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FFmpeg/libavcodec/audiodsp.c
Anton Khirnov 683da86aab audiodsp: reorder arguments for vector_clipf
This will make the x86 asm simpler.

ARM conversion by Martin Storsjö <martin@martin.st> and Janne Grunau
<janne-libav@jannau.net>
2016-09-22 09:47:52 +02:00

119 lines
3.9 KiB
C

/*
* This file is part of Libav.
*
* Libav is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 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
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser 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 <stdint.h>
#include "libavutil/attributes.h"
#include "libavutil/common.h"
#include "audiodsp.h"
static inline uint32_t clipf_c_one(uint32_t a, uint32_t mini,
uint32_t maxi, uint32_t maxisign)
{
if (a > mini)
return mini;
else if ((a ^ (1U << 31)) > maxisign)
return maxi;
else
return a;
}
static void vector_clipf_c_opposite_sign(float *dst, const float *src,
float *min, float *max, int len)
{
int i;
uint32_t mini = *(uint32_t *) min;
uint32_t maxi = *(uint32_t *) max;
uint32_t maxisign = maxi ^ (1U << 31);
uint32_t *dsti = (uint32_t *) dst;
const uint32_t *srci = (const uint32_t *) src;
for (i = 0; i < len; i += 8) {
dsti[i + 0] = clipf_c_one(srci[i + 0], mini, maxi, maxisign);
dsti[i + 1] = clipf_c_one(srci[i + 1], mini, maxi, maxisign);
dsti[i + 2] = clipf_c_one(srci[i + 2], mini, maxi, maxisign);
dsti[i + 3] = clipf_c_one(srci[i + 3], mini, maxi, maxisign);
dsti[i + 4] = clipf_c_one(srci[i + 4], mini, maxi, maxisign);
dsti[i + 5] = clipf_c_one(srci[i + 5], mini, maxi, maxisign);
dsti[i + 6] = clipf_c_one(srci[i + 6], mini, maxi, maxisign);
dsti[i + 7] = clipf_c_one(srci[i + 7], mini, maxi, maxisign);
}
}
static void vector_clipf_c(float *dst, const float *src, int len,
float min, float max)
{
int i;
if (min < 0 && max > 0) {
vector_clipf_c_opposite_sign(dst, src, &min, &max, len);
} else {
for (i = 0; i < len; i += 8) {
dst[i] = av_clipf(src[i], min, max);
dst[i + 1] = av_clipf(src[i + 1], min, max);
dst[i + 2] = av_clipf(src[i + 2], min, max);
dst[i + 3] = av_clipf(src[i + 3], min, max);
dst[i + 4] = av_clipf(src[i + 4], min, max);
dst[i + 5] = av_clipf(src[i + 5], min, max);
dst[i + 6] = av_clipf(src[i + 6], min, max);
dst[i + 7] = av_clipf(src[i + 7], min, max);
}
}
}
static int32_t scalarproduct_int16_c(const int16_t *v1, const int16_t *v2,
int order)
{
int res = 0;
while (order--)
res += *v1++ **v2++;
return res;
}
static void vector_clip_int32_c(int32_t *dst, const int32_t *src, int32_t min,
int32_t max, unsigned int len)
{
do {
*dst++ = av_clip(*src++, min, max);
*dst++ = av_clip(*src++, min, max);
*dst++ = av_clip(*src++, min, max);
*dst++ = av_clip(*src++, min, max);
*dst++ = av_clip(*src++, min, max);
*dst++ = av_clip(*src++, min, max);
*dst++ = av_clip(*src++, min, max);
*dst++ = av_clip(*src++, min, max);
len -= 8;
} while (len > 0);
}
av_cold void ff_audiodsp_init(AudioDSPContext *c)
{
c->scalarproduct_int16 = scalarproduct_int16_c;
c->vector_clip_int32 = vector_clip_int32_c;
c->vector_clipf = vector_clipf_c;
if (ARCH_ARM)
ff_audiodsp_init_arm(c);
if (ARCH_PPC)
ff_audiodsp_init_ppc(c);
if (ARCH_X86)
ff_audiodsp_init_x86(c);
}