diff options
author | Mans Rullgard <mans@mansr.com> | 2011-05-16 19:52:01 +0400 |
---|---|---|
committer | Mans Rullgard <mans@mansr.com> | 2011-05-19 15:25:34 +0400 |
commit | c4f5c2d6f4ffa3f4b56555059000208a6ba47b55 (patch) | |
tree | f46c4f0d94a1e073ac0dae24fab4d1d972bcb2c6 /libavcodec | |
parent | ea91e77127229015d23a046f1797d3fc6a33e54d (diff) |
Move some mpegaudio functions to new mpegaudiodsp subsystem
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>
Diffstat (limited to 'libavcodec')
-rw-r--r-- | libavcodec/Makefile | 3 | ||||
-rw-r--r-- | libavcodec/mpc.c | 4 | ||||
-rw-r--r-- | libavcodec/mpc.h | 1 | ||||
-rw-r--r-- | libavcodec/mpc7.c | 3 | ||||
-rw-r--r-- | libavcodec/mpc8.c | 3 | ||||
-rw-r--r-- | libavcodec/mpegaudio.h | 24 | ||||
-rw-r--r-- | libavcodec/mpegaudiodec.c | 197 | ||||
-rw-r--r-- | libavcodec/mpegaudiodec_float.c | 19 | ||||
-rw-r--r-- | libavcodec/mpegaudiodsp.c | 40 | ||||
-rw-r--r-- | libavcodec/mpegaudiodsp.h | 63 | ||||
-rw-r--r-- | libavcodec/mpegaudiodsp_fixed.c | 20 | ||||
-rw-r--r-- | libavcodec/mpegaudiodsp_float.c | 20 | ||||
-rw-r--r-- | libavcodec/mpegaudiodsp_template.c | 205 | ||||
-rw-r--r-- | libavcodec/ppc/mpegaudiodec_altivec.c | 9 | ||||
-rw-r--r-- | libavcodec/qdm2.c | 6 | ||||
-rw-r--r-- | libavcodec/x86/mpegaudiodec_mmx.c | 9 |
16 files changed, 377 insertions, 249 deletions
diff --git a/libavcodec/Makefile b/libavcodec/Makefile index fa70216c9c..b26c33de63 100644 --- a/libavcodec/Makefile +++ b/libavcodec/Makefile @@ -40,6 +40,9 @@ OBJS-$(CONFIG_HUFFMAN) += huffman.o OBJS-$(CONFIG_LPC) += lpc.o OBJS-$(CONFIG_LSP) += lsp.o OBJS-$(CONFIG_MDCT) += mdct_fixed.o mdct_float.o +OBJS-$(CONFIG_MPEGAUDIODSP) += mpegaudiodsp.o \ + mpegaudiodsp_fixed.o \ + mpegaudiodsp_float.o RDFT-OBJS-$(CONFIG_HARDCODED_TABLES) += sin_tables.o OBJS-$(CONFIG_RDFT) += rdft.o $(RDFT-OBJS-yes) OBJS-$(CONFIG_SINEWIN) += sinewin.o diff --git a/libavcodec/mpc.c b/libavcodec/mpc.c index 15febefe0b..4573860525 100644 --- a/libavcodec/mpc.c +++ b/libavcodec/mpc.c @@ -29,6 +29,7 @@ #include "avcodec.h" #include "get_bits.h" #include "dsputil.h" +#include "mpegaudiodsp.h" #include "mpegaudio.h" #include "mpc.h" @@ -51,7 +52,8 @@ static void mpc_synth(MPCContext *c, int16_t *out, int channels) for(ch = 0; ch < channels; ch++){ samples_ptr = samples + ch; for(i = 0; i < SAMPLES_PER_BAND; i++) { - ff_mpa_synth_filter_fixed(c->synth_buf[ch], &(c->synth_buf_offset[ch]), + ff_mpa_synth_filter_fixed(&c->mpadsp, + c->synth_buf[ch], &(c->synth_buf_offset[ch]), ff_mpa_synth_window_fixed, &dither_state, samples_ptr, channels, c->sb_samples[ch][i]); diff --git a/libavcodec/mpc.h b/libavcodec/mpc.h index 67fc7feed0..eea4b6df36 100644 --- a/libavcodec/mpc.h +++ b/libavcodec/mpc.h @@ -52,6 +52,7 @@ typedef struct { typedef struct { DSPContext dsp; + MPADSPContext mpadsp; GetBitContext gb; int IS, MSS, gapless; int lastframelen; diff --git a/libavcodec/mpc7.c b/libavcodec/mpc7.c index 6a4bf57043..dbfa3c8636 100644 --- a/libavcodec/mpc7.c +++ b/libavcodec/mpc7.c @@ -29,7 +29,7 @@ #include "avcodec.h" #include "get_bits.h" #include "dsputil.h" -#include "mpegaudio.h" +#include "mpegaudiodsp.h" #include "libavutil/audioconvert.h" #include "mpc.h" @@ -68,6 +68,7 @@ static av_cold int mpc7_decode_init(AVCodecContext * avctx) memset(c->oldDSCF, 0, sizeof(c->oldDSCF)); av_lfg_init(&c->rnd, 0xDEADBEEF); dsputil_init(&c->dsp, avctx); + ff_mpadsp_init(&c->mpadsp); c->dsp.bswap_buf((uint32_t*)buf, (const uint32_t*)avctx->extradata, 4); ff_mpc_init(); init_get_bits(&gb, buf, 128); diff --git a/libavcodec/mpc8.c b/libavcodec/mpc8.c index 5de8c15b4c..81de9cf500 100644 --- a/libavcodec/mpc8.c +++ b/libavcodec/mpc8.c @@ -29,7 +29,7 @@ #include "avcodec.h" #include "get_bits.h" #include "dsputil.h" -#include "mpegaudio.h" +#include "mpegaudiodsp.h" #include "libavutil/audioconvert.h" #include "mpc.h" @@ -120,6 +120,7 @@ static av_cold int mpc8_decode_init(AVCodecContext * avctx) memset(c->oldDSCF, 0, sizeof(c->oldDSCF)); av_lfg_init(&c->rnd, 0xDEADBEEF); dsputil_init(&c->dsp, avctx); + ff_mpadsp_init(&c->mpadsp); ff_mpc_init(); diff --git a/libavcodec/mpegaudio.h b/libavcodec/mpegaudio.h index 3422b6df68..c33960e987 100644 --- a/libavcodec/mpegaudio.h +++ b/libavcodec/mpegaudio.h @@ -33,7 +33,6 @@ #include "avcodec.h" #include "get_bits.h" #include "dsputil.h" -#include "dct.h" /* max frame size, in samples */ #define MPA_FRAME_SIZE 1152 @@ -69,7 +68,6 @@ typedef float OUT_INT; #else typedef int16_t OUT_INT; -#define OUT_SHIFT (WFRAC_BITS + FRAC_BITS - 15) #endif #if CONFIG_FLOAT @@ -142,11 +140,7 @@ typedef struct MPADecodeContext { int dither_state; int error_recognition; AVCodecContext* avctx; -#if CONFIG_FLOAT - DCTContext dct; -#endif - void (*apply_window_mp3)(MPA_INT *synth_buf, MPA_INT *window, - int *dither_state, OUT_INT *samples, int incr); + MPADSPContext mpadsp; } MPADecodeContext; /* layer 3 huffman tables */ @@ -158,22 +152,6 @@ typedef struct HuffTable { int ff_mpa_l2_select_table(int bitrate, int nb_channels, int freq, int lsf); int ff_mpa_decode_header(AVCodecContext *avctx, uint32_t head, int *sample_rate, int *channels, int *frame_size, int *bitrate); -extern MPA_INT ff_mpa_synth_window_fixed[]; -void ff_mpa_synth_init_fixed(MPA_INT *window); -void ff_mpa_synth_filter_fixed(MPA_INT *synth_buf_ptr, int *synth_buf_offset, - MPA_INT *window, int *dither_state, - OUT_INT *samples, int incr, - INTFLOAT sb_samples[SBLIMIT]); - -void ff_mpa_synth_init_float(MPA_INT *window); -void ff_mpa_synth_filter_float(MPADecodeContext *s, - MPA_INT *synth_buf_ptr, int *synth_buf_offset, - MPA_INT *window, int *dither_state, - OUT_INT *samples, int incr, - INTFLOAT sb_samples[SBLIMIT]); - -void ff_mpegaudiodec_init_mmx(MPADecodeContext *s); -void ff_mpegaudiodec_init_altivec(MPADecodeContext *s); /* fast header check for resync */ static inline int ff_mpa_check_header(uint32_t header){ diff --git a/libavcodec/mpegaudiodec.c b/libavcodec/mpegaudiodec.c index 4802a04bc8..cc193c68d0 100644 --- a/libavcodec/mpegaudiodec.c +++ b/libavcodec/mpegaudiodec.c @@ -29,7 +29,7 @@ #include "get_bits.h" #include "dsputil.h" #include "mathops.h" -#include "dct32.h" +#include "mpegaudiodsp.h" /* * TODO: @@ -68,8 +68,6 @@ #include "mpegaudiodectab.h" static void RENAME(compute_antialias)(MPADecodeContext *s, GranuleDef *g); -static void apply_window_mp3_c(MPA_INT *synth_buf, MPA_INT *window, - int *dither_state, OUT_INT *samples, int incr); /* vlc structure for decoding layer 3 huffman tables */ static VLC huff_vlc[16]; @@ -119,8 +117,6 @@ static const int32_t scale_factor_mult2[3][3] = { SCALE_GEN(4.0 / 9.0), /* 9 steps */ }; -DECLARE_ALIGNED(16, MPA_INT, RENAME(ff_mpa_synth_window))[512+256]; - /** * Convert region offsets to region sizes and truncate * size to big_values. @@ -259,14 +255,8 @@ static av_cold int decode_init(AVCodecContext * avctx) int i, j, k; s->avctx = avctx; - s->apply_window_mp3 = apply_window_mp3_c; -#if HAVE_MMX && CONFIG_FLOAT - ff_mpegaudiodec_init_mmx(s); -#endif -#if CONFIG_FLOAT - ff_dct_init(&s->dct, 5, DCT_II); -#endif - if (HAVE_ALTIVEC && CONFIG_FLOAT) ff_mpegaudiodec_init_altivec(s); + + ff_mpadsp_init(&s->mpadsp); avctx->sample_fmt= OUT_FMT; s->error_recognition= avctx->error_recognition; @@ -461,183 +451,6 @@ static av_cold int decode_init(AVCodecContext * avctx) return 0; } - -#if CONFIG_FLOAT -static inline float round_sample(float *sum) -{ - float sum1=*sum; - *sum = 0; - return sum1; -} - -/* signed 16x16 -> 32 multiply add accumulate */ -#define MACS(rt, ra, rb) rt+=(ra)*(rb) - -/* signed 16x16 -> 32 multiply */ -#define MULS(ra, rb) ((ra)*(rb)) - -#define MLSS(rt, ra, rb) rt-=(ra)*(rb) - -#else - -static inline int round_sample(int64_t *sum) -{ - int sum1; - sum1 = (int)((*sum) >> OUT_SHIFT); - *sum &= (1<<OUT_SHIFT)-1; - return av_clip_int16(sum1); -} - -# define MULS(ra, rb) MUL64(ra, rb) -# define MACS(rt, ra, rb) MAC64(rt, ra, rb) -# define MLSS(rt, ra, rb) MLS64(rt, ra, rb) -#endif - -#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]); \ -} - -#define SUM8P2(sum1, op1, sum2, op2, w1, w2, p) \ -{ \ - INTFLOAT tmp;\ - tmp = p[0 * 64];\ - op1(sum1, (w1)[0 * 64], tmp);\ - op2(sum2, (w2)[0 * 64], tmp);\ - tmp = p[1 * 64];\ - op1(sum1, (w1)[1 * 64], tmp);\ - op2(sum2, (w2)[1 * 64], tmp);\ - tmp = p[2 * 64];\ - op1(sum1, (w1)[2 * 64], tmp);\ - op2(sum2, (w2)[2 * 64], tmp);\ - tmp = p[3 * 64];\ - op1(sum1, (w1)[3 * 64], tmp);\ - op2(sum2, (w2)[3 * 64], tmp);\ - tmp = p[4 * 64];\ - op1(sum1, (w1)[4 * 64], tmp);\ - op2(sum2, (w2)[4 * 64], tmp);\ - tmp = p[5 * 64];\ - op1(sum1, (w1)[5 * 64], tmp);\ - op2(sum2, (w2)[5 * 64], tmp);\ - tmp = p[6 * 64];\ - op1(sum1, (w1)[6 * 64], tmp);\ - op2(sum2, (w2)[6 * 64], tmp);\ - tmp = p[7 * 64];\ - op1(sum1, (w1)[7 * 64], tmp);\ - op2(sum2, (w2)[7 * 64], tmp);\ -} - -void av_cold RENAME(ff_mpa_synth_init)(MPA_INT *window) -{ - int i, j; - - /* max = 18760, max sum over all 16 coefs : 44736 */ - for(i=0;i<257;i++) { - INTFLOAT v; - v = ff_mpa_enwindow[i]; -#if CONFIG_FLOAT - v *= 1.0 / (1LL<<(16 + FRAC_BITS)); -#endif - window[i] = v; - if ((i & 63) != 0) - v = -v; - if (i != 0) - window[512 - i] = v; - } - - // Needed for avoiding shuffles in ASM implementations - for(i=0; i < 8; i++) - for(j=0; j < 16; j++) - window[512+16*i+j] = window[64*i+32-j]; - - for(i=0; i < 8; i++) - for(j=0; j < 16; j++) - window[512+128+16*i+j] = window[64*i+48-j]; -} - -static void apply_window_mp3_c(MPA_INT *synth_buf, MPA_INT *window, - int *dither_state, OUT_INT *samples, int incr) -{ - register const MPA_INT *w, *w2, *p; - int j; - OUT_INT *samples2; -#if CONFIG_FLOAT - float sum, sum2; -#else - int64_t sum, sum2; -#endif - - /* copy to avoid wrap */ - memcpy(synth_buf + 512, synth_buf, 32 * sizeof(*synth_buf)); - - samples2 = samples + 31 * incr; - w = window; - w2 = window + 31; - - sum = *dither_state; - p = synth_buf + 16; - SUM8(MACS, sum, w, p); - p = synth_buf + 48; - SUM8(MLSS, sum, w + 32, p); - *samples = round_sample(&sum); - samples += incr; - w++; - - /* we calculate two samples at the same time to avoid one memory - access per two sample */ - for(j=1;j<16;j++) { - sum2 = 0; - p = synth_buf + 16 + j; - SUM8P2(sum, MACS, sum2, MLSS, w, w2, p); - p = synth_buf + 48 - j; - SUM8P2(sum, MLSS, sum2, MLSS, w + 32, w2 + 32, p); - - *samples = round_sample(&sum); - samples += incr; - sum += sum2; - *samples2 = round_sample(&sum); - samples2 -= incr; - w++; - w2--; - } - - p = synth_buf + 32; - SUM8(MLSS, sum, w + 32, p); - *samples = round_sample(&sum); - *dither_state= sum; -} - - -/* 32 sub band synthesis filter. Input: 32 sub band samples, Output: - 32 samples. */ -/* XXX: optimize by avoiding ring buffer usage */ -#if !CONFIG_FLOAT -void ff_mpa_synth_filter_fixed(MPA_INT *synth_buf_ptr, int *synth_buf_offset, - MPA_INT *window, int *dither_state, - OUT_INT *samples, int incr, - INTFLOAT sb_samples[SBLIMIT]) -{ - register MPA_INT *synth_buf; - int offset; - - offset = *synth_buf_offset; - synth_buf = synth_buf_ptr + offset; - - ff_dct32_fixed(synth_buf, sb_samples); - apply_window_mp3_c(synth_buf, window, dither_state, samples, incr); - - offset = (offset - 32) & 511; - *synth_buf_offset = offset; -} -#endif - #define C3 FIXHR(0.86602540378443864676/2) /* 0.5 / cos(pi*(2*i+1)/36) */ @@ -1915,9 +1728,7 @@ static int mp_decode_frame(MPADecodeContext *s, samples_ptr = samples + ch; for(i=0;i<nb_frames;i++) { RENAME(ff_mpa_synth_filter)( -#if CONFIG_FLOAT - s, -#endif + &s->mpadsp, s->synth_buf[ch], &(s->synth_buf_offset[ch]), RENAME(ff_mpa_synth_window), &s->dither_state, samples_ptr, s->nb_channels, diff --git a/libavcodec/mpegaudiodec_float.c b/libavcodec/mpegaudiodec_float.c index 0ef85d19c1..94463a824e 100644 --- a/libavcodec/mpegaudiodec_float.c +++ b/libavcodec/mpegaudiodec_float.c @@ -22,25 +22,6 @@ #define CONFIG_FLOAT 1 #include "mpegaudiodec.c" -void ff_mpa_synth_filter_float(MPADecodeContext *s, float *synth_buf_ptr, - int *synth_buf_offset, - float *window, int *dither_state, - float *samples, int incr, - float sb_samples[SBLIMIT]) -{ - float *synth_buf; - int offset; - - offset = *synth_buf_offset; - synth_buf = synth_buf_ptr + offset; - - s->dct.dct32(synth_buf, sb_samples); - s->apply_window_mp3(synth_buf, window, dither_state, samples, incr); - - offset = (offset - 32) & 511; - *synth_buf_offset = offset; -} - static void compute_antialias_float(MPADecodeContext *s, GranuleDef *g) { diff --git a/libavcodec/mpegaudiodsp.c b/libavcodec/mpegaudiodsp.c new file mode 100644 index 0000000000..57fe962b91 --- /dev/null +++ b/libavcodec/mpegaudiodsp.c @@ -0,0 +1,40 @@ +/* + * Copyright (c) 2011 Mans Rullgard + * + * This file is part of Libav. + * + * Libav 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. + * + * Libav 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 Libav; if not, write to the Free Software + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA + */ + +#include "config.h" +#include "mpegaudiodsp.h" +#include "dct.h" +#include "dct32.h" + +void ff_mpadsp_init(MPADSPContext *s) +{ + DCTContext dct; + + ff_dct_init(&dct, 5, DCT_II); + + s->apply_window_float = ff_mpadsp_apply_window_float; + s->apply_window_fixed = ff_mpadsp_apply_window_fixed; + + s->dct32_float = dct.dct32; + s->dct32_fixed = ff_dct32_fixed; + + if (HAVE_MMX) ff_mpadsp_init_mmx(s); + if (HAVE_ALTIVEC) ff_mpadsp_init_altivec(s); +} diff --git a/libavcodec/mpegaudiodsp.h b/libavcodec/mpegaudiodsp.h new file mode 100644 index 0000000000..7b05b68eee --- /dev/null +++ b/libavcodec/mpegaudiodsp.h @@ -0,0 +1,63 @@ +/* + * This file is part of Libav. + * + * Libav 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. + * + * Libav 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 Libav; if not, write to the Free Software + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA + */ + +#ifndef AVCODEC_MPEGAUDIODSP_H +#define AVCODEC_MPEGAUDIODSP_H + +#include <stdint.h> + +typedef struct MPADSPContext { + void (*apply_window_float)(float *synth_buf, float *window, + int *dither_state, float *samples, int incr); + void (*apply_window_fixed)(int32_t *synth_buf, int32_t *window, + int *dither_state, int16_t *samples, int incr); + void (*dct32_float)(float *dst, const float *src); + void (*dct32_fixed)(int *dst, const int *src); +} MPADSPContext; + +void ff_mpadsp_init(MPADSPContext *s); + +extern int32_t ff_mpa_synth_window_fixed[]; +extern float ff_mpa_synth_window_float[]; + +void ff_mpa_synth_filter_fixed(MPADSPContext *s, + int32_t *synth_buf_ptr, int *synth_buf_offset, + int32_t *window, int *dither_state, + int16_t *samples, int incr, + int *sb_samples); + +void ff_mpa_synth_filter_float(MPADSPContext *s, + float *synth_buf_ptr, int *synth_buf_offset, + float *window, int *dither_state, + float *samples, int incr, + float *sb_samples); + +void ff_mpadsp_init_mmx(MPADSPContext *s); +void ff_mpadsp_init_altivec(MPADSPContext *s); + +void ff_mpa_synth_init_float(float *window); +void ff_mpa_synth_init_fixed(int32_t *window); + +void ff_mpadsp_apply_window_float(float *synth_buf, float *window, + int *dither_state, float *samples, + int incr); +void ff_mpadsp_apply_window_fixed(int32_t *synth_buf, int32_t *window, + int *dither_state, int16_t *samples, + int incr); + +#endif diff --git a/libavcodec/mpegaudiodsp_fixed.c b/libavcodec/mpegaudiodsp_fixed.c new file mode 100644 index 0000000000..3c49a568b7 --- /dev/null +++ b/libavcodec/mpegaudiodsp_fixed.c @@ -0,0 +1,20 @@ +/* + * This file is part of Libav. + * + * Libav 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. + * + * Libav 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 Libav; if not, write to the Free Software + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA + */ + +#define CONFIG_FLOAT 0 +#include "mpegaudiodsp_template.c" diff --git a/libavcodec/mpegaudiodsp_float.c b/libavcodec/mpegaudiodsp_float.c new file mode 100644 index 0000000000..2d8d53ea26 --- /dev/null +++ b/libavcodec/mpegaudiodsp_float.c @@ -0,0 +1,20 @@ +/* + * This file is part of Libav. + * + * Libav 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. + * + * Libav 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 Libav; if not, write to the Free Software + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA + */ + +#define CONFIG_FLOAT 1 +#include "mpegaudiodsp_template.c" diff --git a/libavcodec/mpegaudiodsp_template.c b/libavcodec/mpegaudiodsp_template.c new file mode 100644 index 0000000000..5561c46135 --- /dev/null +++ b/libavcodec/mpegaudiodsp_template.c @@ -0,0 +1,205 @@ +/* + * Copyright (c) 2001, 2002 Fabrice Bellard + * + * This file is part of Libav. + * + * Libav 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. + * + * Libav 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 Libav; if not, write to the Free Software + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA + */ + +#include <stdint.h> + +#include "libavutil/mem.h" +#include "dct32.h" +#include "mathops.h" +#include "mpegaudiodsp.h" +#include "mpegaudio.h" +#include "mpegaudiodata.h" + +#if CONFIG_FLOAT +#define RENAME(n) n##_float + +static inline float round_sample(float *sum) +{ + float sum1=*sum; + *sum = 0; + return sum1; +} + +#define MACS(rt, ra, rb) rt+=(ra)*(rb) +#define MULS(ra, rb) ((ra)*(rb)) +#define MLSS(rt, ra, rb) rt-=(ra)*(rb) + +#else + +#define RENAME(n) n##_fixed +#define OUT_SHIFT (WFRAC_BITS + FRAC_BITS - 15) + +static inline int round_sample(int64_t *sum) +{ + int sum1; + sum1 = (int)((*sum) >> OUT_SHIFT); + *sum &= (1<<OUT_SHIFT)-1; + return av_clip_int16(sum1); +} + +# define MULS(ra, rb) MUL64(ra, rb) +# define MACS(rt, ra, rb) MAC64(rt, ra, rb) +# define MLSS(rt, ra, rb) MLS64(rt, ra, rb) +#endif + +DECLARE_ALIGNED(16, MPA_INT, RENAME(ff_mpa_synth_window))[512+256]; + +#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]); \ +} + +#define SUM8P2(sum1, op1, sum2, op2, w1, w2, p) \ +{ \ + INTFLOAT tmp;\ + tmp = p[0 * 64];\ + op1(sum1, (w1)[0 * 64], tmp);\ + op2(sum2, (w2)[0 * 64], tmp);\ + tmp = p[1 * 64];\ + op1(sum1, (w1)[1 * 64], tmp);\ + op2(sum2, (w2)[1 * 64], tmp);\ + tmp = p[2 * 64];\ + op1(sum1, (w1)[2 * 64], tmp);\ + op2(sum2, (w2)[2 * 64], tmp);\ + tmp = p[3 * 64];\ + op1(sum1, (w1)[3 * 64], tmp);\ + op2(sum2, (w2)[3 * 64], tmp);\ + tmp = p[4 * 64];\ + op1(sum1, (w1)[4 * 64], tmp);\ + op2(sum2, (w2)[4 * 64], tmp);\ + tmp = p[5 * 64];\ + op1(sum1, (w1)[5 * 64], tmp);\ + op2(sum2, (w2)[5 * 64], tmp);\ + tmp = p[6 * 64];\ + op1(sum1, (w1)[6 * 64], tmp);\ + op2(sum2, (w2)[6 * 64], tmp);\ + tmp = p[7 * 64];\ + op1(sum1, (w1)[7 * 64], tmp);\ + op2(sum2, (w2)[7 * 64], tmp);\ +} + +void RENAME(ff_mpadsp_apply_window)(MPA_INT *synth_buf, MPA_INT *window, + int *dither_state, OUT_INT *samples, + int incr) +{ + register const MPA_INT *w, *w2, *p; + int j; + OUT_INT *samples2; +#if CONFIG_FLOAT + float sum, sum2; +#else + int64_t sum, sum2; +#endif + + /* copy to avoid wrap */ + memcpy(synth_buf + 512, synth_buf, 32 * sizeof(*synth_buf)); + + samples2 = samples + 31 * incr; + w = window; + w2 = window + 31; + + sum = *dither_state; + p = synth_buf + 16; + SUM8(MACS, sum, w, p); + p = synth_buf + 48; + SUM8(MLSS, sum, w + 32, p); + *samples = round_sample(&sum); + samples += incr; + w++; + + /* we calculate two samples at the same time to avoid one memory + access per two sample */ + for(j=1;j<16;j++) { + sum2 = 0; + p = synth_buf + 16 + j; + SUM8P2(sum, MACS, sum2, MLSS, w, w2, p); + p = synth_buf + 48 - j; + SUM8P2(sum, MLSS, sum2, MLSS, w + 32, w2 + 32, p); + + *samples = round_sample(&sum); + samples += incr; + sum += sum2; + *samples2 = round_sample(&sum); + samples2 -= incr; + w++; + w2--; + } + + p = synth_buf + 32; + SUM8(MLSS, sum, w + 32, p); + *samples = round_sample(&sum); + *dither_state= sum; +} + +/* 32 sub band synthesis filter. Input: 32 sub band samples, Output: + 32 samples. */ +void RENAME(ff_mpa_synth_filter)(MPADSPContext *s, MPA_INT *synth_buf_ptr, + int *synth_buf_offset, + MPA_INT *window, int *dither_state, + OUT_INT *samples, int incr, + MPA_INT *sb_samples) +{ + MPA_INT *synth_buf; + int offset; + + offset = *synth_buf_offset; + synth_buf = synth_buf_ptr + offset; + + s->RENAME(dct32)(synth_buf, sb_samples); + s->RENAME(apply_window)(synth_buf, window, dither_state, samples, incr); + + offset = (offset - 32) & 511; + *synth_buf_offset = offset; +} + +void av_cold RENAME(ff_mpa_synth_init)(MPA_INT *window) +{ + int i, j; + + /* max = 18760, max sum over all 16 coefs : 44736 */ + for(i=0;i<257;i++) { + INTFLOAT v; + v = ff_mpa_enwindow[i]; +#if CONFIG_FLOAT + v *= 1.0 / (1LL<<(16 + FRAC_BITS)); +#endif + window[i] = v; + if ((i & 63) != 0) + v = -v; + if (i != 0) + window[512 - i] = v; + } + + // Needed for avoiding shuffles in ASM implementations + for(i=0; i < 8; i++) + for(j=0; j < 16; j++) + window[512+16*i+j] = window[64*i+32-j]; + + for(i=0; i < 8; i++) + for(j=0; j < 16; j++) + window[512+128+16*i+j] = window[64*i+48-j]; +} diff --git a/libavcodec/ppc/mpegaudiodec_altivec.c b/libavcodec/ppc/mpegaudiodec_altivec.c index af94276e8a..5df0fdafe4 100644 --- a/libavcodec/ppc/mpegaudiodec_altivec.c +++ b/libavcodec/ppc/mpegaudiodec_altivec.c @@ -21,9 +21,8 @@ #include "dsputil_altivec.h" #include "util_altivec.h" - -#define CONFIG_FLOAT 1 -#include "libavcodec/mpegaudio.h" +#include "libavcodec/dsputil.h" +#include "libavcodec/mpegaudiodsp.h" #define MACS(rt, ra, rb) rt+=(ra)*(rb) #define MLSS(rt, ra, rb) rt-=(ra)*(rb) @@ -124,7 +123,7 @@ static void apply_window_mp3(float *in, float *win, int *unused, float *out, *out = sum; } -void ff_mpegaudiodec_init_altivec(MPADecodeContext *s) +void ff_mpadsp_init_altivec(MPADSPContext *s) { - s->apply_window_mp3 = apply_window_mp3; + s->apply_window_float = apply_window_mp3; } diff --git a/libavcodec/qdm2.c b/libavcodec/qdm2.c index 0f4dd18966..f74cfd9258 100644 --- a/libavcodec/qdm2.c +++ b/libavcodec/qdm2.c @@ -39,6 +39,7 @@ #include "get_bits.h" #include "dsputil.h" #include "rdft.h" +#include "mpegaudiodsp.h" #include "mpegaudio.h" #include "qdm2data.h" @@ -170,6 +171,7 @@ typedef struct { float output_buffer[1024]; /// Synthesis filter + MPADSPContext mpadsp; DECLARE_ALIGNED(16, MPA_INT, synth_buf)[MPA_MAX_CHANNELS][512*2]; int synth_buf_offset[MPA_MAX_CHANNELS]; DECLARE_ALIGNED(16, int32_t, sb_samples)[MPA_MAX_CHANNELS][128][SBLIMIT]; @@ -1616,7 +1618,8 @@ static void qdm2_synthesis_filter (QDM2Context *q, int index) OUT_INT *samples_ptr = samples + ch; for (i = 0; i < 8; i++) { - ff_mpa_synth_filter_fixed(q->synth_buf[ch], &(q->synth_buf_offset[ch]), + ff_mpa_synth_filter_fixed(&q->mpadsp, + q->synth_buf[ch], &(q->synth_buf_offset[ch]), ff_mpa_synth_window_fixed, &dither_state, samples_ptr, q->nb_channels, q->sb_samples[ch][(8 * index) + i]); @@ -1863,6 +1866,7 @@ static av_cold int qdm2_decode_init(AVCodecContext *avctx) } ff_rdft_init(&s->rdft_ctx, s->fft_order, IDFT_C2R); + ff_mpadsp_init(&s->mpadsp); qdm2_init(s); diff --git a/libavcodec/x86/mpegaudiodec_mmx.c b/libavcodec/x86/mpegaudiodec_mmx.c index ce5b7d6df8..b64461513e 100644 --- a/libavcodec/x86/mpegaudiodec_mmx.c +++ b/libavcodec/x86/mpegaudiodec_mmx.c @@ -21,9 +21,8 @@ #include "libavutil/cpu.h" #include "libavutil/x86_cpu.h" - -#define CONFIG_FLOAT 1 -#include "libavcodec/mpegaudio.h" +#include "libavcodec/dsputil.h" +#include "libavcodec/mpegaudiodsp.h" #define MACS(rt, ra, rb) rt+=(ra)*(rb) #define MLSS(rt, ra, rb) rt-=(ra)*(rb) @@ -148,11 +147,11 @@ static void apply_window_mp3(float *in, float *win, int *unused, float *out, *out = sum; } -void ff_mpegaudiodec_init_mmx(MPADecodeContext *s) +void ff_mpadsp_init_mmx(MPADSPContext *s) { int mm_flags = av_get_cpu_flags(); if (mm_flags & AV_CPU_FLAG_SSE2) { - s->apply_window_mp3 = apply_window_mp3; + s->apply_window_float = apply_window_mp3; } } |