前言:
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基于自编码器的语音信号降噪 - 哥廷根数学学派的文章 - 知乎
本篇讲一些稍微复杂的基于深度学习的语音降噪方法,并比较了应用于同一任务的两种的网络:全连接层网络和卷积网络。
完整代码如下
考虑以下以 8 kHz 采样的语音信号
[cleanAudio,fs] = audioread("SpeechDFT.wav");sound(cleanAudio,fs)
将洗衣机噪声添加到上述的语音信号中,设置噪声功率,使信噪比 (SNR) 为0dB
noise = audioread("WashingMachine.mp3");
接下来从噪声文件中的随机位置提取噪声段
ind = randi(numel(noise) - numel(cleanAudio) + 1, 1, 1);noiseSegment = noise(ind:ind + numel(cleanAudio) - 1);speechPower = sum(cleanAudio.^2);noisePower = sum(noiseSegment.^2);noisyAudio = cleanAudio + sqrt(speechPower/noisePower) * noiseSegment;%播放信号sound(noisyAudio,fs)
可视化原始信号和噪声信号
t = (1/fs) * (0:numel(cleanAudio)-1);subplot(2,1,1)plot(t,cleanAudio)title("Clean Audio")grid onsubplot(2,1,2)plot(t,noisyAudio)title("Noisy Audio")xlabel("Time (s)")grid on
语音降噪的目的是从语音信号中去除洗衣机噪声,同时最大限度地减少输出语音信号中不希望的所谓的artifacts。
检查数据集
本例使用的数据集包含 48 kHz 的短句录音
训练集,测试集,验证集文件
训练集部分数据
使用 audioDatastore 为训练集创建数据存储
adsTrain = audioDatastore(fullfile(dataFolder,'train'),'IncludeSubfolders',true);
读取datastore中第一个文件的内容
[audio,adsTrainInfo] = read(adsTrain);
播放语音信号
sound(audio,adsTrainInfo.SampleRate)
绘制语音信号
figuret = (1/adsTrainInfo.SampleRate) * (0:numel(audio)-1);plot(t,audio)title("Example Speech Signal")xlabel("Time (s)")grid on
深度学习系统概述
基本的深度学习训练方案如下图所示。
对于熟悉语音信号处理的同学肯定是小case。注意,由于语音通常低于 4 kHz,因此首先将干净和嘈杂的语音信号下采样到 8 kHz,以减少网络的计算负担。 网络的输出是降噪信号的幅度谱,使用输出幅度谱和噪声信号的相位将降噪后的音频转换回时域[1]。
可以使用短时傅里叶变换 (STFT) 将音频信号转换到频域,使用Hamming窗,窗口长度为 256 ,重叠率为 75%。Predicter的输入由 8 个连续的噪声 STFT 向量组成,因此每个 STFT 输出估计值都是基于当前的噪声 STFT 和 7 个先前的噪声 STFT 向量计算的。
如何从一个训练文件生成Target和Predicter(直接用英文单词了,否则容易引起误解)?首先,定义系统参数:
windowLength = 256;win = hamming(windowLength,"periodic");overlap = round(0.75 * windowLength);ffTLength = windowLength;inputFs = 48e3;fs = 8e3;numFeatures = ffTLength/2 + 1;numSegments = 8;
创建一个 dsp.SampleRateConverter 对象以将 48 kHz 音频转换为 8 kHz,
src = dsp.SampleRateConverter("InputSampleRate",inputFs, ... "OutputSampleRate",fs, ... "Bandwidth",7920);
从datastore中读入音频文件的内容
audio = read(adsTrain);
注意:要确保音频长度是采样率转换器抽取因子的倍数
decimationFactor = inputFs/fs;L = floor(numel(audio)/decimationFactor);audio = audio(1:decimationFactor*L);
将音频信号转换为 8 kHz
audio = src(audio);reset(src)
从洗衣机噪声向量中创建一个随机噪声段
randind = randi(numel(noise) - numel(audio),[1 1]);noiseSegment = noise(randind : randind + numel(audio) - 1);
向语音信号中添加噪声,使 SNR 为 0 dB。
noisePower = sum(noiseSegment.^2);cleanPower = sum(audio.^2);noiseSegment = noiseSegment .* sqrt(cleanPower/noisePower);noisyAudio = audio + noiseSegment;
使用STFT从原始和嘈杂的音频信号生成幅值STFT向量
cleanSTFT = stft(audio,'Window',win,'OverlapLength',overlap,'FFTLength',ffTLength);cleanSTFT = abs(cleanSTFT(numFeatures-1:end,:));noisySTFT = stft(noisyAudio,'Window',win,'OverlapLength',overlap,'FFTLength',ffTLength);noisySTFT = abs(noisySTFT(numFeatures-1:end,:));
从嘈杂的 STFT 生成 8 段训练Predicter信号,对应深度学习训练方案图
noisySTFT = [noisySTFT(:,1:numSegments - 1), noisySTFT];stftSegments = zeros(numFeatures, numSegments , size(noisySTFT,2) - numSegments + 1);for index = 1:size(noisySTFT,2) - numSegments + 1 stftSegments(:,:,index) = (noisySTFT(:,index:index + numSegments - 1)); end
设置Target和Predicter,每个Predicter维度为 129×8,每个Target为 129×1,这些都是可以根据信号调整的
targets = cleanSTFT;size(targets)predictors = stftSegments;size(predictors)
为了加快处理速度,使用 tall 数组从datastore中所有音频文件的语音片段中提取特征序列,要是有matlab的并行工具箱和GPU。
首先,将datastore转换为 tall 数组,此处需要的时间较长
reset(adsTrain)T = tall(adsTrain)
从 tall 表中提取Target和Predicter的幅值STFT,这一步很重要,GenerateSpeechDenoisingFeatures是提取Target和Predicter的幅值STFT的函数,我还需要优化一点
[targets,predictors] = cellfun(@(x)GenerateSpeechDenoisingFeatures(x,noise,src),T,"UniformOutput",false);[targets,predictors] = gather(targets,predictors);
将所有特征进行归一化,分别计算Target和Predicter的均值和标准差,并使用它们对数据进行归一化。
predictors = cat(3,predictors{:});noisyMean = mean(predictors(:));noisyStd = std(predictors(:));predictors(:) = (predictors(:) - noisyMean)/noisyStd;targets = cat(2,targets{:});cleanMean = mean(targets(:));cleanStd = std(targets(:));targets(:) = (targets(:) - cleanMean)/cleanStd;
将Target和Predicter进行维度重塑,即reshape为与深度学习网络相对应的维度
predictors = reshape(predictors,size(predictors,1),size(predictors,2),1,size(predictors,3));targets = reshape(targets,1,1,size(targets,1),size(targets,2));
将数据随机分成训练集和验证集。
inds = randperm(size(predictors,4));L = round(0.99 * size(predictors,4));trainPredictors = predictors(:,:,:,inds(1:L));trainTargets = targets(:,:,:,inds(1:L));validatePredictors = predictors(:,:,:,inds(L+1:end));validateTargets = targets(:,:,:,inds(L+1:end));
下面开始步入正题,进行第一个全连接层深层网络的语音降噪
全连接网络是什么就不讲了,看图
定义网络层,将输入大小指定为大小为 NumFeatures-by-NumSegments(本例中为 129-by-8)的图像。 定义两个隐藏的全连接层,每个层有 1024 个神经元。 由于是纯线性系统,因此在每个隐藏的全连接层之后都有一个整流线性单元 (ReLU) 层。 批量归一化层对输出的均值和标准差进行归一化,添加一个具有 129 个神经元的全连接层,然后是一个回归层。
layers = [ imageInputLayer([numFeatures,numSegments]) fullyConnectedLayer(1024) batchNormalizationLayer reluLayer fullyConnectedLayer(1024) batchNormalizationLayer reluLayer fullyConnectedLayer(numFeatures) regressionLayer ];
然后设置网络的训练选项,很好理解
miniBatchSize = 128;options = trainingOptions("adam", ... "MaxEpochs",3, ... "InitialLearnRate",1e-5,... "MiniBatchSize",miniBatchSize, ... "Shuffle","every-epoch", ... "Plots","training-progress", ... "Verbose",false, ... "ValidationFrequency",floor(size(trainPredictors,4)/miniBatchSize), ... "LearnRateSchedule","piecewise", ... "LearnRateDropFactor",0.9, ... "LearnRateDropPeriod",1, ... "ValidationData",{validatePredictors,validateTargets});
利用trainNetwork 使用指定的训练选项和网络层训练深层网络。 因为训练集很大,训练过程较为耗时
denoiseNetFullyConnected = trainNetwork(trainPredictors,trainTargets,layers,options);
计算网络全连接层的权重数量
numWeights = 0;for index = 1:numel(denoiseNetFullyConnected.Layers) if isa(denoiseNetFullyConnected.Layers(index),"nnet.cnn.layer.FullyConnectedLayer") numWeights = numWeights + numel(denoiseNetFullyConnected.Layers(index).Weights); endendfprintf("The number of weights is %d.\n",numWeights);
然后进行卷积层神经网络的语音去噪
卷积层通常比全连接层包含更少的参数,根据文献[2]中描述的全卷积网络的层数,包括 16 个卷积层。 前 15 个卷积层为3层的组,重复 5 次,滤波器宽度分别为 9、5 和 9,滤波器数量分别为 18、30 和 8。 最后一个卷积层的滤波器宽度为129 。 在这个网络中,卷积只在一个方向(沿着频率维度)执行,并且除了第一层之外的所有层,沿着时间维度的滤波器宽度设置为 1。 与全连接网络类似,卷积层之后是 ReLu 和批量归一化层。
layers = [imageInputLayer([numFeatures,numSegments]) convolution2dLayer([9 8],18,"Stride",[1 100],"Padding","same") batchNormalizationLayer reluLayer repmat( ... [convolution2dLayer([5 1],30,"Stride",[1 100],"Padding","same") batchNormalizationLayer reluLayer convolution2dLayer([9 1],8,"Stride",[1 100],"Padding","same") batchNormalizationLayer reluLayer convolution2dLayer([9 1],18,"Stride",[1 100],"Padding","same") batchNormalizationLayer reluLayer],4,1) convolution2dLayer([5 1],30,"Stride",[1 100],"Padding","same") batchNormalizationLayer reluLayer convolution2dLayer([9 1],8,"Stride",[1 100],"Padding","same") batchNormalizationLayer reluLayer convolution2dLayer([129 1],1,"Stride",[1 100],"Padding","same") regressionLayer ];
训练选项与全连接网络的选项类似
options = trainingOptions("adam", ... "MaxEpochs",3, ... "InitialLearnRate",1e-5, ... "MiniBatchSize",miniBatchSize, ... "Shuffle","every-epoch", ... "Plots","training-progress", ... "Verbose",false, ... "ValidationFrequency",floor(size(trainPredictors,4)/miniBatchSize), ... "LearnRateSchedule","piecewise", ... "LearnRateDropFactor",0.9, ... "LearnRateDropPeriod",1, ... "ValidationData",{validatePredictors,permute(validateTargets,[3 1 2 4])});
利用 trainNetwork 使用指定的训练选项和层架构训练网络
denoiseNetFullyConvolutional = trainNetwork(trainPredictors,permute(trainTargets,[3 1 2 4]),layers,options);
计算网络全连接层的权重数量
numWeights = 0;for index = 1:numel(denoiseNetFullyConvolutional.Layers) if isa(denoiseNetFullyConvolutional.Layers(index),"nnet.cnn.layer.Convolution2DLayer") numWeights = numWeights + numel(denoiseNetFullyConvolutional.Layers(index).Weights); endendfprintf("The number of weights in convolutional layers is %d\n",numWeights);
测试降噪网络
读入测试集
adsTest = audioDatastore(fullfile(dataFolder,'test'),'IncludeSubfolders',true);
从datastore中读取文件
[cleanAudio,adsTestInfo] = read(adsTest);
确保音频长度是采样率转换器抽取因子的倍数
L = floor(numel(cleanAudio)/decimationFactor);cleanAudio = cleanAudio(1:decimationFactor*L);
将音频信号转换为 8 kHz。
cleanAudio = src(cleanAudio);reset(src)
测试阶段使用训练阶段未使用的洗衣机噪声来给语音信号加噪
noise = audioread("WashingMachine-16-8-mono-200secs.mp3");
从洗衣机噪声向量中创建一个随机噪声段
randind = randi(numel(noise) - numel(cleanAudio), [1 1]);noiseSegment = noise(randind : randind + numel(cleanAudio) - 1);
向语音信号中添加噪声,使 SNR为0 dB
noisePower = sum(noiseSegment.^2);cleanPower = sum(cleanAudio.^2);noiseSegment = noiseSegment .* sqrt(cleanPower/noisePower);noisyAudio = cleanAudio + noiseSegment;
同样使用STFT从嘈杂的语音信号中生成幅值STFT向量
noisySTFT = stft(noisyAudio,'Window',win,'OverlapLength',overlap,'FFTLength',ffTLength);noisyPhase = angle(noisySTFT(numFeatures-1:end,:));noisySTFT = abs(noisySTFT(numFeatures-1:end,:));
同样从STFT 生成 8 段训练Predicter信号
noisySTFT = [noisySTFT(:,1:numSegments-1) noisySTFT];predictors = zeros( numFeatures, numSegments , size(noisySTFT,2) - numSegments + 1);for index = 1:(size(noisySTFT,2) - numSegments + 1) predictors(:,:,index) = noisySTFT(:,index:index + numSegments - 1); end
通过在训练阶段计算的均值和标准差对Predicter进行归一化。
predictors(:) = (predictors(:) - noisyMean) / noisyStd;
通过对两个经过训练的网络计算降噪幅值STFT
predictors = reshape(predictors, [numFeatures,numSegments,1,size(predictors,3)]);STFTFullyConnected = predict(denoiseNetFullyConnected, predictors);STFTFullyConvolutional = predict(denoiseNetFullyConvolutional, predictors);
通过训练阶段使用的平均值和标准差来恢复输出
STFTFullyConnected(:) = cleanStd * STFTFullyConnected(:) + cleanMean;STFTFullyConvolutional(:) = cleanStd * STFTFullyConvolutional(:) + cleanMean;
将单边STFT 转换为“居中的” STFT,这在信号处理中很好理解
STFTFullyConnected = STFTFullyConnected.' .* exp(1j*noisyPhase);STFTFullyConnected = [conj(STFTFullyConnected(end-1:-1:2,:)); STFTFullyConnected];STFTFullyConvolutional = squeeze(STFTFullyConvolutional) .* exp(1j*noisyPhase);STFTFullyConvolutional = [conj(STFTFullyConvolutional(end-1:-1:2,:)) ; STFTFullyConvolutional];
计算降噪语音信号。 istft 执行逆 STFT,使用带噪声STFT相位的相位来重建时域信号
denoisedAudioFullyConnected = istft(STFTFullyConnected, ... 'Window',win,'OverlapLength',overlap, ... 'FFTLength',ffTLength,'ConjugateSymmetric',true); denoisedAudioFullyConvolutional = istft(STFTFullyConvolutional, ... 'Window',win,'OverlapLength',overlap, ... 'FFTLength',ffTLength,'ConjugateSymmetric',true);
绘制干净、嘈杂和降噪后的音频信号
t = (1/fs) * (0:numel(denoisedAudioFullyConnected)-1);figuresubplot(4,1,1)plot(t,cleanAudio(1:numel(denoisedAudioFullyConnected)))title("Clean Speech")grid onsubplot(4,1,2)plot(t,noisyAudio(1:numel(denoisedAudioFullyConnected)))title("Noisy Speech")grid onsubplot(4,1,3)plot(t,denoisedAudioFullyConnected)title("Denoised Speech (Fully Connected Layers)")grid onsubplot(4,1,4)plot(t,denoisedAudioFullyConvolutional)title("Denoised Speech (Convolutional Layers)")grid onxlabel("Time (s)")
绘制干净、嘈杂和降噪后的频谱图。
h = figure;subplot(4,1,1)spectrogram(cleanAudio,win,overlap,ffTLength,fs);title("Clean Speech")grid onsubplot(4,1,2)spectrogram(noisyAudio,win,overlap,ffTLength,fs);title("Noisy Speech")grid onsubplot(4,1,3)spectrogram(denoisedAudioFullyConnected,win,overlap,ffTLength,fs);title("Denoised Speech (Fully Connected Layers)")grid onsubplot(4,1,4)spectrogram(denoisedAudioFullyConvolutional,win,overlap,ffTLength,fs);title("Denoised Speech (Convolutional Layers)")grid onp = get(h,'Position');set(h,'Position',[p(1) 65 p(3) 800]);
sound(noisyAudio,fs)
播放全连接层神经网络降噪后的语音信号
sound(denoisedAudioFullyConnected,fs)
播放卷积层神经网络降噪后的语音信号
sound(denoisedAudioFullyConvolutional,fs)
播放干净的语音信号
sound(cleanAudio,fs)
测试更多文件,且生成时域图和频域图,同时返回干净、加噪和降噪后的语音信号
[cleanAudio,noisyAudio,denoisedAudioFullyConnected,denoisedAudioFullyConvolutional] = testDenoisingNets(adsTest,denoiseNetFullyConnected,denoiseNetFullyConvolutional,noisyMean,noisyStd,cleanMean,cleanStd);
训练非常耗时,必须要配备matlab并行工具箱和GPU
另外,此方法可迁移至其他的一维信号,比如微震信号,机械振动信号,心电信号等,但要特别注意所加噪声信号的相位问题。
参考文献
[1] "Experiments on Deep Learning for Speech Denoising", Ding Liu, Paris Smaragdis, Minje Kim, INTERSPEECH, 2014.
[2] "A Fully Convolutional Neural Network for Speech Enhancement", Se Rim Park, Jin Won Lee, INTERSPEECH, 2017.
标签: #matlab降噪信号处理