File: comp_dgt_walnut.m

package info (click to toggle)
octave-ltfat 2.3.1%2Bdfsg-8
  • links: PTS, VCS
  • area: main
  • in suites: bullseye
  • size: 11,712 kB
  • sloc: ansic: 30,379; cpp: 8,808; java: 1,499; objc: 345; makefile: 248; xml: 182; python: 124; sh: 18; javascript: 12
file content (184 lines) | stat: -rw-r--r-- 4,301 bytes parent folder | download | duplicates (2)
1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64
65
66
67
68
69
70
71
72
73
74
75
76
77
78
79
80
81
82
83
84
85
86
87
88
89
90
91
92
93
94
95
96
97
98
99
100
101
102
103
104
105
106
107
108
109
110
111
112
113
114
115
116
117
118
119
120
121
122
123
124
125
126
127
128
129
130
131
132
133
134
135
136
137
138
139
140
141
142
143
144
145
146
147
148
149
150
151
152
153
154
155
156
157
158
159
160
161
162
163
164
165
166
167
168
169
170
171
172
173
174
175
176
177
178
179
180
181
182
183
184
function cout=comp_dgt_walnut(f,gf,a,M)
%-*- texinfo -*-
%@deftypefn {Function} comp_dgt_walnut
%@verbatim
%COMP_DGT_WALNUT  First step of full-window factorization of a Gabor matrix.
%   Usage:  c=comp_dgt_walnut(f,gf,a,M);
%
%   Input parameters:
%         f      : Factored input data
%         gf     : Factorization of window (from facgabm).
%         a      : Length of time shift.
%         M      : Number of channels.
%   Output parameters:
%         c      : M x N*W*R array of coefficients, where N=L/a
%
%   Do not call this function directly, use DGT instead.
%   This function does not check input parameters!
%
%   The length of f and gamma must match.
%
%   If input is a matrix, the transformation is applied to
%   each column.
%
%   This function does not handle the multidim case. Take care before
%   calling this.
%
%   References:
%     T. Strohmer. Numerical algorithms for discrete Gabor expansions. In
%     H. G. Feichtinger and T. Strohmer, editors, Gabor Analysis and
%     Algorithms, chapter 8, pages 267--294. Birkhauser, Boston, 1998.
%     
%     P. L. Soendergaard. An efficient algorithm for the discrete Gabor
%     transform using full length windows. IEEE Signal Process. Letters,
%     submitted for publication, 2007.
%     
%@end verbatim
%@strong{Url}: @url{http://ltfat.github.io/doc/comp/comp_dgt_walnut.html}
%@end deftypefn

% Copyright (C) 2005-2016 Peter L. Soendergaard <peter@sonderport.dk>.
% This file is part of LTFAT version 2.3.1
%
% This program is free software: you can redistribute it and/or modify
% it under the terms of the GNU General Public License as published by
% the Free Software Foundation, either version 3 of the License, or
% (at your option) any later version.
%
% This program is distributed in the hope that it will be useful,
% but WITHOUT ANY WARRANTY; without even the implied warranty of
% MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
% GNU General Public License for more details.
%
% You should have received a copy of the GNU General Public License
% along with this program.  If not, see <http://www.gnu.org/licenses/>.

%   AUTHOR : Peter L. Soendergaard.
%   TESTING: OK
%   REFERENCE: OK

L=size(f,1);
W=size(f,2);
LR=numel(gf);
R=LR/L;

N=L/a;

[c,h_a,h_m]=gcd(a,M);
h_a=-h_a;
p=a/c;
q=M/c;
d=N/q;

ff=zeros(p,q*W,c,d,assert_classname(f,gf));

if p==1
  % --- integer oversampling ---

  if (c==1) && (d==1) && (W==1) && (R==1)
    % --- Short time Fourier transform of single signal ---
    % This is used for spectrograms of short signals.      
      ff(1,:,1,1)=f(:);
  else
    for s=0:d-1
      for r=0:c-1    
	for l=0:q-1
	  ff(1,l+1:q:W*q,r+1,s+1)=f(r+s*M+l*c+1,:);
	end;
      end;
    end;    
  end;

else
  % --- rational oversampling ---
  % Set up the small matrices
  % The r-loop (runs up to c) has been vectorized
  for w=0:W-1
    for s=0:d-1
      for l=0:q-1
	for k=0:p-1	    	  
	  ff(k+1,l+1+w*q,:,s+1)=f((1:c)+mod(k*M+s*p*M-l*h_a*a,L),w+1);
	end;
      end;
    end;
  end;
end;

% This version uses matrix-vector products and ffts

% fft them
if d>1
  ff=fft(ff,[],4);
end;

C=zeros(q*R,q*W,c,d,assert_classname(f,gf));

for r=0:c-1    
  for s=0:d-1
    GM=reshape(gf(:,r+s*c+1),p,q*R);
    FM=reshape(ff(:,:,r+1,s+1),p,q*W);
    
    C(:,:,r+1,s+1)=GM'*FM;
  end;
end;

% Inverse fft
if d>1
  C=ifft(C,[],4);
end;

% Place the result

cout=zeros(M,N,R,W,assert_classname(f,gf));

if p==1
  % --- integer oversampling ---

  if (c==1) && (d==1) && (W==1) && (R==1)
    
    % --- Short time Fourier transform of single signal ---
    % This is used for spectrograms of short signals.      
    for l=0:q-1
      cout(l+1,mod((0:q-1)+l,N)+1,1,1)=C(:,l+1,1,1);
    end;
    
  else

    % The r-loop (runs up to c) has been vectorized
    for rw=0:R-1
      for w=0:W-1    
	for s=0:d-1
	  for l=0:q-1
	    for u=0:q-1
	      cout((1:c)+l*c,mod(u+s*q+l,N)+1,rw+1,w+1)=C(u+1+rw*q,l+1+w*q,:,s+1);
	    end;
	  end;
	end;
      end; 
    end;
  end;

else

  % Rational oversampling
  % The r-loop (runs up to c) has been vectorized
  for rw=0:R-1
    for w=0:W-1    
      for s=0:d-1
	for l=0:q-1
	  for u=0:q-1
	    cout((1:c)+l*c,mod(u+s*q-l*h_a,N)+1,rw+1,w+1)=C(u+1+rw*q,l+1+w*q,:,s+1);
	  end;
	end;
      end;
    end; 
  end;

end;

cout=reshape(cout,M,N*W*R);