mirror of
https://github.com/FFmpeg/FFmpeg.git
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fe2ff6d247
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>
(cherry picked from commit c73d99e672
)
143 lines
4.4 KiB
C
143 lines
4.4 KiB
C
/*
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* Copyright (c) 2006 Luca Barbato <lu_zero@gentoo.org>
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg 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 GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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#include "libavcodec/fmtconvert.h"
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#include "dsputil_altivec.h"
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#include "util_altivec.h"
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static void int32_to_float_fmul_scalar_altivec(float *dst, const int *src, float mul, int len)
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{
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union {
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vector float v;
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float s[4];
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} mul_u;
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int i;
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vector float src1, src2, dst1, dst2, mul_v, zero;
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zero = (vector float)vec_splat_u32(0);
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mul_u.s[0] = mul;
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mul_v = vec_splat(mul_u.v, 0);
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for(i=0; i<len; i+=8) {
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src1 = vec_ctf(vec_ld(0, src+i), 0);
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src2 = vec_ctf(vec_ld(16, src+i), 0);
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dst1 = vec_madd(src1, mul_v, zero);
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dst2 = vec_madd(src2, mul_v, zero);
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vec_st(dst1, 0, dst+i);
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vec_st(dst2, 16, dst+i);
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}
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}
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static vector signed short
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float_to_int16_one_altivec(const float *src)
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{
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vector float s0 = vec_ld(0, src);
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vector float s1 = vec_ld(16, src);
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vector signed int t0 = vec_cts(s0, 0);
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vector signed int t1 = vec_cts(s1, 0);
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return vec_packs(t0,t1);
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}
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static void float_to_int16_altivec(int16_t *dst, const float *src, long len)
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{
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int i;
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vector signed short d0, d1, d;
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vector unsigned char align;
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if(((long)dst)&15) //FIXME
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for(i=0; i<len-7; i+=8) {
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d0 = vec_ld(0, dst+i);
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d = float_to_int16_one_altivec(src+i);
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d1 = vec_ld(15, dst+i);
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d1 = vec_perm(d1, d0, vec_lvsl(0,dst+i));
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align = vec_lvsr(0, dst+i);
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d0 = vec_perm(d1, d, align);
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d1 = vec_perm(d, d1, align);
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vec_st(d0, 0, dst+i);
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vec_st(d1,15, dst+i);
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}
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else
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for(i=0; i<len-7; i+=8) {
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d = float_to_int16_one_altivec(src+i);
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vec_st(d, 0, dst+i);
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}
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}
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static void
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float_to_int16_interleave_altivec(int16_t *dst, const float **src,
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long len, int channels)
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{
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int i;
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vector signed short d0, d1, d2, c0, c1, t0, t1;
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vector unsigned char align;
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if(channels == 1)
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float_to_int16_altivec(dst, src[0], len);
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else
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if (channels == 2) {
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if(((long)dst)&15)
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for(i=0; i<len-7; i+=8) {
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d0 = vec_ld(0, dst + i);
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t0 = float_to_int16_one_altivec(src[0] + i);
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d1 = vec_ld(31, dst + i);
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t1 = float_to_int16_one_altivec(src[1] + i);
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c0 = vec_mergeh(t0, t1);
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c1 = vec_mergel(t0, t1);
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d2 = vec_perm(d1, d0, vec_lvsl(0, dst + i));
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align = vec_lvsr(0, dst + i);
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d0 = vec_perm(d2, c0, align);
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d1 = vec_perm(c0, c1, align);
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vec_st(d0, 0, dst + i);
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d0 = vec_perm(c1, d2, align);
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vec_st(d1, 15, dst + i);
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vec_st(d0, 31, dst + i);
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dst+=8;
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}
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else
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for(i=0; i<len-7; i+=8) {
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t0 = float_to_int16_one_altivec(src[0] + i);
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t1 = float_to_int16_one_altivec(src[1] + i);
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d0 = vec_mergeh(t0, t1);
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d1 = vec_mergel(t0, t1);
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vec_st(d0, 0, dst + i);
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vec_st(d1, 16, dst + i);
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dst+=8;
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}
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} else {
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DECLARE_ALIGNED(16, int16_t, tmp)[len];
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int c, j;
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for (c = 0; c < channels; c++) {
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float_to_int16_altivec(tmp, src[c], len);
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for (i = 0, j = c; i < len; i++, j+=channels) {
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dst[j] = tmp[i];
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}
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}
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}
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}
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void ff_fmt_convert_init_ppc(FmtConvertContext *c, AVCodecContext *avctx)
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{
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c->int32_to_float_fmul_scalar = int32_to_float_fmul_scalar_altivec;
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if(!(avctx->flags & CODEC_FLAG_BITEXACT)) {
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c->float_to_int16 = float_to_int16_altivec;
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c->float_to_int16_interleave = float_to_int16_interleave_altivec;
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}
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}
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