tests: Move all test programs to a subdirectory
[libav.git] / libavresample / tests / avresample.c
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1/*
2 * Copyright (c) 2002 Fabrice Bellard
3 * Copyright (c) 2012 Justin Ruggles <justin.ruggles@gmail.com>
4 *
5 * This file is part of Libav.
6 *
7 * Libav is free software; you can redistribute it and/or
8 * modify it under the terms of the GNU Lesser General Public
9 * License as published by the Free Software Foundation; either
10 * version 2.1 of the License, or (at your option) any later version.
11 *
12 * Libav is distributed in the hope that it will be useful,
13 * but WITHOUT ANY WARRANTY; without even the implied warranty of
14 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
15 * Lesser General Public License for more details.
16 *
17 * You should have received a copy of the GNU Lesser General Public
18 * License along with Libav; if not, write to the Free Software
19 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
20 */
21
22#include <stdint.h>
23#include <stdio.h>
24
25#include "libavutil/avstring.h"
26#include "libavutil/common.h"
27#include "libavutil/lfg.h"
28#include "libavutil/libm.h"
29#include "libavutil/log.h"
30#include "libavutil/mem.h"
31#include "libavutil/opt.h"
32#include "libavutil/samplefmt.h"
33
34#include "libavresample/avresample.h"
35
36static double dbl_rand(AVLFG *lfg)
37{
38 return 2.0 * (av_lfg_get(lfg) / (double)UINT_MAX) - 1.0;
39}
40
41#define PUT_FUNC(name, fmt, type, expr) \
42static void put_sample_ ## name(void **data, enum AVSampleFormat sample_fmt,\
43 int channels, int sample, int ch, \
44 double v_dbl) \
45{ \
46 type v = expr; \
47 type **out = (type **)data; \
48 if (av_sample_fmt_is_planar(sample_fmt)) \
49 out[ch][sample] = v; \
50 else \
51 out[0][sample * channels + ch] = v; \
52}
53
54PUT_FUNC(u8, AV_SAMPLE_FMT_U8, uint8_t, av_clip_uint8 ( lrint(v_dbl * (1 << 7)) + 128))
55PUT_FUNC(s16, AV_SAMPLE_FMT_S16, int16_t, av_clip_int16 ( lrint(v_dbl * (1 << 15))))
56PUT_FUNC(s32, AV_SAMPLE_FMT_S32, int32_t, av_clipl_int32(llrint(v_dbl * (1U << 31))))
57PUT_FUNC(flt, AV_SAMPLE_FMT_FLT, float, v_dbl)
58PUT_FUNC(dbl, AV_SAMPLE_FMT_DBL, double, v_dbl)
59
60static void put_sample(void **data, enum AVSampleFormat sample_fmt,
61 int channels, int sample, int ch, double v_dbl)
62{
63 switch (av_get_packed_sample_fmt(sample_fmt)) {
64 case AV_SAMPLE_FMT_U8:
65 put_sample_u8(data, sample_fmt, channels, sample, ch, v_dbl);
66 break;
67 case AV_SAMPLE_FMT_S16:
68 put_sample_s16(data, sample_fmt, channels, sample, ch, v_dbl);
69 break;
70 case AV_SAMPLE_FMT_S32:
71 put_sample_s32(data, sample_fmt, channels, sample, ch, v_dbl);
72 break;
73 case AV_SAMPLE_FMT_FLT:
74 put_sample_flt(data, sample_fmt, channels, sample, ch, v_dbl);
75 break;
76 case AV_SAMPLE_FMT_DBL:
77 put_sample_dbl(data, sample_fmt, channels, sample, ch, v_dbl);
78 break;
79 }
80}
81
82static void audiogen(AVLFG *rnd, void **data, enum AVSampleFormat sample_fmt,
83 int channels, int sample_rate, int nb_samples)
84{
85 int i, ch, k;
86 double v, f, a, ampa;
87 double tabf1[AVRESAMPLE_MAX_CHANNELS];
88 double tabf2[AVRESAMPLE_MAX_CHANNELS];
89 double taba[AVRESAMPLE_MAX_CHANNELS];
90
91#define PUT_SAMPLE put_sample(data, sample_fmt, channels, k, ch, v);
92
93 k = 0;
94
95 /* 1 second of single freq sine at 1000 Hz */
96 a = 0;
97 for (i = 0; i < 1 * sample_rate && k < nb_samples; i++, k++) {
98 v = sin(a) * 0.30;
99 for (ch = 0; ch < channels; ch++)
100 PUT_SAMPLE
101 a += M_PI * 1000.0 * 2.0 / sample_rate;
102 }
103
104 /* 1 second of varying frequency between 100 and 10000 Hz */
105 a = 0;
106 for (i = 0; i < 1 * sample_rate && k < nb_samples; i++, k++) {
107 v = sin(a) * 0.30;
108 for (ch = 0; ch < channels; ch++)
109 PUT_SAMPLE
110 f = 100.0 + (((10000.0 - 100.0) * i) / sample_rate);
111 a += M_PI * f * 2.0 / sample_rate;
112 }
113
114 /* 0.5 second of low amplitude white noise */
115 for (i = 0; i < sample_rate / 2 && k < nb_samples; i++, k++) {
116 v = dbl_rand(rnd) * 0.30;
117 for (ch = 0; ch < channels; ch++)
118 PUT_SAMPLE
119 }
120
121 /* 0.5 second of high amplitude white noise */
122 for (i = 0; i < sample_rate / 2 && k < nb_samples; i++, k++) {
123 v = dbl_rand(rnd);
124 for (ch = 0; ch < channels; ch++)
125 PUT_SAMPLE
126 }
127
128 /* 1 second of unrelated ramps for each channel */
129 for (ch = 0; ch < channels; ch++) {
130 taba[ch] = 0;
131 tabf1[ch] = 100 + av_lfg_get(rnd) % 5000;
132 tabf2[ch] = 100 + av_lfg_get(rnd) % 5000;
133 }
134 for (i = 0; i < 1 * sample_rate && k < nb_samples; i++, k++) {
135 for (ch = 0; ch < channels; ch++) {
136 v = sin(taba[ch]) * 0.30;
137 PUT_SAMPLE
138 f = tabf1[ch] + (((tabf2[ch] - tabf1[ch]) * i) / sample_rate);
139 taba[ch] += M_PI * f * 2.0 / sample_rate;
140 }
141 }
142
143 /* 2 seconds of 500 Hz with varying volume */
144 a = 0;
145 ampa = 0;
146 for (i = 0; i < 2 * sample_rate && k < nb_samples; i++, k++) {
147 for (ch = 0; ch < channels; ch++) {
148 double amp = (1.0 + sin(ampa)) * 0.15;
149 if (ch & 1)
150 amp = 0.30 - amp;
151 v = sin(a) * amp;
152 PUT_SAMPLE
153 a += M_PI * 500.0 * 2.0 / sample_rate;
154 ampa += M_PI * 2.0 / sample_rate;
155 }
156 }
157}
158
159/* formats, rates, and layouts are ordered for priority in testing.
160 e.g. 'avresample-test 4 2 2' will test all input/output combinations of
161 S16/FLTP/S16P/FLT, 48000/44100, and stereo/mono */
162
163static const enum AVSampleFormat formats[] = {
164 AV_SAMPLE_FMT_S16,
165 AV_SAMPLE_FMT_FLTP,
166 AV_SAMPLE_FMT_S16P,
167 AV_SAMPLE_FMT_FLT,
168 AV_SAMPLE_FMT_S32P,
169 AV_SAMPLE_FMT_S32,
170 AV_SAMPLE_FMT_U8P,
171 AV_SAMPLE_FMT_U8,
172 AV_SAMPLE_FMT_DBLP,
173 AV_SAMPLE_FMT_DBL,
174};
175
176static const int rates[] = {
177 48000,
178 44100,
179 16000
180};
181
182static const uint64_t layouts[] = {
183 AV_CH_LAYOUT_STEREO,
184 AV_CH_LAYOUT_MONO,
185 AV_CH_LAYOUT_5POINT1,
186 AV_CH_LAYOUT_7POINT1,
187};
188
189int main(int argc, char **argv)
190{
191 AVAudioResampleContext *s;
192 AVLFG rnd;
193 int ret = 0;
194 uint8_t *in_buf = NULL;
195 uint8_t *out_buf = NULL;
196 unsigned int in_buf_size;
197 unsigned int out_buf_size;
198 uint8_t *in_data[AVRESAMPLE_MAX_CHANNELS] = { 0 };
199 uint8_t *out_data[AVRESAMPLE_MAX_CHANNELS] = { 0 };
200 int in_linesize;
201 int out_linesize;
202 uint64_t in_ch_layout;
203 int in_channels;
204 enum AVSampleFormat in_fmt;
205 int in_rate;
206 uint64_t out_ch_layout;
207 int out_channels;
208 enum AVSampleFormat out_fmt;
209 int out_rate;
210 int num_formats, num_rates, num_layouts;
211 int i, j, k, l, m, n;
212
213 num_formats = 2;
214 num_rates = 2;
215 num_layouts = 2;
216 if (argc > 1) {
217 if (!av_strncasecmp(argv[1], "-h", 3)) {
218 av_log(NULL, AV_LOG_INFO, "Usage: avresample-test [<num formats> "
219 "[<num sample rates> [<num channel layouts>]]]\n"
220 "Default is 2 2 2\n");
221 return 0;
222 }
223 num_formats = strtol(argv[1], NULL, 0);
224 num_formats = av_clip(num_formats, 1, FF_ARRAY_ELEMS(formats));
225 }
226 if (argc > 2) {
227 num_rates = strtol(argv[2], NULL, 0);
228 num_rates = av_clip(num_rates, 1, FF_ARRAY_ELEMS(rates));
229 }
230 if (argc > 3) {
231 num_layouts = strtol(argv[3], NULL, 0);
232 num_layouts = av_clip(num_layouts, 1, FF_ARRAY_ELEMS(layouts));
233 }
234
235 av_log_set_level(AV_LOG_DEBUG);
236
237 av_lfg_init(&rnd, 0xC0FFEE);
238
239 in_buf_size = av_samples_get_buffer_size(&in_linesize, 8, 48000 * 6,
240 AV_SAMPLE_FMT_DBLP, 0);
241 out_buf_size = in_buf_size;
242
243 in_buf = av_malloc(in_buf_size);
244 if (!in_buf)
245 goto end;
246 out_buf = av_malloc(out_buf_size);
247 if (!out_buf)
248 goto end;
249
250 s = avresample_alloc_context();
251 if (!s) {
252 av_log(NULL, AV_LOG_ERROR, "Error allocating AVAudioResampleContext\n");
253 ret = 1;
254 goto end;
255 }
256
257 for (i = 0; i < num_formats; i++) {
258 in_fmt = formats[i];
259 for (k = 0; k < num_layouts; k++) {
260 in_ch_layout = layouts[k];
261 in_channels = av_get_channel_layout_nb_channels(in_ch_layout);
262 for (m = 0; m < num_rates; m++) {
263 in_rate = rates[m];
264
265 ret = av_samples_fill_arrays(in_data, &in_linesize, in_buf,
266 in_channels, in_rate * 6,
267 in_fmt, 0);
268 if (ret < 0) {
269 av_log(s, AV_LOG_ERROR, "failed in_data fill arrays\n");
270 goto end;
271 }
272 audiogen(&rnd, (void **)in_data, in_fmt, in_channels, in_rate, in_rate * 6);
273
274 for (j = 0; j < num_formats; j++) {
275 out_fmt = formats[j];
276 for (l = 0; l < num_layouts; l++) {
277 out_ch_layout = layouts[l];
278 out_channels = av_get_channel_layout_nb_channels(out_ch_layout);
279 for (n = 0; n < num_rates; n++) {
280 out_rate = rates[n];
281
282 av_log(NULL, AV_LOG_INFO, "%s to %s, %d to %d channels, %d Hz to %d Hz\n",
283 av_get_sample_fmt_name(in_fmt), av_get_sample_fmt_name(out_fmt),
284 in_channels, out_channels, in_rate, out_rate);
285
286 ret = av_samples_fill_arrays(out_data, &out_linesize,
287 out_buf, out_channels,
288 out_rate * 6, out_fmt, 0);
289 if (ret < 0) {
290 av_log(s, AV_LOG_ERROR, "failed out_data fill arrays\n");
291 goto end;
292 }
293
294 av_opt_set_int(s, "in_channel_layout", in_ch_layout, 0);
295 av_opt_set_int(s, "in_sample_fmt", in_fmt, 0);
296 av_opt_set_int(s, "in_sample_rate", in_rate, 0);
297 av_opt_set_int(s, "out_channel_layout", out_ch_layout, 0);
298 av_opt_set_int(s, "out_sample_fmt", out_fmt, 0);
299 av_opt_set_int(s, "out_sample_rate", out_rate, 0);
300
301 av_opt_set_int(s, "internal_sample_fmt", AV_SAMPLE_FMT_FLTP, 0);
302
303 ret = avresample_open(s);
304 if (ret < 0) {
305 av_log(s, AV_LOG_ERROR, "Error opening context\n");
306 goto end;
307 }
308
309 ret = avresample_convert(s, out_data, out_linesize, out_rate * 6,
310 in_data, in_linesize, in_rate * 6);
311 if (ret < 0) {
312 char errbuf[256];
313 av_strerror(ret, errbuf, sizeof(errbuf));
314 av_log(NULL, AV_LOG_ERROR, "%s\n", errbuf);
315 goto end;
316 }
317 av_log(NULL, AV_LOG_INFO, "Converted %d samples to %d samples\n",
318 in_rate * 6, ret);
319 if (avresample_get_delay(s) > 0)
320 av_log(NULL, AV_LOG_INFO, "%d delay samples not converted\n",
321 avresample_get_delay(s));
322 if (avresample_available(s) > 0)
323 av_log(NULL, AV_LOG_INFO, "%d samples available for output\n",
324 avresample_available(s));
325 av_log(NULL, AV_LOG_INFO, "\n");
326
327 avresample_close(s);
328 }
329 }
330 }
331 }
332 }
333 }
334
335 ret = 0;
336
337end:
338 av_freep(&in_buf);
339 av_freep(&out_buf);
340 avresample_free(&s);
341 return ret;
342}