【语音增强】Coherence-based语音反混响【含Matlab源码 1026期】
2021/7/21 14:42:04
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一、简介
对语音信号进行基于相干性的去混响
二、源代码
%DEMO_CDR_DEREVERB % % Demonstration of CDR-based noise and reverberation suppression. %演示CDR-based噪声和混响抑制 % To use this with your own recordings: % 1. Change wave filename % 2. Adapt microphone spacing (cfg.d) % 2. Adapt cfg.TDOA, or use the DOA-independent estimator (estimate_cdr_nodoa) % % Reference: % Andreas Schwarz, Walter Kellermann, "Coherent-to-Diffuse Power Ratio % Estimation for Dereverberation", IEEE/ACM Trans. on Audio, Speech and % Lang. Proc., 2015 (under review); preprint available: arXiv:1502.03784 % PDF: http://arxiv.org/pdf/1502.03784 % % Andreas Schwarz (schwarz@lnt.de) % Multimedia Communications and Signal Processing % Friedrich-Alexander-Universitaet Erlangen-Nuernberg (FAU) % Cauerstr. 7, 91058 Erlangen, Germany addpath(genpath('lib')); %% filterbank initialization cfg.K = 512; % FFT size cfg.N = 128; % frame shift cfg.Lp = 1024; % prototype filter length %p=IterLSDesign(cfg.Lp,cfg.K,cfg.N); load('lib/filterbank/prototype_K512_N128_Lp1024.mat'); %% algorithm and scenario configuration cfg.fs = 16000; % sampling rate [Hz] cfg.c = 342; % speed of sound [m/s] cfg.d_mic = 0.08; % mic spacing [m] % all estimators except estimate_cdr_nodoa require the TDOA of the signal; make sure % to adapt this when loading another wave file cfg.TDOA = 2.15e-04; % ground truth for wav/roomC-2m-75deg.wav cfg.nr.lambda = 0.68; % smoothing factor for PSD estimation cfg.nr.mu = 1.3; % noise overestimation factor cfg.nr.floor = 0.1; % minimum gain %cfg.nr.alpha = 1; cfg.nr.beta = 1; % power subtraction cfg.nr.alpha = 2; cfg.nr.beta = 0.5; % magnitude subtraction %cfg.nr.alpha = 2; cfg.nr.beta = 1; % Wiener filter %cfg.estimator = @estimate_cdr_unbiased; % unbiased estimator (CDRprop1) cfg.estimator = @estimate_cdr_robust_unbiased; % unbiased, "robust" estimator (CDRprop2) %cfg.estimator = @estimate_cdr_nodoa; % DOA-independent estimator (CDRprop3) %cfg.estimator = @estimate_cdr_nodiffuse; % noise coherence-independent estimator (CDRprop4; does not work for TDOA -> 0!) %% preparation [x,fs_in] = audioread('wav/roomC-2m-75deg.wav'); x = resample(x,cfg.fs,fs_in); %% Signal processing % The algorithm itself is real-time capable, i.e., no processing of the entire % utterance is necessary. Here however, for efficiency of the MATLAB implementation, % the entire signal is processed at once. fprintf('Performing signal enhancement... ');tic; % analysis filterbank X=DFTAnaRealEntireSignal(x,cfg.K,cfg.N,p); % estimate PSD and coherence Pxx = estimate_psd(X,cfg.nr.lambda); Cxx = estimate_cpsd(X(:,:,1),X(:,:,2),cfg.nr.lambda)./sqrt(Pxx(:,:,1).*Pxx(:,:,2)); frequency = linspace(0,cfg.fs/2,cfg.K/2+1)'; % frequency axis % define coherence models Css = exp(1j * 2 * pi * frequency * cfg.TDOA); % target signal coherence; not required for estimate_cdr_nodoa Cnn = sinc(2 * frequency * cfg.d_mic/cfg.c); % diffuse noise coherence; not required for estimate_cdr_nodiffuse % apply CDR estimator (=SNR) weights = min(weights,1); % postfilter input is computed from averaged PSDs of both microphones Postfilter_input = sqrt(mean(abs(X).^2,3)) .* exp(1j*angle(X(:,:,1))); % apply postfilter Processed = weights .* Postfilter_input; % synthesis filterbank y = DFTSynRealEntireSignal(Processed,cfg.K,cfg.N,p); fprintf('done (%.2fs).\n', toc); %% output % write output file audiowrite('wav/out.wav',y,cfg.fs); %% visualization figure(1) subplot(211) imagesc(10*log10(SNR)) set(gca,'YDir','normal') caxis([-15 15]) colorbar title('Estimated CDR (=SNR) [dB]') xlabel('frame index') ylabel('subband index') subplot(212) imagesc(weights) set(gca,'YDir','normal') caxis([0 1]) colorbar title('Filter gain') xlabel('frame index') ylabel('subband index')
三、运行结果
四、备注
版本:2014a
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