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