麦克风阵列仿真环境的搭建 1. 引言 之前,我在语音增强一文中,提到了有关麦克风阵列语音增强的介绍,当然,麦克风阵列能做的东西远远不只是在语音降噪上的应用,它还可以用来做声源定位、声源估计、波束形成、回声抑制等。个人认为,麦克风阵列在声源定位和波束形成(多指抑制干扰语音方面)的优势是单通道麦克风算法无法比拟的。因为,利用多麦克风以后,就会将空间信息考虑到算法中,这样就特别适合解决一些与空间相关性很强的语音处理问题。 然而,在做一些麦克风阵列相关的算法研究的时候,最先遇到的问题就是:实验环境的搭建。很多做麦克风阵列的爱好者并没有实际的硬件实验环境,这也就成了很多人进行麦克风阵列入门的难题。这里,我要分享的是爱丁堡大学语音实验室开源的基于MATLAB的麦克风阵列实验仿真环境。利用该仿真环境,我们就可以随意的设置房间的大小,混响程度,声源方向以及噪声等基本参数,然后得到我们想要的音频文件去测试你自己相应的麦克风阵列算法。 2. 代码介绍 原始的代码被我加以修改,也是为了更好的运行,如果有兴趣的话,大家还可以参考爱丁堡大学最初的源码,并且我也上传到我的CSDN码云上了,链接是:https://gitee.com/wind_hit/Microphone-Array-Simulation-Environment。 这套MATLAB代码的主函数是multichannelSignalGenerator(),具体如下: function [mcSignals,setup] =multichannelSignalGenerator(setup) %----------------------------------------------------------------------- % Producing the multi_noisy_signals for Mic array Beamforming. % % Usage: multichannelSignalGenerator(setup) % % setup.nRirLength: The length of Room Impulse Response Filter % setup.hpFilterFlag : use 'false' to disable high-pass filter,the high-%pass filter is enabled by default % setup.reflectionOrder: reflection order, default is -1, i.e. maximum order. % setup.micType: [omnidirectional, subcardioid, cardioid, hypercardioid, bidirectional],default is omnidirectional. % % setup.nSensors: The numbers of the Mic % setup.sensorDistance: The distance between the adjacent Mics (m) % setup.reverbTime: The reverberation time of room % setup.speedOfSound: sound velocity (m/s) % % setup.noiseField: Two kinds of Typical noise field, 'spherical' and 'cylindrical' % setup.sdnr: The target mixing snr for diffuse noise and clean siganl. % setup.ssnr: The approxiated mixing snr for sensor noise and clean siganl. % % setup.roomDim: 1 x 3 array specifying the (x,y,z) coordinates of the room (m). % setup.micPoints: 3 x M array, the rows specifying the (x,y,z) coordinates of the mic postions(m). % setup.srcPoint : 3 x M array, the rows specifying the(x,y,z) coordinates of the audio source postion(m). % % srcHeight: The height of target audio source % arrayHeight: The height of mic array % % arrayCenter: The Center Postion of mic array % % arrayToSrcDistInt:The distance between the array and audio source on the xy axis % % % % % % % How To Use : JUST RUN % % % % Code From: Audio analysis Lab of AalborgUniversity (Website: https://audio.create.aau.dk/), % slightly modified by Wind at Harbin Institute of Technology, Shenzhen, in 2018.3.24 % % Copyright (C) 1989, 1991 Free SoftwareFoundation, Inc. %------------------------------------------------------------------------- addpath([cd,'\..\rirGen\']); %-----------------------------------------------initialparameters----------------------------------- setup.nRirLength = 2048; setup.hpFilterFlag = 1; setup.reflectionOrder = -1; setup.micType = 'omnidirectional'; setup.nSensors = 4; setup.sensorDistance = 0.05; setup.reverbTime = 0.1; setup.speedOfSound = 340; setup.noiseField = 'spherical'; setup.sdnr = 20; setup.ssnr = 25; setup.roomDim = [3;4;3]; srcHeight = 1; arrayHeight = 1; arrayCenter = [setup.roomDim(1:2)/2;1]; arrayToSrcDistInt = [1,1]; setup.srcPoint = [1.5;1;1]; setup.micPoints = generateUlaCoords(arrayCenter,setup.nSensors,setup.sensorDistance,0,arrayHeight); [cleanSignal,setup.sampFreq] =audioread('..\data\twoMaleTwoFemale20Seconds.wav'); %---------------------------------------------------initialend---------------------------------------- %-------------------------------algorithmprocessing-------------------------------------------------- if setup.reverbTime == 0, setup.reverbTime = 0.2; reflectionOrder = 0; else reflectionOrder = -1; end rirMatrix = rir_generator(setup.speedOfSound,setup.sampFreq,setup.micPoints',setup.srcPoint',setup.roomDim',... setup.reverbTime,setup.nRirLength,setup.micType,setup.reflectionOrder,[],[],setup.hpFilterFlag); for iSens = 1:setup.nSensors, tmpCleanSignal(:,iSens) = fftfilt(rirMatrix(iSens,  ',cleanSignal); end mcSignals.clean =tmpCleanSignal(setup.nRirLength:end,  ; setup.nSamples = length(mcSignals.clean); mcSignals.clean = mcSignals.clean -ones(setup.nSamples,1)*mean(mcSignals.clean); %-------produce the microphone recievedclean signals--------------------------------------------- mic_clean1=10*mcSignals.clean(:,1);%Because of the attenuation of the recievd signals,Amplify the signals recievedby Mics with tenfold mic_clean2=10*mcSignals.clean(:,2); mic_clean3=10*mcSignals.clean(:,3); mic_clean4=10*mcSignals.clean(:,4); audiowrite('mic_clean1.wav',mic_clean1,setup.sampFreq); audiowrite('mic_clean2.wav',mic_clean2,setup.sampFreq); audiowrite('mic_clean3.wav',mic_clean3,setup.sampFreq); audiowrite('mic_clean4.wav',mic_clean4,setup.sampFreq); %----------------------------------end-------------------------------------------------- addpath([cd,'\..\nonstationaryMultichanNoiseGenerator\']); cleanSignalPowerMeas =var(mcSignals.clean); mcSignals.diffNoise =generateMultichanBabbleNoise(setup.nSamples,setup.nSensors,setup.sensorDistance,... setup.speedOfSound,setup.noiseField); diffNoisePowerMeas =var(mcSignals.diffNoise); diffNoisePowerTrue =cleanSignalPowerMeas/10^(setup.sdnr/10); mcSignals.diffNoise =mcSignals.diffNoise*... diag(sqrt(diffNoisePowerTrue)./sqrt(diffNoisePowerMeas)); mcSignals.sensNoise =randn(setup.nSamples,setup.nSensors); sensNoisePowerMeas =var(mcSignals.sensNoise); sensNoisePowerTrue = cleanSignalPowerMeas/10^(setup.ssnr/10); mcSignals.sensNoise =mcSignals.sensNoise*... diag(sqrt(sensNoisePowerTrue)./sqrt(sensNoisePowerMeas)); mcSignals.noise = mcSignals.diffNoise +mcSignals.sensNoise; mcSignals.observed = mcSignals.clean +mcSignals.noise; %------------------------------processingend----------------------------------------------------------- %----------------produce the noisy speechof MIc in the specific ervironment sets------------------------ noisy_mix1=10*mcSignals.observed(:,1);%Amplify the signals recieved by Mics with tenfold noisy_mix2=10*mcSignals.observed(:,2); noisy_mix3=10*mcSignals.observed(:,3); noisy_mix4=10*mcSignals.observed(:,4); l1=size(noisy_mix1); l2=size(noisy_mix2); l3=size(noisy_mix3); l4=size(noisy_mix4); audiowrite('diffused_babble_noise1_20dB.wav',noisy_mix1,setup.sampFreq); audiowrite('diffused_babble_noise2_20dB.wav',noisy_mix2,setup.sampFreq); audiowrite('diffused_babble_noise3_20dB.wav',noisy_mix3,setup.sampFreq); audiowrite('diffused_babble_noise4_20dB.wav',noisy_mix4,setup.sampFreq); %-----------------------------end------------------------------------------------------------------------- 这个是主函数,直接运行尽可以得到想要的音频文件,但是你需要先给出你的纯净音频文件和噪声音频,分别对应着:multichannelSignalGenerator()函数中的语句:[cleanSignal,setup.sampFreq]= audioread('..\data\twoMaleTwoFemale20Seconds.wav'),和generateMultichanBabbleNoise()函数中的语句:[singleChannelData,samplingFreq]= audioread('babble_8kHz.wav') 。 直接把它们替换成你想要处理的音频文件即可。 除此之外,还有一些基本实验环境参数设置,包括:麦克风的形状为线性麦克风阵列(该代码只能对线性阵列进行仿真建模,并且还是均匀线性阵列,这个不需要设置);麦克风的类型(micType),有全指向型(omnidirectional),心型指向(cardioid),亚心型指向(subcardioid,不知道咋翻译,请见谅) , 超心型(hypercardioid), 双向型(bidirectional),一般默认是全指向型,如下图1所示;麦克风的数量(nSensors);各麦克风之间的间距(sensorDistance);麦克风阵列的中心位置(arrayCenter),用(x,y,z)坐标来表示;麦克风阵列的高度(arrayHeight),感觉和前面的arrayCenter有所重复,不知道为什么还要设置这么一个参数;目标声源的位置(srcPoint),也是用(x,y,z)坐标来表示;目标声源的高度(srcHeight);麦克风阵列距离目标声源的距离(arrayToSrcDistInt),是在xy平面上的投影距离;房间的大小(roomDim),另外房间的(x,y,z)坐标系如图2所示;房间的混响时间(reverbTime);散漫噪声场的类型(noiseField),分为球形场(spherical)和圆柱形场(cylindrical)。 file:///C:/Users/ADMINI~1/AppData/Local/Temp/msohtmlclip1/01/clip_image002.gif 图1 麦克风类型图 file:///C:/Users/ADMINI~1/AppData/Local/Temp/msohtmlclip1/01/clip_image004.gif 图二房间的坐标系 以上便是整个仿真实验环境的参数配置,虽然只能对均匀线性的麦克风阵列进行实验测试,但是这对满足我们进行线阵阵列算法的测试是有很大的帮助。说到底,这种麦克风阵列环境的音频数据产生方法还是基于数学模型的仿真,并不可能取代实际的硬件实验环境测试,所以要想在工程上实现麦克风阵列的一些算法,仍然避免不了在实际的环境中进行测试。最后,希望分享的这套代码对大家进行麦克风阵列算法的入门提供帮助。 ———————————————— 版权声明:本文为CSDN博主「Mr_Researcher」的原创文章,遵循CC 4.0 BY-SA版权协议,转载请附上原文出处链接及本声明。 原文链接:https://blog.csdn.net/zhanglu_wind/article/details/79674998
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