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Compare Speaker Separation Models

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Compare the performance, size, and speed of deep learning speaker separation models.

Introduction

Speaker separation is a challenging and critical speech processing task. Modern speaker separation methods use deep learning to achieve strong results. In this example, you compare four speaker separation models:

  • Time-frequency masking model [2]

  • Conv-TasNet model trained to discriminate two speakers [1]

  • Conv-TasNet model trained to discriminate one speaker and put the "rest" in a separate channel. This type of model is known as "one-and-rest". [1] [5]

  • SepFormer model trained to discriminate two speakers [4]

You can find the training recipes for the time-frequency masking model and the 2-speaker ConvTas-Net model in Cocktail Party Source Separation Using Deep Learning Networks (Audio Toolbox) and Train End-to-End Speaker Separation Model (Audio Toolbox), respectively. You can perform speaker separation using the one-and-rest Conv-TasNet (Conv-TasNet OR) model and the SepFormer model with the separateSpeakers (Audio Toolbox) function.

Load Test Data

To spot-check model performance, load test data consisting of two speakers and their mix. Listen to the speakers individually and mixed. Plot the mix and individual speakers using the supporting function, plotSpeakerSeparation.

[audioIn,fs] = audioread("MultipleSpeakers-16-8-4channel-5secs.flac");
t1 = audioIn(:,2);
t2 = audioIn(:,3);
x = t1 + t2;
x = x/max(abs(x));

plotSpeakerSeparation(t1,t2,x)

sound(t1,fs),pause(5)
sound(t2,fs),pause(5)
sound(x,fs),pause(5)

Load Models

Time-Frequency Masking

Load the pretrained speaker separation weights for the time-frequency masking model. The inference model is defined in the supporting function separateSpeakersTimeFrequency. To examine and train this model, see Cocktail Party Source Separation Using Deep Learning Networks (Audio Toolbox).

downloadFolder = matlab.internal.examples.downloadSupportFile("audio/examples","cocktailpartyfc.zip");
dataFolder = tempdir;
tfNetFolder = fullfile(dataFolder,"CocktailPartySourceSeparation");
unzip(downloadFolder,tfNetFolder)

Separate the mixed test signal and then plot and listen to the results.

y = separateSpeakersTimeFrequency(x,tfNetFolder);

plotSpeakerSeparation(t1,t2,x,y)

sound(y(:,1),fs),pause(5)
sound(y(:,2),fs),pause(5)

Conv-TasNet

Load the pretrained speaker separation weights for the Conv-TasNet model. The inference model is defined in the supporting function separateSpeakersConvTasNet. To examine and train this model, see Train End-to-End Speaker Separation Model (Audio Toolbox).

downloadFolder = matlab.internal.examples.downloadSupportFile("audio","speechSeparation.zip");
dataFolder = tempdir;
unzip(downloadFolder,dataFolder)
convtasNetFolder = fullfile(dataFolder,"speechSeparation");

Separate the mixed test signal and then plot and listen to the results.

y = separateSpeakersConvTasNet(x,convtasNetFolder);

plotSpeakerSeparation(t1,t2,x,y);

sound(y(:,1),fs),pause(5)
sound(y(:,2),fs),pause(5)

Conv-TasNet with One-and-Rest Permutation Invariant Training

The separateSpeakers (Audio Toolbox) function uses three models under-the-hood: a 2-speaker SepFormer model, a 3-speaker SepFormer model, and a one-and-rest Conv-TasNet model. To use the one-and-rest Conv-TasNet model, specify NumSpeakers as 1 or do not specify the NumSpeakers. When NumSpeakers is not specified, the function passes the "rest" from the separation back through the model until no more speakers are detected. For the purposes of this example, call separateSpeakers twice with NumSpeakers=1 for both calls.

Separate the mixed test signal and then plot and listen to the results. If you have not downloaded the required files to use separateSpeakers, an error is thrown with the link to the download.

[y1,r] = separateSpeakers(x,fs,NumSpeakers=1);
[y2,r] = separateSpeakers(r,fs,NumSpeakers=1);

plotSpeakerSeparation(t1,t2,x,[y1,y2])

sound(y1,fs),pause(5)
sound(y2,fs),pause(5)

SepFormer

Call separateSpeakers (Audio Toolbox) with NumSpeakers=2 to perform speaker separation using the 2-speaker SepFormer model.

Separate the mixed test signal and then plot and listen to the results.

y = separateSpeakers(x,fs,NumSpeakers=2);

plotSpeakerSeparation(t1,t2,x,y)

sound(y(:,1),fs),pause(5)
sound(y(:,2),fs),pause(5)

Compare Models

Compare the computation time, model size, and performance of the models.

Computation Time

To compare execution times for different duration inputs, use the supporting function compareComputationTime. If the execution time is less than the input duration, then the model can run in real time (without dropping samples).

compareComputationTime(DurationToTest=[1,5,10], ...
    CompareCPU=true, ...
    CompareGPU=true, ...
    TimeFrequencyMaskNetFolder=tfNetFolder, ...
    ConvTasNetFolder=convtasNetFolder)

Model Size

Compare the size of all models. Note that the Conv-TasNet model trained in the example and the Conv-TasNet OR model provided with the separateSpeakers (Audio Toolbox) function are quite different in size. In addition to different loss functions and training recipes, Conv-TasNet OR and Conv-TasNet are both variations on the architecture described in [1]. Most noticeably, the Conv-TasNet OR model uses 24 convolutional blocks while the example Conv-TasNet model uses 32.

timefrequency_size = dir(fullfile(tfNetFolder,"CocktailPartyNet.mat")).bytes/1e6;
convtasnet_size = dir(fullfile(convtasNetFolder,"paramsBest.mat")).bytes/1e6;
convtasnet_or_size = dir(which("convtasnet-librimix-orpit.mat")).bytes/1e6;
sepformer_size = dir(which("sepformer-libri2mix-upit.mat")).bytes/1e6;

n = categorical(["Time-Frequency Mask","Conv-TasNet","Conv-TasNet OR","SepFormer"]);
n = reordercats(n,["Time-Frequency Mask","Conv-TasNet","Conv-TasNet OR","SepFormer"]);

figure
bar(n,[timefrequency_size,convtasnet_size,convtasnet_or_size,sepformer_size])
grid on
ylabel("Size (MB)")
title("Disk Memory")

Compare Scale Invariant Signal to Noise Ratio (SNR)

To compare model performance, download the LibriSpeech [3] test-clean dataset. The dataset consists of files of single speakers reading.

downloadDatasetFolder = tempdir;
datasetFolder = fullfile(downloadDatasetFolder,"LibriSpeech","test-clean");
filename = "test-clean.tar.gz";
url = "http://www.openSLR.org/resources/12/" + filename;
if ~datasetExists(datasetFolder)
    gunzip(url,downloadDatasetFolder);
    unzippedFile = fullfile(downloadDatasetFolder,filename);
    untar(unzippedFile{1}(1:end-3),downloadDatasetFolder);
end

ads = audioDatastore(datasetFolder,IncludeSubfolders=true);

Test the model scale-invariant signal-to-noise ratio (SI-SNR) [6] performances on a sampling of the dataset. SI-SNR is a popular objective metric for the quality of speaker separation algorithms. If a GPU and Parallel Computing Toolbox™ are available, use the GPU to speed up processing.

The testModel supporting function combines randomly selected audio files, mixes them, passes the mixed data through the specified model, and then calculates the permutation-invariant SI-SNR.

The SepFormer model achieves the best results (higher SNR is better).

tf_sisnr = testModel(ads,@(x)separateSpeakersTimeFrequency(x,tfNetFolder),UseGPU=canUseGPU);
convtasnet_sisnr = testModel(ads,@(x)separateSpeakersConvTasNet(x,convtasNetFolder),UseGPU=canUseGPU);
convtasnet_orpit_sisnr = testModel(ads,@(x)separateSpeakers(x,8e3,NumSpeakers=1),UseGPU=canUseGPU,OneAndRest=true);
sepformer_sisnr = testModel(ads,@(x)separateSpeakers(x,8e3,NumSpeakers=2),UseGPU=canUseGPU);

figure
bar(n,[tf_sisnr,convtasnet_sisnr,convtasnet_orpit_sisnr,sepformer_sisnr])
grid on
ylabel("SI-SNR")
title("Separation Performance (Test Dataset)")

Supporting Functions

Test Model

function testSISNR = testModel(ads,model,options)
%testModel Calculate SISNR over dataset

arguments
    ads
    model
    options.OneAndRest = false
    options.UseGPU = false
    options.NumTestPoints = 50
    options.TestDuration = []
    options.SignalRatio = [0.6 0.75 0.85 1]
end
total_sisnr = zeros(options.NumTestPoints,1);

fn = ads.Files;
spkids = filenames2labels(fn,ExtractBefore="-");

rng default
for ii = 1:options.NumTestPoints
    % Choose a random file for speaker 1
    idx1 = randi(numel(fn));
    fn1 = fn{idx1};

    % Choose another random file for speaker 2, exclude speaker 1 from
    % possibilities
    spk1 = spkids(idx1);
    fnsubset = fn(spkids~=spk1);
    idx2 = randi(numel(fnsubset));
    fn2 = fnsubset{idx2};

    % Read audio file and resample to 8 kHz.
    [t1,fs1] = audioread(fn1);
    [t2,fs2] = audioread(fn2);
    t1 = resample(t1,8e3,fs1);
    t2 = resample(t2,8e3,fs2);

    % Make files the same length
    if isempty(options.TestDuration)
        N = min(numel(t1),numel(t2));
    else
        N = 8e3*options.TestDuration;
    end
    t1 = trimOrPad(t1,N);
    t2 = trimOrPad(t2,N);

    % Scale audio to abs max of 1
    t1 = t1./max(abs(t1));
    t2 = t2./max(abs(t2));

    % Scale the second speaker according to the scaling ratio
    t2 = options.SignalRatio(randi(numel(options.SignalRatio)))*t2;

    % Mix
    x = t1 + t2;
    x = x./max(abs(x));

    % Use GPU if requested
    if options.UseGPU
        x = gpuArray(x);
    end

    % Pass signal through model
    if options.OneAndRest
        [y1,r] = model(x);
        y2 = model(r);
        y = [y1,y2];
    else
        y = model(x);
    end

    % Perform permutation invariant signal to noise ratio
    total_sisnr(ii) = permutationInvariantSISNR(y,[t1,t2]);

end
testSISNR = gather(mean(total_sisnr));
end

Plot Speaker Separation

function plotSpeakerSeparation(t1,t2,x,y)
%plotSpeakerSeparation Plot the ground truth and predictions

arguments
    t1
    t2
    x
    y = []
end
fs = 8e3;
timeVector = ((0:size(t1,1)-1)/fs)';

tiledlayout(3,1)

nexttile()
plot(timeVector,x)
xlabel("Time (s)")
ylabel("Mix")
grid on
xlim tight
ylim([-1 1])

% Match the targets and predictions based on which set of pairs results in
% the best SI-SNR
if ~isempty(y)
    [~,reorderidx] = permutationInvariantSISNR(y,[t1,t2]);
    y = y(:,reorderidx);
end

nexttile()
if ~isempty(y)
    plot(timeVector,t1,"-",timeVector,y(:,1),"--")
    legend("Target","Prediction")
else
    plot(timeVector,t1)
end
ylabel("Speaker 1",FontWeight="bold")
xlabel("Time (s)")
grid on
xlim tight
ylim([-1 1])

nexttile()
if ~isempty(y)
    plot(timeVector,t2,"-",timeVector,y(:,2),"--")
    legend("Target","Prediction",Location="best")
else
    plot(timeVector,t2)
end
ylabel("Speaker 2",FontWeight="bold")
xlabel("Time (s)")
grid on
xlim tight
ylim([-1 1])
end

Separate Speakers Using Time-Frequency Masking

function output = separateSpeakersTimeFrequency(mix,pathToNet)
%separateSpeakersTimeFrequency STFT-based speaker separation function

persistent CocktailPartyNet
if isempty(CocktailPartyNet)
    s = load(fullfile(pathToNet,"CocktailPartyNet.mat"));
    CocktailPartyNet = s.CocktailPartyNet;
end

WindowLength  = 128;
FFTLength = 128;
OverlapLength = 128-1;
win = hann(WindowLength,"periodic");

% Downsample to 4 kHz
mixR = resample(mix,1,2);

P0 = stft(mixR, ...
    Window=win, ...
    OverlapLength=OverlapLength,...
    FFTLength=FFTLength, ...
    FrequencyRange="onesided");
P = log(abs(P0) + eps);
MP = mean(P(:));
SP = std(P(:));
P = (P-MP)/SP;

seqLen = 20;
PSeq = zeros(1 + FFTLength/2,seqLen,1,0);
seqOverlap = seqLen;

loc = 1;
while loc < size(P,2)-seqLen
    PSeq(:,:,:,end+1) = P(:,loc:loc+seqLen-1); %#ok
    loc = loc + seqOverlap;
end

PSeq  = reshape(PSeq, [1 1 (1 + FFTLength/2)*seqLen size(PSeq,4)]);

estimatedMasks = predict(CocktailPartyNet,PSeq);

estimatedMasks = estimatedMasks.';
estimatedMasks = reshape(estimatedMasks,1 + FFTLength/2,numel(estimatedMasks)/(1 + FFTLength/2));

mask1 = estimatedMasks; 
mask2 = 1 - mask1;

P0 = P0(:,1:size(mask1,2));

P_speaker1 = P0.*mask1;

speaker1 = istft(P_speaker1, ...
    Window=win, ...
    OverlapLength=OverlapLength,...
    FFTLength=FFTLength, ...
    ConjugateSymmetric=true,...
    FrequencyRange="onesided");
speaker1 = speaker1 / max(abs(speaker1));

P_speaker2 = P0.*mask2;

speaker2 = istft(P_speaker2, ...
    Window=win, ...
    OverlapLength=OverlapLength,...
    FFTLength=FFTLength, ...
    ConjugateSymmetric=true,...
    FrequencyRange="onesided");
speaker2 = speaker2/max(speaker2);

speaker1 = resample(double(speaker1),2,1);
speaker2 = resample(double(speaker2),2,1);

N = numel(mix) - numel(speaker1);
mixToAdd = mix(end-N+1:end);

speaker1 = [speaker1;mixToAdd];
speaker2 = [speaker2;mixToAdd];

output = [speaker1,speaker2];
end

Separate Speakers Using uPIT ConvTasNet

function output = separateSpeakersConvTasNet(input,netFolder)
%separateSpeakersConvTasNet Separate two speaker signals from a mixture
% input using 2-speaker Conv-TasNet.

persistent learnables states
if isempty(learnables)
    M = load(fullfile(netFolder,"paramsBest.mat"));
    learnables = M.learnables;
    states = M.states;
end

if ~isdlarray(input)
    input = dlarray(input,"SCB");
end

x = dlconv(input,learnables.Conv1W,learnables.Conv1B,Stride=10);

x = relu(x);
x0 = x;

x = x - mean(x,2);
x = x./sqrt(mean(x.^2, 2) + 1e-5);
x = x.*learnables.ln_weight + learnables.ln_bias;

encoderOut = dlconv(x,learnables.Conv2W,learnables.Conv2B);

for index = 1:32
    encoderOut = convBlock(encoderOut,index-1,learnables.Blocks(index),states(index));
end

masks = dlconv(encoderOut,learnables.Conv3W,learnables.Conv3B);
masks = relu(masks);

mask1 = masks(:,1:256,:);
mask2 = masks(:,257:512,:);

out1 = x0.*mask1;
out2 = x0.*mask2;

weights = learnables.TransConv1W;
bias = learnables.TransConv1B;
output2 = dltranspconv(out1,weights,bias,Stride=10);
output1 = dltranspconv(out2,weights,bias,Stride=10);

output1 = gather(extractdata(output1));
output2 = gather(extractdata(output2));

output1 = output1./max(abs(output1));
output2 = output2./max(abs(output2));

output1 = trimOrPad(output1,numel(input));
output2 = trimOrPad(output2,numel(input));

output = [output1,output2];
end

ConvTasNet - Conv Block

function output = convBlock(input,count,learnables,state)
%convBlock - Convolutional block for ConvTasNet

% Conv:
conv1Out = dlconv(input,learnables.Conv1W,learnables.Conv1B);

% PRelu:
conv1Out = relu(conv1Out) - learnables.Prelu1.*relu(-conv1Out);

% BatchNormalization:
batchOut = batchnorm(conv1Out,learnables.BN1Offset,learnables.BN1Scale,state.BN1Mean,state.BN1Var);

% Conv:
padding = [1 1] * 2^(mod(count,8));
dilationFactor = 2^(mod(count,8));
convOut = dlconv(batchOut,learnables.Conv2W,learnables.Conv2B,DilationFactor=dilationFactor,Padding=padding);

% PRelu:
convOut = relu(convOut) - learnables.Prelu2.*relu(-convOut);

% BatchNormalization:
batchOut = batchnorm(convOut,learnables.BN2Offset,learnables.BN2Scale,state.BN2Mean,state.BN2Var);

% Conv:
output = dlconv(batchOut,learnables.Conv3W,learnables.Conv3B);

% Skip connection
output = output + input;

end

Trim or Pad Audio Signal to Desired Length

function y = trimOrPad(x,n)
%trimOrPad Trim or pad to desired length

% Trim or expand to match desired size
if size(x,1)>=n
    % Choose a random starting index such that you still have numSamples
    % after indexing the noise.
    start = randi(size(x,1) - n + 1);
    y = x(start:start+n-1);
else
    numReps = ceil(n/size(x,1));
    temp = repmat(x,numReps,1);
    start = randi(size(temp,1) - n + 1);
    y = temp(start:start+n-1);
end
 
end

Compare Execution Time

function compareComputationTime(options)
%compareComputationTime Compare computation time

arguments
    options.DurationToTest
    options.CompareCPU
    options.CompareGPU
    options.TimeFrequencyMaskNetFolder
    options.ConvTasNetFolder
end
fs = 8e3;
dur = options.DurationToTest;

if options.CompareCPU
    tf.CPU = zeros(numel(dur),1);
    convtas.CPU = zeros(numel(dur),1);
    convtas_orpit.CPU = zeros(numel(dur),1);
    sepformer.CPU = zeros(numel(dur),1);
    for ii = 1:numel(dur)
        x = pinknoise(dur(ii)*fs,"single");

        tf.CPU(ii) = timeit(@()separateSpeakersTimeFrequency(x,options.TimeFrequencyMaskNetFolder));
        convtas.CPU(ii) = timeit(@()separateSpeakersConvTasNet(x,options.ConvTasNetFolder));
        convtas_orpit.CPU(ii) = timeit(@()separateSpeakers(x,8e3,NumSpeakers=1,ConserveEnergy=false));
        sepformer.CPU(ii) = timeit(@()separateSpeakers(x,8e3,NumSpeakers=2,ConserveEnergy=false));
    end
    convtas_orpit.CPU = 2*convtas_orpit.CPU; % Double to adjust for two-passes of one-and-rest.
end
if options.CompareGPU
    tf.GPU = zeros(numel(dur),1);
    convtas.GPU = zeros(numel(dur),1);
    convtas_orpit.GPU = zeros(numel(dur),1);
    sepformer.GPU = zeros(numel(dur),1);
    for ii = 1:numel(dur)
        x = gpuArray(pinknoise(dur(ii)*fs,"single"));

        tf.GPU(ii) = gputimeit(@()separateSpeakersTimeFrequency(x,options.TimeFrequencyMaskNetFolder));
        convtas.GPU(ii) = gputimeit(@()separateSpeakersConvTasNet(x,options.ConvTasNetFolder));
        convtas_orpit.GPU(ii) = gputimeit(@()separateSpeakers(x,8e3,NumSpeakers=1,ConserveEnergy=false));
        sepformer.GPU(ii) = gputimeit(@()separateSpeakers(x,8e3,NumSpeakers=2,ConserveEnergy=false));
    end
    convtas_orpit.GPU = 2*convtas_orpit.GPU; % Double to adjust for two-passes of one-and-rest.
end

numTiles = double(options.CompareCPU)+double(options.CompareGPU);
tlh = tiledlayout(numTiles,1);

environments = ["CPU","GPU"];
environments = environments([options.CompareCPU,options.CompareGPU]);

for ii = 1:numel(environments)
    nexttile(tlh)
    ee = environments(ii);
    plot(dur,tf.(ee),'b',dur,convtas.(ee),'r',dur,convtas_orpit.(ee),'g',dur,sepformer.(ee),'k', ...
        dur,tf.(ee),'bo',dur,convtas.(ee),'ro',dur,convtas_orpit.(ee),'go',dur,sepformer.(ee),'ko')
    legend("Time-Frequency Mask","Conv-TasNet","Conv-TasNet OR","SepFormer",Location="best")
    xlabel("Input Duration (s)")
    ylabel("Execution Time (s)")
    title(ee + " Execution Time")
    grid on
end
end

References

[1] Luo, Yi, and Nima Mesgarani. "Conv-TasNet: Surpassing Ideal Time-Frequency Magnitude Masking for Speech Separation." IEEE/ACM Transactions on Audio, Speech, and Language Processing 27, no. 8 (August 2019): 1256-66. https://doi.org/10.1109/TASLP.2019.2915167.

[2] Simpson, Andrew J. R. "Probabilistic Binary-Mask Cocktail-Party Source Separation in a Convolutional Deep Neural Network." arXiv.org, March 24, 2015. https://arxiv.org/abs/1503.06962.

[3] Panayotov, Vassil, Guoguo Chen, Daniel Povey, and Sanjeev Khudanpur. "Librispeech: An ASR Corpus Based on Public Domain Audio Books." In 2015 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP), 5206-10. South Brisbane, Queensland, Australia: IEEE, 2015. https://doi.org/10.1109/ICASSP.2015.7178964.

[4] Subakan, Cem, Mirco Ravanelli, Samuele Cornell, Mirko Bronzi, and Jianyuan Zhong. "Attention Is All You Need In Speech Separation." In ICASSP 2021 - 2021 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP), 21-25. Toronto, ON, Canada: IEEE, 2021. https://doi.org/10.1109/ICASSP39728.2021.9413901.

[5] Takahashi, Naoya, Sudarsanam Parthasaarathy, Nabarun Goswami, and Yuki Mitsufuji. "Recursive Speech Separation for Unknown Number of Speakers." In Interspeech 2019, 1348-52. ISCA, 2019. https://doi.org/10.21437/Interspeech.2019-1550.

[6] Roux, Jonathan Le, et al. "SDR – Half-Baked or Well Done?" ICASSP 2019 - 2019 IEEE International Conference on Acoustics, Speech and Signal Processing (ICASSP), IEEE, 2019, pp. 626–30. DOI.org (Crossref), https://doi.org/10.1109/ICASSP.2019.8683855.

See Also

(Audio Toolbox)

Related Topics