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comparison spandsp-0.0.6pre17/tests/plc_tests.c @ 4:26cd8f1ef0b1

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import spandsp-0.0.6pre17
author Peter Meerwald <pmeerw@cosy.sbg.ac.at>
date Fri, 25 Jun 2010 15:50:58 +0200
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3:c6c5a16ce2f2 4:26cd8f1ef0b1
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 1.26 2009/05/30 15:23:14 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 #if defined(HAVE_CONFIG_H)
46 #include "config.h"
47 #endif
48
49 #include <stdlib.h>
50 #include <stdio.h>
51 #include <unistd.h>
52 #include <string.h>
53
54 #include <sndfile.h>
55
56 #include "spandsp.h"
57 #include "spandsp-sim.h"
58
59 #define INPUT_FILE_NAME "../test-data/local/short_nb_voice.wav"
60 #define OUTPUT_FILE_NAME "post_plc.wav"
61
62 int main(int argc, char *argv[])
63 {
64 SNDFILE *inhandle;
65 SNDFILE *outhandle;
66 plc_state_t plc;
67 int inframes;
68 int outframes;
69 int16_t amp[1024];
70 int block_no;
71 int lost_blocks;
72 int block_len;
73 int loss_rate;
74 int dropit;
75 int block_real;
76 int block_synthetic;
77 int tone;
78 int i;
79 uint32_t phase_acc;
80 int32_t phase_rate;
81 int opt;
82
83 loss_rate = 25;
84 block_len = 160;
85 block_real = FALSE;
86 block_synthetic = FALSE;
87 tone = -1;
88 while ((opt = getopt(argc, argv, "b:l:rst:")) != -1)
89 {
90 switch (opt)
91 {
92 case 'b':
93 block_len = atoi(optarg);
94 break;
95 case 'l':
96 loss_rate = atoi(optarg);
97 break;
98 case 'r':
99 block_real = TRUE;
100 break;
101 case 's':
102 block_synthetic = TRUE;
103 break;
104 case 't':
105 tone = atoi(optarg);
106 break;
107 }
108 }
109 phase_rate = 0;
110 inhandle = NULL;
111 if (tone < 0)
112 {
113 if ((inhandle = sf_open_telephony_read(INPUT_FILE_NAME, 1)) == NULL)
114 {
115 fprintf(stderr, " Failed to open audio file '%s'\n", INPUT_FILE_NAME);
116 exit(2);
117 }
118 }
119 else
120 {
121 phase_rate = dds_phase_ratef((float) tone);
122 }
123 if ((outhandle = sf_open_telephony_write(OUTPUT_FILE_NAME, 1)) == NULL)
124 {
125 fprintf(stderr, " Failed to open audio file '%s'\n", OUTPUT_FILE_NAME);
126 exit(2);
127 }
128 plc_init(&plc);
129 lost_blocks = 0;
130 for (block_no = 0; ; block_no++)
131 {
132 if (tone < 0)
133 {
134 inframes = sf_readf_short(inhandle, amp, block_len);
135 if (inframes != block_len)
136 break;
137 }
138 else
139 {
140 if (block_no > 10000)
141 break;
142 for (i = 0; i < block_len; i++)
143 amp[i] = (int16_t) dds_modf(&phase_acc, phase_rate, 10000.0, 0);
144 inframes = block_len;
145 }
146 dropit = rand()/(RAND_MAX/100);
147 if (dropit > loss_rate)
148 {
149 plc_rx(&plc, amp, inframes);
150 if (block_real)
151 memset(amp, 0, sizeof(int16_t)*inframes);
152 }
153 else
154 {
155 lost_blocks++;
156 plc_fillin(&plc, amp, inframes);
157 if (block_synthetic)
158 memset(amp, 0, sizeof(int16_t)*inframes);
159 }
160 outframes = sf_writef_short(outhandle, amp, inframes);
161 if (outframes != inframes)
162 {
163 fprintf(stderr, " Error writing out sound\n");
164 exit(2);
165 }
166 }
167 printf("Dropped %d of %d blocks\n", lost_blocks, block_no);
168 if (tone < 0)
169 {
170 if (sf_close(inhandle) != 0)
171 {
172 fprintf(stderr, " Cannot close audio file '%s'\n", INPUT_FILE_NAME);
173 exit(2);
174 }
175 }
176 if (sf_close(outhandle) != 0)
177 {
178 fprintf(stderr, " Cannot close audio file '%s'\n", OUTPUT_FILE_NAME);
179 exit(2);
180 }
181 return 0;
182 }
183 /*- End of function --------------------------------------------------------*/
184 /*- End of file ------------------------------------------------------------*/

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