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Move some mpegaudio functions to new mpegaudiodsp subsystem

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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
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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
View File

@ -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
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@ -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;
}
}