ffmpeg/libavcodec/ppc/fmtconvert_altivec.c
Justin Ruggles c73d99e672 Separate format conversion DSP functions from DSPContext.
This will be beneficial for use with the audio conversion API without
requiring it to depend on all of dsputil.

Signed-off-by: Mans Rullgard <mans@mansr.com>
2011-02-02 02:44:53 +00:00

143 lines
4.4 KiB
C

/*
* Copyright (c) 2006 Luca Barbato <lu_zero@gentoo.org>
*
* This file is part of FFmpeg.
*
* FFmpeg 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.
*
* 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
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser 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 "libavcodec/fmtconvert.h"
#include "dsputil_altivec.h"
#include "util_altivec.h"
static void int32_to_float_fmul_scalar_altivec(float *dst, const int *src, float mul, int len)
{
union {
vector float v;
float s[4];
} mul_u;
int i;
vector float src1, src2, dst1, dst2, mul_v, zero;
zero = (vector float)vec_splat_u32(0);
mul_u.s[0] = mul;
mul_v = vec_splat(mul_u.v, 0);
for(i=0; i<len; i+=8) {
src1 = vec_ctf(vec_ld(0, src+i), 0);
src2 = vec_ctf(vec_ld(16, src+i), 0);
dst1 = vec_madd(src1, mul_v, zero);
dst2 = vec_madd(src2, mul_v, zero);
vec_st(dst1, 0, dst+i);
vec_st(dst2, 16, dst+i);
}
}
static vector signed short
float_to_int16_one_altivec(const float *src)
{
vector float s0 = vec_ld(0, src);
vector float s1 = vec_ld(16, src);
vector signed int t0 = vec_cts(s0, 0);
vector signed int t1 = vec_cts(s1, 0);
return vec_packs(t0,t1);
}
static void float_to_int16_altivec(int16_t *dst, const float *src, long len)
{
int i;
vector signed short d0, d1, d;
vector unsigned char align;
if(((long)dst)&15) //FIXME
for(i=0; i<len-7; i+=8) {
d0 = vec_ld(0, dst+i);
d = float_to_int16_one_altivec(src+i);
d1 = vec_ld(15, dst+i);
d1 = vec_perm(d1, d0, vec_lvsl(0,dst+i));
align = vec_lvsr(0, dst+i);
d0 = vec_perm(d1, d, align);
d1 = vec_perm(d, d1, align);
vec_st(d0, 0, dst+i);
vec_st(d1,15, dst+i);
}
else
for(i=0; i<len-7; i+=8) {
d = float_to_int16_one_altivec(src+i);
vec_st(d, 0, dst+i);
}
}
static void
float_to_int16_interleave_altivec(int16_t *dst, const float **src,
long len, int channels)
{
int i;
vector signed short d0, d1, d2, c0, c1, t0, t1;
vector unsigned char align;
if(channels == 1)
float_to_int16_altivec(dst, src[0], len);
else
if (channels == 2) {
if(((long)dst)&15)
for(i=0; i<len-7; i+=8) {
d0 = vec_ld(0, dst + i);
t0 = float_to_int16_one_altivec(src[0] + i);
d1 = vec_ld(31, dst + i);
t1 = float_to_int16_one_altivec(src[1] + i);
c0 = vec_mergeh(t0, t1);
c1 = vec_mergel(t0, t1);
d2 = vec_perm(d1, d0, vec_lvsl(0, dst + i));
align = vec_lvsr(0, dst + i);
d0 = vec_perm(d2, c0, align);
d1 = vec_perm(c0, c1, align);
vec_st(d0, 0, dst + i);
d0 = vec_perm(c1, d2, align);
vec_st(d1, 15, dst + i);
vec_st(d0, 31, dst + i);
dst+=8;
}
else
for(i=0; i<len-7; i+=8) {
t0 = float_to_int16_one_altivec(src[0] + i);
t1 = float_to_int16_one_altivec(src[1] + i);
d0 = vec_mergeh(t0, t1);
d1 = vec_mergel(t0, t1);
vec_st(d0, 0, dst + i);
vec_st(d1, 16, dst + i);
dst+=8;
}
} else {
DECLARE_ALIGNED(16, int16_t, tmp)[len];
int c, j;
for (c = 0; c < channels; c++) {
float_to_int16_altivec(tmp, src[c], len);
for (i = 0, j = c; i < len; i++, j+=channels) {
dst[j] = tmp[i];
}
}
}
}
void ff_fmt_convert_init_ppc(FmtConvertContext *c, AVCodecContext *avctx)
{
c->int32_to_float_fmul_scalar = int32_to_float_fmul_scalar_altivec;
if(!(avctx->flags & CODEC_FLAG_BITEXACT)) {
c->float_to_int16 = float_to_int16_altivec;
c->float_to_int16_interleave = float_to_int16_interleave_altivec;
}
}