diff toolboxes/MIRtoolbox1.3.2/AuditoryToolbox/MakeVowel.m @ 0:e9a9cd732c1e tip

first hg version after svn

--- /dev/null	Thu Jan 01 00:00:00 1970 +0000
+++ b/toolboxes/MIRtoolbox1.3.2/AuditoryToolbox/MakeVowel.m	Tue Feb 10 15:05:51 2015 +0000
@@ -0,0 +1,130 @@
+function y=MakeVowel(len, pitch, sampleRate, f1, f2, f3)
+%  MakeVowel(len, pitch [, sampleRate, f1, f2, f3]) - Make a vowel with
+%    "len" samples and the given pitch.  The sample rate defaults to
+%    be 22254.545454 Hz (the native Mactinosh Sampling Rate).  The
+%    formant frequencies are f1, f2 & f3.  Some common vowels are
+%               Vowel       f1      f2      f3
+%                /a/        730    1090    2440
+%                /i/        270    2290    3010
+%                /u/        300     870    2240
+%
+% The pitch variable can either be a scalar indicating the actual
+%      pitch frequency, or an array of impulse locations. Using an
+%      array of impulses allows this routine to compute vowels with
+%      varying pitch.
+%
+% Alternatively, f1 can be replaced with one of the following strings
+%      'a', 'i', 'u' and the appropriate formant frequencies are
+%      automatically selected.
+%  Modified by R. Duda, 3/13/94
+
+% (c) 1998 Interval Research Corporation  
+
+if nargin < 2,
+   fprintf('Format: y = MakeVowel(len, pitch [, sampleRate, f1, f2, f3])\n');
+   return;
+end;
+
+if nargin < 6; f3 = 0; end;
+if nargin < 5; f2 = 0; end;
+if nargin < 4,
+   f1 = 0;
+else
+    if isstr(f1)
+        if f1 == 'a' | f1 == '/a/'
+               f1=730; f2=1090; f3=2440;
+        elseif f1 == 'i' | f1 == '/i/'
+               f1=270; f2=2290; f3=3010;
+        elseif f1 == 'u' | f1 == '/u/'
+               f1=300; f2=870; f3=2240;
+        end
+   end;
+end;
+
+if nargin < 3,
+   sampleRate = 22254.545454;
+elseif sampleRate < 1000,         % Apparently for test purposes
+   sampleRate = 22254.545454;
+end;
+
+%  GlottalPulses(pitch, fs, len) - Generate a stream of
+%    glottal pulses with the given pitch (in Hz) and sampling
+%    frequency (sampleRate).  A vector of the requested length is returned.
+y=zeros(1,len);
+if length(pitch) > 1,            % If true, use to determine points
+   points=pitch;                 % Check for valid sequence of points
+   if any(points~=sort(points)),
+      error('Values in pitch array must be in ascending order.')
+   end;
+   if points(1) < 1,
+      error('Values in pitch array cannot be less than 1.');
+   end;
+   kmax=sum(points <= len);
+   if kmax == 0,
+      error('All values in pitch array exceed "len"; none should.');
+   elseif kmax < length(points),
+      fprintf('Some values in pitch array exceed "len"; truncating.\n');
+      points=points(1:kmax);
+   end;
+else
+    points=1:sampleRate/pitch:len;
+end;
+indices=floor(points);
+
+%  Use a triangular approximation to an impulse function.  The important
+%  part is to keep the total amplitude the same.
+y(indices) = (indices+1)-points;
+y(indices+1) = points-indices;
+
+%  GlottalFilter(x,fs) - Filter an impulse train and simulate the glottal
+%    transfer function.  The sampling interval (sampleRate) is given in Hz.
+%    The filtering performed by this function is two first-order filters
+%    at 250Hz.
+a = exp(-250*2*pi/sampleRate);
+%y=filter([1,0,-1],[1,-2*a,a*a],y);      %  Not as good as one below....
+y=filter([1],[1,0,-a*a],y);
+
+%  FormantFilter(input, f, fs) - Filter an input sequence to model one
+%    formant in a speech signal.  The formant frequency (in Hz) is given
+%    by f and the bandwidth of the formant is a constant 50Hz.  The
+%    sampling frequency in Hz is given by fs.
+if f1 > 0
+        cft = f1/sampleRate;
+        bw = 50;
+        q = f1/bw;
+        rho = exp(-pi * cft / q);
+        theta = 2 * pi * cft * sqrt(1-1/(4 * q*q));
+        a2 = -2*rho*cos(theta);
+        a3 = rho*rho;
+        y=filter([1+a2+a3],[1,a2,a3],y);
+end;
+
+%  FormantFilter(input, f, fs) - Filter an input sequence to model one
+%    formant in a speech signal.  The formant frequency (in Hz) is given
+%    by f and the bandwidth of the formant is a constant 50Hz.  The
+%    sampling frequency in Hz is given by fs.
+if f2 > 0
+        cft = f2/sampleRate;
+        bw = 50;
+        q = f2/bw;
+        rho = exp(-pi * cft / q);
+        theta = 2 * pi * cft * sqrt(1-1/(4 * q*q));
+        a2 = -2*rho*cos(theta);
+        a3 = rho*rho;
+        y=filter([1+a2+a3],[1,a2,a3],y);
+end;
+
+%  FormantFilter(input, f, fs) - Filter an input sequence to model one
+%    formant in a speech signal.  The formant frequency (in Hz) is given
+%    by f and the bandwidth of the formant is a constant 50Hz.  The
+%    sampling frequency in Hz is given by fs.
+if f3 > 0
+        cft = f3/sampleRate;
+        bw = 50;
+        q = f3/bw;
+        rho = exp(-pi * cft / q);
+        theta = 2 * pi * cft * sqrt(1-1/(4 * q*q));
+        a2 = -2*rho*cos(theta);
+        a3 = rho*rho;
+        y=filter([1+a2+a3],[1,a2,a3],y);
+end;