updated beat histogram computation and replace matlab spectrogram fun…

…ction

ComputeBeatHisto.m

@@ -7,15 +7,15 @@
 %>
 %> @param afAudioData: time domain sample data, dimension channels X samples
 %> @param f_s: sample rate of audio data
-%> @param method: preferred method of beat histogram computation (default: Corr), can be [] empty
+%> @param cMethod: method of beat histogram computation (default: 'FFT')
 %> @param afWindow: FFT window of length iBlockLength (default: hann), can be [] empty
 %> @param iBlockLength: internal block length (default: 4096 samples)
 %> @param iHopLength: internal hop length (default: 512 samples)
 %>
 %> @retval T: Beat histogram
 %> @retval Bpm: tempo axis
 % ======================================================================
-function [T, Bpm] = ComputeBeatHisto (afAudioData, f_s, method, afWindow, iBlockLength, iHopLength)
+function [T, Bpm] = ComputeBeatHisto (afAudioData, f_s, cMethod, afWindow, iBlockLength, iHopLength)
 
     % set default parameters if necessary
     if (nargin < 6)
@@ -28,7 +28,7 @@
         afWindow    = hann(iBlockLength,'periodic');
     end
     if (nargin < 3)
-        method    = 'FFT';
+        cMethod    = 'FFT';
     end
     % compute FFT window function
     if (length(afWindow) ~= iBlockLength)
@@ -39,14 +39,14 @@
     afAudioData = ToolDownmix(afAudioData);
 
     % novelty function
-    [d,t,peaks] = ComputeNoveltyFunction (  'Flux', ...
+    [d,t, G_T,peaks] = ComputeNoveltyFunction (  'Flux', ...
                                             afAudioData, ...
                                             f_s, ...
                                             afWindow, ...
                                             iBlockLength, ...
                                             iHopLength);
 
-    if strcmp(method,'Corr')
+    if strcmp(cMethod,'Corr')
         % compute autocorrelation
         afCorr  = xcorr(d,'coeff');
         afCorr  = afCorr((ceil((length(afCorr)/2))+1):end);
@@ -59,15 +59,14 @@
         if length(d)< 2*iLength
             d = [d zeros(1,2*iLength-length(d))];
         end
-        [X,f,tf] = spectrogram( d, ...
+        [X,f,tf] = ComputeSpectrogram( d, ...
+                                f_s, ...
                                 [hann(iLength); zeros(iLength,1)], ...
-                                iLength-iHopLength, ...
                                 2*iLength, ...
-                                f_s);
+                                iLength/4 );
 
         % adjust output BPM range
         T       = mean(abs(X),2);
-        T(1:8)  = 0;
         Bpm     = f*60;
         lIdx    = max(find(Bpm < 30));
         hIdx    = min(find(Bpm > 200));

ComputeSpectrogram.m

@@ -0,0 +1,61 @@
+% ======================================================================
+%> @brief computes a mel spectrogram from the audio data
+%>
+%> @param x: time domain sample data, dimension channels X samples
+%> @param f_s: sample rate of audio data
+%> @param afWindow: FFT window of length iBlockLength (default: hann), can be [] empty
+%> @param iBlockLength: internal block length (default: 4096 samples)
+%> @param iHopLength: internal hop length (default: 2048 samples)
+%>
+%> @retval X spectrogram
+%> @retval f frequency bands
+%> @retval t time stamps
+% ======================================================================
+function [X, f, t] = ComputeSpectrogram (x, f_s, afWindow, iBlockLength, iHopLength, bNormalize)
+
+    % set default parameters if necessary
+    if (nargin < 6)
+        bNormalize = true;
+    end
+    if (nargin < 5)
+        iHopLength = 2048;
+    end
+    if (nargin < 4)
+        iBlockLength = 4096;
+    end
+    if (nargin < 3 || isempty(afWindow))
+        afWindow = hann(iBlockLength,'periodic');
+    end
+
+    if (length(afWindow) ~= iBlockLength)
+        error('window length mismatch');
+    end
+
+    if (size(afWindow,1) < size(afWindow,2))
+        afWindow = afWindow';
+    end
+    if (size(x,1) < size(x,2))
+        x = x';
+    end
+
+    % pre-processing: down-mixing
+    x = ToolDownmix(x);
+
+    % pre-processing: normalization 
+    if bNormalize
+        x = ToolNormalizeAudio(x);
+    end
+
+    [x_b, t] = ToolBlockAudio (x, iBlockLength, iHopLength, f_s);
+
+    X = zeros(size(x_b,2)/2+1, size(x_b,1));
+    f = linspace(0,(size(X,1)-1), f_s/2);
+
+    for n=1:size(X,2)
+        tmp = fft(x_b(n,:)' .* afWindow);
+        X(:,n) = abs(tmp(1:size(X,1))) * 2 / iBlockLength;
+    end
+
+    % normalization
+    X([1 end],:)= X([1 end],:)/sqrt(2);
+end

ToolBlockAudio.m

@@ -0,0 +1,27 @@
+% ======================================================================
+%> @brief blocks audio signal into overlapping blocks
+%>
+%> @param x: audio signal (dimension length x 1)
+%> @param iBlockLength: target block size
+%> @param iHopLength: target hopsize
+%> @param f_s: sample rate
+%>
+%> @retval x_b (dimension iNumOfBlocks x iBlockLength)
+%> @retval t time stamps for blocks
+% ======================================================================
+function [x_b, t] = ToolBlockAudio(x, iBlockLength, iHopLength, f_s)
+
+    iNumBlocks = ceil(size(x,1) / iHopLength );
+
+    % time stamp vector
+    t = (0:(iNumBlocks-1)) * iHopLength / f_s;
+
+    % pad with block length zeros just to make sure it runs for weird inputs, too
+    xPadded = [x; zeros(iBlockLength+iHopLength, 1)];
+
+    x_b = zeros(iNumBlocks, iBlockLength);
+
+    for n=1:iNumBlocks
+        x_b(n,:) = xPadded((n-1)*iHopLength+1:(n-1)*iHopLength+iBlockLength);
+    end
+end

ToolNormalizeAudio.m

@@ -9,5 +9,7 @@
 
     if (length(x)> 1)
         x_norm = x/max(abs(x),[],'all');
+    else
+        x_norm = x;
     end
 end