CN105103219A

CN105103219A - Noise reduction method

Info

Publication number: CN105103219A
Application number: CN201380075550.3A
Authority: CN
Inventors: 赵春宁
Original assignee: Individual
Current assignee: Individual
Priority date: 2013-11-11
Filing date: 2013-11-11
Publication date: 2015-11-25
Anticipated expiration: 2033-11-11
Also published as: WO2015066926A1; 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 for reducing noise

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

There are two main methods for reducing noise: active noise reduction and passive noise reduction. The active noise reduction is realized by generating reverse sound waves equal to external noise through a noise reduction system and offsetting noise vibration. Passive noise reduction is mainly achieved by using sound absorbing materials to absorb sound or by forming an enclosed space and sound insulating materials to block outside noise.

The existing noise reduction earphone combines the active noise reduction mode and the passive noise reduction mode to reduce noise. On one hand, the noise reduction earphone forms a closed space by surrounding ears, and adopts sound insulation materials such as silica gel earplugs and the like to block external noise; on the other hand, a signal microphone is arranged in the noise reduction earphone and can be used for detecting low-frequency noise (100-1000 Hz) in the environment which can be heard by the ear. The signal microphone transmits the noise signal to the control circuit, and the control circuit performs real-time operation to counteract the noise through the sound wave with opposite phase (180 DEG difference) and same amplitude emitted by the Hi-Fi loudspeaker.

However, if the noise reduction method of the noise reduction headphone is applied to a place with a large space such as a production shop, the following defects still exist:

1. in the case of a noisy environment with a large space such as a production workshop, noise is often generated by a large-sized production machine, and people need to work on the production machine, and unless the ears are completely wrapped, a closed space is formed to isolate a sound source of the noise, which is hardly realized.

2. The noise reduction earphone mostly utilizes the masking effect of human ears and achieves the noise reduction effect through the dual-MIC identification, voice filtering, noise separation and voice amplification technology. The noise is mainly covered by amplifying the sound source, the noise reduction is not really realized by a sound wave cancellation mode, the noise generated by a large production machine is covered by generating a sound, and only serious harm is brought to the ears of people.

3. The distance between a signal collecting microphone and noise in the noise reduction earphone is not fixed and is not used as a reference point for sound phase cancellation, and meanwhile, the sound production direction of the Hi-Fi loudspeaker is the same as that of an original signal, and a cancellation condition cannot be formed under the condition that sound velocities are equal.

4. Since the speed of sound is consistent with each frequency, there must be a certain time difference between the sound source used for cancellation and the propagation process of noise, and since the space in the earphone is small, cancellation is also possible. However, in a spatially large environment, this time difference is amplified. Thus, the sound wave emitted by the Hi-Fi speaker is difficult to match with noise in amplitude, time and space, and cannot reach the same position at the same time, so that the sound wave cannot be offset with the sound wave of the noise, and the noise can be strengthened. Disclosure of the inventionit 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 an aspect of the present invention, there is provided a method of reducing noise, comprising the steps of:

(1) analyzing the noise and determining the sound source characteristics of the noise, wherein the sound source characteristics refer to inherent characteristics such as the propagation direction, frequency, wavelength and amplitude of the sound and transient characteristics such as the instantaneous propagation direction, frequency, wavelength and amplitude of the sound;

(2) collecting a noise source signal (generally collecting on one side close to a noise sound production point and in a noise propagation direction), and performing the following processing on the collected noise source signal according to a noise source characteristic: correcting, delaying, reversing and converting to obtain a first reproduced sound source;

(3) and canceling the sound and the noise transmitted by the first reproduced sound source at a first canceling point to obtain first-level noise reduction.

The present invention creates a reproduced sound source that vibrates in a direction opposite to and has the same other sound source characteristics as the noise source except for the direction of vibration by collecting the noise source signal and a series of processes. The method has the advantages that the path for meeting the noise and the reconstructed sound source is artificially arranged, so that the two sound waves meet at the same time and the same position, the time difference between the noise and the reconstructed sound source reaching the counteracting point is eliminated (the time difference is made to tend to zero), and the counteracting of the noise is realized. The correction processing of the invention can effectively cancel the noise of instantaneous change, so that the noise reduction method of the invention can be suitable for the environment of large space such as industrial production and the like, and the harm caused by the noise is reduced.

In some embodiments, a noise source signal may be collected by a cardioid microphone, and a first reproduced sound source may be propagated by an acoustic transducer (e.g., a speaker, a sound box, a flat panel sounder, a hel sounder, a piezoelectric sounder, etc.). The high-performance heart-shaped directional microphone and the sound transducer can more effectively finish the sound restoration work, reduce the loop squeal and prevent the sound distortion.

In some embodiments, the first cancellation point may be close to the sound emission point of the first reproduced sound source, i.e. the first cancellation point is located at a distance from the sound emission point of the first reproduced sound source that is less than the distance from the first cancellation point to the sound emission point of the noise. The distance between the first canceling point and the sound emitting point of the first reproduced sound source is preferably 1/2, which is smaller than the high frequency wavelength (λ) of the noise, and 1/4, which is smaller than the high frequency wavelength (λ) of the noise, is most preferably used. The collection of noise source signals is carried out close to a noise sound production point and in the transmission direction of noise, and the offset point is close to the sound production point of the reconstructed sound source, so that enough time is provided for relevant correction and processing of the collected sound source, the noise and the reconstructed sound source can reach the offset point at the same time, the time difference is reduced, and the complete ^ elimination is realized.

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

carrying out A/D conversion on the collected sound source signals;

correcting errors brought to sound source signals in the acquisition process;

the collected sound source signals are subjected to delay processing, and the sound source signals can reach a first offset point simultaneously with noise when being converted into sound energy for transmission by loading the delay signals; the conversion process is performed using an acoustic transducer (the distance between the cardioid microphone and the acoustic transducer is fixed), and before the conversion process, the acoustic transducer is subjected to a correction process;

carrying out reverse processing on the sound source signal to enable the vibration direction of the sound source signal to be opposite to the vibration direction of the noise;

correcting errors brought to sound source signals in delay processing, reverse processing and conversion processing;

the sound source signal is D/a converted into sound energy, thereby being propagated.

Because hardware devices (such as a cardioid microphone, an acoustic transducer, an ADC, a DAC, a DSP processing chip, a register, a memory, a power amplifier, a connector, a transmission link, etc.) used in the processes of acquisition, playing, transmission, etc., can change the acquired sound source signal (i.e., "distortion", which mainly refers to additional or missing sound waves caused by the defects of the devices themselves, or the factors of the transfer process between the devices and the inertia characteristics of sound movement, various reflections, diffraction, hardware and software, etc.), and at the same time, the system may have slight changes in the electrical performance of the components after working for a period of time. For sound wave cancellation, these slight differences will not only make the two sound sources non-canceling, but may even result in the two sound sources overlapping each other with undesirable consequences. Therefore, only through the series of precise correction and adjustment, the sound source signal can strictly accord with the transient characteristics of noise vibration, and the reconstructed sound source and the noise can be ensured to simultaneously reach the offset point and be offset at the offset point.

In some embodiments, in the above method for reducing noise, the following processing may be further performed on the first-stage noise reduction:

collecting a primary noise reduction sound source signal to measure the counteracting effect of the first reproduced sound source and noise, and correcting the sound source signal processing in the step (2) according to the measuring result, adjusting the loading delaying length of the delayed signal or adjusting the position of a first counteracting point. The collected noise signal and whether the processed sound source signal is distorted can be obtained through the step. Meanwhile, through the step, the consistency degree of the noise at the position of the cancellation point and the reproduced sound source can be tested and known.

Through the measurement, the counteracting result can be monitored, and the characteristics of the reconstructed sound source, the position of the counteracting point and other elements can be adjusted in time, so that the noise and the reconstructed sound source can reach the counteracting point at the same time and complete counteracting is realized.

In some embodiments, the following may also be performed on the first order noise reduction:

analyzing the primary noise reduction, and determining the sound source characteristics of the primary noise reduction;

collecting a sound source signal of the primary noise reduction, and processing the collected sound source signal of the primary noise reduction according to the sound source characteristics of the primary noise reduction as follows: correcting, delaying, reversing and converting to obtain a second reconstructed sound source;

and offsetting the sound transmitted by the second reconstructed sound source and the primary noise reduction at a second offset point to obtain secondary noise reduction.

The above process can be further cycled, for example, the following process can be performed on the second-stage noise reduction: analyzing the secondary noise reduction, and determining the sound source characteristics of the secondary noise reduction;

collecting a sound source signal of the secondary noise reduction, and processing the collected sound source signal of the secondary noise reduction according to the sound source characteristics of the secondary noise reduction as follows: correcting, delaying, reversing and converting to obtain a third reconstructed sound source;

and the sound transmitted by the third reproduced sound source and the secondary noise reduction are counteracted at a third counteracting point.

For some complex noises, it is difficult to achieve the required noise reduction requirement through one-time noise reduction treatment, so that the frequency band and coverage of the noise can be split, the split noise is gradually reduced, and the purpose of noise reduction is achieved through multi-stage treatment. Drawings

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

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

Fig. 3 is a response graph of amplitude magnitude at various frequency points of the acoustic transducer before correction.

Fig. 4 is a response graph of the magnitude of the amplitude at each frequency point of the sound transducer after correction.

Fig. 5 is a graph of the impulse response of the acoustic transducer before calibration.

Fig. 6 is a graph of the impulse response of the corrected acoustic transducer.

Fig. 7 is a graph showing the change in noise before and after the processing by the flow shown in fig. 2. Detailed Description

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

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

the noise 101 is analyzed and the source characteristics of the noise 101 are determined.

A sound source signal of the noise 101 is collected by a cardioid directional microphone near a sound emission point of the noise 101 and on one side of a propagation direction of the noise.

And selecting proper hardware and software according to the sound source characteristics of the noise 101 to perform a series of processes such as correction, time delay, inversion and the like on the acquired sound source signal.

The processed signal is converted into acoustic energy by an acoustic transducer to obtain a first reproduced sound source 103, and the first reproduced sound source 103 propagates and cancels the noise 101 at a first cancellation point 102 (a sound emission point close to the first reproduced sound source 103) to obtain a first-order noise drop 104.

Collecting a sound source signal of the primary noise reduction 104 to measure the counteracting effect of the first reproduced sound source 103 and the noise 101, and selecting appropriate software and hardware (even replacing the software and the hardware) for processing the collected sound source signal of the noise according to the measurement result, correcting the sound source signal processing, and adjusting the loading delaying length of the delaying signal or the position of a first counteracting point.

The primary noise reduction 104 is analyzed to determine the source characteristics of the primary noise reduction 104. The acquisition of the source signal of the primary noise drop 104 is performed by a cardioid directional microphone near the point where the primary noise drop 104 originates and on one side of the propagation direction of the primary noise drop 104.

And selecting proper hardware and software according to the sound source characteristics of the primary noise reduction 104 to perform a series of processes such as correction, time delay, inversion and the like on the acquired sound source signal of the primary noise reduction 104.

The processed signal is converted into sound energy through an acoustic transducer to obtain a second reproduced sound source 106, the second reproduced sound source 106 is propagated, and the second reproduced sound source 106 and the first-level noise reduction 104 are subjected to cancellation at a second cancellation point 105 (a sound production point close to the second reproduced sound source 106), so that a second-level noise reduction 107 is obtained.

Collecting the sound source signal of the secondary noise reduction 107 to measure the counteracting effect of the second reproduced sound source 106 and the primary noise reduction 104, and selecting appropriate software and hardware (even replacing the software and hardware) for processing the collected sound source signal of the primary noise reduction 104 according to the measurement result, and adjusting various parameters in the processing process.

And analyzing the secondary noise reduction 107 and determining the sound source characteristics of the secondary noise reduction 107.

The acquisition of the sound source signal of the secondary noise drop 107 is performed by a cardioid directional microphone near the point where the secondary noise drop 107 is emitted and on one side of the propagation direction of the secondary noise drop 107.

The acquired sound source signal of the secondary noise reduction 107 is subjected to a series of processes by selecting appropriate hardware and software according to the sound source characteristics of the secondary noise reduction 107.

The processed signals are converted into sound energy through an acoustic transducer to obtain a third reproduced sound source 109, the third reproduced sound source 109 is transmitted, and the third reproduced sound source 109 and the second-level noise reduction 10 are counteracted at a third counteracting point 108 (a sound producing point close to the third reproduced sound source 109) to obtain a standard mute environment, namely a silencing sound field 110.

Collecting sound source signals of the muffled sound field 110 to measure the 4 ℃ muffling effect of the third reproduced sound source 109 and the secondary muffling noise 10, and based on the measurement result, selecting appropriate software and hardware (even replacing software and hardware) for processing the collected sound source signals of the secondary muffling noise 10, and adjusting various parameters in the processing process.

In this embodiment, noise 101 is subjected to three-level noise reduction processing. However, in other embodiments, the noise 101 may be subjected to multiple levels (e.g., two, four, five) of noise reduction processing as needed until the required noise reduction requirement is met.

Fig. 2 shows a flow chart of a process for obtaining the first order noise reduction 104.

As shown in fig. 2, a sound source signal of the noise 101 is collected with a cardioid microphone 201 at a position close to the sound emission point of the noise 101 and on one side of the propagation direction of the noise 101 (step S201).

The acquired sound source signal is converted into a digital signal by a/D conversion (step S202). At this point, some intrinsic and transient characteristics of the noise 101 can be read, such as: frequency, amplitude, phase, etc. Based on these features, suitable sound processing software and hardware may be selected, such as: a bass sound source must not respond to a treble sound transducer, and sometimes even requires processing of the source signal with a combination of multiple transducers.

Since a cardioid directional microphone is used to collect the sound source signal, the resulting signal will have more or less some errors. Typically, before the cardioid microphone is used, it is measured by an audio analysis system to obtain compensation and correction values for the intrinsic properties of the cardioid microphone. The acquired sound source signal is corrected by DSP processing using this compensation value and correction value to correct an error of the sound source signal by the cardioid microphone 201 (step S203). The corrected information includes: the sound pressure correction data of each frequency point in the frequency band and the phase data among the frequency points can make the signal captured by the cardioid microphone consistent with the original noise signal through correction.

Since the acoustic transducer (the acoustic transducer used in this embodiment is an electrodynamic speaker) mainly implements the function of converting an electric signal into an acoustic signal, it is necessary to perform frequency conversion of one frequency band. The sound production positions and the starting time of the sound transducers are different at each frequency point, so that the amplitude and the relative phase of the frequency are different. Therefore, in addition to correcting the error of the cardioid microphone 201 to the sound source signal, the error of the sound transducer to the sound source signal also needs to be corrected (step S204) to achieve a faithful recovery of the original signal.

In addition to the cardioid pointing microphone and acoustic transducer, the processor, ADC, in the overall processing system,

The DAC, memory, register, power amplifier, transmission link, etc. may cause delay, frequency variation, or amplitude attenuation of the signal, which needs to be corrected (step S205).

The processing of correcting the cardioid directional microphone, the acoustic transducer, various hardware and transmission links of the system can be realized by digital processing hardware and software, and a processor or a professional DSP chip and other hardware can be selected to be matched with a software algorithm for unified processing.

The present embodiment performs transducer and system testing and calibration by way of an audio analysis system and audio DSP processing. The audio test is divided into a steady state test and a transient test, and the embodiment selects SMAARTLIVE7 software for testing. The steady state test method comprises the following steps: the system itself emits a continuous test signal which is a wide frequency noise signal and which serves as a test reference which is represented by a channel datum. This reference is a loop and is reflected on the tester, and since the input and output are loops, the system will appear as a straight line. The same reference signal is sent to a system needing to be measured, after the system is measured to respond, the FFT (fast Fourier transform) is converted through a heart-shaped directional microphone and other acquisition devices (which can be electric signals and acoustic signals), the signals are displayed in other channels, the obtained result is compared with the original signals, namely, two or more channels are compared, and the problem of the comparison can be visually seen. The steady state test is a continuous signal measurement, and the transient test is a pulse signal test. The same principle is also adopted for relative phase test, and different phase responses, namely time response differences, can be obtained by taking fixed frequency as a phase starting point and other frequencies as comparison.

And the signal processing process is carried out by combining a general-purpose processor or a professional DSP processing chip with corresponding software. The present example adopts a hardware system composed of SHARC ADSP-21448 processing chips of ADI company and combines corresponding software to process signals. The signal processing process comprises the following steps: the method includes (1) performing input compensation processing (including amplitude-frequency response and phase response characteristics thereof) on the A/D converted sound source signal according to the previously tested cardioid directional microphone calibration data to correct errors introduced by the measuring microphone (step S203), (2) correcting errors introduced by the sound transducer according to the above audio test, including compensation and correction of amplitude-frequency response, phase response and transient response (step S204), and (3) compensating and correcting errors introduced in the whole system according to the above audio test, including amplitude-frequency response, phase response and transient response (step S205).

In addition, in this embodiment, the amplitude and phase are corrected by using a comprehensive algorithm such as an FIR filter and an ALLPASS filter, and the response to the transient is corrected by applying an inverse signal.

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

Fig. 5 is a graph of the impulse response of the acoustic transducer before correction and fig. 6 is a graph of the impulse response of the acoustic transducer after correction. As can be seen from fig. 5 and 6, before optimization, a few large amplitude gegen additional pulses are added below the main pulse and then a large ringing pulse at time; the main clutter is corrected in a mode of a reverse filter, the main clutter becomes smaller, the additional clutter becomes less, and the transient angle is closer to the original waveform.

The above step S203-S205 adds the corrected correction results to the signal input by the original cardioid-oriented microphone, thus forming a composite corrected signal.

The modified signal is loaded with a delay signal using the hardware and software described above (step S206) and then processed in reverse (step S207). Then, the signal is D/a-processed, converted into an analog signal (step S208), output to a power amplifier (step S209), and finally propagated through the speaker 206, thus obtaining the first reproduced sound source 103 (step S210).

The first reproduced sound source 103 propagates through a path set in the air, the first cancel point 102 is close to the speaker 206, and the first reproduced sound source 103 cancels the noise 101 at the first cancel point 102, thereby obtaining the first-order noise reduction sound 104. In this embodiment, the distance between the first canceling point 102 and the sound emitting point of the first reproduced sound source 103 is less than 1/2 of the noise wavelength, and in other embodiments, the distance between the first canceling point 102 and the sound emitting point of the first reproduced sound source 103 is less than 1/4 of the noise wavelength.

A first measurement microphone 301 may be disposed at the first cancellation point 102, and a second measurement microphone 303 may be disposed at the first noise reduction 104 (noise reduction field), and a sound source signal collected by the first measurement microphone 301 and a noise signal may be compared by the test system 302, and a sound source signal collected by the second measurement microphone 303 and the noise signal may be compared to judge the noise reduction effect. At the same time, further corrections can be made to the processing system based on this effect.

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

The first regulation loop is: firstly, an audio analysis system is used for comprehensively testing a reconstructed sound source system consisting of a cardioid directional microphone 201, signal processing hardware, a power amplifier, an acoustic transducer, a link, a plug-in and the like. That is, a signal is sent to the cardioid microphone 201, so that the result of the measurement of the reconstructed sound source system by the first measuring microphone 301 can be obtained, and the result is compared with the original test signal to determine the degree of coincidence between the two signals, thereby adjusting the correction parameters of the whole system. The debugging process of the parameters of the whole system comprises the following steps:

1. the cardioid directional microphone 201 is adjusted individually, mainly by correction with the same standard sound source method. Using a standard sound source (loudspeaker with good response) as the measurement sound source, and using the obtained and calibrated measurement microphone as the comparison standard, the difference between the two microphones can be obtained by the audio analysis system, and the difference generally includes two items of data, i.e. frequency response and phase response, which are obtained as the calibration parameters of the cardioid pointing microphone 201. This parameter is directly input into the compensation data of the DSP software program to complete the adjustment of the parameters of the cardioid microphone 201.

2. The acoustic transducer is adjusted independently, and the measurement of the acoustic transducer is mainly carried out by an audio analysis system, wherein the measurement comprises two main categories of steady state and transient state. The amplitude frequency, the phase and the transient are corrected by the DSP and software thereof, and the corrected data are stored in a software program of the DSP.

3. And adjusting the conformity degree of the reconstructed sound source by an isolation method. A sound source (which may be emitted by a speaker) of a defined frequency band is used to separate the sound source to an isolated location which is isolated from the acoustic transducer. The signal is captured by a measuring microphone to form a reference, and simultaneously the acoustic signal is captured by a cardioid directional microphone at the same position as the measuring microphone, and the signal is sent to a reproduced sound source system, and the acoustic signal of the reproduced sound source is captured by another microphone, and the measurement result is compared with the reference, so that the difference of the whole circuit of the reproduced sound source is formed, and the correction data is written into the program of the DSP again.

The second debug loop is: the time delay matching of the reproduced sound source and the original noise is realized by the first measuring microphone

301 are obtained via a test system 302. In this way, the loaded delay signal can be corrected. The short-time pulse is mainly used as a debugging sound source to adjust the time delay. That is, a short pulse is emitted from another speaker as a tuning sound source, and the time to the cancellation point is measured. And (5) fixing the position of the debugging sound source, and measuring the time of the sound source reaching the offset point after the sound is emitted by the reconstructed sound source again. The cancellation point is a position as close as possible to the loudspeaker. The position of the cancellation point is also the position of the measuring microphone. The above two data are measured by the measuring microphone. And adjusting the position of the cardioid directional microphone away from the offset point to ensure that the time for the reconstructed sound source to reach the offset point is slightly less than the time for the debugging sound source to reach the offset point. The time for the two sound waves to reach the 4's vanishing point can be equalized by adding precise delay time to the reconstructed sound source.

The third debug loop is: the difference between the reproduced sound source and the original noise can be obtained by the second measuring microphone 303 of the testing system 302, and at this time, the parameters of the whole system can be corrected and adjusted by comparing the consistency of the two sound sources and repeating the counteracting effect of the reproduced sound source in the opposite direction, and the counteracting and the scheme optimization can be performed for a plurality of times.

Because the time for the noise 101 to reach the first cancellation point 102 is fixed and can be measured by the first measurement microphone 301. Therefore, the matching with the time delay of the reconstructed sound source system can be realized only by adjusting the distance between the heart-shaped pointing microphone 201 and the offset point, namely the time delay of the reconstructed sound source system is large, the distance between the heart-shaped pointing microphone 201 and the offset point is long, and the aim is to ensure that the original noise and the reconstructed sound source are respectively arranged at two sides of the offset point and have opposite directions of propagation in the air. The time for the reconstructed sound source to reach the cancellation point is slightly shorter than the original noise. By adjusting the time delay of the reproduced sound source after the measurement, it is possible to achieve that the time when the first reproduced sound source 103 reaches the first low dissipation point 102 arrives at the same time as the noise 101.

If the noise 101 is measured by the cardioid microphone 201 as a reference point, the time to reach the first cancellation point 102 is T1, the time T2 of the transducer space propagating sound wave to reach the first cancellation point 102, and the total propagation and digital processing and correction process delay and all analog and digital device fixed delays of the noise 101 captured by the cardioid microphone 201 are collectively called the total system integrated delay time T3. Since the sound propagation speed is slower than the propagation speed of the electrical signal, it is possible to adjust the distance from the cardioid pointing microphone 201 to the cancellation point, and it is ensured that t 1> t 2+ t3, and a time delay t 4 is loaded to the acquired sound source in step 204 for fine adjustment, so that t 1= t 2+ t 3+ t 4, thereby enabling the first reproduced sound source 103 and the noise 101 to reach the first cancellation point 102 at the same time.

Fig. 7 is a graph showing the change in noise before and after the processing by the flow shown in fig. 2. As shown in fig. 7, line a is an ambient noise curve and line B is a noise source in the 30 Hz-2 KHz frequency range which is continuous, a reconstructed source is obtained by the processing of step S202-S210 in fig. 2, the noise reduction processing in fig. 2 is performed, and line C is obtained after cancellation. As can be seen from fig. 7, the noise is already significantly reduced after the noise reduction method of the present invention. What has been described above is merely one embodiment of the present invention. It will be apparent to those skilled in the art that various changes and modifications can be made without departing from the inventive concept thereof, and these changes and modifications can be made without departing from the spirit and scope of the invention.

Claims

Claims

1. A method of reducing noise comprising the steps of:

(1) analyzing the noise and determining the sound source characteristics of the noise;

(2) collecting a noise source signal, and processing the collected sound source signal according to the sound source characteristics of noise as follows: correcting, delaying, reversing and converting to obtain a first reproduced sound source;

(3) and canceling the sound transmitted by the first reproduced sound source and the noise at a first canceling point to obtain a first-level noise reduction.

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

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

carrying out A/D conversion on the collected sound source signals;

correcting errors brought to sound source signals in the acquisition process;

the collected sound source signals are subjected to delay processing, and the sound source signals can reach a first offset point simultaneously with noise when being converted into sound energy for transmission by loading the delay signals;

the conversion processing is carried out by using an acoustic transducer, and before the conversion processing, the acoustic transducer is subjected to correction processing;

carrying out reverse processing on the sound source signal to enable the vibration direction of the sound source signal to be opposite to the vibration direction of the noise;

correcting errors brought to sound source signals in the time delay processing, the reverse processing and the conversion processing; the sound source signal is D/a converted into sound energy, thereby being propagated.

4. A method for reducing noise according to any one of claims 1-3, wherein in step (3) the first reproduced sound source is propagated by an acoustic transducer.

5. The method of reducing noise of claim 4, wherein said collecting the noise source signal is performed near a point of sound emission of the noise and in a direction of propagation of the noise, and wherein said first cancellation point is located near a point of sound emission of the first reproduced sound source.

6. The method of reducing noise of claim 4, wherein the first cancellation point is located less than 1/2 of the high frequency wavelength of the noise from the point of the first reproduced sound source.

7. The method of reducing noise of claim 4, wherein the first cancellation point is located less than 1/4 of the high frequency wavelength of the noise from the point of the first reproduced sound source.

8. The method of reducing noise according to claim 3, further comprising the following for primary noise reduction:

collecting a primary noise reduction sound source signal, measuring the counteracting effect of the first reproduced sound source and noise, and correcting the sound source signal in the step (2) according to the measuring result, and adjusting the loading delaying length of the delay signal or adjusting the position of a first counteracting point.

9. The method of reducing noise according to claim 8, further comprising the following for primary noise reduction:

analyzing the primary noise reduction, and determining the sound source characteristics of the primary noise reduction;

collecting a sound source signal of the primary noise reduction, and processing the collected sound source signal of the primary noise reduction according to the sound source characteristics of the primary noise reduction as follows: correcting, delaying, reversing and converting to obtain a second reconstructed sound source;

and offsetting the sound transmitted by the second reconstructed sound source and the primary noise reduction at a second offset point to obtain secondary noise reduction.

10. The method of reducing noise according to claim 9, further comprising the step of:

analyzing the secondary noise reduction, and determining the sound source characteristics of the secondary noise reduction;

collecting a sound source signal of the secondary noise reduction, and processing the collected sound source signal of the secondary noise reduction according to the sound source characteristics of the secondary noise reduction as follows: correcting, delaying, reversing and converting to obtain a third reconstructed sound source;

and canceling the sound transmitted by the third reproduced sound source and the secondary noise reduction at a third canceling point.