view ppc/mpegaudiodec_altivec.c @ 12454:f4355cd85faa libavcodec

Port latest x264 deblock asm (before they moved to using NV12 as internal format), LGPL'ed with permission from Jason and Loren. This includes mmx2 code, so remove inline asm from h264dsp_mmx.c accordingly.

/*
 * 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;
}