1
0
mirror of https://github.com/FFmpeg/FFmpeg.git synced 2024-12-23 12:43:46 +02:00
FFmpeg/tests/checkasm/af_afir.c
Andreas Rheinhardt 0df18f29ae avfilter/af_afir: Only keep DSP stuff in header
Only the AudioFIRDSPContext and the functions for its initialization
are needed outside of lavfi/af_afir.c.
Also rename the header to af_afirdsp.h to reflect the change.

Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@outlook.com>
2022-05-06 05:19:49 +02:00

97 lines
3.2 KiB
C

/*
* This file is part of FFmpeg.
*
* FFmpeg is free software; you can redistribute it and/or modify
* it under the terms of the GNU General Public License as published by
* the Free Software Foundation; either version 2 of the License, or
* (at your option) any later version.
*
* FFmpeg is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
* GNU General Public License for more details.
*
* You should have received a copy of the GNU General Public License along
* with FFmpeg; if not, write to the Free Software Foundation, Inc.,
* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
*/
#include "config.h"
#include <float.h>
#include <stdint.h>
#include "libavfilter/af_afirdsp.h"
#include "libavutil/internal.h"
#include "libavutil/mem_internal.h"
#include "checkasm.h"
#define LEN 256
#define randomize_buffer(buf) \
do { \
int i; \
double bmg[2], stddev = 10.0, mean = 0.0; \
\
for (i = 0; i < LEN*2+8; i += 2) { \
av_bmg_get(&checkasm_lfg, bmg); \
buf[i] = bmg[0] * stddev + mean; \
buf[i + 1] = bmg[1] * stddev + mean; \
} \
} while(0);
static void test_fcmul_add(const float *src0, const float *src1, const float *src2)
{
LOCAL_ALIGNED_32(float, cdst, [LEN*2+8]);
LOCAL_ALIGNED_32(float, odst, [LEN*2+8]);
int i;
declare_func(void, float *sum, const float *t, const float *c,
ptrdiff_t len);
memcpy(cdst, src0, (LEN*2+8) * sizeof(float));
memcpy(odst, src0, (LEN*2+8) * sizeof(float));
call_ref(cdst, src1, src2, LEN);
call_new(odst, src1, src2, LEN);
for (i = 0; i <= LEN*2; i++) {
int idx = i & ~1;
float cre = src2[idx];
float cim = src2[idx + 1];
float tre = src1[idx];
float tim = src1[idx + 1];
double t = fabs(src0[i]) +
fabs(tre) + fabs(tim) + fabs(cre) + fabs(cim) +
fabs(tre * cre) + fabs(tim * cim) +
fabs(tre * cim) + fabs(tim * cre) +
fabs(tre * cre - tim * cim) +
fabs(tre * cim + tim * cre) +
fabs(cdst[i]) + 1.0;
if (!float_near_abs_eps(cdst[i], odst[i], t * 2 * FLT_EPSILON)) {
fprintf(stderr, "%d: %- .12f - %- .12f = % .12g\n",
i, cdst[i], odst[i], cdst[i] - odst[i]);
fail();
break;
}
}
memcpy(odst, src0, (LEN*2+8) * sizeof(float));
bench_new(odst, src1, src2, LEN);
}
void checkasm_check_afir(void)
{
LOCAL_ALIGNED_32(float, src0, [LEN*2+8]);
LOCAL_ALIGNED_32(float, src1, [LEN*2+8]);
LOCAL_ALIGNED_32(float, src2, [LEN*2+8]);
AudioFIRDSPContext fir = { 0 };
ff_afir_init(&fir);
randomize_buffer(src0);
randomize_buffer(src1);
randomize_buffer(src2);
if (check_func(fir.fcmul_add, "fcmul_add"))
test_fcmul_add(src0, src1, src2);
report("fcmul_add");
}