mirror of https://github.com/FFmpeg/FFmpeg.git
This separation allows these functions to be used in a cleaner fashion from other codecs (e.g. qdm2) and simplifies creating optimised versions of them. Signed-off-by: Mans Rullgard <mans@mansr.com>pull/2/head
Mans Rullgard
14 years ago
parent
ea91e77127
commit
c4f5c2d6f4
17 changed files with 390 additions and 255 deletions
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/*
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* Copyright (c) 2011 Mans Rullgard |
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* |
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* This file is part of Libav. |
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* |
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* Libav 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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* Libav 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 Libav; 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 "config.h" |
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#include "mpegaudiodsp.h" |
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#include "dct.h" |
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#include "dct32.h" |
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void ff_mpadsp_init(MPADSPContext *s) |
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{ |
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DCTContext dct; |
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ff_dct_init(&dct, 5, DCT_II); |
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s->apply_window_float = ff_mpadsp_apply_window_float; |
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s->apply_window_fixed = ff_mpadsp_apply_window_fixed; |
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s->dct32_float = dct.dct32; |
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s->dct32_fixed = ff_dct32_fixed; |
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if (HAVE_MMX) ff_mpadsp_init_mmx(s); |
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if (HAVE_ALTIVEC) ff_mpadsp_init_altivec(s); |
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} |
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/*
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* This file is part of Libav. |
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* |
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* Libav 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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* Libav 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 Libav; 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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#ifndef AVCODEC_MPEGAUDIODSP_H |
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#define AVCODEC_MPEGAUDIODSP_H |
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#include <stdint.h> |
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typedef struct MPADSPContext { |
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void (*apply_window_float)(float *synth_buf, float *window, |
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int *dither_state, float *samples, int incr); |
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void (*apply_window_fixed)(int32_t *synth_buf, int32_t *window, |
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int *dither_state, int16_t *samples, int incr); |
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void (*dct32_float)(float *dst, const float *src); |
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void (*dct32_fixed)(int *dst, const int *src); |
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} MPADSPContext; |
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void ff_mpadsp_init(MPADSPContext *s); |
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extern int32_t ff_mpa_synth_window_fixed[]; |
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extern float ff_mpa_synth_window_float[]; |
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void ff_mpa_synth_filter_fixed(MPADSPContext *s, |
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int32_t *synth_buf_ptr, int *synth_buf_offset, |
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int32_t *window, int *dither_state, |
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int16_t *samples, int incr, |
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int *sb_samples); |
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void ff_mpa_synth_filter_float(MPADSPContext *s, |
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float *synth_buf_ptr, int *synth_buf_offset, |
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float *window, int *dither_state, |
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float *samples, int incr, |
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float *sb_samples); |
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void ff_mpadsp_init_mmx(MPADSPContext *s); |
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void ff_mpadsp_init_altivec(MPADSPContext *s); |
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void ff_mpa_synth_init_float(float *window); |
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void ff_mpa_synth_init_fixed(int32_t *window); |
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void ff_mpadsp_apply_window_float(float *synth_buf, float *window, |
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int *dither_state, float *samples, |
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int incr); |
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void ff_mpadsp_apply_window_fixed(int32_t *synth_buf, int32_t *window, |
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int *dither_state, int16_t *samples, |
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int incr); |
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#endif |
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@ -0,0 +1,20 @@ |
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/*
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* This file is part of Libav. |
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* |
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* Libav 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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* Libav 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 Libav; 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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#define CONFIG_FLOAT 0 |
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#include "mpegaudiodsp_template.c" |
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/*
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* This file is part of Libav. |
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* |
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* Libav 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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* Libav 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 Libav; 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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#define CONFIG_FLOAT 1 |
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#include "mpegaudiodsp_template.c" |
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@ -0,0 +1,205 @@ |
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/*
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* Copyright (c) 2001, 2002 Fabrice Bellard |
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* |
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* This file is part of Libav. |
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* |
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* Libav 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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* Libav 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 Libav; 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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Reference in new issue