Mercurial > hg > audiostuff
comparison spandsp-0.0.6pre17/tests/line_model_tests.c @ 4:26cd8f1ef0b1
import spandsp-0.0.6pre17
author | Peter Meerwald <pmeerw@cosy.sbg.ac.at> |
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date | Fri, 25 Jun 2010 15:50:58 +0200 |
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3:c6c5a16ce2f2 | 4:26cd8f1ef0b1 |
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1 /* | |
2 * SpanDSP - a series of DSP components for telephony | |
3 * | |
4 * line_model_tests.c - Tests for the telephone line model. | |
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: line_model_tests.c,v 1.28 2009/09/23 16:02:59 steveu Exp $ | |
26 */ | |
27 | |
28 /*! \page line_model_tests_page Telephony line model tests | |
29 \section line_model_tests_page_sec_1 What does it do? | |
30 ???. | |
31 | |
32 \section line_model_tests_page_sec_2 How does it work? | |
33 ???. | |
34 */ | |
35 | |
36 #if defined(HAVE_CONFIG_H) | |
37 #include "config.h" | |
38 #endif | |
39 | |
40 #include <stdlib.h> | |
41 #include <stdio.h> | |
42 #include <fcntl.h> | |
43 #include <unistd.h> | |
44 #include <string.h> | |
45 #include <time.h> | |
46 #include <sndfile.h> | |
47 | |
48 //#if defined(WITH_SPANDSP_INTERNALS) | |
49 #define SPANDSP_EXPOSE_INTERNAL_STRUCTURES | |
50 //#endif | |
51 | |
52 #include "spandsp.h" | |
53 #include "spandsp-sim.h" | |
54 | |
55 #if !defined(NULL) | |
56 #define NULL (void *) 0 | |
57 #endif | |
58 | |
59 #define BLOCK_LEN 160 | |
60 | |
61 #define OUT_FILE_COMPLEXIFY "complexify.wav" | |
62 #define IN_FILE_NAME1 "line_model_test_in1.wav" | |
63 #define IN_FILE_NAME2 "line_model_test_in2.wav" | |
64 #define OUT_FILE_NAME1 "line_model_one_way_test_out.wav" | |
65 #define OUT_FILE_NAME2 "line_model_two_way_test_out.wav" | |
66 | |
67 int channel_codec; | |
68 int rbs_pattern; | |
69 | |
70 static void complexify_tests(void) | |
71 { | |
72 complexify_state_t *s; | |
73 complexf_t cc; | |
74 int16_t amp; | |
75 int i; | |
76 SNDFILE *outhandle; | |
77 int outframes; | |
78 int16_t out[40000]; | |
79 awgn_state_t noise1; | |
80 | |
81 if ((outhandle = sf_open_telephony_write(OUT_FILE_COMPLEXIFY, 2)) == NULL) | |
82 { | |
83 fprintf(stderr, " Cannot create audio file '%s'\n", OUT_FILE_COMPLEXIFY); | |
84 exit(2); | |
85 } | |
86 awgn_init_dbm0(&noise1, 1234567, -10.0f); | |
87 s = complexify_init(); | |
88 for (i = 0; i < 20000; i++) | |
89 { | |
90 amp = awgn(&noise1); | |
91 cc = complexify(s, amp); | |
92 out[2*i] = cc.re; | |
93 out[2*i + 1] = cc.im; | |
94 } | |
95 outframes = sf_writef_short(outhandle, out, 20000); | |
96 if (outframes != 20000) | |
97 { | |
98 fprintf(stderr, " Error writing audio file\n"); | |
99 exit(2); | |
100 } | |
101 if (sf_close(outhandle)) | |
102 { | |
103 fprintf(stderr, " Cannot close audio file '%s'\n", OUT_FILE_COMPLEXIFY); | |
104 exit(2); | |
105 } | |
106 } | |
107 /*- End of function --------------------------------------------------------*/ | |
108 | |
109 static void test_one_way_model(int line_model_no, int speech_test) | |
110 { | |
111 one_way_line_model_state_t *model; | |
112 int16_t input1[BLOCK_LEN]; | |
113 int16_t output1[BLOCK_LEN]; | |
114 int16_t amp[2*BLOCK_LEN]; | |
115 SNDFILE *inhandle1; | |
116 SNDFILE *outhandle; | |
117 int outframes; | |
118 int samples; | |
119 int i; | |
120 int j; | |
121 awgn_state_t noise1; | |
122 | |
123 if ((model = one_way_line_model_init(line_model_no, -50, channel_codec, rbs_pattern)) == NULL) | |
124 { | |
125 fprintf(stderr, " Failed to create line model\n"); | |
126 exit(2); | |
127 } | |
128 | |
129 awgn_init_dbm0(&noise1, 1234567, -10.0f); | |
130 | |
131 if (speech_test) | |
132 { | |
133 if ((inhandle1 = sf_open_telephony_read(IN_FILE_NAME1, 1)) == NULL) | |
134 { | |
135 fprintf(stderr, " Cannot open audio file '%s'\n", IN_FILE_NAME1); | |
136 exit(2); | |
137 } | |
138 } | |
139 else | |
140 { | |
141 inhandle1 = NULL; | |
142 } | |
143 if ((outhandle = sf_open_telephony_write(OUT_FILE_NAME1, 1)) == NULL) | |
144 { | |
145 fprintf(stderr, " Cannot create audio file '%s'\n", OUT_FILE_NAME1); | |
146 exit(2); | |
147 } | |
148 for (i = 0; i < 10000; i++) | |
149 { | |
150 if (speech_test) | |
151 { | |
152 samples = sf_readf_short(inhandle1, input1, BLOCK_LEN); | |
153 if (samples == 0) | |
154 break; | |
155 } | |
156 else | |
157 { | |
158 for (j = 0; j < BLOCK_LEN; j++) | |
159 input1[j] = awgn(&noise1); | |
160 samples = BLOCK_LEN; | |
161 } | |
162 for (j = 0; j < samples; j++) | |
163 { | |
164 one_way_line_model(model, | |
165 &output1[j], | |
166 &input1[j], | |
167 1); | |
168 amp[j] = output1[j]; | |
169 } | |
170 outframes = sf_writef_short(outhandle, amp, samples); | |
171 if (outframes != samples) | |
172 { | |
173 fprintf(stderr, " Error writing audio file\n"); | |
174 exit(2); | |
175 } | |
176 } | |
177 if (speech_test) | |
178 { | |
179 if (sf_close(inhandle1)) | |
180 { | |
181 fprintf(stderr, " Cannot close audio file '%s'\n", IN_FILE_NAME1); | |
182 exit(2); | |
183 } | |
184 } | |
185 if (sf_close(outhandle)) | |
186 { | |
187 fprintf(stderr, " Cannot close audio file '%s'\n", OUT_FILE_NAME1); | |
188 exit(2); | |
189 } | |
190 one_way_line_model_release(model); | |
191 } | |
192 /*- End of function --------------------------------------------------------*/ | |
193 | |
194 static void test_both_ways_model(int line_model_no, int speech_test) | |
195 { | |
196 both_ways_line_model_state_t *model; | |
197 int16_t input1[BLOCK_LEN]; | |
198 int16_t input2[BLOCK_LEN]; | |
199 int16_t output1[BLOCK_LEN]; | |
200 int16_t output2[BLOCK_LEN]; | |
201 int16_t amp[2*BLOCK_LEN]; | |
202 SNDFILE *inhandle1; | |
203 SNDFILE *inhandle2; | |
204 SNDFILE *outhandle; | |
205 int outframes; | |
206 int samples; | |
207 int i; | |
208 int j; | |
209 awgn_state_t noise1; | |
210 awgn_state_t noise2; | |
211 | |
212 if ((model = both_ways_line_model_init(line_model_no, -50, line_model_no + 1, -35, channel_codec, rbs_pattern)) == NULL) | |
213 { | |
214 fprintf(stderr, " Failed to create line model\n"); | |
215 exit(2); | |
216 } | |
217 | |
218 awgn_init_dbm0(&noise1, 1234567, -10.0f); | |
219 awgn_init_dbm0(&noise2, 1234567, -10.0f); | |
220 | |
221 if (speech_test) | |
222 { | |
223 if ((inhandle1 = sf_open_telephony_read(IN_FILE_NAME1, 1)) == NULL) | |
224 { | |
225 fprintf(stderr, " Cannot open audio file '%s'\n", IN_FILE_NAME1); | |
226 exit(2); | |
227 } | |
228 if ((inhandle2 = sf_open_telephony_read(IN_FILE_NAME2, 1)) == NULL) | |
229 { | |
230 fprintf(stderr, " Cannot open audio file '%s'\n", IN_FILE_NAME2); | |
231 exit(2); | |
232 } | |
233 } | |
234 else | |
235 { | |
236 inhandle1 = | |
237 inhandle2 = NULL; | |
238 } | |
239 if ((outhandle = sf_open_telephony_write(OUT_FILE_NAME2, 2)) == NULL) | |
240 { | |
241 fprintf(stderr, " Cannot create audio file '%s'\n", OUT_FILE_NAME2); | |
242 exit(2); | |
243 } | |
244 for (i = 0; i < 10000; i++) | |
245 { | |
246 if (speech_test) | |
247 { | |
248 samples = sf_readf_short(inhandle1, input1, BLOCK_LEN); | |
249 if (samples == 0) | |
250 break; | |
251 samples = sf_readf_short(inhandle2, input2, samples); | |
252 if (samples == 0) | |
253 break; | |
254 } | |
255 else | |
256 { | |
257 for (j = 0; j < BLOCK_LEN; j++) | |
258 { | |
259 input1[j] = awgn(&noise1); | |
260 input2[j] = awgn(&noise2); | |
261 } | |
262 samples = BLOCK_LEN; | |
263 } | |
264 for (j = 0; j < samples; j++) | |
265 { | |
266 both_ways_line_model(model, | |
267 &output1[j], | |
268 &input1[j], | |
269 &output2[j], | |
270 &input2[j], | |
271 1); | |
272 amp[2*j] = output1[j]; | |
273 amp[2*j + 1] = output2[j]; | |
274 } | |
275 outframes = sf_writef_short(outhandle, amp, samples); | |
276 if (outframes != samples) | |
277 { | |
278 fprintf(stderr, " Error writing audio file\n"); | |
279 exit(2); | |
280 } | |
281 } | |
282 if (speech_test) | |
283 { | |
284 if (sf_close(inhandle1)) | |
285 { | |
286 fprintf(stderr, " Cannot close audio file '%s'\n", IN_FILE_NAME1); | |
287 exit(2); | |
288 } | |
289 if (sf_close(inhandle2)) | |
290 { | |
291 fprintf(stderr, " Cannot close audio file '%s'\n", IN_FILE_NAME2); | |
292 exit(2); | |
293 } | |
294 } | |
295 if (sf_close(outhandle)) | |
296 { | |
297 fprintf(stderr, " Cannot close audio file '%s'\n", OUT_FILE_NAME2); | |
298 exit(2); | |
299 } | |
300 both_ways_line_model_release(model); | |
301 } | |
302 /*- End of function --------------------------------------------------------*/ | |
303 | |
304 static void test_line_filter(int line_model_no) | |
305 { | |
306 float out; | |
307 double sumin; | |
308 double sumout; | |
309 int i; | |
310 int j; | |
311 int p; | |
312 int ptr; | |
313 int len; | |
314 swept_tone_state_t *s; | |
315 float filter[129]; | |
316 int16_t buf[BLOCK_LEN]; | |
317 | |
318 s = swept_tone_init(NULL, 200.0f, 3900.0f, -10.0f, 120*SAMPLE_RATE, 0); | |
319 for (j = 0; j < 129; j++) | |
320 filter[j] = 0.0f; | |
321 ptr = 0; | |
322 for (;;) | |
323 { | |
324 if ((len = swept_tone(s, buf, BLOCK_LEN)) <= 0) | |
325 break; | |
326 sumin = 0.0; | |
327 sumout = 0.0; | |
328 for (i = 0; i < len; i++) | |
329 { | |
330 /* Add the sample in the filter buffer */ | |
331 p = ptr; | |
332 filter[p] = buf[i]; | |
333 if (++p == 129) | |
334 p = 0; | |
335 ptr = p; | |
336 | |
337 /* Apply the filter */ | |
338 out = 0.0f; | |
339 for (j = 0; j < 129; j++) | |
340 { | |
341 out += line_models[line_model_no][128 - j]*filter[p]; | |
342 if (++p >= 129) | |
343 p = 0; | |
344 } | |
345 sumin += buf[i]*buf[i]; | |
346 sumout += out*out; | |
347 } | |
348 /*endfor*/ | |
349 printf("%7.1f %f\n", swept_tone_current_frequency(s), 10.0*log10(sumout/sumin)); | |
350 } | |
351 /*endfor*/ | |
352 swept_tone_free(s); | |
353 } | |
354 /*- End of function --------------------------------------------------------*/ | |
355 | |
356 int main(int argc, char *argv[]) | |
357 { | |
358 int line_model_no; | |
359 int speech_test; | |
360 int log_audio; | |
361 int opt; | |
362 | |
363 channel_codec = MUNGE_CODEC_NONE; | |
364 log_audio = FALSE; | |
365 line_model_no = 0; | |
366 rbs_pattern = 0; | |
367 speech_test = FALSE; | |
368 while ((opt = getopt(argc, argv, "c:lm:r:s:")) != -1) | |
369 { | |
370 switch (opt) | |
371 { | |
372 case 'c': | |
373 channel_codec = atoi(optarg); | |
374 break; | |
375 case 'l': | |
376 log_audio = TRUE; | |
377 break; | |
378 case 'm': | |
379 line_model_no = atoi(optarg); | |
380 break; | |
381 case 'r': | |
382 rbs_pattern = atoi(optarg); | |
383 break; | |
384 case 's': | |
385 speech_test = atoi(optarg); | |
386 break; | |
387 default: | |
388 //usage(); | |
389 exit(2); | |
390 } | |
391 } | |
392 complexify_tests(); | |
393 test_one_way_model(line_model_no, speech_test); | |
394 test_both_ways_model(line_model_no, speech_test); | |
395 test_line_filter(line_model_no); | |
396 } | |
397 /*- End of function --------------------------------------------------------*/ | |
398 /*- End of file ------------------------------------------------------------*/ |