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/*
* This file is part of Libav.
*
* Libav 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.
*
* Libav 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 Libav; if not, write to the Free Software Foundation, Inc.,
* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
*/
#include <math.h>
#include <string.h>
#include <stdio.h>
#include <stdlib.h>
#include "libavcodec/audiodsp.h"
#include "libavutil/common.h"
#include "libavutil/intreadwrite.h"
#include "checkasm.h"
#define MAX_SIZE (32 * 128)
#define randomize_float(buf, len) \
do { \
int i; \
for (i = 0; i < len; i++) { \
float f = (float)rnd() / (UINT_MAX >> 5) - 16.0f; \
buf[i] = f; \
} \
} while (0)
#define randomize_int(buf, len, size, bits) \
do { \
int i; \
for (i = 0; i < len; i++) { \
uint ## size ## _t r = rnd() & ((1LL << bits) - 1); \
AV_WN ## size ## A(buf + i, -(1LL << (bits - 1)) + r); \
} \
} while (0)
void checkasm_check_audiodsp(void)
{
AudioDSPContext adsp;
ff_audiodsp_init(&adsp);
if (check_func(adsp.scalarproduct_int16, "audiodsp.scalarproduct_int16")) {
LOCAL_ALIGNED(32, int16_t, v1, [MAX_SIZE]);
LOCAL_ALIGNED(32, int16_t, v2, [MAX_SIZE]);
unsigned int len_bits_minus4, v1_bits, v2_bits, len;
int32_t res0, res1;
declare_func_emms(AV_CPU_FLAG_MMX, int32_t, const int16_t *v1, const int16_t *v2, int len);
// generate random 5-12bit vector length
len_bits_minus4 = rnd() % 8;
len = rnd() & ((1 << len_bits_minus4) - 1);
len = 16 * FFMAX(len, 1);
// generate the bit counts for each of the vectors such that the result
// fits into int32
v1_bits = 1 + rnd() % 15;
v2_bits = FFMIN(32 - (len_bits_minus4 + 4) - v1_bits - 1, 15);
randomize_int(v1, MAX_SIZE, 16, v1_bits + 1);
randomize_int(v2, MAX_SIZE, 16, v2_bits + 1);
res0 = call_ref(v1, v2, len);
res1 = call_new(v1, v2, len);
if (res0 != res1)
fail();
bench_new(v1, v2, MAX_SIZE);
}
if (check_func(adsp.vector_clip_int32, "audiodsp.vector_clip_int32")) {
LOCAL_ALIGNED(32, int32_t, src, [MAX_SIZE]);
LOCAL_ALIGNED(32, int32_t, dst0, [MAX_SIZE]);
LOCAL_ALIGNED(32, int32_t, dst1, [MAX_SIZE]);
int32_t val1, val2, min, max;
int len;
declare_func_emms(AV_CPU_FLAG_MMX, void, int32_t *dst, const int32_t *src,
int32_t min, int32_t max, unsigned int len);
val1 = ((int32_t)rnd());
val1 = FFSIGN(val1) * (val1 & ((1 << 24) - 1));
val2 = ((int32_t)rnd());
val2 = FFSIGN(val2) * (val2 & ((1 << 24) - 1));
min = FFMIN(val1, val2);
max = FFMAX(val1, val2);
randomize_int(src, MAX_SIZE, 32, 32);
len = rnd() % 128;
len = 32 * FFMAX(len, 1);
call_ref(dst0, src, min, max, len);
call_new(dst1, src, min, max, len);
if (memcmp(dst0, dst1, len * sizeof(*dst0)))
fail();
bench_new(dst1, src, min, max, MAX_SIZE);
}
if (check_func(adsp.vector_clipf, "audiodsp.vector_clipf")) {
LOCAL_ALIGNED(32, float, src, [MAX_SIZE]);
LOCAL_ALIGNED(32, float, dst0, [MAX_SIZE]);
LOCAL_ALIGNED(32, float, dst1, [MAX_SIZE]);
float val1, val2, min, max;
int i, len;
declare_func_emms(AV_CPU_FLAG_MMX, void, float *dst, const float *src,
int len, float min, float max);
val1 = (float)rnd() / (UINT_MAX >> 1) - 1.0f;
val2 = (float)rnd() / (UINT_MAX >> 1) - 1.0f;
min = FFMIN(val1, val2);
max = FFMAX(val1, val2);
randomize_float(src, MAX_SIZE);
len = rnd() % 128;
len = 16 * FFMAX(len, 1);
call_ref(dst0, src, len, min, max);
call_new(dst1, src, len, min, max);
for (i = 0; i < len; i++) {
if (!float_near_ulp_array(dst0, dst1, 3, len))
fail();
}
bench_new(dst1, src, MAX_SIZE, min, max);
}
report("audiodsp");
}
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