mirror of https://github.com/FFmpeg/FFmpeg.git
* qatar/master: qdm2: Use floating point synthesis filter. h264: correct border check. h264: fix loopfilter with threading at slice boundaries. Fix ff_mpa_synth_filter_fixed() prototype Rename costablegen.c ---> cos_tablegen.c. Collapse tableprint.c into tableprint.h. Simplify trig table rules Remove potentially unstable filenames from comments in generated files. Ignore generated tables and generated table generator programs. Simplify CLEANFILES make variable by using wildcards. Remove silly insults from avformat_version() Doxygen documentation. mpegaudiodsp: fix x86 and ppc makefiles configure: Adjust AVX assembler check. mpegaudio: remove unused version of SAME_HEADER_MASK mpegaudio: remove useless #undef at end of file asfdec: add missing #include for av_bswap32() mpegaudio: merge two #if CONFIG_FLOAT blocks mpegaudio: move some struct definitions from mpegaudio.h Move some mpegaudio functions to new mpegaudiodsp subsystem Conflicts: libavcodec/h264.c libavcodec/x86/Makefile Merged-by: Michael Niedermayer <michaelni@gmx.at>pull/2/head
Michael Niedermayer
14 years ago
commit
80d156d7fd
28 changed files with 489 additions and 408 deletions
@ -0,0 +1,63 @@ |
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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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|
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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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int32_t *sb_samples); |
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|
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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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|
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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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@ -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 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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