fate: qtrle: disable audio in all tests
[libav.git] / tests / audiogen.c
1 /*
2 * Generate a synthetic stereo sound.
3 * NOTE: No floats are used to guarantee bitexact output.
4 *
5 * Copyright (c) 2002 Fabrice Bellard
6 *
7 * This file is part of Libav.
8 *
9 * Libav is free software; you can redistribute it and/or
10 * modify it under the terms of the GNU Lesser General Public
11 * License as published by the Free Software Foundation; either
12 * version 2.1 of the License, or (at your option) any later version.
13 *
14 * Libav is distributed in the hope that it will be useful,
15 * but WITHOUT ANY WARRANTY; without even the implied warranty of
16 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
17 * Lesser General Public License for more details.
18 *
19 * You should have received a copy of the GNU Lesser General Public
20 * License along with Libav; if not, write to the Free Software
21 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
22 */
23
24 #include <stdlib.h>
25 #include <stdint.h>
26 #include <stdio.h>
27 #include <string.h>
28
29 #define MAX_CHANNELS 8
30
31 static unsigned int myrnd(unsigned int *seed_ptr, int n)
32 {
33 unsigned int seed, val;
34
35 seed = *seed_ptr;
36 seed = (seed * 314159) + 1;
37 if (n == 256) {
38 val = seed >> 24;
39 } else {
40 val = seed % n;
41 }
42 *seed_ptr = seed;
43 return val;
44 }
45
46 #define FRAC_BITS 16
47 #define FRAC_ONE (1 << FRAC_BITS)
48
49 #define COS_TABLE_BITS 7
50
51 /* integer cosinus */
52 static const unsigned short cos_table[(1 << COS_TABLE_BITS) + 2] = {
53 0x8000, 0x7ffe, 0x7ff6, 0x7fea, 0x7fd9, 0x7fc2, 0x7fa7, 0x7f87,
54 0x7f62, 0x7f38, 0x7f0a, 0x7ed6, 0x7e9d, 0x7e60, 0x7e1e, 0x7dd6,
55 0x7d8a, 0x7d3a, 0x7ce4, 0x7c89, 0x7c2a, 0x7bc6, 0x7b5d, 0x7aef,
56 0x7a7d, 0x7a06, 0x798a, 0x790a, 0x7885, 0x77fb, 0x776c, 0x76d9,
57 0x7642, 0x75a6, 0x7505, 0x7460, 0x73b6, 0x7308, 0x7255, 0x719e,
58 0x70e3, 0x7023, 0x6f5f, 0x6e97, 0x6dca, 0x6cf9, 0x6c24, 0x6b4b,
59 0x6a6e, 0x698c, 0x68a7, 0x67bd, 0x66d0, 0x65de, 0x64e9, 0x63ef,
60 0x62f2, 0x61f1, 0x60ec, 0x5fe4, 0x5ed7, 0x5dc8, 0x5cb4, 0x5b9d,
61 0x5a82, 0x5964, 0x5843, 0x571e, 0x55f6, 0x54ca, 0x539b, 0x5269,
62 0x5134, 0x4ffb, 0x4ec0, 0x4d81, 0x4c40, 0x4afb, 0x49b4, 0x486a,
63 0x471d, 0x45cd, 0x447b, 0x4326, 0x41ce, 0x4074, 0x3f17, 0x3db8,
64 0x3c57, 0x3af3, 0x398d, 0x3825, 0x36ba, 0x354e, 0x33df, 0x326e,
65 0x30fc, 0x2f87, 0x2e11, 0x2c99, 0x2b1f, 0x29a4, 0x2827, 0x26a8,
66 0x2528, 0x23a7, 0x2224, 0x209f, 0x1f1a, 0x1d93, 0x1c0c, 0x1a83,
67 0x18f9, 0x176e, 0x15e2, 0x1455, 0x12c8, 0x113a, 0x0fab, 0x0e1c,
68 0x0c8c, 0x0afb, 0x096b, 0x07d9, 0x0648, 0x04b6, 0x0324, 0x0192,
69 0x0000, 0x0000,
70 };
71
72 #define CSHIFT (FRAC_BITS - COS_TABLE_BITS - 2)
73
74 static int int_cos(int a)
75 {
76 int neg, v, f;
77 const unsigned short *p;
78
79 a = a & (FRAC_ONE - 1); /* modulo 2 * pi */
80 if (a >= (FRAC_ONE / 2))
81 a = FRAC_ONE - a;
82 neg = 0;
83 if (a > (FRAC_ONE / 4)) {
84 neg = -1;
85 a = (FRAC_ONE / 2) - a;
86 }
87 p = cos_table + (a >> CSHIFT);
88 /* linear interpolation */
89 f = a & ((1 << CSHIFT) - 1);
90 v = p[0] + (((p[1] - p[0]) * f + (1 << (CSHIFT - 1))) >> CSHIFT);
91 v = (v ^ neg) - neg;
92 v = v << (FRAC_BITS - 15);
93 return v;
94 }
95
96 FILE *outfile;
97
98 static void put16(int16_t v)
99 {
100 fputc( v & 0xff, outfile);
101 fputc((v >> 8) & 0xff, outfile);
102 }
103
104 static void put32(uint32_t v)
105 {
106 fputc( v & 0xff, outfile);
107 fputc((v >> 8) & 0xff, outfile);
108 fputc((v >> 16) & 0xff, outfile);
109 fputc((v >> 24) & 0xff, outfile);
110 }
111
112 #define HEADER_SIZE 46
113 #define FMT_SIZE 18
114 #define SAMPLE_SIZE 2
115 #define WFORMAT_PCM 0x0001
116
117 static void put_wav_header(int sample_rate, int channels, int nb_samples)
118 {
119 int block_align = SAMPLE_SIZE * channels;
120 int data_size = block_align * nb_samples;
121
122 fputs("RIFF", outfile);
123 put32(HEADER_SIZE + data_size);
124 fputs("WAVEfmt ", outfile);
125 put32(FMT_SIZE);
126 put16(WFORMAT_PCM);
127 put16(channels);
128 put32(sample_rate);
129 put32(block_align * sample_rate);
130 put16(block_align);
131 put16(SAMPLE_SIZE * 8);
132 put16(0);
133 fputs("data", outfile);
134 put32(data_size);
135 }
136
137 int main(int argc, char **argv)
138 {
139 int i, a, v, j, f, amp, ampa;
140 unsigned int seed = 1;
141 int tabf1[MAX_CHANNELS], tabf2[MAX_CHANNELS];
142 int taba[MAX_CHANNELS];
143 int sample_rate = 44100;
144 int nb_channels = 2;
145 char *ext;
146
147 if (argc < 2 || argc > 4) {
148 printf("usage: %s file [<sample rate> [<channels>]]\n"
149 "generate a test raw 16 bit audio stream\n"
150 "If the file extension is .wav a WAVE header will be added.\n"
151 "default: 44100 Hz stereo\n", argv[0]);
152 exit(1);
153 }
154
155 if (argc > 2) {
156 sample_rate = atoi(argv[2]);
157 if (sample_rate <= 0) {
158 fprintf(stderr, "invalid sample rate: %d\n", sample_rate);
159 return 1;
160 }
161 }
162
163 if (argc > 3) {
164 nb_channels = atoi(argv[3]);
165 if (nb_channels < 1 || nb_channels > MAX_CHANNELS) {
166 fprintf(stderr, "invalid number of channels: %d\n", nb_channels);
167 return 1;
168 }
169 }
170
171 outfile = fopen(argv[1], "wb");
172 if (!outfile) {
173 perror(argv[1]);
174 return 1;
175 }
176
177 if ((ext = strrchr(argv[1], '.')) != NULL && !strcmp(ext, ".wav"))
178 put_wav_header(sample_rate, nb_channels, 6 * sample_rate);
179
180 /* 1 second of single freq sinus at 1000 Hz */
181 a = 0;
182 for (i = 0; i < 1 * sample_rate; i++) {
183 v = (int_cos(a) * 10000) >> FRAC_BITS;
184 for (j = 0; j < nb_channels; j++)
185 put16(v);
186 a += (1000 * FRAC_ONE) / sample_rate;
187 }
188
189 /* 1 second of varing frequency between 100 and 10000 Hz */
190 a = 0;
191 for (i = 0; i < 1 * sample_rate; i++) {
192 v = (int_cos(a) * 10000) >> FRAC_BITS;
193 for (j = 0; j < nb_channels; j++)
194 put16(v);
195 f = 100 + (((10000 - 100) * i) / sample_rate);
196 a += (f * FRAC_ONE) / sample_rate;
197 }
198
199 /* 0.5 second of low amplitude white noise */
200 for (i = 0; i < sample_rate / 2; i++) {
201 v = myrnd(&seed, 20000) - 10000;
202 for (j = 0; j < nb_channels; j++)
203 put16(v);
204 }
205
206 /* 0.5 second of high amplitude white noise */
207 for (i = 0; i < sample_rate / 2; i++) {
208 v = myrnd(&seed, 65535) - 32768;
209 for (j = 0; j < nb_channels; j++)
210 put16(v);
211 }
212
213 /* 1 second of unrelated ramps for each channel */
214 for (j = 0; j < nb_channels; j++) {
215 taba[j] = 0;
216 tabf1[j] = 100 + myrnd(&seed, 5000);
217 tabf2[j] = 100 + myrnd(&seed, 5000);
218 }
219 for (i = 0; i < 1 * sample_rate; i++) {
220 for (j = 0; j < nb_channels; j++) {
221 v = (int_cos(taba[j]) * 10000) >> FRAC_BITS;
222 put16(v);
223 f = tabf1[j] + (((tabf2[j] - tabf1[j]) * i) / sample_rate);
224 taba[j] += (f * FRAC_ONE) / sample_rate;
225 }
226 }
227
228 /* 2 seconds of 500 Hz with varying volume */
229 a = 0;
230 ampa = 0;
231 for (i = 0; i < 2 * sample_rate; i++) {
232 for (j = 0; j < nb_channels; j++) {
233 amp = ((FRAC_ONE + int_cos(ampa)) * 5000) >> FRAC_BITS;
234 if (j & 1)
235 amp = 10000 - amp;
236 v = (int_cos(a) * amp) >> FRAC_BITS;
237 put16(v);
238 a += (500 * FRAC_ONE) / sample_rate;
239 ampa += (2 * FRAC_ONE) / sample_rate;
240 }
241 }
242
243 fclose(outfile);
244 return 0;
245 }