added headphone equilization back

applications/Binaural-HRTF/HpTF_class/ita_headphone_equalization.m

+function output = ita_headphone_equalization(HPTF,type)
+
+
+% <ITA-Toolbox>
+% This file is part of the application HRTF_Measurement for the ITA-Toolbox. All rights reserved.
+% You can find the license for this m-file in the application folder.
+% </ITA-Toolbox>
+
+output = HPTF(1,1);
+for cdx = 1:HPTF(1,1).nChannels
+    
+    hp = merge(HPTF.ch(cdx));
+    
+    switch type
+        case 'mean'
+            hp.freqData = mean(abs(hp.freqData),2);
+        case 'max'
+            hp.freqData = max(abs(hp.freqData),[],2);
+        case 'mSTD'
+            hp.freqData = mean(abs(hp.freqData),2) + 2*std(abs(hp.freqData),0,2);
+        otherwise
+            error('Unknown type');
+    end
+    
+    %% Short Filter with no correction for low freqs and minimun phase
+    R = hp;
+    % R = ke4;
+    n = R.nSamples;
+    aux = max(abs(R.freqData),[],2);
+    
+    % find first maximum and truncate low freq correction at this point
+    idx1 = find(R.freqVector > 100,1,'first');
+    idx2 = find(R.freqVector < 300,1,'last');
+    d_aux = diff(aux(idx1:idx2));
+    idx = find(diff(sign(d_aux)) ~= 0,1,'first');
+    aux(1:idx1+idx+1) = aux(idx1+idx+2);
+    aux(aux==0) = rand(1)*eps;
+    R.freqData = aux;
+    
+    % do smoothing
+    %R = ita_smooth(R,'LogFreqOctave1',1/6,'Abs');
+    R = ita_invert_spk_regularization(R,[0 18000],'beta',.01);
+    
+    % minimum phase
+    R = ita_time_shift(R,n/2,'samples');
+    R = ita_extend_dat(R,R.nSamples*2);
+    R = ita_time_shift(R,-n/2,'samples');
+    N = 2*n;
+    aux = R.timeData;
+    R.timeData = ifft(log(abs(fft(aux))));
+    T = 2^16;
+    if T >= N-3
+        T = N-3;
+    end
+    if rem(T-1,2) == 1
+        T = T+1;
+    end
+    u = [1; 2*ones((N-T-1)/2,1); cos(pi*(0:T-1)'/T)+1; zeros((N-T-1)/2,1)]; %pode colocar uma transicao mais suave no meio.
+    R.timeData = R.timeData.*u;
+    H = R;
+    H.timeData = ifft(exp(fft(R.timeData)),'symmetric');
+    H = ita_time_crop(H,[1 n],'samples');
+    % H = ita_time_window(H,[2^12 2^13],'samples');
+    % H = ita_time_window(H,[0.1 0.2],'time','dc',true);
+    H = ita_time_window(H,[2^9 2^10],'samples');
+    H.channelUnits = {''};
+    % It is necessary to correct the overall level of the filter
+    % we can either guaranty no gain
+    % H = H/max(abs(H.freqData));
+    
+       
+%     spl = ita_spk2level(H,3,'averaged');
+%     
+%     H = H/mean(spl.data);
+
+    output.freqData(:,cdx) = H.freqData;
+end
+
+% or guaranty that the average level of the signal is not altered
+% this also means that the overall loudness of the signal will not be
+% considerably altered.
+% output = output/norm(output.rms);
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