mirror of
https://github.com/xenia-project/FFmpeg.git
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bed12e24ff
To the best of my knowledge the author has not agreed to the change from ffmpeg->libav thus i revert it.
206 lines
5.6 KiB
C
206 lines
5.6 KiB
C
/*
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* Copyright (c) 2001, 2002 Fabrice Bellard
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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 <stdint.h>
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#include "libavutil/mem.h"
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#include "dct32.h"
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#include "mathops.h"
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#include "mpegaudiodsp.h"
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#include "mpegaudio.h"
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#include "mpegaudiodata.h"
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#if CONFIG_FLOAT
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#define RENAME(n) n##_float
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static inline float round_sample(float *sum)
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{
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float sum1=*sum;
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*sum = 0;
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return sum1;
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}
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#define MACS(rt, ra, rb) rt+=(ra)*(rb)
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#define MULS(ra, rb) ((ra)*(rb))
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#define MLSS(rt, ra, rb) rt-=(ra)*(rb)
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#else
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#define RENAME(n) n##_fixed
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#define OUT_SHIFT (WFRAC_BITS + FRAC_BITS - 15)
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static inline int round_sample(int64_t *sum)
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{
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int sum1;
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sum1 = (int)((*sum) >> OUT_SHIFT);
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*sum &= (1<<OUT_SHIFT)-1;
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return av_clip_int16(sum1);
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}
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# define MULS(ra, rb) MUL64(ra, rb)
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# define MACS(rt, ra, rb) MAC64(rt, ra, rb)
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# define MLSS(rt, ra, rb) MLS64(rt, ra, rb)
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#endif
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DECLARE_ALIGNED(16, MPA_INT, RENAME(ff_mpa_synth_window))[512+256];
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#define SUM8(op, sum, w, p) \
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{ \
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op(sum, (w)[0 * 64], (p)[0 * 64]); \
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op(sum, (w)[1 * 64], (p)[1 * 64]); \
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op(sum, (w)[2 * 64], (p)[2 * 64]); \
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op(sum, (w)[3 * 64], (p)[3 * 64]); \
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op(sum, (w)[4 * 64], (p)[4 * 64]); \
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op(sum, (w)[5 * 64], (p)[5 * 64]); \
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op(sum, (w)[6 * 64], (p)[6 * 64]); \
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op(sum, (w)[7 * 64], (p)[7 * 64]); \
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}
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#define SUM8P2(sum1, op1, sum2, op2, w1, w2, p) \
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{ \
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INTFLOAT tmp;\
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tmp = p[0 * 64];\
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op1(sum1, (w1)[0 * 64], tmp);\
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op2(sum2, (w2)[0 * 64], tmp);\
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tmp = p[1 * 64];\
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op1(sum1, (w1)[1 * 64], tmp);\
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op2(sum2, (w2)[1 * 64], tmp);\
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tmp = p[2 * 64];\
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op1(sum1, (w1)[2 * 64], tmp);\
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op2(sum2, (w2)[2 * 64], tmp);\
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tmp = p[3 * 64];\
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op1(sum1, (w1)[3 * 64], tmp);\
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op2(sum2, (w2)[3 * 64], tmp);\
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tmp = p[4 * 64];\
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op1(sum1, (w1)[4 * 64], tmp);\
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op2(sum2, (w2)[4 * 64], tmp);\
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tmp = p[5 * 64];\
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op1(sum1, (w1)[5 * 64], tmp);\
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op2(sum2, (w2)[5 * 64], tmp);\
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tmp = p[6 * 64];\
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op1(sum1, (w1)[6 * 64], tmp);\
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op2(sum2, (w2)[6 * 64], tmp);\
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tmp = p[7 * 64];\
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op1(sum1, (w1)[7 * 64], tmp);\
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op2(sum2, (w2)[7 * 64], tmp);\
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}
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void RENAME(ff_mpadsp_apply_window)(MPA_INT *synth_buf, MPA_INT *window,
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int *dither_state, OUT_INT *samples,
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int incr)
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{
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register const MPA_INT *w, *w2, *p;
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int j;
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OUT_INT *samples2;
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#if CONFIG_FLOAT
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float sum, sum2;
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#else
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int64_t sum, sum2;
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#endif
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/* copy to avoid wrap */
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memcpy(synth_buf + 512, synth_buf, 32 * sizeof(*synth_buf));
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samples2 = samples + 31 * incr;
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w = window;
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w2 = window + 31;
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sum = *dither_state;
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p = synth_buf + 16;
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SUM8(MACS, sum, w, p);
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p = synth_buf + 48;
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SUM8(MLSS, sum, w + 32, p);
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*samples = round_sample(&sum);
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samples += incr;
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w++;
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/* we calculate two samples at the same time to avoid one memory
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access per two sample */
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for(j=1;j<16;j++) {
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sum2 = 0;
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p = synth_buf + 16 + j;
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SUM8P2(sum, MACS, sum2, MLSS, w, w2, p);
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p = synth_buf + 48 - j;
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SUM8P2(sum, MLSS, sum2, MLSS, w + 32, w2 + 32, p);
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*samples = round_sample(&sum);
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samples += incr;
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sum += sum2;
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*samples2 = round_sample(&sum);
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samples2 -= incr;
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w++;
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w2--;
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}
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p = synth_buf + 32;
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SUM8(MLSS, sum, w + 32, p);
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*samples = round_sample(&sum);
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*dither_state= sum;
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}
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/* 32 sub band synthesis filter. Input: 32 sub band samples, Output:
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32 samples. */
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void RENAME(ff_mpa_synth_filter)(MPADSPContext *s, MPA_INT *synth_buf_ptr,
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int *synth_buf_offset,
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MPA_INT *window, int *dither_state,
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OUT_INT *samples, int incr,
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MPA_INT *sb_samples)
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{
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MPA_INT *synth_buf;
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int offset;
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offset = *synth_buf_offset;
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synth_buf = synth_buf_ptr + offset;
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s->RENAME(dct32)(synth_buf, sb_samples);
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s->RENAME(apply_window)(synth_buf, window, dither_state, samples, incr);
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offset = (offset - 32) & 511;
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*synth_buf_offset = offset;
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}
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void av_cold RENAME(ff_mpa_synth_init)(MPA_INT *window)
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{
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int i, j;
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/* max = 18760, max sum over all 16 coefs : 44736 */
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for(i=0;i<257;i++) {
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INTFLOAT v;
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v = ff_mpa_enwindow[i];
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#if CONFIG_FLOAT
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v *= 1.0 / (1LL<<(16 + FRAC_BITS));
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#endif
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window[i] = v;
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if ((i & 63) != 0)
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v = -v;
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if (i != 0)
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window[512 - i] = v;
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}
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// Needed for avoiding shuffles in ASM implementations
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for(i=0; i < 8; i++)
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for(j=0; j < 16; j++)
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window[512+16*i+j] = window[64*i+32-j];
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for(i=0; i < 8; i++)
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for(j=0; j < 16; j++)
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window[512+128+16*i+j] = window[64*i+48-j];
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}
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