ksbai123 commited on
Commit
796ff25
·
1 Parent(s): 6327bf7

Upload 7 files

Browse files
tools/utils/extract.py ADDED
@@ -0,0 +1,128 @@
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
1
+ #!/usr/bin/env python
2
+ """ extract.py
3
+
4
+ Extract CHiME 3 audio segments from continuous audio
5
+
6
+ Usage:
7
+ extract.py [-f] [-p pad] [-c channel] <segfilenm> <inwavroot> <outwavroot>
8
+ extract.py --help
9
+
10
+ Options:
11
+ <segfilenm> Name of the segmentation file
12
+ <inwavroot> Name of the root dir for the input audio file
13
+ <outwavroot> Name of the root dir for the output segments
14
+ -p <pad>, --padding=<pad> padding at start and end in seconds [default: 0]
15
+ -f, --fullname Use fullname for outfile
16
+ -c <chan>, --channel=<chan> Recording channel (defaults to all)
17
+ --help print this help screen
18
+
19
+ """
20
+
21
+ from __future__ import print_function
22
+ import json
23
+ import os
24
+ import subprocess
25
+ import argparse
26
+ import sys
27
+
28
+
29
+ def extract(segment, in_root, out_root, padding=0.0, channel=0, fullname=False):
30
+ """use sox to extract segment from wav file
31
+
32
+ in_root - root directory for unsegmented audio files
33
+ out_root - root directory for output audio segments
34
+ """
35
+ infilenm = '{}/{}.CH{}.wav'.format(in_root, segment['wavfile'], channel)
36
+
37
+ if fullname:
38
+ outtemplate = '{}/{}.{}.{}.{}.{:02d}.{:03d}.ch{}.wav'
39
+ outfilenm = outtemplate.format(out_root,
40
+ segment['wavfile'],
41
+ segment['wsj_name'],
42
+ segment['environment'],
43
+ segment['speaker'],
44
+ segment['repeat'],
45
+ segment['index'],
46
+ channel)
47
+ else:
48
+ outfilenm = '{}/{}_{}_{}.CH{}.wav'.format(out_root,
49
+ segment['speaker'],
50
+ segment['wsj_name'],
51
+ segment['environment'],
52
+ channel)
53
+
54
+ subprocess.call(['sox', infilenm, outfilenm,
55
+ 'trim',
56
+ str(segment['start'] - padding),
57
+ '=' + str(segment['end'] + padding)])
58
+
59
+
60
+ def to_string(segment):
61
+ return "{}:{}-{}:{:03d}({:03d})".format(segment['wavfile'],
62
+ segment['start'],
63
+ segment['end'],
64
+ segment['index'],
65
+ segment['repeat'])
66
+
67
+
68
+ def do_extract(seg_filenm, in_root, out_root,
69
+ padding=0.0, channel=0, fullname=False):
70
+ """
71
+ Extract segments listed in seg file from recording channel, 'channel'
72
+ """
73
+
74
+ with open(seg_filenm, 'r') as infile:
75
+ json_string = infile.read()
76
+ segments = json.loads(json_string)
77
+
78
+ if not os.path.isdir(out_root):
79
+ os.makedirs(out_root)
80
+
81
+ print('Extracting audio in channel {}...'.format(channel))
82
+
83
+ for i, segment in enumerate(segments):
84
+ sys.stdout.write(' Processing segment {: 5}/{: <5}\r'.format(i+1, len(segments)))
85
+ sys.stdout.flush()
86
+
87
+ extract(segment, in_root, out_root, padding=padding,
88
+ channel=channel, fullname=fullname)
89
+ sys.stdout.write('\n')
90
+ sys.stdout.flush()
91
+
92
+
93
+ def main():
94
+ """Main method called from commandline."""
95
+ parser = argparse.ArgumentParser(description='Extract CHiME 3 audio segments from continuous audio.')
96
+ parser.add_argument('segfilenm', metavar='<segfilenm>',
97
+ help='Name of the segmentation file', type=str)
98
+ parser.add_argument('inwavroot', metavar='<inwavroot>',
99
+ help='Name of the root dir for the input audio file', type=str)
100
+ parser.add_argument('outwavroot', metavar='<outwavroot>',
101
+ help='Name of the root dir for the output segments', type=str)
102
+ parser.add_argument('-p', '--padding', metavar='pad',
103
+ help='Padding at start and end in seconds [default: 0]', type=float, default=0)
104
+ parser.add_argument('-f', '--fullname',
105
+ help='Use fullname for outfile', action='store_true')
106
+ parser.add_argument('-c', '--channel', metavar='channel',
107
+ help='Recording channel (defaults to all).', action='append', type=int, default=[])
108
+
109
+ args = parser.parse_args()
110
+
111
+ segfilenm = args.segfilenm
112
+ in_root = args.inwavroot
113
+ out_root = args.outwavroot
114
+ padding = args.padding
115
+ fullname = args.fullname
116
+ channels = args.channel
117
+
118
+ if len(channels) == 0:
119
+ channels = [0, 1, 2, 3, 4, 5, 6]
120
+
121
+ for channel in channels:
122
+ do_extract(segfilenm, in_root, out_root, padding, channel, fullname)
123
+
124
+
125
+ if __name__ == '__main__':
126
+ main()
127
+
128
+ # ./extract.py ../../data/annotations/utterance/LR_141103_01.json ../../data/16khz16bit xxx
tools/utils/istft_multi.m ADDED
@@ -0,0 +1,59 @@
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
1
+ function x=istft_multi(X,nsampl)
2
+
3
+ % ISTFT_MULTI Multichannel inverse short-time Fourier transform (ISTFT)
4
+ % using half-overlapping sine windows.
5
+ %
6
+ % x=istft_multi(X,nsampl)
7
+ %
8
+ % Inputs:
9
+ % X: nbin x nfram x nsrc matrix containing STFT coefficients for nsrc
10
+ % sources with nbin frequency bins and nfram time frames or nbin x nfram x
11
+ % nsrc x nchan matrix containing the STFT coefficients of nsrc spatial
12
+ % source images over nchan channels
13
+ % nsampl: number of samples to which the corresponding time-domain signals
14
+ % are to be truncated
15
+ %
16
+ % Output:
17
+ % x: nsrc x nsampl matrix or nsrc x nsampl x nchan matrix containing the
18
+ % corresponding time-domain signals
19
+ % If x is a set of signals of length nsampl and X=stft_multi(x), then
20
+ % x=istft_multi(X,nsampl).
21
+ %
22
+ %%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%
23
+ % Copyright 2008 Emmanuel Vincent
24
+ % This software is distributed under the terms of the GNU Public License
25
+ % version 3 (http://www.gnu.org/licenses/gpl.txt)
26
+ %%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%
27
+
28
+
29
+ %%% Errors and warnings %%%
30
+ if nargin<2, error('Not enough input arguments.'); end
31
+ [nbin,nfram,nsrc,nchan]=size(X);
32
+ if nbin==2*floor(nbin/2), error('The number of frequency bins must be odd.'); end
33
+ wlen=2*(nbin-1);
34
+
35
+ %%% Computing inverse STFT signal %%%
36
+ % Defining sine window
37
+ win=sin((.5:wlen-.5)/wlen*pi);
38
+ % Pre-processing for edges
39
+ swin=zeros(1,(nfram+1)*wlen/2);
40
+ for t=0:nfram-1,
41
+ swin(t*wlen/2+1:t*wlen/2+wlen)=swin(t*wlen/2+1:t*wlen/2+wlen)+win.^2;
42
+ end
43
+ swin=sqrt(swin/wlen);
44
+ x=zeros(nsrc,(nfram+1)*wlen/2,nchan);
45
+ for i=1:nchan,
46
+ for j=1:nsrc,
47
+ for t=0:nfram-1,
48
+ % IFFT
49
+ fframe=[X(:,t+1,j,i);conj(X(wlen/2:-1:2,t+1,j,i))];
50
+ frame=real(ifft(fframe));
51
+ % Overlap-add
52
+ x(j,t*wlen/2+1:t*wlen/2+wlen,i)=x(j,t*wlen/2+1:t*wlen/2+wlen,i)+frame.'.*win./swin(t*wlen/2+1:t*wlen/2+wlen);
53
+ end
54
+ end
55
+ end
56
+ % Truncation
57
+ x=x(:,wlen/4+1:wlen/4+nsampl,:);
58
+
59
+ return;
tools/utils/json2mat.m ADDED
@@ -0,0 +1,65 @@
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
1
+ function mat=json2mat(filename)
2
+
3
+ % JSON2MAT Reads a JSON file
4
+ %
5
+ % mat=json2mat(filename)
6
+ %
7
+ % Input:
8
+ % filename: JSON filename (.json extension)
9
+ %
10
+ % Output:
11
+ % mat: Matlab cell array whose entries are Matlab structures containing the
12
+ % value for each JSON field
13
+ %
14
+ % Note: all numeric fields are rounded to double precision. Digits beyond
15
+ % double precision are lost.
16
+ %
17
+ % If you use this software in a publication, please cite:
18
+ %
19
+ % Jon Barker, Ricard Marxer, Emmanuel Vincent, and Shinji Watanabe, The
20
+ % third 'CHiME' Speech Separation and Recognition Challenge: Dataset,
21
+ % task and baselines, submitted to IEEE 2015 Automatic Speech Recognition
22
+ % and Understanding Workshop (ASRU), 2015.
23
+ %
24
+ %%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%
25
+ % Copyright 2015 University of Sheffield (Jon Barker, Ricard Marxer)
26
+ % Inria (Emmanuel Vincent)
27
+ % Mitsubishi Electric Research Labs (Shinji Watanabe)
28
+ % This software is distributed under the terms of the GNU Public License
29
+ % version 3 (http://www.gnu.org/licenses/gpl.txt)
30
+ %%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%
31
+
32
+ fid=fopen(filename,'r');
33
+ fgetl(fid); % [
34
+ txt=fgetl(fid); % { or ]
35
+ txt=fgetl(fid); % first field
36
+ mat=cell(1);
37
+ ind=1; % entry index
38
+ while txt~=-1, % end of file
39
+ if strcmp(txt,' }, ') || strcmp(txt,' }'), % next entry
40
+ ind=ind+1;
41
+ txt=fgetl(fid); % { or ]
42
+ else
43
+ try
44
+ pos=strfind(txt,'"');
45
+ field=txt(pos(1)+1:pos(2)-1);
46
+ catch
47
+ keyboard;
48
+ end
49
+ if ~strcmp(txt(end-1:end),', '), % last field
50
+ txt=txt(pos(2)+3:end);
51
+ else
52
+ txt=txt(pos(2)+3:end-2);
53
+ end
54
+ if strcmp(txt(1),'"') && strcmp(txt(end),'"'), % text value
55
+ value=txt(2:end-1);
56
+ else % boolean or numerical value
57
+ value=eval(txt);
58
+ end
59
+ mat{ind}.(field)=value;
60
+ end
61
+ txt=fgetl(fid); % next field
62
+ end
63
+ fclose(fid);
64
+
65
+ return
tools/utils/localize.m ADDED
@@ -0,0 +1,158 @@
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
1
+ function [path,TDOA]=localize(X,chanlist)
2
+
3
+ % LOCALIZE Tracks the speaker spatial position over time and computes the
4
+ % corresponding TDOA using SRP-PHAT and the Viterbi algorithm
5
+ %
6
+ % [path,TDOA]=localize(X,chanlist)
7
+ %
8
+ % Inputs:
9
+ % Y: nbin x nfram x nchan STFT of the inpu signal
10
+ % chanlist: list of input channels (from 1 to 6)
11
+ %
12
+ % Output:
13
+ % path: 3 x nfram position of the speaker over time in centimeters
14
+ % TDOA: nchan x nfram corresponding TDOAs between the speaker position and
15
+ % the microphone positions
16
+ %
17
+ % Note: for computational efficiency, the position on the z-axis is assumed
18
+ % to be constant over time.
19
+ %
20
+ % If you use this software in a publication, please cite:
21
+ %
22
+ % Jon Barker, Ricard Marxer, Emmanuel Vincent, and Shinji Watanabe, The
23
+ % third 'CHiME' Speech Separation and Recognition Challenge: Dataset,
24
+ % task and baselines, submitted to IEEE 2015 Automatic Speech Recognition
25
+ % and Understanding Workshop (ASRU), 2015.
26
+ %
27
+ %%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%
28
+ % Copyright 2015-2016 University of Sheffield (Jon Barker, Ricard Marxer)
29
+ % Inria (Emmanuel Vincent)
30
+ % Mitsubishi Electric Research Labs (Shinji Watanabe)
31
+ % This software is distributed under the terms of the GNU Public License
32
+ % version 3 (http://www.gnu.org/licenses/gpl.txt)
33
+ %%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%
34
+
35
+ if nargin < 2,
36
+ chanlist=[1 3:6];
37
+ end
38
+
39
+ % Define hyper-parameters
40
+ pow_thresh=-20; % threshold in dB below which a microphone is considered to fail
41
+ center_factor=0.05; % weight given to the prior that the speaker's horizontal position is close to the center
42
+ smoothing_factor=3; % weight given to the transition probabilities
43
+
44
+ % Remove zero frequency
45
+ X = X(2:end,:,:);
46
+ [nbin,nfram,nchan] = size(X);
47
+ wlen=2*nbin;
48
+ f=16000/wlen*(1:nbin).';
49
+
50
+ % Compute relative channel power
51
+ if length(chanlist) > 2,
52
+ xpow=shiftdim(sum(sum(abs(X).^2,2),1));
53
+ xpow=10*log10(xpow/max(xpow));
54
+ else
55
+ xpow=zeros(1,2);
56
+ end
57
+
58
+ % Define microphone positions in centimeters
59
+ xmic=[-10 0 10 -10 0 10]; % left to right axis
60
+ ymic=[9.5 9.5 9.5 -9.5 -9.5 -9.5]; % bottom to top axis
61
+ zmic=[0 -2 0 0 0 0]; % back to front axis
62
+ xmic=xmic(chanlist);
63
+ ymic=ymic(chanlist);
64
+ zmic=zmic(chanlist);
65
+
66
+ % Define grid of possible speaker positions in centimeters
67
+ xres=46;
68
+ xpos=linspace(-45,45,xres);
69
+ yres=46;
70
+ ypos=linspace(-45,45,yres);
71
+ zres=4;
72
+ zpos=linspace(15,45,zres);
73
+ ngrid=xres*yres*zres;
74
+
75
+ % Compute horizontal distances between grid points
76
+ xvect=reshape(repmat(xpos.',[1 yres]),xres*yres,1);
77
+ yvect=reshape(repmat(ypos,[xres 1]),xres*yres,1);
78
+ pair_dist=sqrt((repmat(xvect,[1 xres*yres])-repmat(xvect.',[xres*yres 1])).^2+(repmat(yvect,[1 xres*yres])-repmat(yvect.',[xres*yres 1])).^2);
79
+
80
+ % Compute horizontal distances to the center
81
+ center_dist=sqrt((xvect-mean(xpos)).^2+(yvect-mean(ypos)).^2);
82
+
83
+ % Compute theoretical TDOAs between front pairs
84
+ d_grid=zeros(nchan,xres,yres,zres); % speaker-to-microphone distances
85
+ for c=1:nchan,
86
+ d_grid(c,:,:,:)=sqrt(repmat((xpos.'-xmic(c)).^2,[1 yres zres])+repmat((ypos-ymic(c)).^2,[xres 1 zres])+repmat((permute(zpos,[3 1 2])-zmic(c)).^2,[xres yres 1]));
87
+ end
88
+ d_grid=reshape(d_grid,nchan,ngrid);
89
+ pairs=[];
90
+ for c=1:nchan,
91
+ pairs=[pairs [c*ones(1,nchan-c); c+1:nchan]]; % microphone pairs
92
+ end
93
+ npairs=size(pairs,2);
94
+ tau_grid=zeros(npairs,ngrid); % TDOAs
95
+ for p=1:npairs,
96
+ c1=pairs(1,p);
97
+ c2=pairs(2,p);
98
+ tau_grid(p,:)=(d_grid(c2,:)-d_grid(c1,:))/343/100;
99
+ end
100
+
101
+ % Compute the SRP-PHAT pseudo-spectrum
102
+ srp=zeros(nfram,ngrid);
103
+ for p=1:npairs, % Loop over front pairs
104
+ c1=pairs(1,p);
105
+ c2=pairs(2,p);
106
+ d=sqrt((xmic(c1)-xmic(c2))^2+(ymic(c1)-ymic(c2))^2+(zmic(c1)-zmic(c2))^2);
107
+ alpha=10*343/(d*16000);
108
+ lin_grid=linspace(min(tau_grid(p,:)),max(tau_grid(p,:)),100);
109
+ lin_spec=zeros(nbin,nfram,100); % GCC-PHAT pseudo-spectrum over a uniform interval
110
+ if (xpow(c1)>pow_thresh) && (xpow(c2)>pow_thresh), % discard channels with low power (microphone failure)
111
+ P=X(:,:,c1).*conj(X(:,:,c2));
112
+ P=P./abs(P);
113
+ for ind=1:100,
114
+ EXP=repmat(exp(-2*1i*pi*lin_grid(ind)*f),1,nfram);
115
+ lin_spec(:,:,ind)=ones(nbin,nfram)-tanh(alpha*real(sqrt(2-2*real(P.*EXP))));
116
+ end
117
+ end
118
+ lin_spec=shiftdim(sum(lin_spec,1));
119
+ tau_spec=zeros(nfram,ngrid); % GCC-PHAT pseudo-spectrum over the whole grid
120
+ for t=1:nfram,
121
+ tau_spec(t,:)=interp1(lin_grid,lin_spec(t,:),tau_grid(p,:));
122
+ end
123
+ srp=srp+tau_spec; % sum over the microphone pairs
124
+ end
125
+
126
+ % Loop over possible z-axis positions
127
+ path=zeros(zres,nfram);
128
+ logpost=zeros(zres,1);
129
+ xpath=zeros(zres,nfram);
130
+ ypath=zeros(zres,nfram);
131
+ zpath=zeros(zres,nfram);
132
+ srp=reshape(srp,nfram,xres*yres,zres);
133
+ for zind=1:zres,
134
+
135
+ % Weight by distance to the center
136
+ weighted_srp=srp(:,:,zind)-center_factor*repmat(center_dist.',[nfram 1]);
137
+
138
+ % Track the source position over time
139
+ [path(zind,:),logpost(zind)]=viterbi(weighted_srp.',zeros(xres*yres,1),zeros(xres*yres,1),-smoothing_factor*pair_dist);
140
+ for t=1:nfram,
141
+ [xpath(zind,t),ypath(zind,t)]=ind2sub([xres yres],path(zind,t));
142
+ zpath(zind,t)=zind;
143
+ end
144
+ end
145
+
146
+ % Select the best z-axis position
147
+ [~,zind]=max(logpost);
148
+ path=(zind-1)*xres*yres+path(zind,:);
149
+ xpath=xpos(xpath(zind,:));
150
+ ypath=ypos(ypath(zind,:));
151
+ zpath=zpos(zpath(zind,:));
152
+
153
+ % Derive TDOA
154
+ d_path=d_grid(:,path);
155
+ TDOA=d_path/343/100;
156
+ path=[xpath; ypath; zpath];
157
+
158
+ return
tools/utils/mat2json.m ADDED
@@ -0,0 +1,66 @@
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
1
+ function mat2json(mat,filename)
2
+
3
+ % MAT2JSON Writes a JSON file
4
+ %
5
+ % mat2json(mat,filename)
6
+ %
7
+ % Inputs:
8
+ % mat: Matlab cell array whose entries are Matlab structures containing the
9
+ % value for each JSON field
10
+ % filename: JSON filename (.json extension)
11
+ %
12
+ % Note: using JSON2MAT followed by MAT2JSON will generally not lead back to
13
+ % the original JSON file due to the loss of digits beyond double precision
14
+ % and to the handling of trailing zeros.
15
+ %
16
+ % If you use this software in a publication, please cite:
17
+ %
18
+ % Jon Barker, Ricard Marxer, Emmanuel Vincent, and Shinji Watanabe, The
19
+ % third 'CHiME' Speech Separation and Recognition Challenge: Dataset,
20
+ % task and baselines, submitted to IEEE 2015 Automatic Speech Recognition
21
+ % and Understanding Workshop (ASRU), 2015.
22
+ %
23
+ %%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%
24
+ % Copyright 2015 University of Sheffield (Jon Barker, Ricard Marxer)
25
+ % Inria (Emmanuel Vincent)
26
+ % Mitsubishi Electric Research Labs (Shinji Watanabe)
27
+ % This software is distributed under the terms of the GNU Public License
28
+ % version 3 (http://www.gnu.org/licenses/gpl.txt)
29
+ %%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%
30
+
31
+ fid=fopen(filename,'w');
32
+ fprintf(fid,'%s\n','[');
33
+ for ind=1:length(mat), % loop over entries
34
+ fprintf(fid,' %s\n','{'); % entry delimiter
35
+ fields=fieldnames(mat{ind});
36
+ for f=1:length(fields), % loop over fields
37
+ field=fields{f};
38
+ value=mat{ind}.(field);
39
+ if ischar(value), % text field
40
+ fprintf(fid,' "%s": "%s"',field,value);
41
+ elseif islogical(value), % boolean field
42
+ if value,
43
+ fprintf(fid,' "%s": true',field);
44
+ else
45
+ fprintf(fid,' "%s": false',field);
46
+ end
47
+ elseif value==floor(value), % integer field
48
+ fprintf(fid,' "%s": %d',field,value);
49
+ else % double field
50
+ fprintf(fid,' "%s": %17.*f',field,15-max(0,floor(log10(value))),value);
51
+ end
52
+ if f~=length(fields), % field delimiter
53
+ fprintf(fid,', ');
54
+ end
55
+ fprintf(fid,'\n');
56
+ end
57
+ fprintf(fid,' }'); % entry delimiter
58
+ if ind~=length(mat),
59
+ fprintf(fid,', ');
60
+ end
61
+ fprintf(fid,'\n');
62
+ end
63
+ fprintf(fid,']');
64
+ fclose(fid);
65
+
66
+ return
tools/utils/stft_multi.m ADDED
@@ -0,0 +1,58 @@
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
1
+ function X=stft_multi(x,wlen)
2
+
3
+ % STFT_MULTI Multichannel short-time Fourier transform (STFT) using
4
+ % half-overlapping sine windows.
5
+ %
6
+ % X=stft_multi(x)
7
+ % X=stft_multi(x,wlen)
8
+ %
9
+ % Inputs:
10
+ % x: nchan x nsampl matrix containing nchan time-domain mixture signals
11
+ % with nsampl samples
12
+ % wlen: window length (default: 1024 samples or 64ms at 16 kHz, which is
13
+ % optimal for speech source separation via binary time-frequency masking)
14
+ %
15
+ % Output:
16
+ % X: nbin x nfram x nchan matrix containing the STFT coefficients with nbin
17
+ % frequency bins and nfram time frames
18
+ %
19
+ %%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%
20
+ % Copyright 2008 Emmanuel Vincent
21
+ % This software is distributed under the terms of the GNU Public License
22
+ % version 3 (http://www.gnu.org/licenses/gpl.txt)
23
+ %%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%
24
+
25
+
26
+ %%% Errors and warnings %%%
27
+ if nargin<1, error('Not enough input arguments.'); end
28
+ if nargin<2, wlen=1024; end
29
+ [nchan,nsampl]=size(x);
30
+ if nchan>nsampl, error('The signals must be within rows.'); end
31
+ if wlen~=4*floor(wlen/4), error('The window length must be a multiple of 4.'); end
32
+
33
+ %%% Computing STFT coefficients %%%
34
+ % Defining sine window
35
+ win=sin((.5:wlen-.5)/wlen*pi).';
36
+ % Zero-padding
37
+ nfram=ceil(nsampl/wlen*2);
38
+ x=[x,zeros(nchan,nfram*wlen/2-nsampl)];
39
+ % Pre-processing for edges
40
+ x=[zeros(nchan,wlen/4),x,zeros(nchan,wlen/4)];
41
+ swin=zeros((nfram+1)*wlen/2,1);
42
+ for t=0:nfram-1,
43
+ swin(t*wlen/2+1:t*wlen/2+wlen)=swin(t*wlen/2+1:t*wlen/2+wlen)+win.^2;
44
+ end
45
+ swin=sqrt(wlen*swin);
46
+ nbin=wlen/2+1;
47
+ X=zeros(nbin,nfram,nchan);
48
+ for i=1:nchan,
49
+ for t=0:nfram-1,
50
+ % Framing
51
+ frame=x(i,t*wlen/2+1:t*wlen/2+wlen).'.*win./swin(t*wlen/2+1:t*wlen/2+wlen);
52
+ % FFT
53
+ fframe=fft(frame);
54
+ X(:,t+1,i)=fframe(1:nbin);
55
+ end
56
+ end
57
+
58
+ return;
tools/utils/viterbi.m ADDED
@@ -0,0 +1,45 @@
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
 
1
+ function [path,logpost]=viterbi(loglik,loginitp,logfinalp,logtransp)
2
+
3
+ % VITERBI Viterbi algorithm
4
+ %
5
+ % path=viterbi(loglik,loginitp,logfinalp,logtransp)
6
+ %
7
+ % Inputs:
8
+ % loglik: nstates x nfram matrix of log-likelihood values
9
+ % loginitp: nstates x 1 vector of initial log-probability values
10
+ % logfinalp: nstates x 1 vector of final log-probability values
11
+ % logtransp: nstates x nstates matrix of transition log-probabilities
12
+ %
13
+ % Output:
14
+ % path: 1 x nfram best state sequence
15
+ % logpost: log-posterior probability of the best state sequence
16
+ %
17
+ %%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%
18
+ % Copyright 2015 University of Sheffield (Jon Barker, Ricard Marxer)
19
+ % Inria (Emmanuel Vincent)
20
+ % Mitsubishi Electric Research Labs (Shinji Watanabe)
21
+ % This software is distributed under the terms of the GNU Public License
22
+ % version 3 (http://www.gnu.org/licenses/gpl.txt)
23
+ %%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%
24
+
25
+ [nstates,nfram]=size(loglik);
26
+
27
+ % Forward pass
28
+ logalpha=loglik(:,1)+loginitp;
29
+ prev=zeros(nstates,nfram-1);
30
+ for t=2:nfram,
31
+ logalphaprev=logalpha;
32
+ for n=1:nstates,
33
+ [logalpha(n),prev(n,t-1)]=max(logalphaprev+logtransp(:,n));
34
+ end
35
+ logalpha=logalpha+loglik(:,t);
36
+ end
37
+
38
+ % Backward pass
39
+ path=zeros(1,nfram);
40
+ [logpost,path(nfram)]=max(logalpha+logfinalp);
41
+ for t=nfram-1:-1:1,
42
+ path(t)=prev(path(t+1),t);
43
+ end
44
+
45
+ return