view toolboxes/MIRtoolbox1.3.2/AuditoryToolbox/synlpc.m @ 0:e9a9cd732c1e tip

first hg version after svn

function synWave = synlpc(aCoeff,source,sr,G,fr,fs,preemp)
% USAGE: synWave = synlpc(aCoeff,source,sr,G,fr,fs,preemp);
%
% This function synthesizes a (speech) signal based on a LPC (linear-
% predictive coding) model of the signal.  The LPC coefficients are a 
% short-time measure of the speech signal which describe the signal as the 
% output of an all-pole filter.  This all-pole filter provides a good 
% description of the speech articulators; thus LPC analysis is often used in 
% speech recognition and speech coding systems.  The LPC analysis is done
% using the proclpc routine.  This routine can be used to verify that the 
% LPC analysis produces the correct answer, or as a synthesis stage after
% first modifying the LPC model.
%
% The results of LPC analysis are a new representation of the signal
%	s(n) = G e(n) - sum from 1 to L a(i)s(n-i)
% where s(n) is the original data.  a(i) and e(n) are the outputs of the LPC 
% analysis with a(i) representing the LPC model. The e(n) term represents 
% either the speech source's excitation, or the residual: the details of the 
% signal that are not captured by the LPC coefficients.  The G factor is a
% gain term.
%
% LPC synthesis produces a monaural sound vector (synWave) which is 
% sampled at a sampling rate of "sr".  The following parameters are mandatory
% aCoeff - The LPC analysis results, a(i).  One column of L+1 numbers for each
%	frame of data.  The number of rows of aCoeff determines L.
% source - The LPC residual, e(n).  One column of sr*fs samples representing
%	the excitation or residual of the LPC filter.
% G - The LPC gain for each frame.
%
% The following parameters are optional and default to the indicated values.
% fr - Frame time increment, in ms.  The LPC analysis is done starting every
% 	fr ms in time.  Defaults to 20ms (50 LPC vectors a second)
% fs - Frame size in ms.  The LPC analysis is done by windowing the speech
% 	data with a rectangular window that is fs ms long.  Defaults to 30ms
% preemp - This variable is the epsilon in a digital one-zero filter which 
%	serves to preemphasize the speech signal and compensate for the 6dB
%	per octave rolloff in the radiation function.  Defaults to .9378.
%
% This code was graciously provided by:
% 	Delores Etter (University of Colorado, Boulder) and 
%	Professor Geoffrey Orsak (Southern Methodist University) 
% It was first published in
%	Orsak, G.C. et al. "Collaborative SP education using the Internet and
%	MATLAB" IEEE SIGNAL PROCESSING MAGAZINE  Nov. 1995. vol.12, no.6, pp.
%	23-32.
% Modified and debugging plots added by Kate Nguyen and Malcolm Slaney

% (c) 1998 Interval Research Corporation  
% A more complete set of routines for LPC analysis can be found at
%	http://www.ee.ic.ac.uk/hp/staff/dmb/voicebox/voicebox.html
   


if (nargin < 5), fr = 20; end;
if (nargin < 6), fs = 30; end;
if (nargin < 7), preemp = .9378; end;

msfs = round(sr*fs/1000);
msfr = round(sr*fr/1000);
msoverlap = msfs - msfr;
ramp = [0:1/(msoverlap-1):1]';
[L1 nframe] = size(aCoeff); 			% L1 = 1+number of LPC coeffs 

[row col] = size(source);
if(row==1 | col==1)				% continous stream; must be 
						% windowed
  postFilter = 0; duration = length(source); frameIndex = 1;
  for sampleIndex=1:msfr:duration-msfs+1
    resid(:,frameIndex) = source(sampleIndex:(sampleIndex+msfs-1))';
    frameIndex = frameIndex+1;    
  end
else
  postFilter = 1; resid = source;
end

[row col] = size(resid);
%if ~(col==nframe)
%  error('synLPC: numbers of LPC frames and source frames do not match');
if col<nframe  
  nframe=col;
end

for frameIndex=1:nframe
 						% Calculate the filter response
						% by evaluating the z-transform
%  if 1
%    gain=0;
%    cft=0:(1/255):1;
%    for index=1:L1-1
%      gain = gain + aCoeff(index,frameIndex)*exp(-i*2*pi*cft).^index;
%    end
%    gain = abs(1./gain);
%    spec(:,frameIndex) = 20*log10(gain(1:128))';
    % plot(20*log10(gain));
    % title(frameIndex);
    % drawnow;
%  end

						% Calculate the filter response
						% from the filter's impulse
						% response (to check above).
%  if 0
%    impulseResponse = filter(1, aCoeff(:,frameIndex), [1 zeros(1,255)]);
%    freqResp = 20*log10(abs(fft(impulseResponse)));
%    plot(freqResp);
%  end


  A = aCoeff(:,frameIndex);
  residFrame = resid(:,frameIndex)*G(frameIndex);
  synFrame = filter(1, A', residFrame);		% synthesize speech from LPC 
						% coeffs
  if(frameIndex==1)				% add synthesize frames using a
    synWave = synFrame(1:msfr);			% trapezoidal window
  else
    synWave = [synWave; overlap+synFrame(1:msoverlap).*ramp; ...
			synFrame(msoverlap+1:msfr)];
  end
  if(frameIndex==nframe)
    synWave = [synWave; synFrame(msfr+1:msfs)];
  else
    overlap = synFrame(msfr+1:msfs).*flipud(ramp);	
  end	
  %length(synWave)						
end;

if(postFilter)
  synWave = filter(1, [1 -preemp], synWave);
end