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-rwxr-xr-xconfigure19
-rw-r--r--libavcodec/Makefile3
-rw-r--r--libavcodec/mpc.c4
-rw-r--r--libavcodec/mpc.h1
-rw-r--r--libavcodec/mpc7.c3
-rw-r--r--libavcodec/mpc8.c3
-rw-r--r--libavcodec/mpegaudio.h24
-rw-r--r--libavcodec/mpegaudiodec.c197
-rw-r--r--libavcodec/mpegaudiodec_float.c19
-rw-r--r--libavcodec/mpegaudiodsp.c40
-rw-r--r--libavcodec/mpegaudiodsp.h63
-rw-r--r--libavcodec/mpegaudiodsp_fixed.c20
-rw-r--r--libavcodec/mpegaudiodsp_float.c20
-rw-r--r--libavcodec/mpegaudiodsp_template.c205
-rw-r--r--libavcodec/ppc/mpegaudiodec_altivec.c9
-rw-r--r--libavcodec/qdm2.c6
-rw-r--r--libavcodec/x86/mpegaudiodec_mmx.c9
17 files changed, 390 insertions, 255 deletions
diff --git a/configure b/configure
index 5b81e0b599..6becb09724 100755
--- a/configure
+++ b/configure
@@ -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"
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;
}
}
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