int _tmain(int argc, _TCHAR* argv[])
{
#if TEST_INT
// 整数のテスト
printf("fp32_to_double\tfp32_to_int\tinner\n");
for (int i =INT_MIN; i < INT_MAX; i++) {
fp32 fv = int_to_fp32(i);
printf("%d\t%lf\t%d\t%d\n", i, fp32_to_double(fv), fp32_to_int(fv), fv);
}
#else
// 浮動小数点のテスト
printf("expr\tdouble\tfp32\terr\n");
calc(INT_MAX, INT_MIN);
for (int i = 0; i < CALC_REPEAT; i++) {
double dx = dbl_rand();
double dy = dbl_rand();
// 除算エラー回避
if (dy == 0.0f)
continue;
calc(dx, dy);
}
#endif
return 0;
}
static void audiogen(void *data, enum AVSampleFormat sample_fmt,
int channels, int sample_rate, int nb_samples)
{
int i, ch, k;
double v, f, a, ampa;
double tabf1[SWR_CH_MAX];
double tabf2[SWR_CH_MAX];
double taba[SWR_CH_MAX];
unsigned static rnd;
#define PUT_SAMPLE set(data, ch, k, channels, sample_fmt, v);
#define uint_rand(x) ((x) = (x) * 1664525 + 1013904223)
#define dbl_rand(x) (uint_rand(x)*2.0 / (double)UINT_MAX - 1)
k = 0;
/* 1 second of single freq sinus at 1000 Hz */
a = 0;
for (i = 0; i < 1 * sample_rate && k < nb_samples; i++, k++) {
v = sin(a) * 0.30;
for (ch = 0; ch < channels; ch++)
PUT_SAMPLE
a += M_PI * 1000.0 * 2.0 / sample_rate;
}
/* 1 second of varying frequency between 100 and 10000 Hz */
a = 0;
for (i = 0; i < 1 * sample_rate && k < nb_samples; i++, k++) {
v = sin(a) * 0.30;
for (ch = 0; ch < channels; ch++)
PUT_SAMPLE
f = 100.0 + (((10000.0 - 100.0) * i) / sample_rate);
a += M_PI * f * 2.0 / sample_rate;
}
/* 0.5 second of low amplitude white noise */
for (i = 0; i < sample_rate / 2 && k < nb_samples; i++, k++) {
v = dbl_rand(rnd) * 0.30;
for (ch = 0; ch < channels; ch++)
PUT_SAMPLE
}
/* 0.5 second of high amplitude white noise */
for (i = 0; i < sample_rate / 2 && k < nb_samples; i++, k++) {
v = dbl_rand(rnd);
for (ch = 0; ch < channels; ch++)
PUT_SAMPLE
}
/* 1 second of unrelated ramps for each channel */
for (ch = 0; ch < channels; ch++) {
taba[ch] = 0;
tabf1[ch] = 100 + uint_rand(rnd) % 5000;
tabf2[ch] = 100 + uint_rand(rnd) % 5000;
}
for (i = 0; i < 1 * sample_rate && k < nb_samples; i++, k++) {
for (ch = 0; ch < channels; ch++) {
v = sin(taba[ch]) * 0.30;
PUT_SAMPLE
f = tabf1[ch] + (((tabf2[ch] - tabf1[ch]) * i) / sample_rate);
taba[ch] += M_PI * f * 2.0 / sample_rate;
}
}
/* 2 seconds of 500 Hz with varying volume */
a = 0;
ampa = 0;
for (i = 0; i < 2 * sample_rate && k < nb_samples; i++, k++) {
for (ch = 0; ch < channels; ch++) {
double amp = (1.0 + sin(ampa)) * 0.15;
if (ch & 1)
amp = 0.30 - amp;
v = sin(a) * amp;
PUT_SAMPLE
a += M_PI * 500.0 * 2.0 / sample_rate;
ampa += M_PI * 2.0 / sample_rate;
}
}
}
