frequency component corresponds to a signal component; in addition, at the same frequency, different durations also correspond to a signal component; for example: at the current moment, there are three frequencies of 100HZ, 500HZ and 10KHZ, which correspond to three signal components respectively; in the 100HZ frequency, there are 100HZ sounds in the three durations of 8:00:00-9:00:00; 9:01:01-9:01:02; 10:10:00-10:11:00, and the three durations correspond to three signal components respectively; according to the different time periods of the components at each frequency, the signal components are divided, and the divided signal components meet the conditions of different frequencies and / or different durations. That is, each signal component meets any one of the following three conditions: the frequency is the same, but the duration is different; the frequency is different, and the duration is regarded as the same; the frequency is different, and the duration is also different.

[0069] Afterwards, the duration corresponding to each duration segment is determined respectively. If the duration exceeds a preset reference duration, the signal component corresponding to the duration is determined to be a stationary noise component, and the stationary noise component is filtered out.

[0070] Specifically, the frequency of human voice determines the tone of human voice, and the volume of human voice is related to the level of human voice signal; the higher the tone of human voice, the greater the frequency of the corresponding signal; the higher the volume, the greater the level. The tone and volume of human voice when speaking are usually not constant. When people are explaining, there will be pauses, the tone will change, and the volume of the voice will also change; while for the stationary noise component, its frequency usually remains basically unchanged for a period of time; based on the difference between the human voice component and the stationary noise component, the stationary noise component is identified.

[0071] After determining each signal component, the duration corresponding to the duration segment can be determined according to the duration segment corresponding to each signal component. The duration is the value of the duration segment. For example: the 1KHZ signal component corresponds to a duration segment of 8:00:01-8:00:07; the duration is 6 seconds.

[0072] The preset reference duration is a preset reference value. When the duration is greater than the reference duration, the signal component is characterized as a stationary noise component. Continuing with the previous example, the reference duration is set to 5 seconds, and the duration corresponding to the 1KHZ signal component is 6 seconds, then the signal component is judged to be a stationary noise component. After the stationary noise component is determined, the stationary noise component is filtered out, which weakens noise such as mechanical vibration and improves the clarity of the speech.

[0073] In a possible implementation of the embodiment of the present application, before the first processing module 21 identifies each signal component in the first audio signal according to the difference in frequency and / or duration of each component in the first audio signal, it is further configured to:

[0074] Each signal component in the first audio signal that is within the normal frequency range of human voice is amplified according to a preset first ratio.

[0075] Among them, the preset first ratio is determined by the value of resistor R3. After all signal components within the normal frequency range of human voice are amplified, the amplitude of each signal within the normal frequency range of human voice will increase. For some stationary noise components with small amplitude and frequency within the normal frequency range of human voice, since their amplitude is very small, they are difficult to be identified and will be ignored during filtering. Some of these components will still exist in the generated first audio processing signal, affecting the call quality. Therefore, the signal within the normal frequency range of human voice is amplified first, and then the stationary noise component in the signal is filtered, which can improve the purity of the filtered signal.

[0076] In a possible implementation of the embodiment of the present application, after the first processing module 21 amplifies the first audio signal, it is further used to:

[0077] Detect the amplitude of each signal component; if the amplitude of any signal component is less than a preset first reference voltage value, filter out each signal component whose amplitude is less than the preset first reference voltage value.

[0078] Specifically, in the first audio signal obtained by the first-stage filtering module 1, there are still some high-frequency noises and low-frequency noises that have not been completely filtered out. After the first processing module 21 amplifies the signal within the normal frequency range of human voice, the noise signal outside the normal frequency range of human voice is kept at the original amplitude unchanged. After the signal is amplified, the signal less than the preset first reference voltage value is filtered out. On the one hand, the remaining noise outside the normal frequency range of human voice is filtered out, and the signal within the normal frequency range of human voice is largely retained, which improves the clarity and retains the useful human voice signal to a greater extent; on the other hand, the signal within the normal frequency range of human voice but with very small sound is filtered out, for example, the whispering sound of other people in the surrounding environment, the volume is very small, but it will affect the clear transmission of the caller's voice; for example, the very small buzzing sound generated by mechanical vibration will also affect the transmission of sound; therefore, the signal component with extremely small volume is filtered out, which also improves the clarity.

[0079] In a possible implementation manner of the embodiment of the present application, the first processing module 21 is further used to identify the amplitude of each signal component and filter out all signal components with amplitudes greater than a preset second reference voltage value.

[0080] Among them, the preset second reference voltage value is greater than the preset first reference voltage value. When the callers are talking, the amplitude of the sound is limited. When the volume is extremely high, it is basically a huge noise coming from the surrounding environment, which is very interfering. Therefore, when the amplitude of the signal component is greater than the preset second reference voltage value, the signal component is characterized as a loud noise component, and the signal component is filtered out, which is beneficial to improving the call quality.

[0081] After being processed by the primary filtering module 1 and the first processing module 21, most of the noise is filtered out, such as: most of the high-frequency noise, low-frequency noise, stable noise components, extremely small noise components and extremely large noise components. The first audio processing signal generated after most of the noise is filtered out has less noise and can have better call quality during calls. In order to provide clearer call quality, the first processing module 21 is also connected to the secondary filtering module 22, and the secondary filtering module 22 is connected to the second processing module 23.

[0082] The secondary filtering module 22 is used to receive the first audio processing signal, filter the high-frequency noise and low-frequency noise in the first audio processing signal, and generate a second audio processing signal; the second processing module 23 receives the second audio processing signal, amplifies the second audio processing signal according to a preset second ratio, and after amplification, filters the steady noise component, the bass noise component with extremely low volume, and the treble noise component with extremely high volume in the second audio processing signal to generate a second audio signal.

[0083] Among them, refer to Figure 2 The secondary filtering module 22 includes a resistor R8, a resistor R9, a resistor R10, a capacitor C12 and a capacitor C13, wherein one end of the resistor R8 is connected to the negative electrode of the polarity capacitor C11 of the first processing module 21 to receive the first audio processing signal, and the other end is connected to one end of the capacitor C12, the other end of the capacitor C12 is connected to one end of the resistor R9, and the other end of the resistor R9 outputs the second audio processing signal; the resistor R10 is connected in parallel with the capacitor C20, one end of the resistor R10 is connected to one end of the resistor R8 connected to the capacitor C12, and the other end is grounded.

[0084] Specifically, when the first audio processing signal passes through resistor R8, resistor R10 and capacitor C13, since capacitor C13 has the characteristics of blocking DC and passing AC, it blocks the low-frequency noise component from passing through; the low-frequency noise component in the first audio processing signal is suppressed; when passing through capacitor C12 and resistor R9, it blocks the high-frequency noise component from passing through, so as to suppress the high-frequency noise component.

[0085] Compared with the primary filtering module 1, the secondary filtering module 22 receives signals mixed with less noise signal components. When the secondary filtering module 22 performs filtering, it can achieve more refined suppression of redundant components in the signal, and the filtering accuracy and filtering effect are better.

[0086] The second processing module 23 also uses the MC33219 chip as a processing chip, and its peripheral circuit is similar to that of the first processing module 21, which will not be described in detail. Figure 2The second processing module 23 receives the second audio processing signal, first amplifies the second audio processing signal according to a preset second ratio, and then identifies the stationary noise component, the bass noise component with too low volume, and the treble noise component with too high volume in the amplified second audio processing signal, and filters out the stationary noise component, the bass noise component and the treble noise component. The second processing module 23 will identify and process the signal again, and identify and filter out some noise components that are not identified and filtered out in the first processing module 21 again, so that the output second audio signal is purer and has better call quality.

[0087] It is worth noting that the audio processing module 2 can also include a three-stage filtering module and a third processing module, but considering the economy and purity of the audio signal, the audio processing module 2 includes a first processing module 21, a second processing module and a second processing module 23, which can have higher clarity and is more economical.

[0088] Furthermore, after filtering the noise behind the audio signal, the power amplification module 3 shapes and amplifies the filtered signal and then outputs it to the audio output device for playback.

[0089] Specifically, refer to Figure 2 The power amplifier module 3 receives the second audio signal, amplifies the second audio signal according to a preset amplification ratio, and generates an audio power amplifier signal to the audio output device.

[0090] The power amplification module 3 includes an operational amplifier, a resistor R15, a resistor R16, a resistor R17, a resistor R18, a resistor R19, a resistor R20, a resistor R21, a resistor R22, a resistor R23, a resistor R25, a resistor RX, a capacitor C23, a capacitor C24, a polar capacitor C25, a polar capacitor C26 and a polar capacitor C27.

[0091] Among them, one end of the resistor R15 receives the second audio signal, and the other end is connected to the negative electrode of the polarity capacitor C25, the positive electrode of the polarity capacitor C25 is connected to the positive input terminal of the operational amplifier, one end of the capacitor C23 is connected to the negative electrode of the polarity capacitor C25, and the other end is grounded; the resistor RX is connected in parallel to the two ends of the capacitor C23; one end of the resistor R17 is connected to the positive input terminal of the operational amplifier, and the other end is grounded; one end of the resistor R16 is connected to the positive input terminal of the operational amplifier, and the other end is connected to the 26V power supply; the negative input terminal of the operational amplifier is connected to one end of the resistor R18, and the resistor R18 The other end of the resistor R19 is connected to the positive electrode of the capacitor C27, and the negative electrode of the capacitor C27 is grounded; one end of the resistor R19 is connected to the negative input terminal, and the other end is connected to one end of the resistor R20, the other end of the resistor R20 is connected to the output end of the operational amplifier, and the capacitor C24 is connected in parallel to both ends of the resistor R19; the output end of the operational amplifier is connected to the positive electrode of the polarity capacitor C26, the negative electrode of the polarity capacitor C26 is connected to one end of the resistor R21, and the other end of the resistor R21 outputs the audio power amplifier signal; one end of the resistor R22 is connected to one end of the resistor R21 outputting the audio power amplifier signal, and the other end is grounded.

[0092] The integration circuit formed by the power amplifier module is used to amplify the second audio signal while reducing the offset and signal distortion, so that the generated audio power amplifier signal is clearer during playback.

[0093] In a possible implementation of the embodiment of the present application, the resistor RX is a variable resistor. More specifically, the resistor RX is a sliding variable resistor. The amplification ratio can be adjusted by adjusting the effective resistance of the resistor RX connected to the circuit to meet the requirements of different occasions.

[0094] A possible implementation method of the embodiment of the present application is that, in order to improve the convenience of use, the voice call noise reduction circuit is usually integrated into a module, and the module is encapsulated in a shielding box to protect the internal circuit. When used in a complex environment, it can protect the various components in the voice call circuit and improve applicability.

[0095] Specifically, the shielding box is provided with an input through hole, an output through hole and a power through hole, wherein the power through hole is used to provide an interface for connecting the noise reduction circuit with the power supply, and the voice call noise reduction circuit will be connected to the power supply through the power through hole, the input through hole is used to provide an interface for connecting the noise reduction circuit with the audio input device, and the output through hole is used to provide an interface for connecting the noise reduction circuit with the audio output device. The shielding box is made of metal, specifically aluminum alloy. The voice call noise reduction circuit is encapsulated inside the shielding box, which can protect the circuit components and shield the surrounding signal interference, thereby reducing noise.

[0096] The embodiments of this specific implementation are all preferred embodiments of the present application, and are not intended to limit the protection scope of the present application. The same components are represented by the same figure marks. Therefore, any equivalent changes made based on the structure, shape, and principle of the present application should be included in the protection scope of the present application.

Claims

1. A voice call noise reduction circuit, characterized in that: include: A primary filtering module (1) is connected to an audio input device and is used to receive an audio signal, filter out high-frequency components in the audio signal, and generate a first audio signal; The high-frequency component is a signal in the audio signal whose frequency is higher than the upper limit of the normal frequency range of human voice; An audio processing module (2) is used to receive the first audio signal, filter out a stationary noise component in the first audio signal, and generate a second audio signal, wherein the stationary noise component is a component of a signal whose frequency remains unchanged and remains unchanged within a preset reference time length; A power amplifier module (3) connected to the audio processing module (2) and used for receiving the second audio signal and amplifying the second audio signal according to a preset amplification ratio to generate an audio power amplifier signal; The power amplification module (3) is connected to an audio output device, and the audio output device receives an audio power amplifier signal to play audio.

2. A voice call noise reduction circuit according to claim 1, characterized in that: The audio processing module (2) comprises a first processing module (21), a secondary filtering module (22) and a second processing module (23); The first processing module (21) is connected to the first-level filtering module (1), and is used to receive the first audio signal, filter the stationary noise component in the first audio signal, and generate a first audio processing signal; The secondary filtering module (22) is connected to the first processing module (21) and is used to receive a first audio processing signal to attenuate high-frequency components in the first audio processing signal and then generate a second audio processing signal; The second processing module (23) is connected to the secondary filtering module (22) and is used to receive a second audio processing signal and perform secondary filtering on the stationary noise component in the second audio processing signal to generate a second audio signal.

3. A voice call noise reduction circuit according to claim 2, characterized in that: When filtering the stationary noise component, the first processing module (21) is specifically used to: identifying each signal component in the first audio signal according to differences in frequency and / or duration in the first audio signal; According to each of the duration periods, the duration corresponding to each signal component is determined respectively. If the duration exceeds a preset reference duration, the signal component corresponding to the duration is determined to be a stationary noise component, and the stationary noise component is filtered.

4. A voice call noise reduction circuit according to claim 3, characterized in that: The first processing module (21) is further configured to: identify each signal component in the first audio signal according to the difference in frequency and / or duration in the first audio signal; Each signal component in the first audio signal that is within a normal frequency range of human voice is amplified according to a preset first ratio.

5. A voice call noise reduction circuit according to claim 4, characterized in that: The first processing module (21) is further configured to amplify each signal component in the first audio signal within the normal frequency range of human voice according to a preset first ratio, and then: Detecting the amplitude of each signal component; If the amplitude of any one of the signal components is smaller than the preset first reference voltage value, each signal component with an amplitude smaller than the preset first reference voltage value is filtered out.

6. A voice call noise reduction circuit according to claim 4, characterized in that: The first processing module (21) is further configured to amplify each signal component in the first audio signal within the normal frequency range of human voice according to a preset first ratio, and then: Identify the amplitude of each signal component; All signal components with amplitudes greater than a preset second reference voltage value are filtered out.

7. The voice call noise reduction circuit according to claim 1, characterized in that: The power amplification module (3) comprises an operational amplifier, a resistor R15, a resistor R16, a resistor R17, a resistor R18, a resistor R19, a resistor R20, a resistor R21, a resistor R22, a resistor R23, a resistor R25, a resistor RX, a capacitor C23, a capacitor C24, a polar capacitor C25, a polar capacitor C26 and a polar capacitor C27; One end of the resistor R15 is used to receive the second audio signal, and the other end is connected to the negative electrode of the polarity capacitor C25, the positive electrode of the polarity capacitor C25 is connected to the positive input terminal of the operational amplifier, one end of the capacitor C23 is connected to the negative electrode of the polarity capacitor C25, and the other end is grounded; the resistor RX is connected in parallel to both ends of the capacitor C23; one end of the resistor R17 is connected to the positive input terminal of the operational amplifier, and the other end is grounded; one end of the resistor R16 is connected to the positive input terminal of the operational amplifier, and the other end is connected to the 26V power supply; the negative input terminal of the operational amplifier is connected to one end of the resistor R18, and the positive input terminal of the resistor R18 is connected to the positive input terminal of the operational amplifier. The other end is connected to the positive electrode of capacitor C27, and the negative electrode of capacitor C27 is grounded; one end of resistor R19 is connected to the negative input end, and the other end is connected to one end of resistor R20, the other end of resistor R20 is connected to the output end of the operational amplifier, and capacitor C24 is connected in parallel to both ends of resistor R19; the output end of the operational amplifier is connected to the positive electrode of polarity capacitor C26, the negative electrode of polarity capacitor C26 is connected to one end of resistor R21, and the other end of resistor R21 outputs the audio power amplifier signal; one end of resistor R22 is connected to one end of resistor R21 outputting the audio power amplifier signal, and the other end is grounded.

8. A voice call noise reduction circuit according to claim 7, characterized in that: The resistor RX is a variable resistor.

9. The voice call noise reduction circuit according to claim 1, characterized in that: The primary filtering module (1) comprises a first resistor R1, a second resistor R2, a first capacitor C1 and a second capacitor C2; One end of the first resistor R1 is connected to the audio input device, and the other end is grounded; one end of the first capacitor C1 is connected to the end of the first resistor R1 that is not grounded, and the other end is connected to one end of the second resistor R2, and the other end of the second resistor R2 is connected to the audio processing module (2) for outputting the first audio signal; one end of the second capacitor C2 is connected to the end of the first capacitor C1 that is connected to the second resistor R2, and the other end is grounded.

Citation Information

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