WO2015066926A1

WO2015066926A1 - Noise reduction method

Info

Publication number: WO2015066926A1
Application number: PCT/CN2013/086875
Authority: WO
Inventors: 赵春宁
Original assignee: 赵春宁
Priority date: 2013-11-11
Filing date: 2013-11-11
Publication date: 2015-05-14
Also published as: CN105103219A; CN105103219B

Abstract

Disclosed is a noise reduction method. The method comprises: analyzing noise, and determining a sound source characteristic of the noise; acquiring a sound source signal, the performing the following processing on the acquired sound source signal: correction processing, time delay processing, backward processing and conversion processing, so as to obtain a first reconstructed sound source; and performing cancellation between sound of the first reconstructed sound source and the noise on a first cancellation point. In the present invention, by acquiring noise sound source and performing a series of processing, and a reconstructed sound source that has a vibration direction opposite to that of a noise sound source and has other characteristics exactly the same as those of the noise sound source when the reconstructed sound source reaches a cancellation point is created, by artificially creating a met path, the noise and the reconstructed sound source meet at same time and a same position, a variety of errors caused in the creating the reconstructed sound source, and a "time difference" between the noise and the reconstructed sound source approaches to zero, thereby cancelling the noise. The correction processing in the present invention can effectively cancel transiently changed noise, and can be applied in a huge space environment such as industrial production.

Description

Method of reducing noise

The present invention relates to a method of reducing noise. Background technique

There are two main methods for reducing noise: active noise reduction and passive noise reduction. Active noise reduction is to generate a reverse sound wave equal to the external noise through the noise reduction system, and cancel the noise vibration to achieve noise reduction. Passive noise reduction mainly achieves noise reduction by using sound absorbing materials to absorb sound, or by forming an enclosed space and soundproofing material to block external noise.

The existing noise canceling headphones combine the methods of active noise reduction and passive noise reduction for noise reduction. On the one hand, the noise canceling earphone forms an enclosed space by surrounding the ear, and uses soundproof materials such as silicone earplugs to block external noise; on the other hand, the noise canceling earphone is provided with a signal microphone, which can be used for detecting low frequency noise in the environment that the ear can hear. (100 ~ 1000Hz). The signal microphone transmits the noise signal to the control circuit, and the control circuit performs real-time operation to cancel the noise by Hi-Fi La8 transmitting sound waves with the opposite phase of noise (180° difference) and the same amplitude.

However, if the noise reduction method of the above noise canceling earphone is applied to a place with a large space such as a production workshop, the following defects are still present:

1. For a large noise environment such as a production workshop, the noise is often formed by large production machines, and people need to work on these production machines. Unless the ears are completely covered, a closed space is formed to block the noise. The sound source is almost impossible to achieve.

2, noise canceling headphones mostly use the masking effect of the human ear, as well as through dual MIC identification, filtering voice, separating noise and amplifying voice technology to achieve noise reduction. Mainly by amplifying this sound source to cover up the noise, it is not really to achieve noise reduction by means of sound wave cancellation, and by generating a sound to cover the noise generated by large production machines, it will only bring serious harm to human ears. .

3. The distance between the collected signal microphone and the noise in the noise canceling headphone is not fixed. There is no reference point for the sound phase cancellation. At the same time, the Hi-Fi sound direction is the same as the original signal, and the sound speed cannot be equal. Form the offset condition.

4. Since the speed of the sound is consistent with each frequency, there must be a certain time difference between the sound source and the noise used for the cancellation. Since the space inside the earphone is small, it is possible to achieve offset. However, in a large space environment, this time difference is amplified. In this way, the sound waves and noise emitted by the Hi-Fi speakers are difficult to match in amplitude, time, and space, and cannot reach the same position at the same time, and not only cannot cancel the sound waves of the noise, but also may cause the noise to be enhanced. Summary of the invention It is an object of the present invention to provide a method of reducing noise to solve at least one of the above technical problems.

According to one aspect of the invention, a method of reducing noise is provided, comprising the steps of:

(1) Analyze the noise and determine the sound source characteristics of the noise. The sound source characteristics refer to the inherent characteristics of the sound propagation direction, frequency, wavelength, amplitude, etc., as well as the instantaneous propagation characteristics, frequency, wavelength, amplitude, etc. of the sound;

(2) Acquiring the noise source signal (generally collected near the noise vocalization point and on the side of the noise propagation direction), and processing the collected sound source signal according to the sound source characteristics of the noise as follows: correction processing, delay Processing and reverse processing and conversion processing to obtain a first reconstructed sound source;

(3) The sound and noise propagating from the first reconstructed sound source are cancelled at the first offset point to obtain a first-order noise reduction.

The present invention creates a reconstructed sound source by collecting a noise source signal and a series of processes, the reconstructed sound source having the same direction of vibration as the noise source and having the same other source characteristics as the noise source except for the direction of the vibration. By artificially setting a path where the noise meets the reconstructed sound source, the two sound waves meet at the same time and at the same position, eliminating the "time difference" between the noise and the reconstructed sound source reaching the offset point (making the "time difference" tend to zero ), to achieve the cancellation of noise. The correction processing of the present invention can effectively cancel the instantaneously varying noise, so that the noise reduction method of the present invention can be applied to an environment of a large space such as industrial production, and the harm caused by noise can be reduced.

In some embodiments, the noise source signal can be collected by a cardioid pointing microphone, and the first re-enactment is performed by an acoustic transducer (eg, a speaker, a speaker, a flat sounder, a Haier sounder, a piezoelectric sounder, etc.) The source is spread. The high-performance heart-shaped microphone and sound transducer can more effectively complete the sound restoration, which can reduce the howling of the loop and prevent the sound from being distorted.

In some embodiments, the first cancellation point may be close to the utterance point of the first reconstructed sound source, that is, the distance between the first cancellation point and the utterance point of the first reconstructed sound source is smaller than the first cancellation point and the utterance point of the noise. the distance. Wherein, the distance between the first cancellation point and the utterance point of the first reconstructed sound source is less than 1/2 of the high frequency wavelength (λ) of the noise, and the distance between the first cancellation point and the utterance point of the first reconstructed sound source is better. The 1/4 effect of the high frequency wavelength smaller than the noise is optimal. The noise source signal is collected close to the noise sounding point and in the direction of noise propagation, and the offset point is close to the sounding point of the reconstructed sound source, so that there is enough time to perform relevant correction and processing on the collected sound source, so that the noise And the re-creation of the sound source can achieve the simultaneous arrival of the offset point, reduce the "time difference", and achieve complete ^^ elimination.

In some embodiments, the above step (2) may include:

Performing A/D conversion on the collected sound source signal;

Correcting the error caused to the sound source signal during the acquisition process;

Delaying the collected sound source signal, and loading the delayed signal to make the sound source signal reach the first offset point simultaneously with the noise when it is converted into sound energy for propagation; The conversion process is performed using an acoustic transducer (the distance between the heart-shaped pointing microphone and the acoustic transducer is fixed), and the acoustic transducer is corrected before the conversion process;

Reverse processing the sound source signal such that the vibration direction of the sound source signal is opposite to the vibration direction of the noise;

Correcting errors in the delay signal processing, reverse processing, and conversion processing to the sound source signal;

The sound source signal is D/A converted and converted into sound energy for propagation.

Hardware devices used in acquisition, playback, transmission, etc. (eg, heart-shaped microphones, acoustic transducers, ADCs, DACs, DSP processing chips, registers, memory, power amplifiers, connectors, transmission links, etc.) More or less changes in the collected sound source signal (that is, "distortion", mainly refers to the transfer process between the equipment itself or the inertia of the sound movement, various reflections, diffraction, hardware and Additional or missing sound waves generated by factors such as software). At the same time, after a period of operation of the system, slight changes in the electrical performance of the components may occur. For acoustic cancellation, these subtle differences will not only make the two sources unable to cancel, but may even cause the two sources to overlap each other, with undesirable consequences. Therefore, only after this series of precise corrections and adjustments can the sound source signal strictly conform to the transient characteristics of noise vibration, and also ensure that the reconstructed sound source and noise simultaneously reach the offset point and cancel at the offset point.

In some embodiments, in the noise reduction method described above, the following processing can also be performed for the first level noise reduction:

Collecting a first-level noise reduction sound source signal to measure the cancellation effect of the first reconstructed sound source and noise, and correcting the sound source signal processing in step (2) according to the measurement result, adjusting the delay signal loading delay Length or adjust the position of the first offset point. The acquired noise signal and the processed source signal are distorted and can be obtained by this step. At the same time, through this step, it is also possible to test the degree of compliance between the position noise of the offset point and the reconstructed sound source.

Through the above measurement, not only the offset result can be monitored, but also the characteristics of the reconstructed sound source and the position of the offset point can be adjusted in time to ensure that the noise and the reconstructed sound source can reach the offset point at the same time and achieve complete cancellation.

In some embodiments, the following processing can also be performed on the first level noise reduction:

Analyze the noise reduction at the first level and determine the sound source characteristics of the first-level noise reduction;

The first-level noise-reducing sound source signal is collected, and the collected first-level noise-reduced sound source signal is processed according to the sound source characteristics of the first-order noise reduction: correction processing, delay processing, reverse processing, and conversion processing, Second reconstructed sound source;

The sound propagated by the second reconstructed sound source and the first-order noise reduction are cancelled at the second offset point to obtain a second-stage noise reduction.

The above process can be further cycled. For example, the second-level noise reduction can be further processed as follows: Analyze the secondary noise reduction and determine the sound source characteristics of the secondary noise reduction;

The sound source signal of the second-level noise reduction is collected, and the sound source signal of the second-level noise reduction is processed according to the sound source characteristics of the second-level noise reduction: correction processing, delay processing, reverse processing, and conversion processing, Third re-created sound source;

The sound propagated by the third reconstructed sound source and the secondary noise reduction are cancelled at the third offset point.

For some complex noises, it is more difficult to achieve the required noise reduction requirements through one noise reduction process. Therefore, it is possible to split the frequency band and coverage of the noise, and gradually reduce the noise of the split, and realize the multi-stage processing. The purpose of noise reduction. DRAWINGS

1 is a schematic diagram of the principle of a method for reducing noise according to an embodiment of the present invention.

Fig. 2 is a flow chart showing the process of obtaining the first-order noise reduction in Fig. 1.

Figure 3 is a response diagram of the magnitude of the amplitude at each frequency point of the acoustic transducer before correction.

Figure 4 is a response diagram of the magnitude of the amplitude at each frequency point of the corrected acoustic transducer.

Figure 5 is an impulse response diagram of the pre-corrected acoustic transducer.

Figure 6 is an impulse response diagram of the corrected acoustic transducer.

Fig. 7 is a graph showing changes in noise before and after the processing in Fig. 2. detailed description

The invention will now be described in further detail with reference to the accompanying drawings.

Fig. 1 schematically shows a schematic diagram of the principle of a method for reducing noise according to an embodiment of the present invention. As shown in FIG. 1, a method for reducing noise includes the following steps:

The noise 101 is analyzed to determine the sound source characteristics of the noise 101.

The sound source signal of the noise 101 is collected by a cardioid pointing microphone at a side close to the noise 101 sounding point and on the side of the noise propagation direction.

According to the sound source characteristics of the noise 101, suitable hardware and software are selected to perform a series of processing such as correction, delay, and reverse on the collected sound source signals.

The processed signal is converted into acoustic energy by the acoustic transducer, and the first reconstructed sound source 103 is obtained. The first reconstructed sound source 103 propagates at the first offset point 102 (close to the sound of the first reconstructed sound source 103). Point) and noise 101 are eliminated, and the first-order noise reduction 104 is obtained.

Acquiring the sound source signal of the first-level noise reduction 104 to measure the cancellation effect of the first reconstructed sound source 103 and the noise 101, and selecting appropriate software and hardware for processing the sound source signal of the collected noise according to the measurement result (or even replacing Software and hardware), correct the processing of the sound source signal, adjust the length of the delay signal loading delay or adjust the position of the first offset point.

The primary noise reduction 104 is analyzed to determine the sound source characteristics of the primary noise reduction 104. A sound source signal for collecting the primary noise reduction 104 is performed by a cardioid pointing microphone at a side close to the first-order noise reduction 104 sounding point and on the side of the first-order noise reduction 104.

According to the sound source characteristics of the first-level noise reduction 104, suitable hardware and software are selected to perform a series of processing such as correction, delay, and reverse of the collected sound source signals of the first-order noise reduction 104.

The processed signal is converted into acoustic energy by the acoustic transducer to obtain a second reconstructed sound source 106, and the second reconstructed sound source 106 propagates, at the second offset point 105 (close to the sound of the second reconstructed sound source 106) Point) and the first-level noise reduction 104 are eliminated, and the second-level noise reduction 107 is obtained.

Acquiring the sound source signal of the second-level noise reduction 107 to measure the cancellation effect of the second reconstructed sound source 106 and the first-stage noise reduction 104, and selecting an appropriate processing method to collect the sound source signal of the first-level noise reduction 104 according to the measurement result. Software and hardware (even replacing software and hardware), as well as adjusting various parameters during processing.

Analyze the secondary noise reduction 107, and determine the sound source characteristics of the secondary noise reduction 107.

The sound source signal of the secondary noise reduction 107 is acquired by a cardioid pointing microphone at a side close to the secondary noise reduction 107 sounding point and on the side of the secondary noise reduction 107.

According to the sound source characteristics of the secondary noise reduction 107, suitable hardware and software are selected to perform a series of processing on the collected sound source signals of the secondary noise reduction 107.

The processed signal is converted into acoustic energy by an acoustic transducer to obtain a third reconstructed sound source 109, and the third reconstructed sound source 109 propagates, at a third offset point 108 (close to the sound of the third reconstructed sound source 109) The point is offset by the secondary noise reduction 10 to obtain a silent environment that meets the standard, that is, the muffling sound field 110.

The sound source signal of the anechoic sound field 110 is collected to measure the effect of the third reconstructed sound source 109 and the second-level noise reduction 10, and according to the measurement result, the sound source of the second-level noise reduction 10 collected by the appropriate processing is selected. Signal software and hardware (even replacement software and hardware), and adjust various parameters during processing.

In the present embodiment, the noise 101 is subjected to three levels of noise reduction processing. However, in other embodiments, noise 101 may be subjected to multiple levels of noise reduction (e.g., secondary, quadruple, and fifth) as needed to achieve the desired noise reduction requirements.

Figure 2 shows a process flow diagram for obtaining a first level noise reduction 104.

As shown in Fig. 2, at a position close to the noise 101 sounding point and on the side of the propagation direction of the noise 101, the sound source signal of the noise 101 is collected by the heart-shaped pointing microphone 201 (step S201).

The collected sound source signal is converted into a digital signal by A/D conversion (step S202). At this point, some of the inherent characteristics and transient characteristics of noise 101 can be read, such as: frequency, amplitude, phase, and so on. Based on these characteristics, you can select the appropriate sound processing software and hardware, such as: A bass sound source, which can't respond with a high-pitched sound transducer, and sometimes even a multi-transducer combination of sound source signals for processing.

Since the heart-shaped pointing microphone is used to collect the sound source signal, the resulting signal will be more or less There are some errors. Generally, before the heart-shaped pointing microphone is used, it is measured by an audio analysis system to obtain a compensation value and a correction value for the inherent characteristic of the heart-shaped pointing microphone. The acquired sound source signal is corrected by the DSP processing using this compensation value and the correction value to correct the error caused by the cardioid pointing microphone 201 to the sound source signal (step S203). The corrected information includes: sound pressure correction data at each frequency point in the frequency band, and phase data between the respective frequency points, and the signal captured by the heart shape pointing to the microphone can be made coincident with the original noise signal by correction.

Since the acoustic transducer (the acoustic transducer used in the embodiment is an electric speaker) mainly realizes the function of converting an electrical signal into an acoustic signal, it is necessary to complete frequency conversion of one frequency band. The sounding position and starting time of the sound transducers at different frequency points are not the same, which results in the difference in amplitude and frequency relative phase of each frequency. Therefore, in addition to correcting the error caused by the cardioid pointing microphone 201 to the sound source signal, it is also necessary to correct the error caused by the acoustic transducer to the sound source signal (step S204) to achieve faithful restoration of the original signal.

In addition to the heart-shaped pointing microphone and acoustic transducer, the processor, ADC,

The DAC, memory, registers, power amplifier, transmission link, etc. all have the potential to cause signal delay, frequency change or amplitude attenuation, and need to be corrected (step S205).

Correction of the heart-shaped microphone, acoustic transducer, system hardware and transmission link can be realized by digital processing hardware and software. It can be processed by hardware or software algorithms such as processor or professional DSP chip.

This embodiment performs the testing and correction of the transducer and system by means of an audio analysis system and audio DSP processing. The audio test is divided into steady state test and transient test. In this embodiment, SMAARTLIVE7 software is used for testing. The steady-state test method is: The system itself sends out a continuous test signal, which is a wide-frequency noise signal used as a test basis. This reference is represented by a channel data. This reference is a loop that is reflected on the tester. Since the input and output are loops, the system will display a straight line. The same reference signal is sent to the system that needs to be measured. After the response of the measurement system, it is converted by an FFT (Fast Fourier), which is a heart-shaped pointing microphone (which can be an electrical signal and an acoustic signal), and displayed in another channel. The obtained result is compared with the original signal, that is, two or more channels are compared, and the problem of comparison can be visually seen. The steady state test is measured with a continuous signal, and the transient test is tested with a pulse signal. The same principle is used for the relative phase test. The fixed frequency is the phase start point, and the other frequencies are used as comparisons to obtain different phase responses, that is, the response differences of time.

The process of signal processing is performed by a general-purpose processor or a professional DSP processing chip in combination with corresponding software. In this example, the hardware system consisting of ADI's SHARC ADSP-21448 processing chip is used and the corresponding software is used for signal processing. The signal processing process is: (1) performing input compensation processing (including amplitude frequency response and phase response characteristics) on the A/D converted sound source signal according to the pre-tested heart-shaped pointing microphone calibration data, to correct the measurement microphone Bringing in error (step S203); (2) According to the above audio test to correct the error brought in by the acoustic transducer, including amplitude frequency response, phase response, compensation and correction of transient response (step S204); (3) taking the entire system according to the above audio test The error includes amplitude frequency response, phase response, compensation and correction of transient response (step S205).

In addition, the present embodiment also uses an integrated algorithm such as FIR filter and ALLPASS filter to correct the amplitude and phase, and uses a reverse signal to correct the response of the transient.

Figure 3 is a response diagram of the magnitude of the amplitude at each frequency point of the acoustic transducer before correction, and Figure 4 is a response plot of the magnitude of the amplitude at each frequency point of the corrected transducer. It can be seen from Fig. 3 and Fig. 4 that after correction, the amplitude response of each frequency point of the original transducer is corrected, and the phase response is also straight, so that the input and output signals are consistent.

Figure 5 is an impulse response diagram of the pre-corrected acoustic transducer, and Figure 6 is an impulse response diagram of the corrected acoustic transducer. It can be seen from Fig. 5 and Fig. 6 that before the optimization, several additional pulses with a large amplitude and a large residual vibration pulse at a later time are added under the main pulse; the reverse filter is applied to the main clutter. Corrected, the main clutter becomes smaller, the additional clutter becomes less, and it is closer to the original waveform from the transient point of view.

The corrected correction results in the above steps S203 to S205 are all superimposed on the original heart-shaped signal input to the microphone, thus forming a comprehensively corrected signal.

With the above hardware and software, a delayed signal is applied to the corrected signal (step S206), and then reverse processing is performed (step S207). Then, the signal is D/A processed, converted into an analog signal (step S208), the analog signal is output to the power amplifier (step S209), and finally, it is propagated through the speaker 206, thus obtaining the first reconstructed sound source 103 ( Step S210).

The first reconstructed sound source 103 propagates in a path set in the air, the first canceling point 102 is close to the speaker 206, and the first reconstructed sound source 103 and the noise 101 are cancelled at the first canceling point 102 to obtain a first-level noise reduction 104. . In this embodiment, the distance between the first cancellation point 102 and the utterance point of the first reconstructed sound source 103 is less than 1/2 of the noise wavelength. In other embodiments, the first cancellation point 102 and the first reconstructed sound source 103 are audible. The distance of the point is less than 1/4 of the wavelength of the noise.

A first measurement microphone 301 can be provided at the first cancellation point 102, and a second measurement microphone 303 is provided at the primary noise reduction 104 (anechoic sound field), and the sound source signal collected by the test system 302 for the first measurement microphone 301 is The noise signal is compared, and the sound source signal collected by the second measurement microphone 303 is compared with the noise signal to determine the noise reduction effect. At the same time, according to this effect, the processing system can be further corrected.

In summary, the present invention performs system adjustment and matching through three loops and steps.

The first adjustment loop is: First, a comprehensive analysis of the reconstructed sound source system consisting of a cardioid pointing microphone 201, a signal processing hardware, a power amplifier, an acoustic transducer, a link, and a plug-in is performed through an audio analysis system. That is, a signal is sent to the heart-shaped pointing microphone 201, so that the result of the measurement by the reconstructed sound source system through the first measuring microphone 301 can be obtained, and the result is compared with the original test signal to determine the degree of agreement between the two signals, thereby being adjusted. Correction parameters for the entire system. The debugging process for the parameters of the entire system is:

1. Adjust the cardioid pointing microphone 201 separately, mainly by correcting with the same standard sound source method. That is, a standard sound source (resonant speaker) is used as the measurement sound source, and the measurement microphone that has been obtained and corrected is used as a comparison standard. The difference between the two microphones can be obtained by the audio analysis system. Two data of frequency response and phase response are included, and this data is obtained as a correction parameter of the heart shape pointing to the microphone 201. This parameter is directly input into the compensation data of the DSP software program, and the adjustment of the parameters of the cardioid pointing microphone 201 is completed.

2. Adjust the acoustic transducer separately, mainly using the audio analysis system to measure the acoustic transducer, including two categories of steady state and transient. The amplitude, phase, and transient are corrected by the DSP and its software, and the corrected data is stored in the DSP software program.

3. Adjust the conformity of the reconstructed sound source by the isolation method. With a source of defined frequency (which can be emitted by the speaker), the source is first separated into an isolated position that is isolated from the acoustic transducer. Capture the signal with a measuring microphone to form a reference, and simultaneously capture the acoustic signal with the heart-shaped pointing microphone at the same position of the measuring microphone, send the signal to the reconstructed sound source system, and capture the sound signal of the reconstructed sound source with another microphone. The result is compared with the benchmark, and the difference between the entire loop of the reconstructed sound source is formed, and the correction data is written again to the DSP program.

The second debug loop is: Reconstructing the sound source to match the delay of the original noise through the first measurement microphone

301 is obtained via test system 302. In this way, the loaded delay signal can be corrected. The short-time pulse is mainly used as the debugging sound source to adjust the delay. That is, another speaker emits a short-time pulse as a debugging sound source, and measures the time to the offset point. The position of the debug sound source is fixed, and the time when the sound source reaches the cancel point after the sound source is reproduced is measured again. The offset point is as close as possible to the speaker. The position of the offset point is also the position of the measurement microphone. The above two data are measured by the measuring microphone. Adjust the position of the heart pointing to the offset point of the microphone so that the time at which the reconstructed sound source reaches the offset point is slightly less than the time the debug source reaches the offset point. By adding a precise delay time to the reconstructed sound source, the time at which the two sound waves reach the 4th cancellation point is equal.

The third debugging loop is: The difference between the reconstructed sound source and the original noise can be obtained by the second measuring microphone 303 of the testing system 302. At this time, the consistency of the two sound sources can be compared and the offset effect of the reconstructed sound source can be reversed. Repeated adjustments to the parameters of the entire system, as well as multiple offsets and optimization of the program.

Because the time at which the noise 101 reaches the first cancellation point 102 is fixed, and can be measured by the first measurement microphone 301. Therefore, it is only necessary to adjust the distance of the heart-shaped pointing microphone 201 from the offset point to achieve the matching with the delay of the reconstructed sound source system, that is, the delay of the reconstructed sound source system is large, and the heart-shaped pointing microphone 201 is far away from the offset point, the target It is to ensure that the original noise and the reconstructed sound source are listed on both sides of the offset point, and the direction of propagation in the air is opposite. At this time, the time at which the reconstructed sound source reaches the offset point should be slightly shorter than the original noise. The first reconstructed sound source 103 can be reached by adjusting the delay of the reconstructed sound source after the measurement. The time of the first cancellation point 102 arrives at the same time as the noise 101.

If the noise 101 points in a heart shape to the microphone 201 as a measurable reference point, the time to reach the first cancellation point 102 is T1, the time at which the transducer spatially propagates the sound wave to reach the first cancellation point 102, and the entire direction from the heart shape to the microphone 201 captures the propagation of the noise 101 signal and the digital processing process and the calibration process delay and all analog and digital device fixed delays are collectively referred to as the overall system integrated delay time is T3. Since the sound propagation speed is slower than the propagation speed of the electrical signal, the distance between the microphone 201 and the offset point can be adjusted by the heart shape, and Τ1>Τ2+Τ3 is ensured, and a delay Τ4 is applied to the collected sound source in step 204 for fine adjustment. Let Τ1=Τ2+Τ3+Τ4, so that the first reconstructed sound source 103 and the noise 101 reach the first canceling point 102 at the same time.

Fig. 7 is a graph showing changes in noise before and after the processing in Fig. 2. As shown in FIG. 7, the line is an environmental noise curve, and line B is a noise source continuously from a frequency range of 30 Hz to 2 KHz. After the processing of steps S202 to S210 in FIG. 2, a reconstructed sound source is obtained, and the reproduced sound source is as shown in FIG. Noise processing, after completing the offset, get line C. As can be seen from Fig. 7, after the noise reduction method of the present invention, the noise has been significantly reduced. What has been described above is only one embodiment of the present invention. It will be apparent to those skilled in the art that various modifications and improvements can be made without departing from the spirit and scope of the invention.

Claims

Claim

1. A method of reducing noise, including the following steps:

(1) Analyze the noise and determine the sound source characteristics of the noise;

(2) collecting the noise source signal, and performing the following processing on the collected sound source signal according to the sound source characteristics of the noise: correction processing, delay processing, reverse processing and conversion processing to obtain the first reconstructed sound source;

(3) The sound propagating the first reconstructed sound source is cancelled with the noise at the first canceling point to obtain a first-order noise reduction.

The method of reducing noise according to claim 1, wherein in the step (2), the noise sound source signal is collected by a cardioid directional microphone.

The method of reducing noise according to claim 1, wherein said step (2) comprises:

Performing A/D conversion on the collected sound source signal;

Delaying the collected sound source signal, and loading the delayed signal to make the sound source signal reach the first offset point simultaneously with the noise when it is converted into sound energy for propagation;

The conversion process is performed using an acoustic transducer, and the acoustic transducer is corrected before the conversion process;

Correcting the error caused by the delay signal processing, the reverse processing and the conversion processing to the sound source signal; performing D/A conversion on the sound source signal, converting into sound energy, thereby enabling propagation.

The method of reducing noise according to any one of claims 1 to 3, characterized in that, in the step (3), the first reconstructed sound source is propagated through the acoustic transducer.

The method for reducing noise according to claim 4, wherein the collected noise source signal is performed near a sounding point of the noise and in a noise propagation direction, and the first offset point is close to the first Repeatedly making the sound point of the sound source.

The method of reducing noise according to claim 4, wherein the distance between the first cancellation point and the utterance point of the first reconstructed sound source is less than 1 /2 of the high frequency wavelength of the noise.

The method of reducing noise according to claim 4, wherein the distance between the first canceling point and the sounding point of the first reconstructed sound source is less than 1/4 of the high frequency wavelength of the noise.

8. The method of reducing noise according to claim 3, further comprising the following processing of the first order noise reduction:

Collecting a first-level noise-reducing sound source signal, measuring the offset effect of the first reconstructed sound source and noise, and correcting the sound source signal in step (2) according to the measurement result, adjusting the length of the delay signal loading delay or Adjust the position of the first offset point.

9. The method of reducing noise according to claim 8, further comprising the following processing of the first order noise reduction:

The sound propagated by the second reconstructed sound source is canceled at the second offset point with the first-order noise reduction, and the second-level noise reduction is obtained.

10. The method of reducing noise according to claim 9, further comprising the following processing of the secondary noise reduction:

Analyze the secondary noise reduction and determine the sound source characteristics of the secondary noise reduction;

The sound propagated by the third reconstructed sound source and the second-level noise reduction are cancelled at the third offset point.