Mercurial > libavcodec.hg
view ppc/mpegaudiodec_altivec.c @ 12223:93e27a5401de libavcodec
Convert VP8 macroblock structures to a ring buffer.
Uses a slightly nonintuitive ring buffer size of (width+height*2) to simplify
addressing logic.
Also split out the segmentation map to a separate structure, necessary to
implement the ring buffer.
author | darkshikari |
---|---|
date | Thu, 22 Jul 2010 11:45:18 +0000 |
parents | b4888704c11e |
children |
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/* * Altivec optimized MP3 decoding functions * Copyright (c) 2010 Vitor Sessak * * 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 "dsputil_altivec.h" #include "util_altivec.h" #define CONFIG_FLOAT 1 #include "libavcodec/mpegaudio.h" #define MACS(rt, ra, rb) rt+=(ra)*(rb) #define MLSS(rt, ra, rb) rt-=(ra)*(rb) #define SUM8(op, sum, w, p) \ { \ op(sum, (w)[0 * 64], (p)[0 * 64]); \ op(sum, (w)[1 * 64], (p)[1 * 64]); \ op(sum, (w)[2 * 64], (p)[2 * 64]); \ op(sum, (w)[3 * 64], (p)[3 * 64]); \ op(sum, (w)[4 * 64], (p)[4 * 64]); \ op(sum, (w)[5 * 64], (p)[5 * 64]); \ op(sum, (w)[6 * 64], (p)[6 * 64]); \ op(sum, (w)[7 * 64], (p)[7 * 64]); \ } static void apply_window(const float *buf, const float *win1, const float *win2, float *sum1, float *sum2, int len) { const vector float *win1a = (const vector float *) win1; const vector float *win2a = (const vector float *) win2; const vector float *bufa = (const vector float *) buf; vector float *sum1a = (vector float *) sum1; vector float *sum2a = (vector float *) sum2; vector float av_uninit(v0), av_uninit(v4); vector float v1, v2, v3; len = len >> 2; #define MULT(a, b) \ { \ v1 = vec_ld(a, win1a); \ v2 = vec_ld(b, win2a); \ v3 = vec_ld(a, bufa); \ v0 = vec_madd(v3, v1, v0); \ v4 = vec_madd(v2, v3, v4); \ } while (len--) { v0 = vec_xor(v0, v0); v4 = vec_xor(v4, v4); MULT( 0, 0); MULT( 256, 64); MULT( 512, 128); MULT( 768, 192); MULT(1024, 256); MULT(1280, 320); MULT(1536, 384); MULT(1792, 448); vec_st(v0, 0, sum1a); vec_st(v4, 0, sum2a); sum1a++; sum2a++; win1a++; win2a++; bufa++; } } static void apply_window_mp3(float *in, float *win, int *unused, float *out, int incr) { LOCAL_ALIGNED_16(float, suma, [17]); LOCAL_ALIGNED_16(float, sumb, [17]); LOCAL_ALIGNED_16(float, sumc, [17]); LOCAL_ALIGNED_16(float, sumd, [17]); float sum; int j; float *out2 = out + 32 * incr; /* copy to avoid wrap */ memcpy(in + 512, in, 32 * sizeof(*in)); apply_window(in + 16, win , win + 512, suma, sumc, 16); apply_window(in + 32, win + 48, win + 640, sumb, sumd, 16); SUM8(MLSS, suma[0], win + 32, in + 48); sumc[ 0] = 0; sumb[16] = 0; sumd[16] = 0; out[0 ] = suma[ 0]; out += incr; out2 -= incr; for(j=1;j<16;j++) { *out = suma[ j] - sumd[16-j]; *out2 = -sumb[16-j] - sumc[ j]; out += incr; out2 -= incr; } sum = 0; SUM8(MLSS, sum, win + 16 + 32, in + 32); *out = sum; } void ff_mpegaudiodec_init_altivec(MPADecodeContext *s) { s->apply_window_mp3 = apply_window_mp3; }