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683da86aab
This will make the x86 asm simpler. ARM conversion by Martin Storsjö <martin@martin.st> and Janne Grunau <janne-libav@jannau.net>
147 lines
5.0 KiB
C
147 lines
5.0 KiB
C
/*
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* This file is part of Libav.
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*
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* Libav is free software; you can redistribute it and/or modify
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* it under the terms of the GNU General Public License as published by
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* the Free Software Foundation; either version 2 of the License, or
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* (at your option) any later version.
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*
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* Libav is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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* GNU General Public License for more details.
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*
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* You should have received a copy of the GNU General Public License along
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* with Libav; if not, write to the Free Software Foundation, Inc.,
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* 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
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*/
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#include <math.h>
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#include <string.h>
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#include <stdio.h>
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#include <stdlib.h>
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#include "libavcodec/audiodsp.h"
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#include "libavutil/common.h"
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#include "libavutil/intreadwrite.h"
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#include "checkasm.h"
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#define MAX_SIZE (32 * 128)
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#define randomize_float(buf, len) \
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do { \
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int i; \
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for (i = 0; i < len; i++) { \
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float f = (float)rnd() / (UINT_MAX >> 5) - 16.0f; \
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buf[i] = f; \
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} \
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} while (0)
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#define randomize_int(buf, len, size, bits) \
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do { \
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int i; \
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for (i = 0; i < len; i++) { \
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uint ## size ## _t r = rnd() & ((1LL << bits) - 1); \
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AV_WN ## size ## A(buf + i, -(1LL << (bits - 1)) + r); \
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} \
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} while (0)
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void checkasm_check_audiodsp(void)
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{
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AudioDSPContext adsp;
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ff_audiodsp_init(&adsp);
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if (check_func(adsp.scalarproduct_int16, "audiodsp.scalarproduct_int16")) {
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LOCAL_ALIGNED(32, int16_t, v1, [MAX_SIZE]);
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LOCAL_ALIGNED(32, int16_t, v2, [MAX_SIZE]);
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unsigned int len_bits_minus4, v1_bits, v2_bits, len;
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int32_t res0, res1;
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declare_func_emms(AV_CPU_FLAG_MMX, int32_t, const int16_t *v1, const int16_t *v2, int len);
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// generate random 5-12bit vector length
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len_bits_minus4 = rnd() % 8;
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len = rnd() & ((1 << len_bits_minus4) - 1);
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len = 16 * FFMAX(len, 1);
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// generate the bit counts for each of the vectors such that the result
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// fits into int32
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v1_bits = 1 + rnd() % 15;
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v2_bits = FFMIN(32 - (len_bits_minus4 + 4) - v1_bits - 1, 15);
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randomize_int(v1, MAX_SIZE, 16, v1_bits + 1);
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randomize_int(v2, MAX_SIZE, 16, v2_bits + 1);
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res0 = call_ref(v1, v2, len);
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res1 = call_new(v1, v2, len);
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if (res0 != res1)
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fail();
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bench_new(v1, v2, MAX_SIZE);
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}
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if (check_func(adsp.vector_clip_int32, "audiodsp.vector_clip_int32")) {
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LOCAL_ALIGNED(32, int32_t, src, [MAX_SIZE]);
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LOCAL_ALIGNED(32, int32_t, dst0, [MAX_SIZE]);
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LOCAL_ALIGNED(32, int32_t, dst1, [MAX_SIZE]);
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int32_t val1, val2, min, max;
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int len;
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declare_func_emms(AV_CPU_FLAG_MMX, void, int32_t *dst, const int32_t *src,
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int32_t min, int32_t max, unsigned int len);
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val1 = ((int32_t)rnd());
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val1 = FFSIGN(val1) * (val1 & ((1 << 24) - 1));
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val2 = ((int32_t)rnd());
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val2 = FFSIGN(val2) * (val2 & ((1 << 24) - 1));
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min = FFMIN(val1, val2);
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max = FFMAX(val1, val2);
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randomize_int(src, MAX_SIZE, 32, 32);
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len = rnd() % 128;
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len = 32 * FFMAX(len, 1);
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call_ref(dst0, src, min, max, len);
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call_new(dst1, src, min, max, len);
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if (memcmp(dst0, dst1, len * sizeof(*dst0)))
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fail();
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bench_new(dst1, src, min, max, MAX_SIZE);
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}
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if (check_func(adsp.vector_clipf, "audiodsp.vector_clipf")) {
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LOCAL_ALIGNED(32, float, src, [MAX_SIZE]);
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LOCAL_ALIGNED(32, float, dst0, [MAX_SIZE]);
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LOCAL_ALIGNED(32, float, dst1, [MAX_SIZE]);
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float val1, val2, min, max;
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int i, len;
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declare_func_emms(AV_CPU_FLAG_MMX, void, float *dst, const float *src,
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int len, float min, float max);
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val1 = (float)rnd() / (UINT_MAX >> 1) - 1.0f;
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val2 = (float)rnd() / (UINT_MAX >> 1) - 1.0f;
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min = FFMIN(val1, val2);
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max = FFMAX(val1, val2);
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randomize_float(src, MAX_SIZE);
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len = rnd() % 128;
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len = 16 * FFMAX(len, 1);
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call_ref(dst0, src, len, min, max);
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call_new(dst1, src, len, min, max);
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for (i = 0; i < len; i++) {
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if (!float_near_ulp_array(dst0, dst1, 3, len))
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fail();
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}
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bench_new(dst1, src, MAX_SIZE, min, max);
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}
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report("audiodsp");
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}
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