WO2015066926A1 - Procédé de réduction de bruit - Google Patents

Procédé de réduction de bruit Download PDF

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Publication number
WO2015066926A1
WO2015066926A1 PCT/CN2013/086875 CN2013086875W WO2015066926A1 WO 2015066926 A1 WO2015066926 A1 WO 2015066926A1 CN 2013086875 W CN2013086875 W CN 2013086875W WO 2015066926 A1 WO2015066926 A1 WO 2015066926A1
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WIPO (PCT)
Prior art keywords
sound source
noise
processing
noise reduction
signal
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PCT/CN2013/086875
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English (en)
Chinese (zh)
Inventor
赵春宁
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赵春宁
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Application filed by 赵春宁 filed Critical 赵春宁
Priority to CN201380075550.3A priority Critical patent/CN105103219B/zh
Priority to PCT/CN2013/086875 priority patent/WO2015066926A1/fr
Publication of WO2015066926A1 publication Critical patent/WO2015066926A1/fr

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Classifications

    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10K—SOUND-PRODUCING DEVICES; METHODS OR DEVICES FOR PROTECTING AGAINST, OR FOR DAMPING, NOISE OR OTHER ACOUSTIC WAVES IN GENERAL; ACOUSTICS NOT OTHERWISE PROVIDED FOR
    • G10K11/00—Methods or devices for transmitting, conducting or directing sound in general; Methods or devices for protecting against, or for damping, noise or other acoustic waves in general
    • G10K11/16—Methods or devices for protecting against, or for damping, noise or other acoustic waves in general
    • G10K11/175—Methods or devices for protecting against, or for damping, noise or other acoustic waves in general using interference effects; Masking sound
    • G10K11/178—Methods or devices for protecting against, or for damping, noise or other acoustic waves in general using interference effects; Masking sound by electro-acoustically regenerating the original acoustic waves in anti-phase
    • G10K11/1787—General system configurations
    • G10K11/17873—General system configurations using a reference signal without an error signal, e.g. pure feedforward
    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10K—SOUND-PRODUCING DEVICES; METHODS OR DEVICES FOR PROTECTING AGAINST, OR FOR DAMPING, NOISE OR OTHER ACOUSTIC WAVES IN GENERAL; ACOUSTICS NOT OTHERWISE PROVIDED FOR
    • G10K11/00—Methods or devices for transmitting, conducting or directing sound in general; Methods or devices for protecting against, or for damping, noise or other acoustic waves in general
    • G10K11/16—Methods or devices for protecting against, or for damping, noise or other acoustic waves in general
    • G10K11/175—Methods or devices for protecting against, or for damping, noise or other acoustic waves in general using interference effects; Masking sound
    • G10K11/178—Methods or devices for protecting against, or for damping, noise or other acoustic waves in general using interference effects; Masking sound by electro-acoustically regenerating the original acoustic waves in anti-phase
    • G10K11/1785—Methods, e.g. algorithms; Devices
    • G10K11/17857—Geometric disposition, e.g. placement of microphones
    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10K—SOUND-PRODUCING DEVICES; METHODS OR DEVICES FOR PROTECTING AGAINST, OR FOR DAMPING, NOISE OR OTHER ACOUSTIC WAVES IN GENERAL; ACOUSTICS NOT OTHERWISE PROVIDED FOR
    • G10K2210/00—Details of active noise control [ANC] covered by G10K11/178 but not provided for in any of its subgroups
    • G10K2210/10—Applications
    • G10K2210/111—Directivity control or beam pattern
    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10K—SOUND-PRODUCING DEVICES; METHODS OR DEVICES FOR PROTECTING AGAINST, OR FOR DAMPING, NOISE OR OTHER ACOUSTIC WAVES IN GENERAL; ACOUSTICS NOT OTHERWISE PROVIDED FOR
    • G10K2210/00—Details of active noise control [ANC] covered by G10K11/178 but not provided for in any of its subgroups
    • G10K2210/10—Applications
    • G10K2210/12—Rooms, e.g. ANC inside a room, office, concert hall or automobile cabin
    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10K—SOUND-PRODUCING DEVICES; METHODS OR DEVICES FOR PROTECTING AGAINST, OR FOR DAMPING, NOISE OR OTHER ACOUSTIC WAVES IN GENERAL; ACOUSTICS NOT OTHERWISE PROVIDED FOR
    • G10K2210/00—Details of active noise control [ANC] covered by G10K11/178 but not provided for in any of its subgroups
    • G10K2210/30—Means
    • G10K2210/301—Computational
    • G10K2210/3055—Transfer function of the acoustic system
    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10K—SOUND-PRODUCING DEVICES; METHODS OR DEVICES FOR PROTECTING AGAINST, OR FOR DAMPING, NOISE OR OTHER ACOUSTIC WAVES IN GENERAL; ACOUSTICS NOT OTHERWISE PROVIDED FOR
    • G10K2210/00—Details of active noise control [ANC] covered by G10K11/178 but not provided for in any of its subgroups
    • G10K2210/30—Means
    • G10K2210/321—Physical
    • G10K2210/3215—Arrays, e.g. for beamforming

Definitions

  • the present invention relates to a method of reducing noise. Background technique
  • 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.
  • the noise canceling earphone forms an enclosed space by surrounding the ear, and uses soundproof materials such as silicone earplugs to block external noise;
  • 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.
  • 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. .
  • a method of reducing noise comprising the steps of:
  • 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;
  • 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;
  • 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.
  • 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.
  • 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.)
  • an acoustic transducer eg, a speaker, a speaker, a flat sounder, a Haier sounder, a piezoelectric sounder, etc.
  • 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.
  • 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.
  • 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, 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.
  • the above step (2) may include:
  • 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;
  • 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.
  • disortion 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.
  • the following processing can also be performed for the first level noise reduction:
  • step (2) 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.
  • 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.
  • 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.
  • the following processing can also be performed on 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 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.
  • FIG. 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
  • Fig. 1 schematically shows a schematic diagram of the principle of a method for reducing noise according to an embodiment of the present invention.
  • 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.
  • 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.
  • 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.
  • 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.
  • 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.
  • 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.
  • noise 101 is subjected to three levels of noise reduction processing.
  • 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.
  • 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).
  • 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.
  • the heart-shaped pointing microphone 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.
  • 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.
  • the processor In addition to the heart-shaped pointing microphone and acoustic transducer, the processor, ADC, the processor, ADC, and
  • 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.
  • 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.
  • the response of the measurement system 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.
  • FFT Fast Fourier
  • 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.
  • a general-purpose processor or a professional DSP processing chip in combination with corresponding software.
  • 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).
  • 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.
  • FIR filter and ALLPASS filter 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
  • 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
  • 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.
  • 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.
  • 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.
  • 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.
  • 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:
  • a standard sound source resonant speaker
  • 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.
  • 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
  • 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.
  • 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.
  • 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.
  • the time to reach the first cancellation point 102 is T1
  • Fig. 7 is a graph showing changes in noise before and after the processing in Fig. 2.
  • the line is an environmental noise curve
  • line B is a noise source continuously from a frequency range of 30 Hz to 2 KHz.

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  • Physics & Mathematics (AREA)
  • Engineering & Computer Science (AREA)
  • Acoustics & Sound (AREA)
  • Multimedia (AREA)
  • Soundproofing, Sound Blocking, And Sound Damping (AREA)

Abstract

L'invention concerne un procédé de réduction de bruit. Le procédé consiste à : analyser un bruit, et déterminer une caractéristique de source sonore du bruit ; acquérir un signal de source sonore, la réalisation du traitement suivant sur le signal de source sonore acquis : un traitement de correction, un traitement de retard temporel, un traitement de recul et un traitement de conversion, de façon à obtenir une première source sonore reconstruite ; et réaliser une annulation entre le son de la première source sonore reconstruite et le bruit sur un premier point d'annulation. Dans la présente invention, par acquisition d'une source sonore de bruit et réalisation d'une série de traitements, une source sonore reconstruite qui a une direction de vibration opposée à celle d'une source sonore de bruit et a d'autres caractéristiques exactement identiques à celles de la source sonore de bruit, lorsque la source sonore reconstruite atteint un point d'annulation, est créée et, par création artificielle d'un trajet de rencontre, le bruit et la source sonore reconstruite se rencontrent en même temps et dans une même position, une diversité d'erreurs sont provoquées lors de la création de la source sonore reconstruite, et une « différence temporelle » entre le bruit et la source sonore reconstruite s'approche de zéro, permettant ainsi d'éliminer le bruit. Le traitement de correction de la présente invention peut annuler efficacement un bruit changé de façon transitoire, et peut être appliqué dans un environnement de grand espace, tel qu'une production industrielle.
PCT/CN2013/086875 2013-11-11 2013-11-11 Procédé de réduction de bruit WO2015066926A1 (fr)

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CN201380075550.3A CN105103219B (zh) 2013-11-11 2013-11-11 降低噪音的方法
PCT/CN2013/086875 WO2015066926A1 (fr) 2013-11-11 2013-11-11 Procédé de réduction de bruit

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CN104967717A (zh) * 2015-05-26 2015-10-07 努比亚技术有限公司 终端语音交互模式下的降噪方法及装置
CN105679300A (zh) * 2015-12-29 2016-06-15 努比亚技术有限公司 移动终端和降噪方法
CN107749295A (zh) * 2017-10-17 2018-03-02 张博强 风力发电机组噪声有源噪声控制方法与有源噪声控制系统

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Publication number Priority date Publication date Assignee Title
CN106161780A (zh) * 2016-06-22 2016-11-23 深圳市金悦通机器人技术有限公司 具有扬声器降噪功能的通信装置及其降噪方法
CN107230481A (zh) * 2017-05-31 2017-10-03 宇龙计算机通信科技(深圳)有限公司 噪音处理方法及终端
US10096313B1 (en) * 2017-09-20 2018-10-09 Bose Corporation Parallel active noise reduction (ANR) and hear-through signal flow paths in acoustic devices
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