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