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comparison spandsp-0.0.3/spandsp-0.0.3/tests/plc_tests.c @ 5:f762bf195c4b

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import spandsp-0.0.3
author Peter Meerwald <pmeerw@cosy.sbg.ac.at>
date Fri, 25 Jun 2010 16:00:21 +0200
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4:26cd8f1ef0b1 5:f762bf195c4b
1 /*
2 * SpanDSP - a series of DSP components for telephony
3 *
4 * plc_tests.c
5 *
6 * Written by Steve Underwood <steveu@coppice.org>
7 *
8 * Copyright (C) 2004 Steve Underwood
9 *
10 * All rights reserved.
11 *
12 * This program is free software; you can redistribute it and/or modify
13 * it under the terms of the GNU General Public License version 2, as
14 * published by the Free Software Foundation.
15 *
16 * This program is distributed in the hope that it will be useful,
17 * but WITHOUT ANY WARRANTY; without even the implied warranty of
18 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
19 * GNU General Public License for more details.
20 *
21 * You should have received a copy of the GNU General Public License
22 * along with this program; if not, write to the Free Software
23 * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
24 *
25 * $Id: plc_tests.c,v 1.18 2006/11/19 14:07:27 steveu Exp $
26 */
27
28 /*! \page plc_tests_page Packet loss concealment tests
29 \section plc_tests_page_sec_1 What does it do?
30 These tests run a speech file through the packet loss concealment routines.
31 The loss rate, in percent, and the packet size, in samples, may be specified
32 on the command line.
33
34 \section plc_tests_page_sec_2 How are the tests run?
35 These tests process a speech file called pre_plc.wav. This file should contain
36 8000 sample/second 16 bits/sample linear audio. The tests read this file in
37 blocks, of a size specified on the command line. Some of these blocks are
38 dropped, to simulate packet loss. The rate of loss is also specified on the
39 command line. The PLC module is then used to reconstruct an acceptable
40 approximation to the original signal. The resulting audio is written to a new
41 audio file, called post_plc.wav. This file contains 8000 sample/second
42 16 bits/sample linear audio.
43 */
44
45 #ifdef HAVE_CONFIG_H
46 #include "config.h"
47 #endif
48
49 #include <stdio.h>
50 #include <inttypes.h>
51 #include <stdlib.h>
52 #include <string.h>
53 #if defined(HAVE_TGMATH_H)
54 #include <tgmath.h>
55 #endif
56 #if defined(HAVE_MATH_H)
57 #include <math.h>
58 #endif
59 #include <tiffio.h>
60
61 #include <audiofile.h>
62
63 #include "spandsp.h"
64
65 #define INPUT_FILE_NAME "../localtests/short_nb_voice.wav"
66 #define OUTPUT_FILE_NAME "post_plc.wav"
67
68 int main(int argc, char *argv[])
69 {
70 AFfilehandle inhandle;
71 AFfilehandle outhandle;
72 AFfilesetup filesetup;
73 plc_state_t plc;
74 int inframes;
75 int outframes;
76 int16_t amp[1024];
77 int block_no;
78 int lost_blocks;
79 int block_len;
80 int loss_rate;
81 int dropit;
82 int block_real;
83 int block_synthetic;
84 int tone;
85 int i;
86 uint32_t phase_acc;
87 int32_t phase_rate;
88
89 loss_rate = 25;
90 block_len = 160;
91 block_real = FALSE;
92 block_synthetic = FALSE;
93 tone = -1;
94 for (i = 1; i < argc; i++)
95 {
96 if (strcmp(argv[i], "-l") == 0)
97 {
98 loss_rate = atoi(argv[++i]);
99 continue;
100 }
101 if (strcmp(argv[i], "-b") == 0)
102 {
103 block_len = atoi(argv[++i]);
104 continue;
105 }
106 if (strcmp(argv[i], "-t") == 0)
107 {
108 tone = atoi(argv[++i]);
109 continue;
110 }
111 if (strcmp(argv[i], "-r") == 0)
112 block_real = TRUE;
113 if (strcmp(argv[i], "-s") == 0)
114 block_synthetic = TRUE;
115 }
116 if ((filesetup = afNewFileSetup()) == AF_NULL_FILESETUP)
117 {
118 fprintf(stderr, " Failed to create file setup\n");
119 exit(2);
120 }
121 afInitSampleFormat(filesetup, AF_DEFAULT_TRACK, AF_SAMPFMT_TWOSCOMP, 16);
122 afInitRate(filesetup, AF_DEFAULT_TRACK, (float) SAMPLE_RATE);
123 afInitFileFormat(filesetup, AF_FILE_WAVE);
124 afInitChannels(filesetup, AF_DEFAULT_TRACK, 1);
125
126 phase_rate = 0;
127 inhandle = NULL;
128 if (tone < 0)
129 {
130 if ((inhandle = afOpenFile(INPUT_FILE_NAME, "r", NULL)) == AF_NULL_FILEHANDLE)
131 {
132 fprintf(stderr, " Failed to open wave file '%s'\n", INPUT_FILE_NAME);
133 exit(2);
134 }
135 }
136 else
137 {
138 phase_rate = dds_phase_ratef((float) tone);
139 }
140 if ((outhandle = afOpenFile(OUTPUT_FILE_NAME, "w", filesetup)) == AF_NULL_FILEHANDLE)
141 {
142 fprintf(stderr, " Failed to open wave file '%s'\n", OUTPUT_FILE_NAME);
143 exit(2);
144 }
145 plc_init(&plc);
146 lost_blocks = 0;
147 for (block_no = 0; ; block_no++)
148 {
149 if (tone < 0)
150 {
151 inframes = afReadFrames(inhandle,
152 AF_DEFAULT_TRACK,
153 amp,
154 block_len);
155 if (inframes != block_len)
156 break;
157 }
158 else
159 {
160 if (block_no > 10000)
161 break;
162 for (i = 0; i < block_len; i++)
163 amp[i] = (int16_t) dds_modf(&phase_acc, phase_rate, 10000.0, 0);
164 inframes = block_len;
165 }
166 dropit = rand()/(RAND_MAX/100);
167 if (dropit > loss_rate)
168 {
169 plc_rx(&plc, amp, inframes);
170 if (block_real)
171 memset(amp, 0, sizeof(int16_t)*inframes);
172 }
173 else
174 {
175 lost_blocks++;
176 plc_fillin(&plc, amp, inframes);
177 if (block_synthetic)
178 memset(amp, 0, sizeof(int16_t)*inframes);
179 }
180 outframes = afWriteFrames(outhandle,
181 AF_DEFAULT_TRACK,
182 amp,
183 inframes);
184 if (outframes != inframes)
185 {
186 fprintf(stderr, " Error writing out sound\n");
187 exit(2);
188 }
189 }
190 printf("Dropped %d of %d blocks\n", lost_blocks, block_no);
191 if (tone < 0)
192 {
193 if (afCloseFile(inhandle) != 0)
194 {
195 fprintf(stderr, " Cannot close wave file '%s'\n", INPUT_FILE_NAME);
196 exit(2);
197 }
198 }
199 if (afCloseFile(outhandle) != 0)
200 {
201 fprintf(stderr, " Cannot close wave file '%s'\n", OUTPUT_FILE_NAME);
202 exit(2);
203 }
204 afFreeFileSetup(filesetup);
205 return 0;
206 }
207 /*- End of function --------------------------------------------------------*/
208 /*- End of file ------------------------------------------------------------*/

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