WO2016041247A1 - 下行主动降噪装置、方法及移动终端 - Google Patents

下行主动降噪装置、方法及移动终端 Download PDF

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WO2016041247A1
WO2016041247A1 PCT/CN2014/091049 CN2014091049W WO2016041247A1 WO 2016041247 A1 WO2016041247 A1 WO 2016041247A1 CN 2014091049 W CN2014091049 W CN 2014091049W WO 2016041247 A1 WO2016041247 A1 WO 2016041247A1
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signal
noise reduction
secondary signal
mobile terminal
adaptive controller
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PCT/CN2014/091049
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English (en)
French (fr)
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吴建明
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中兴通讯股份有限公司
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    • 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
    • G10K11/1781—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 characterised by the analysis of input or output signals, e.g. frequency range, modes, transfer functions
    • G10K11/17821—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 characterised by the analysis of input or output signals, e.g. frequency range, modes, transfer functions characterised by the analysis of the input signals only
    • G10K11/17823—Reference signals, e.g. ambient acoustic environment
    • 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
    • G10K11/1783—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 handling or detecting of non-standard events or conditions, e.g. changing operating modes under specific operating conditions
    • 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
    • G10K11/17853—Methods, e.g. algorithms; Devices of the filter
    • G10K11/17854—Methods, e.g. algorithms; Devices of the filter the filter being an adaptive filter
    • 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
    • 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
    • G10K11/17879—General system configurations using both a reference signal and an error signal
    • G10K11/17881—General system configurations using both a reference signal and an error signal the reference signal being an acoustic signal, e.g. recorded with a microphone

Definitions

  • the present invention relates to the field of communications technologies, and in particular, to a downlink active noise reduction device, method, and mobile terminal.
  • Active noise reduction technology was first used as a new technology for military communications. Before the active noise reduction occurs, it can only adopt the passive noise reduction method, that is, the whole earmuffs are used to wrap the left and right ears, the sound insulation characteristics of the earmuffs are added, and the sound absorbing sponge is added in the earmuffs to reduce the arrival person. Ear noise. However, this method can only reduce the high frequency, and the noise of the low frequency is not much reduced. After the emergence of active noise reduction technology, various series of active noise reduction headwear appeared.
  • the earliest noise reduction principles include two types: one is feedforward noise reduction and the other is feedback noise reduction.
  • the background noise of the human ear is collected by the microphone inside the earmuff, and a reverse noise with a phase difference of 180 degrees is generated through the circuit, and the speaker is released and acoustic interference is superimposed on the background noise, thereby effectively reducing the background.
  • the size of the noise signal is that the human ear is protected while improving the quality of the call, and the hearing damage caused by the human ear being exposed to a high noise environment for a long time is avoided.
  • active noise reduction has been greatly developed, gradually shifting from military to civilian.
  • the noise reduction technology existing on mobile phones mainly focuses on how to improve the voice quality of the mobile phone sending direction, that is, the dual microphone noise reduction technology in the usual sense.
  • the technology focuses on how to improve the signal-to-noise ratio of the uplink voice in places with poor noise conditions (such as public places such as offices, restaurants, and streets), so that the other end of the call can be heard more clearly.
  • the embodiments of the present invention provide a downlink active noise reduction device, a method, and a mobile terminal, so as to at least solve the problem that the smart mobile terminal cannot filter the local call noise and affect the call quality in the prior art.
  • a downlink active noise reduction device comprising:
  • a first acoustic wave collector configured to collect an ambient noise signal, obtain a reference signal, and output the reference signal to an adaptive controller
  • An adaptive controller configured to calculate a secondary signal for performing acoustic wave cancellation based on the received reference signal, and adjust a magnitude and a phase of the secondary signal in a forward or reverse direction based on the received error signal;
  • a secondary sound field transmitter configured to emit the adjusted secondary signal in the form of sound waves
  • the second sound wave collector is configured to collect an ambient noise signal and a secondary signal from the secondary sound field emitter, superimpose the two, obtain an error signal, and output the signal to the adaptive controller.
  • the first sound wave collector is further configured to detect whether the current ambient noise reaches a set threshold, and if yes, output the obtained reference signal to the adaptive controller; otherwise , it is determined that no noise reduction processing is currently required, and the process ends.
  • the adaptive controller is configured to calculate a secondary signal and adjust a magnitude and a phase of the secondary signal by using a preset XLMS adaptive algorithm.
  • the adaptive controller specifically includes:
  • the A/D conversion unit is configured to perform analog-to-digital conversion on the reference signal and the error signal received by the adaptive controller, and output the same to the DSP;
  • a DSP configured to calculate a secondary signal for performing acoustic wave cancellation based on the received reference signal, and adjust a magnitude and a phase of the secondary signal in a forward or reverse direction based on the error signal;
  • the D/A conversion unit is configured to perform digital-to-analog conversion on the adjusted secondary signal and output the same to the power amplifier unit;
  • the power amplifier unit is configured to perform power amplification processing on the secondary signal and output to the secondary sound field transmitter.
  • the second acoustic wave collector is arranged in an infinite proximity to the secondary sound field emitter; the device types of the first acoustic wave collector and the second acoustic wave collector are microphone.
  • a mobile terminal in which the downlink active noise reduction device of the present invention is integrated.
  • a downlink active noise reduction method including:
  • Step 1 collecting an environmental noise signal to obtain a reference signal
  • Step 2 calculating a secondary signal used to implement acoustic wave cancellation based on the reference signal
  • Step 3 using the currently obtained error signal, adjusting the amplitude and phase of the secondary signal in the forward or reverse direction;
  • Step 4 the adjusted secondary signal is sent in the form of sound waves
  • Step 5 collecting an environmental noise signal and a secondary signal sent in the form of sound waves, superimposing the two to obtain an error signal, and providing the error signal to step 3.
  • the method before the step 1, the method further includes: detecting whether the current ambient noise reaches a set threshold, and if yes, continuing to perform step 1; otherwise, determining that the current noise reduction process is not required, and ending.
  • the downlink active noise reduction device of the invention realizes the extraction of the noise signal and the error signal by arranging two sound wave collectors, and real-time correction of the secondary signal by the error signal is realized by the adaptive controller, thereby achieving localized "Quiet zone";
  • the mobile terminal integrated with the active noise reduction device applies active noise reduction to the mobile terminal, improves the local call reception effect, and has a good market prospect.
  • FIG. 1 is a structural diagram of a downlink active noise reduction device provided by the present invention.
  • FIG. 2 is a structural diagram of an adaptive controller in the present invention
  • FIG. 3 is a schematic diagram showing the arrangement of an acoustic wave collector disposed on a mobile terminal according to the present invention
  • FIG. 4 is a flowchart of a downlink active noise reduction method provided by the present invention.
  • the present invention provides a downlink active noise reduction device, method, and mobile terminal, where the device sets two sound wave collectors, A pick-up external noise signal is used as a reference signal, and an error signal is obtained by picking up the noise signal and the secondary signal to control the size of the secondary sound source through the error signal, thereby achieving a local "quiet zone" and making the noise signal Minimize this, which increases the intelligibility of the downstream speech.
  • the embodiment of the invention provides a downlink active noise reduction device. As shown in FIG. 1 , the device specifically includes:
  • the first acoustic wave collector 110 is configured to collect an environmental noise signal, obtain a reference signal, and output the reference signal to the adaptive controller 120;
  • the adaptive controller 120 is configured to calculate a secondary signal for performing acoustic wave cancellation based on the received reference signal, and adjust the amplitude and phase of the secondary signal in a forward or reverse direction based on the received error signal;
  • the calculated secondary signal for implementing the acoustic wave cancellation ideally refers to the signal after the reference signal phase is inverted by 180 degrees, but since the circuit processing and the acoustic wave transmission have delays, it is necessary to reverse the delay information according to the delay information.
  • the phase information of the signal is corrected, and the corrected signal is the calculated secondary signal.
  • the adaptive controller 120 adjusts the amplitude and phase of the secondary signal forward or backward, which is determined by the error signal.
  • the final purpose is to adjust the error of the secondary signal and the noise signal after adjusting the secondary signal. Approaching zero, ideally equal to zero.
  • a secondary sound field emitter 130 configured to emit the secondary signal adjusted by the adaptive controller 120 in the form of sound waves
  • the second acoustic wave collector 140 is configured to collect the ambient noise signal and the secondary signal emitted by the secondary sound field emitter 130, and superimpose the two to obtain an error signal, and then output the signal to the adaptive controller 120.
  • the first acoustic wave collector 110 detects whether the current environmental noise reaches a set threshold when collecting environmental noise, and if so, The obtained reference signal is output to the adaptive controller 120; otherwise, it is determined that no noise reduction processing is currently required, and the process ends. That is to say, the active noise reduction process is only turned on when the background noise is large, and not all cases are subjected to noise reduction processing.
  • the second acoustic wave collector 140 is disposed in an infinite proximity to the secondary sound field emitter 130 to better receive the secondary signal from the secondary sound field emitter 130;
  • the device types of the first sound wave collector 110 and the second sound wave collector 140 may be, but are not limited to, a microphone.
  • the adaptive controller 120 specifically uses a preset XLMS adaptive algorithm to calculate the secondary signal and adjust the amplitude and phase of the secondary signal.
  • XLMS adaptive algorithm is only an exemplary explanation, and is not limited to the implementation of the algorithm.
  • Algorithms that can be implemented by those skilled in the art that can implement the related functions of the adaptive controller 120 are all in the present invention. Within the scope of protection thinking.
  • the adaptive controller 120 specifically includes:
  • the A/D conversion unit 121 is configured to perform analog-to-digital conversion of the reference signal and the error signal received by the adaptive controller 120, and output to the DSP 122;
  • the DSP 122 is configured to calculate a secondary signal for performing acoustic wave cancellation based on the received reference signal, and adjust the amplitude and phase of the secondary signal in a forward or reverse direction based on the error signal;
  • the D/A conversion unit 123 is configured to perform digital-to-analog conversion on the adjusted secondary signal and output to the power amplifier unit 124;
  • the power amplifier unit 124 is configured to perform power amplification processing on the secondary signal and output to the secondary sound field emitter 130.
  • the downlink active noise reduction device realizes the extraction of the noise signal and the error signal by arranging two sound wave collectors, and realizes the use of the error signal to the secondary signal through the adaptive controller. Real-time corrections to achieve a local "quiet zone".
  • the embodiment of the invention provides a mobile terminal, which is integrated with the downlink active noise reduction device of the first embodiment. Since the structure of the downlink active noise reduction device has been described in detail in the first embodiment, the embodiment does not describe the integrated structure.
  • the arrangement positions of the integrated first sound wave collector, the second sound wave collector, and the secondary sound field emitter are also given in this embodiment.
  • the first acoustic wave collector is disposed on the back of the mobile terminal;
  • the secondary sound field transmitter is the earpiece of the mobile terminal; and
  • the second acoustic wave collector is disposed at the position of the earpiece near the mobile terminal.
  • the first sound wave collector is arranged in a position that the mobile terminal is not easily blocked by the user.
  • the specific control process for the mobile terminal to implement downlink active noise reduction is as follows:
  • the noise source emits an acoustic signal P(t);
  • the microphone MIC1 located near the noise source on the back of the earpiece picks up the reference signal x(t) (ie, the noise signal collected by the MIC1) as an input to the adaptive controller;
  • the adaptive controller calculates the secondary signal y(t) under the action of the preset XLMS adaptive algorithm, and outputs it to the earpiece, and the sound wave is emitted by the earpiece in the form of sound waves;
  • the noise source and the earpiece respectively form a primary sound field and a secondary sound field
  • the error microphone MIC2 second sound wave collector located near the earpiece simultaneously receives the sound pressures of the primary sound field and the secondary sound field, and the two superimpose to form an error signal e ( t);
  • the error signal e(t) is input and fed back to the adaptive controller.
  • the adaptive controller uses the preset XLMS adaptive algorithm to adjust the adaptive controller weight coefficient according to the preset control objective function, thereby quickly
  • the amplitude and phase of the currently calculated secondary signal are adjusted forward or reverse, and the adjusted secondary signal is output to the earpiece, and the sound wave is emitted by the earpiece to perform sound wave cancellation. The above process is repeated until the system is stable, forming a local "quiet zone".
  • the adaptive algorithm program is loaded into the DSP of the adaptive controller.
  • the MIC1 located on the back of the earpiece collects ambient noise as a reference signal x(t), which is sampled into the DSP by the A/D conversion unit.
  • the DSP loaded with the adaptive algorithm outputs a secondary signal y(t) to the earpiece, which is emitted by the earpiece in the form of sound waves to perform acoustic wave cancellation.
  • the DSP uses the built-in adaptive algorithm to adjust the controller weight coefficient according to the preset control objective function, thereby rapidly changing the amplitude and phase of the secondary signal, and outputting y(t) to the handset through the D/A conversion unit and the power amplifier unit. Until the system is stable, a local stable active noise reduction is formed.
  • the call voice signal received by the mobile terminal is directly output to the handset through the D/A conversion unit and the power amplifier unit.
  • the mobile terminal according to the embodiment of the present invention applies active noise reduction to the mobile phone through the downlink active noise reduction device, thereby improving the local call reception effect and having a good market prospect.
  • An embodiment of the present invention provides a downlink active noise reduction method, as shown in FIG. 4, including:
  • Step S401 collecting an environmental noise signal to obtain a reference signal
  • the method further includes: detecting whether the current ambient noise reaches a set threshold, and if yes, continuing to perform step S401; otherwise, determining that the current noise reduction process is not required, and ending.
  • Step S402 calculating a secondary signal used to implement acoustic wave cancellation based on the reference signal
  • Step S403 using the currently obtained error signal, adjusting the amplitude and phase of the secondary signal in the forward direction or the reverse direction;
  • step S404 the adjusted secondary signal is sent in the form of sound waves.
  • Step S405 the ambient noise signal and the secondary signal emitted in the form of sound waves are collected, and the two are superimposed to obtain an error signal, and the error signal is supplied to step S403.
  • the method described in this embodiment realizes the extraction of the noise signal and the error signal, and uses the error signal to correct the secondary signal in real time, thereby achieving a local "quiet zone".
  • the downlink active noise reduction apparatus, method, and mobile terminal provided by the embodiments of the present invention have the following beneficial effects: by arranging two acoustic wave collectors, the noise signal and the error signal are extracted, and adaptive control is implemented.
  • the device realizes real-time correction of the secondary signal by using the error signal, thereby achieving a local “quiet zone”; and the mobile terminal integrated with the active noise reduction device applies active noise reduction to the mobile terminal, thereby improving the local call reception effect. With good market prospects.

Abstract

一种下行主动降噪装置、方法及移动终端，所述装置包括：第一声波采集器（110），设置为采集环境噪声信号，得到参考信号，并将所述参考信号输出至自适应控制器（120）；自适应控制器（120），设置为基于接收的参考信号，计算用以实施声波抵消的次级信号，并基于接收到的误差信号，正向或反向调整次级信号的幅值及相位；次级声场发射器（130），设置为将自适应控制器（120）调整后的次级信号以声波形式发出；第二声波采集器（140），设置为采集环境噪声信号和次级信号，将二者进行叠加，得到误差信号后输出至自适应控制器（120）。通过下行主动降噪装置，将主动降噪应用到手机上，提高了本地通话接收效果。

Description

下行主动降噪装置、方法及移动终端 技术领域
本发明涉及通信技术领域，尤其涉及一种下行主动降噪装置、方法及移动终端。
背景技术
在通信领域，经常会遇到高噪声的环境，噪声对语音通信的影响是致命的。因为在高噪声环境下，会出现语音被噪声掩蔽，造成信噪比严重下降，从而导致语音的可听度及清晰度下降。
主动降噪技术最早作为一种新技术用于军事通信上。在主动降噪出现之前，只能是采取被动降噪的方式，即用整个耳罩将左右耳朵包住，同过耳罩的隔声特性，及在耳罩内增加吸音海绵，来降低到达人耳的噪声。但这种方法只能降低高频，对低频的声音降噪不大。主动降噪技术出现后，出现了各系列的有源降噪头戴。
最早降噪原理包括两种：一种是前馈降噪，一种是反馈降噪。具体是通过装置在耳罩内部的麦克风来采集人耳附近的背景噪声大小，通过电路产生一种相位相差180度的反向噪声，喇叭放出后与背景噪声进行声干涉叠加，从而有效的降低背景噪声信号大小。其好处是：在提高的通话质量的同时保护了人耳，避免了人耳长时间暴露在高噪声环境中造成的听力损伤。主动降噪作为一种新技术得到了长足的发展，逐渐由军用转向民用领域。
目前手机上存在的降噪技术，主要集中在如何提高手机发送方向的话音质量，即通常意义上的双麦克降噪技术。该技术把着重点放在了如何在噪声环境比较差的地方(如办公室、餐馆、大街等公共场所)，提高上行话音的信噪比，从而让通话的另一端听音更清晰。
但我们知道手机通话时还有一种现象，即在嘈杂的环境下听不清楚对方的声音，即使你将听筒音量调整到最大，调整各种角度，你依然听的比较模糊。我们知道不能通过一味的提高听筒的电信号大小，来提高音量，这样增加了烧坏听筒的风险。市场上的手机都一致性忽略了噪声对本地通话者的听觉影响，未取得关注。目前为止市场上未有具有下行主动降噪的手机面世。
发明内容
本发明实施例提供了一种下行主动降噪装置、方法及移动终端，以至少解决现有技术中智能移动终端不能滤除本地通话噪音，影响通话质量的问题。
依据本发明的一个方面，提供一种下行主动降噪装置，包括：
第一声波采集器，设置为采集环境噪声信号，得到参考信号，并将所述参考信号输出至自适应控制器；
自适应控制器，设置为基于接收的参考信号，计算用以实施声波抵消的次级信号，并基于接收到的误差信号，正向或反向调整次级信号的幅值及相位；
次级声场发射器，设置为将自适应控制器调整后的次级信号以声波形式发出；
第二声波采集器，设置为采集环境噪声信号和次级声场发射器发出的次级信号，将二者进行叠加，得到误差信号后输出至自适应控制器。
可选地，本发明所述装置中，所述第一声波采集器，还设置为检测当前环境噪声是否达到设定的阈值，若是，则将得到的参考信号输出至自适应控制器；否则，判定当前无需降噪处理，结束。
可选地，本发明所述装置中，所述自适应控制器，设置为采用预置的XLMS自适应算法，计算次级信号以及调整次级信号的幅值及相位。
可选地，本发明所述装置中，所述自适应控制器，具体包括：
A/D转换单元，设置为将自适应控制器接收到的参考信号和误差信号进行模数转换后输出至DSP；
DSP，设置为基于接收的参考信号，计算用以实施声波抵消的次级信号，并基于误差信号，正向或反向调整次级信号的幅值及相位；
D/A转换单元，设置为将调整后的次级信号进行数模转换后输出至功放单元；
功放单元，设置为将次级信号进行功率放大处理后输出至次级声场发射器。
可选地，本发明所述装置中，所述第二声波采集器以无限靠近于次级声场发射器的原则进行布设；所述第一声波采集器、第二声波采集器的器件类型为麦克风。
依据本发明的另一个方面，提供一种移动终端，所述移动终端内集成有本发明所述的下行主动降噪装置。
依据本发明的再一个方面，提供一种下行主动降噪方法，包括：
步骤1，采集环境噪声信号，得到参考信号；
步骤2，基于所述参考信号，计算用以实施声波抵消的次级信号；
步骤3，利用当前得到的误差信号，正向或反向调整次级信号幅值及相位；
步骤4，将调整后的次级信号以声波的形式发出；
步骤5，采集环境噪声信号和以声波形式发出的次级信号，将二者进行叠加后得到误差信号，并将该误差信号提供给步骤3。
可选地，本发明所述方法中，所述步骤1前还包括：检测当前环境噪声是否达到设定的阈值，若是，则继续执行步骤1；否则，判定当前无需降噪处理，结束。
本发明有益效果如下：
本发明所述下行主动降噪装置通过布设两个声波采集器，实现了噪声信号和误差信号的提取，并通过自适应控制器实现了利用误差信号对次级信号的实时修正，从而达到局部的“静区”；
集成有主动降噪装置的移动终端，将主动降噪应用到移动终端上，提高了本地通话接收效果，具有良好的市场前景。
附图说明
为了更清楚地说明本发明实施例或现有技术中的技术方案，下面将对实施例或现有技术描述中所需要使用的附图作一简单地介绍，显而易见地，下面描述中的附图仅仅是本发明的一些实施例，对于本领域普通技术人员来讲，在不付出创造性劳动性的前提下，还可以根据这些附图获得其他的附图。
图1为本发明提供的一种下行主动降噪装置的结构图；
图2为本发明中自适应控制器的结构图；
图3为本发明中移动终端上布设声波采集器的布设示意图；
图4为本发明提供的一种下行主动降噪方法的流程图。
具体实施方式
下面将结合本发明实施例中的附图，对本发明实施例中的技术方案进行清楚、完整地描述，显然，所描述的实施例仅仅是本发明一部分实施例，而不是全部的实施例。基于本发明中的实施例，本领域普通技术人员在没有做出创造性劳动前提下所获得的所有其他实施例，都属于本发明保护的范围。
为了解决现有技术中智能移动终端不能滤除本地通话噪音，影响通话质量的问题，本发明提供了一种下行主动降噪装置、方法及移动终端，所述装置通过设置两个声波采集器，一个拾取外界噪声信号作为参考信号，一个通过拾取噪声信号及次级信号求取二者的误差信号，以通过误差信号，控制次级声源大小，从而达到局部的“静区”，使噪声信号减到最小，此举提高了下行语音的可懂度。下面就通过几个具体实施例对本发明的详细实施过程进行阐述。
实施例一
本发明实施例提供一种下行主动降噪装置，如图1所示，所述装置具体包括：
第一声波采集器110，设置为采集环境噪声信号，得到参考信号，并将所述参考信号输出至自适应控制器120；
自适应控制器120，设置为基于接收的参考信号，计算用以实施声波抵消的次级信号，并基于接收到的误差信号，正向或反向调整次级信号的幅值及相位；
其中，计算的用以实施声波抵消的次级信号理想状态下指将参考信号相位翻转180度后的信号，但由于电路处理及声波传输都存在时延，所以还需根据时延信息对翻转后信号的相位信息进行修正，修正后得到的信号即为计算得到的次级信号。
进一步的，自适应控制器120正向还是反向调整次级信号的幅值及相位，是由误差信号来决定的，最终目的是对次级信号调整后，使得次级信号与噪声信号的误差趋近于零，理想状态下等于零。
次级声场发射器130，设置为将自适应控制器120调整后的次级信号以声波形式发出；
第二声波采集器140，设置为采集环境噪声信号和次级声场发射器130发出的次级信号，将二者进行叠加，得到误差信号后输出至自适应控制器120。
基于上述结构框架及实施原理，下面给出在上述结构下的几个具体及优选实施方式，用以细化和优化本发明所述下行主动降噪装置的功能，具体涉及如下内容：
本发明实施例中，为了降低降噪带来的能量损耗，避免浪费，优选地，第一声波采集器110在采集到环境噪声时，检测当前环境噪声是否达到设定的阈值，若是，则将得到的参考信号输出至自适应控制器120；否则，判定当前无需降噪处理，结束。也就是说，只有背景噪声较大时，才开启主动降噪过程，并非所有情况都进行降噪处理。
进一步地，本发明实施例中，第二声波采集器140以无限靠近于次级声场发射器130的原则进行布设，以更好的接收次级声场发射器130发出的次级信号；
进一步地，本发明实施例中，第一声波采集器110、第二声波采集器140的器件类型可以但不限于为麦克风。
进一步的，本发明实施例中，自适应控制器120，具体采用预置的XLMS自适应算法，计算次级信号以及调整次级信号的幅值及相位。当然，需要说明的是，采用XLMS自适应算法只是一种示例性解释，并非唯一限定采用该算法实现，本领域技术人员容易想到的可以实现自适应控制器120的相关功能的算法都在本发明的保护思想范围之内。
进一步地，本发明实施例中还给出自适应控制器120的具体结构组成，以从结构功能方面更清楚的说明自适应控制器120，如图2所示，自适应控制器120具体包括：
A/D转换单元121，设置为将自适应控制器120接收到的参考信号和误差信号进行模数转换后输出至DSP122；
DSP122，设置为基于接收的参考信号，计算用以实施声波抵消的次级信号，并基于误差信号，正向或反向调整次级信号的幅值及相位；
D/A转换单元123，设置为将调整后的次级信号进行数模转换后输出至功放单元124；
功放单元124，设置为将次级信号进行功率放大处理后输出至次级声场发射器130。
综上所述，本发明实施例所述下行主动降噪装置，通过布设两个声波采集器，实现了噪声信号和误差信号的提取，并通过自适应控制器实现了利用误差信号对次级信号的实时修正，从而达到局部的“静区”。
实施例二
本发明实施例提供一种移动终端，该移动终端集成有实施例一所述的下行主动降噪装置。由于实施例一中已经对下行主动降噪装置的结构进行详细阐述，所以本实施例不对集成后的结构再做赘述。
为了将下行主动降噪装置更好的与移动终端进行融合，本实施例中还给出集成后第一声波采集器、第二声波采集器以及次级声场发射器的布设位置。具体地，如图3所示，第一声波采集器设置在移动终端的背面；次级声场发射器为移动终端的听筒；第二声波采集器设置在靠近移动终端的听筒位置处。
其中，第一声波采集器的布设原则为：布设在移动终端被面不易被用户手持遮挡的位置。
下面给出本发明一个较佳的实施例，并结合对实施例的描述，进一步给出本发明的技术细节，使其能够更好地说明移动终端实现下行主动降噪的过程。
本实施例中：
P(t)--噪声源发出的信号；
x(t)--参考信号；
y(t)--次级信号；
e(t)--误差信号。
本实施例中，移动终端实现下行主动降噪的具体控制过程如下：
(1)噪声源发出声波信号P(t)；
(2)位于听筒背面噪声源附近的麦克风MIC1(第一声波采集器)拾取参考信号x(t)(即MIC1采集的噪声信号)，作为自适应控制器的输入；
(3)自适应控制器在预置的XLMS自适应算法的作用下计算出次级信号y(t)，输出至听筒，由听筒以声波的形式发出实施声波抵消；
(4)噪声源和听筒分别形成初级声场和次级声场，位于听筒附近的误差麦克风MIC2(第二声波采集器)同时接收初级声场和次级声场的声压，两者叠加形成误差信号e(t)；
(5)误差信号e(t)输入反馈至自适应控制器中，自适应控制器利用预置的XLMS自适应算法，根据预先设定的控制目标函数调整自适应控制器权系数，从而快速的正向或反向调整当前计算得到的次级信号的幅值及相位，并将调整后的次级信号输出至听筒，由听筒以声波的形式发出实施声波抵消。重复上述过程，直到系统的稳定，形成局部“静区”。
为了给出更多技术细节，下面结合自适应控制器的具体结构(如图2所示)进行降噪过程说明。
将自适应算法程序装载于自适应控制器的DSP内。位于听筒背面的MIC1采集周围背景噪音，作为参考信号x(t)，通过A/D转换单元采样进入到DSP内。
装载有自适应算法的DSP输出次级信号y(t)至听筒，由听筒以声波的形式发出实施声波抵消。
位于听筒附近的MIC2，作为误差麦克风MIC2同时接受初级声场和次级声场的声压，两者叠加形成误差信号e(t)，误差信号输入反馈至DSP中。
DSP利用内置的自适应算法，根据预先设定的控制目标函数调整控制器权系数，从而快速的改变次级信号幅值及相位，通过D/A转换单元及功放单元输出y(t)至听筒，直到系统的稳定，形成局部稳定主动降噪。
而对于移动终端接收到的通话语音信号直接经D/A转换单元及功放单元输出至听筒。
综上所述，可知本发明实施例所述移动终端，通过下行主动降噪装置，将主动降噪应用到手机上，提高了本地通话接收效果，具有良好的市场前景。
实施例三
本发明实施例提供一种下行主动降噪方法，如图4所示，包括：
步骤S401，采集环境噪声信号，得到参考信号；
优选地，在该步骤前还包括：检测当前环境噪声是否达到设定的阈值，若是，则继续执行步骤S401；否则，判定当前无需降噪处理，结束。
步骤S402，基于所述参考信号，计算用以实施声波抵消的次级信号；
步骤S403，利用当前得到的误差信号，正向或反向调整次级信号幅值及相位；
步骤S404，将调整后的次级信号以声波的形式发出。
步骤S405，采集环境噪声信号和以声波形式发出的次级信号，将二者进行叠加后得到误差信号，并将该误差信号提供给步骤S403。
本实施例所述方法，实现了噪声信号和误差信号的提取，并利用误差信号对次级信号进行实时修正，从而达到局部的“静区”。
显然，本领域的技术人员可以对本发明进行各种改动和变型而不脱离本发明的精神和范围。这样，倘若本发明的这些修改和变型属于本发明权利要求及其等同技术的范围之内，则本发明也意图包含这些改动和变型在内。
工业实用性
如上所述，本发明实施例提供的一种下行主动降噪装置、方法及移动终端具有以下有益效果：通过布设两个声波采集器，实现了噪声信号和误差信号的提取，并通过自适应控制器实现了利用误差信号对次级信号的实时修正，从而达到局部的“静区”；而集成有主动降噪装置的移动终端，将主动降噪应用到移动终端上，提高了本地通话接收效果，具有良好的市场前景。

Claims (10)

  1. 一种下行主动降噪装置，包括：
    第一声波采集器，设置为采集环境噪声信号，得到参考信号，并将所述参考信号输出至自适应控制器；
    自适应控制器，设置为基于接收的参考信号，计算用以实施声波抵消的次级信号，并基于接收到的误差信号，正向或反向调整次级信号的幅值及相位；
    次级声场发射器，设置为将自适应控制器调整后的次级信号以声波形式发出；
    第二声波采集器，设置为采集环境噪声信号和次级声场发射器发出的次级信号，将二者进行叠加，得到误差信号后输出至自适应控制器。
  2. 如权利要求1所述的装置，其中，所述第一声波采集器，还设置为检测当前环境噪声是否达到设定的阈值，若是，则将得到的参考信号输出至自适应控制器；否则，判定当前无需降噪处理，结束。
  3. 如权利要求1所述的装置，其中，所述自适应控制器，设置为采用预置的XLMS自适应算法，计算次级信号以及调整次级信号的幅值及相位。
  4. 如权利要求1至3任意一项所述的装置，其中，所述自适应控制器，具体包括：
    A/D转换单元，设置为将自适应控制器接收到的参考信号和误差信号进行模数转换后输出至DSP；
    DSP，设置为基于接收的参考信号，计算用以实施声波抵消的次级信号，并基于误差信号，正向或反向调整次级信号的幅值及相位；
    D/A转换单元，设置为将调整后的次级信号进行数模转换后输出至功放单元；
    功放单元，设置为将次级信号进行功率放大处理后输出至次级声场发射器。
  5. 如权利要求1所述的装置，其中，
    所述第二声波采集器以无限靠近于次级声场发射器的原则进行布设；
    所述第一声波采集器、第二声波采集器的器件类型为麦克风。
  6. 一种移动终端，所述移动终端内集成有权利要求1至5任意一项所述的下行主动降噪装置。
  7. 如权利要求6所述的移动终端，其中，所述下行主动降噪装置中：
    第一声波采集器设置在移动终端的背面；
    次级声场发射器为移动终端的听筒；
    第二声波采集器设置在靠近移动终端的听筒位置处。
  8. 如权利要求7所述的移动终端，其中，所述第一声波采集器布设在移动终端背面不易被用户手持遮挡的位置。
  9. 一种下行主动降噪方法，包括：
    步骤1，采集环境噪声信号，得到参考信号；
    步骤2，基于所述参考信号，计算用以实施声波抵消的次级信号；
    步骤3，利用当前得到的误差信号，正向或反向调整次级信号幅值及相位；
    步骤4，将调整后的次级信号以声波的形式发出；
    步骤5，采集环境噪声信号和以声波形式发出的次级信号，将二者进行叠加后得到误差信号，并将该误差信号提供给步骤3。
  10. 如权利要求9所述的方法，其中，所述步骤1前还包括：检测当前环境噪声是否达到设定的阈值，若是，则继续执行步骤1；否则，判定当前无需降噪处理，结束。
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