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Move some mpegaudio functions to new mpegaudiodsp subsystem

Browse Source 
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>
This commit is contained in:
Mans Rullgard 2011-05-16 16:52:01 +01:00
parent ea91e77127
commit c4f5c2d6f4
17 changed files with 390 additions and 255 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

19
configure vendored
View File

@ -952,6 +952,7 @@ CONFIG_LIST="
mdct
memalign_hack
mlib
mpegaudiodsp
network
nonfree
pic
@ -1235,6 +1236,7 @@ symver_if_any="symver_asm_label symver_gnu_asm"
dct_select="rdft"
mdct_select="fft"
rdft_select="fft"
mpegaudiodsp_select="dct"
# decoders / encoders / hardware accelerators
aac_decoder_select="mdct sinewin"
@ -1286,11 +1288,16 @@ ljpeg_encoder_select="aandct"
loco_decoder_select="golomb"
mjpeg_encoder_select="aandct"
mlp_decoder_select="mlp_parser"
mp1float_decoder_select="dct"
mp2float_decoder_select="dct"
mp3adufloat_decoder_select="dct"
mp3float_decoder_select="dct"
mp3on4float_decoder_select="dct"
mp1_decoder_select="mpegaudiodsp"
mp2_decoder_select="mpegaudiodsp"
mp3adu_decoder_select="mpegaudiodsp"
mp3_decoder_select="mpegaudiodsp"
mp3on4_decoder_select="mpegaudiodsp"
mp1float_decoder_select="mpegaudiodsp"
mp2float_decoder_select="mpegaudiodsp"
mp3adufloat_decoder_select="mpegaudiodsp"
mp3float_decoder_select="mpegaudiodsp"
mp3on4float_decoder_select="mpegaudiodsp"
mpeg1video_encoder_select="aandct"
mpeg2video_encoder_select="aandct"
mpeg4_decoder_select="h263_decoder mpeg4video_parser"
@ -1315,7 +1322,7 @@ nellymoser_encoder_select="mdct sinewin"
png_decoder_select="zlib"
png_encoder_select="zlib"
qcelp_decoder_select="lsp"
qdm2_decoder_select="mdct rdft"
qdm2_decoder_select="mdct rdft mpegaudiodsp"
ra_144_encoder_select="lpc"
rv10_decoder_select="h263_decoder"
rv10_encoder_select="h263_encoder"

3
libavcodec/Makefile
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@ -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

4
libavcodec/mpc.c
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@ -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]);

1
libavcodec/mpc.h
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@ -52,6 +52,7 @@ typedef struct {
typedef struct {
DSPContext dsp;
MPADSPContext mpadsp;
GetBitContext gb;
int IS, MSS, gapless;
int lastframelen;

3
libavcodec/mpc7.c
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@ -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);

3
libavcodec/mpc8.c
View File

@ -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();

24
libavcodec/mpegaudio.h
View File

@ -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){

197
libavcodec/mpegaudiodec.c
View File

@ -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,

19
libavcodec/mpegaudiodec_float.c
View File

@ -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)
{

40
libavcodec/mpegaudiodsp.c Normal file
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@ -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);
}

63
libavcodec/mpegaudiodsp.h Normal file
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@ -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

20
libavcodec/mpegaudiodsp_fixed.c Normal file
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@ -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"

20
libavcodec/mpegaudiodsp_float.c Normal file
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@ -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"

205
libavcodec/mpegaudiodsp_template.c Normal file
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@ -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];
}

9
libavcodec/ppc/mpegaudiodec_altivec.c
View File

@ -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;
}

6
libavcodec/qdm2.c
View File

@ -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);

9
libavcodec/x86/mpegaudiodec_mmx.c
View File

@ -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;
}
}