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MP3: Support decode mp3 by FFmpeg natively. (#296) (#3340)

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winlin 2022-12-26 13:23:39 +08:00
parent fe3502e6ad
commit 1c5788c638
29 changed files with 6001 additions and 0 deletions
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225
trunk/3rdparty/ffmpeg-4-fit/libavcodec/dct.c vendored Normal file
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/*
* (I)DCT Transforms
* Copyright (c) 2009 Peter Ross <pross@xvid.org>
* Copyright (c) 2010 Alex Converse <alex.converse@gmail.com>
* Copyright (c) 2010 Vitor Sessak
*
* This file is part of FFmpeg.
*
* FFmpeg 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.
*
* FFmpeg 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 FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
* @file
* (Inverse) Discrete Cosine Transforms. These are also known as the
* type II and type III DCTs respectively.
*/
#include <math.h>
#include <string.h>
#include "libavutil/mathematics.h"
#include "dct.h"
#include "dct32.h"
/* sin((M_PI * x / (2 * n)) */
#define SIN(s, n, x) (s->costab[(n) - (x)])
/* cos((M_PI * x / (2 * n)) */
#define COS(s, n, x) (s->costab[x])
static void dst_calc_I_c(DCTContext *ctx, FFTSample *data)
{
int n = 1 << ctx->nbits;
int i;
data[0] = 0;
for (i = 1; i < n / 2; i++) {
float tmp1 = data[i ];
float tmp2 = data[n - i];
float s = SIN(ctx, n, 2 * i);
s *= tmp1 + tmp2;
tmp1 = (tmp1 - tmp2) * 0.5f;
data[i] = s + tmp1;
data[n - i] = s - tmp1;
}
data[n / 2] *= 2;
ctx->rdft.rdft_calc(&ctx->rdft, data);
data[0] *= 0.5f;
for (i = 1; i < n - 2; i += 2) {
data[i + 1] += data[i - 1];
data[i] = -data[i + 2];
}
data[n - 1] = 0;
}
static void dct_calc_I_c(DCTContext *ctx, FFTSample *data)
{
int n = 1 << ctx->nbits;
int i;
float next = -0.5f * (data[0] - data[n]);
for (i = 0; i < n / 2; i++) {
float tmp1 = data[i];
float tmp2 = data[n - i];
float s = SIN(ctx, n, 2 * i);
float c = COS(ctx, n, 2 * i);
c *= tmp1 - tmp2;
s *= tmp1 - tmp2;
next += c;
tmp1 = (tmp1 + tmp2) * 0.5f;
data[i] = tmp1 - s;
data[n - i] = tmp1 + s;
}
ctx->rdft.rdft_calc(&ctx->rdft, data);
data[n] = data[1];
data[1] = next;
for (i = 3; i <= n; i += 2)
data[i] = data[i - 2] - data[i];
}
static void dct_calc_III_c(DCTContext *ctx, FFTSample *data)
{
int n = 1 << ctx->nbits;
int i;
float next = data[n - 1];
float inv_n = 1.0f / n;
for (i = n - 2; i >= 2; i -= 2) {
float val1 = data[i];
float val2 = data[i - 1] - data[i + 1];
float c = COS(ctx, n, i);
float s = SIN(ctx, n, i);
data[i] = c * val1 + s * val2;
data[i + 1] = s * val1 - c * val2;
}
data[1] = 2 * next;
ctx->rdft.rdft_calc(&ctx->rdft, data);
for (i = 0; i < n / 2; i++) {
float tmp1 = data[i] * inv_n;
float tmp2 = data[n - i - 1] * inv_n;
float csc = ctx->csc2[i] * (tmp1 - tmp2);
tmp1 += tmp2;
data[i] = tmp1 + csc;
data[n - i - 1] = tmp1 - csc;
}
}
static void dct_calc_II_c(DCTContext *ctx, FFTSample *data)
{
int n = 1 << ctx->nbits;
int i;
float next;
for (i = 0; i < n / 2; i++) {
float tmp1 = data[i];
float tmp2 = data[n - i - 1];
float s = SIN(ctx, n, 2 * i + 1);
s *= tmp1 - tmp2;
tmp1 = (tmp1 + tmp2) * 0.5f;
data[i] = tmp1 + s;
data[n-i-1] = tmp1 - s;
}
ctx->rdft.rdft_calc(&ctx->rdft, data);
next = data[1] * 0.5;
data[1] *= -1;
for (i = n - 2; i >= 0; i -= 2) {
float inr = data[i ];
float ini = data[i + 1];
float c = COS(ctx, n, i);
float s = SIN(ctx, n, i);
data[i] = c * inr + s * ini;
data[i + 1] = next;
next += s * inr - c * ini;
}
}
static void dct32_func(DCTContext *ctx, FFTSample *data)
{
ctx->dct32(data, data);
}
av_cold int ff_dct_init(DCTContext *s, int nbits, enum DCTTransformType inverse)
{
int n = 1 << nbits;
int i;
int ret;
memset(s, 0, sizeof(*s));
s->nbits = nbits;
s->inverse = inverse;
if (inverse == DCT_II && nbits == 5) {
s->dct_calc = dct32_func;
} else {
ff_init_ff_cos_tabs(nbits + 2);
s->costab = ff_cos_tabs[nbits + 2];
s->csc2 = av_malloc_array(n / 2, sizeof(FFTSample));
if (!s->csc2)
return AVERROR(ENOMEM);
if ((ret = ff_rdft_init(&s->rdft, nbits, inverse == DCT_III)) < 0) {
av_freep(&s->csc2);
return ret;
}
for (i = 0; i < n / 2; i++)
s->csc2[i] = 0.5 / sin((M_PI / (2 * n) * (2 * i + 1)));
switch (inverse) {
case DCT_I : s->dct_calc = dct_calc_I_c; break;
case DCT_II : s->dct_calc = dct_calc_II_c; break;
case DCT_III: s->dct_calc = dct_calc_III_c; break;
case DST_I : s->dct_calc = dst_calc_I_c; break;
}
}
s->dct32 = ff_dct32_float;
if (ARCH_X86)
ff_dct_init_x86(s);
return 0;
}
av_cold void ff_dct_end(DCTContext *s)
{
ff_rdft_end(&s->rdft);
av_freep(&s->csc2);
}

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trunk/3rdparty/ffmpeg-4-fit/libavcodec/dct32.h vendored Normal file
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/*
* This file is part of FFmpeg.
*
* FFmpeg 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.
*
* FFmpeg 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 FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#ifndef AVCODEC_DCT32_H
#define AVCODEC_DCT32_H
void ff_dct32_float(float *dst, const float *src);
void ff_dct32_fixed(int *dst, const int *src);
#endif /* AVCODEC_DCT32_H */

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/*
* This file is part of FFmpeg.
*
* FFmpeg 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.
*
* FFmpeg 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 FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#define DCT32_FLOAT 0
#include "dct32_template.c"

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/*
* This file is part of FFmpeg.
*
* FFmpeg 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.
*
* FFmpeg 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 FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#define DCT32_FLOAT 1
#include "dct32_template.c"

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/*
* Template for the Discrete Cosine Transform for 32 samples
* Copyright (c) 2001, 2002 Fabrice Bellard
*
* This file is part of FFmpeg.
*
* FFmpeg 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.
*
* FFmpeg 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 FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include "dct32.h"
#include "mathops.h"
#include "libavutil/internal.h"
#ifdef CHECKED
#define SUINT int
#define SUINT32 int32_t
#else
#define SUINT unsigned
#define SUINT32 uint32_t
#endif
#if DCT32_FLOAT
# define dct32 ff_dct32_float
# define FIXHR(x) ((float)(x))
# define MULH3(x, y, s) ((s)*(y)*(x))
# define INTFLOAT float
# define SUINTFLOAT float
#else
# define dct32 ff_dct32_fixed
# define FIXHR(a) ((int)((a) * (1LL<<32) + 0.5))
# define MULH3(x, y, s) MULH((s)*(x), y)
# define INTFLOAT int
# define SUINTFLOAT SUINT
#endif
/* tab[i][j] = 1.0 / (2.0 * cos(pi*(2*k+1) / 2^(6 - j))) */
/* cos(i*pi/64) */
#define COS0_0 FIXHR(0.50060299823519630134/2)
#define COS0_1 FIXHR(0.50547095989754365998/2)
#define COS0_2 FIXHR(0.51544730992262454697/2)
#define COS0_3 FIXHR(0.53104259108978417447/2)
#define COS0_4 FIXHR(0.55310389603444452782/2)
#define COS0_5 FIXHR(0.58293496820613387367/2)
#define COS0_6 FIXHR(0.62250412303566481615/2)
#define COS0_7 FIXHR(0.67480834145500574602/2)
#define COS0_8 FIXHR(0.74453627100229844977/2)
#define COS0_9 FIXHR(0.83934964541552703873/2)
#define COS0_10 FIXHR(0.97256823786196069369/2)
#define COS0_11 FIXHR(1.16943993343288495515/4)
#define COS0_12 FIXHR(1.48416461631416627724/4)
#define COS0_13 FIXHR(2.05778100995341155085/8)
#define COS0_14 FIXHR(3.40760841846871878570/8)
#define COS0_15 FIXHR(10.19000812354805681150/32)
#define COS1_0 FIXHR(0.50241928618815570551/2)
#define COS1_1 FIXHR(0.52249861493968888062/2)
#define COS1_2 FIXHR(0.56694403481635770368/2)
#define COS1_3 FIXHR(0.64682178335999012954/2)
#define COS1_4 FIXHR(0.78815462345125022473/2)
#define COS1_5 FIXHR(1.06067768599034747134/4)
#define COS1_6 FIXHR(1.72244709823833392782/4)
#define COS1_7 FIXHR(5.10114861868916385802/16)
#define COS2_0 FIXHR(0.50979557910415916894/2)
#define COS2_1 FIXHR(0.60134488693504528054/2)
#define COS2_2 FIXHR(0.89997622313641570463/2)
#define COS2_3 FIXHR(2.56291544774150617881/8)
#define COS3_0 FIXHR(0.54119610014619698439/2)
#define COS3_1 FIXHR(1.30656296487637652785/4)
#define COS4_0 FIXHR(M_SQRT1_2/2)
/* butterfly operator */
#define BF(a, b, c, s)\
{\
tmp0 = val##a + val##b;\
tmp1 = val##a - val##b;\
val##a = tmp0;\
val##b = MULH3(tmp1, c, 1<<(s));\
}
#define BF0(a, b, c, s)\
{\
tmp0 = tab[a] + tab[b];\
tmp1 = tab[a] - tab[b];\
val##a = tmp0;\
val##b = MULH3(tmp1, c, 1<<(s));\
}
#define BF1(a, b, c, d)\
{\
BF(a, b, COS4_0, 1);\
BF(c, d,-COS4_0, 1);\
val##c += val##d;\
}
#define BF2(a, b, c, d)\
{\
BF(a, b, COS4_0, 1);\
BF(c, d,-COS4_0, 1);\
val##c += val##d;\
val##a += val##c;\
val##c += val##b;\
val##b += val##d;\
}
#define ADD(a, b) val##a += val##b
/* DCT32 without 1/sqrt(2) coef zero scaling. */
void dct32(INTFLOAT *out, const INTFLOAT *tab_arg)
{
const SUINTFLOAT *tab = tab_arg;
SUINTFLOAT tmp0, tmp1;
SUINTFLOAT val0 , val1 , val2 , val3 , val4 , val5 , val6 , val7 ,
val8 , val9 , val10, val11, val12, val13, val14, val15,
val16, val17, val18, val19, val20, val21, val22, val23,
val24, val25, val26, val27, val28, val29, val30, val31;
/* pass 1 */
BF0( 0, 31, COS0_0 , 1);
BF0(15, 16, COS0_15, 5);
/* pass 2 */
BF( 0, 15, COS1_0 , 1);
BF(16, 31,-COS1_0 , 1);
/* pass 1 */
BF0( 7, 24, COS0_7 , 1);
BF0( 8, 23, COS0_8 , 1);
/* pass 2 */
BF( 7, 8, COS1_7 , 4);
BF(23, 24,-COS1_7 , 4);
/* pass 3 */
BF( 0, 7, COS2_0 , 1);
BF( 8, 15,-COS2_0 , 1);
BF(16, 23, COS2_0 , 1);
BF(24, 31,-COS2_0 , 1);
/* pass 1 */
BF0( 3, 28, COS0_3 , 1);
BF0(12, 19, COS0_12, 2);
/* pass 2 */
BF( 3, 12, COS1_3 , 1);
BF(19, 28,-COS1_3 , 1);
/* pass 1 */
BF0( 4, 27, COS0_4 , 1);
BF0(11, 20, COS0_11, 2);
/* pass 2 */
BF( 4, 11, COS1_4 , 1);
BF(20, 27,-COS1_4 , 1);
/* pass 3 */
BF( 3, 4, COS2_3 , 3);
BF(11, 12,-COS2_3 , 3);
BF(19, 20, COS2_3 , 3);
BF(27, 28,-COS2_3 , 3);
/* pass 4 */
BF( 0, 3, COS3_0 , 1);
BF( 4, 7,-COS3_0 , 1);
BF( 8, 11, COS3_0 , 1);
BF(12, 15,-COS3_0 , 1);
BF(16, 19, COS3_0 , 1);
BF(20, 23,-COS3_0 , 1);
BF(24, 27, COS3_0 , 1);
BF(28, 31,-COS3_0 , 1);
/* pass 1 */
BF0( 1, 30, COS0_1 , 1);
BF0(14, 17, COS0_14, 3);
/* pass 2 */
BF( 1, 14, COS1_1 , 1);
BF(17, 30,-COS1_1 , 1);
/* pass 1 */
BF0( 6, 25, COS0_6 , 1);
BF0( 9, 22, COS0_9 , 1);
/* pass 2 */
BF( 6, 9, COS1_6 , 2);
BF(22, 25,-COS1_6 , 2);
/* pass 3 */
BF( 1, 6, COS2_1 , 1);
BF( 9, 14,-COS2_1 , 1);
BF(17, 22, COS2_1 , 1);
BF(25, 30,-COS2_1 , 1);
/* pass 1 */
BF0( 2, 29, COS0_2 , 1);
BF0(13, 18, COS0_13, 3);
/* pass 2 */
BF( 2, 13, COS1_2 , 1);
BF(18, 29,-COS1_2 , 1);
/* pass 1 */
BF0( 5, 26, COS0_5 , 1);
BF0(10, 21, COS0_10, 1);
/* pass 2 */
BF( 5, 10, COS1_5 , 2);
BF(21, 26,-COS1_5 , 2);
/* pass 3 */
BF( 2, 5, COS2_2 , 1);
BF(10, 13,-COS2_2 , 1);
BF(18, 21, COS2_2 , 1);
BF(26, 29,-COS2_2 , 1);
/* pass 4 */
BF( 1, 2, COS3_1 , 2);
BF( 5, 6,-COS3_1 , 2);
BF( 9, 10, COS3_1 , 2);
BF(13, 14,-COS3_1 , 2);
BF(17, 18, COS3_1 , 2);
BF(21, 22,-COS3_1 , 2);
BF(25, 26, COS3_1 , 2);
BF(29, 30,-COS3_1 , 2);
/* pass 5 */
BF1( 0, 1, 2, 3);
BF2( 4, 5, 6, 7);
BF1( 8, 9, 10, 11);
BF2(12, 13, 14, 15);
BF1(16, 17, 18, 19);
BF2(20, 21, 22, 23);
BF1(24, 25, 26, 27);
BF2(28, 29, 30, 31);
/* pass 6 */
ADD( 8, 12);
ADD(12, 10);
ADD(10, 14);
ADD(14, 9);
ADD( 9, 13);
ADD(13, 11);
ADD(11, 15);
out[ 0] = val0;
out[16] = val1;
out[ 8] = val2;
out[24] = val3;
out[ 4] = val4;
out[20] = val5;
out[12] = val6;
out[28] = val7;
out[ 2] = val8;
out[18] = val9;
out[10] = val10;
out[26] = val11;
out[ 6] = val12;
out[22] = val13;
out[14] = val14;
out[30] = val15;
ADD(24, 28);
ADD(28, 26);
ADD(26, 30);
ADD(30, 25);
ADD(25, 29);
ADD(29, 27);
ADD(27, 31);
out[ 1] = val16 + val24;
out[17] = val17 + val25;
out[ 9] = val18 + val26;
out[25] = val19 + val27;
out[ 5] = val20 + val28;
out[21] = val21 + val29;
out[13] = val22 + val30;
out[29] = val23 + val31;
out[ 3] = val24 + val20;
out[19] = val25 + val21;
out[11] = val26 + val22;
out[27] = val27 + val23;
out[ 7] = val28 + val18;
out[23] = val29 + val19;
out[15] = val30 + val17;
out[31] = val31;
}

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/*
* MPEG Audio common code
* Copyright (c) 2001, 2002 Fabrice Bellard
*
* This file is part of FFmpeg.
*
* FFmpeg 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.
*
* FFmpeg 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 FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
* @file
* MPEG Audio common code.
*/
#include "mpegaudio.h"
/* bitrate is in kb/s */
int ff_mpa_l2_select_table(int bitrate, int nb_channels, int freq, int lsf)
{
int ch_bitrate, table;
ch_bitrate = bitrate / nb_channels;
if (!lsf) {
if ((freq == 48000 && ch_bitrate >= 56) ||
(ch_bitrate >= 56 && ch_bitrate <= 80))
table = 0;
else if (freq != 48000 && ch_bitrate >= 96)
table = 1;
else if (freq != 32000 && ch_bitrate <= 48)
table = 2;
else
table = 3;
} else {
table = 4;
}
return table;
}

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/*
* copyright (c) 2001 Fabrice Bellard
*
* This file is part of FFmpeg.
*
* FFmpeg 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.
*
* FFmpeg 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 FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
* @file
* mpeg audio declarations for both encoder and decoder.
*/
#ifndef AVCODEC_MPEGAUDIO_H
#define AVCODEC_MPEGAUDIO_H
#ifndef USE_FLOATS
# define USE_FLOATS 0
#endif
#include <stdint.h>
#include "libavutil/internal.h"
/* max frame size, in samples */
#define MPA_FRAME_SIZE 1152
/* max compressed frame size */
#define MPA_MAX_CODED_FRAME_SIZE 1792
#define MPA_MAX_CHANNELS 2
#define SBLIMIT 32 /* number of subbands */
#define MPA_STEREO 0
#define MPA_JSTEREO 1
#define MPA_DUAL 2
#define MPA_MONO 3
#ifndef FRAC_BITS
#define FRAC_BITS 23 /* fractional bits for sb_samples and dct */
#define WFRAC_BITS 16 /* fractional bits for window */
#endif
#define IMDCT_SCALAR 1.759
#define FRAC_ONE (1 << FRAC_BITS)
#define FIX(a) ((int)((a) * FRAC_ONE))
#if USE_FLOATS
# define INTFLOAT float
# define SUINTFLOAT float
typedef float MPA_INT;
typedef float OUT_INT;
#elif FRAC_BITS <= 15
# define INTFLOAT int
# define SUINTFLOAT SUINT
typedef int16_t MPA_INT;
typedef int16_t OUT_INT;
#else
# define INTFLOAT int
# define SUINTFLOAT SUINT
typedef int32_t MPA_INT;
typedef int16_t OUT_INT;
#endif
int ff_mpa_l2_select_table(int bitrate, int nb_channels, int freq, int lsf);
#endif /* AVCODEC_MPEGAUDIO_H */

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/*
* MPEG Audio parser
* Copyright (c) 2003 Fabrice Bellard
* Copyright (c) 2003 Michael Niedermayer
*
* This file is part of FFmpeg.
*
* FFmpeg 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.
*
* FFmpeg 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 FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include "parser.h"
#include "mpegaudiodecheader.h"
#include "libavutil/common.h"
#include "libavformat/apetag.h" // for APE tag.
#include "libavformat/id3v1.h" // for ID3v1_TAG_SIZE
typedef struct MpegAudioParseContext {
ParseContext pc;
int frame_size;
uint32_t header;
int header_count;
int no_bitrate;
} MpegAudioParseContext;
#define MPA_HEADER_SIZE 4
/* header + layer + freq + lsf/mpeg25 */
#define SAME_HEADER_MASK \
(0xffe00000 | (3 << 17) | (3 << 10) | (3 << 19))
static int mpegaudio_parse(AVCodecParserContext *s1,
AVCodecContext *avctx,
const uint8_t **poutbuf, int *poutbuf_size,
const uint8_t *buf, int buf_size)
{
MpegAudioParseContext *s = s1->priv_data;
ParseContext *pc = &s->pc;
uint32_t state= pc->state;
int i;
int next= END_NOT_FOUND;
int flush = !buf_size;
for(i=0; i<buf_size; ){
if(s->frame_size){
int inc= FFMIN(buf_size - i, s->frame_size);
i += inc;
s->frame_size -= inc;
state = 0;
if(!s->frame_size){
next= i;
break;
}
}else{
while(i<buf_size){
int ret, sr, channels, bit_rate, frame_size;
enum AVCodecID codec_id = avctx->codec_id;
state= (state<<8) + buf[i++];
ret = ff_mpa_decode_header(state, &sr, &channels, &frame_size, &bit_rate, &codec_id);
if (ret < 4) {
if (i > 4)
s->header_count = -2;
} else {
int header_threshold = avctx->codec_id != AV_CODEC_ID_NONE && avctx->codec_id != codec_id;
if((state&SAME_HEADER_MASK) != (s->header&SAME_HEADER_MASK) && s->header)
s->header_count= -3;
s->header= state;
s->header_count++;
s->frame_size = ret-4;
if (s->header_count > header_threshold) {
avctx->sample_rate= sr;
avctx->channels = channels;
s1->duration = frame_size;
avctx->codec_id = codec_id;
if (s->no_bitrate || !avctx->bit_rate) {
s->no_bitrate = 1;
avctx->bit_rate += (bit_rate - avctx->bit_rate) / (s->header_count - header_threshold);
}
}
if (s1->flags & PARSER_FLAG_COMPLETE_FRAMES) {
s->frame_size = 0;
next = buf_size;
} else if (codec_id == AV_CODEC_ID_MP3ADU) {
avpriv_report_missing_feature(avctx,
"MP3ADU full parser");
*poutbuf = NULL;
*poutbuf_size = 0;
return buf_size; /* parsers must not return error codes */
}
break;
}
}
}
}
pc->state= state;
if (ff_combine_frame(pc, next, &buf, &buf_size) < 0) {
*poutbuf = NULL;
*poutbuf_size = 0;
return buf_size;
}
if (flush && buf_size >= ID3v1_TAG_SIZE && memcmp(buf, "TAG", 3) == 0) {
*poutbuf = NULL;
*poutbuf_size = 0;
return next;
}
if (flush && buf_size >= APE_TAG_FOOTER_BYTES && memcmp(buf, APE_TAG_PREAMBLE, 8) == 0) {
*poutbuf = NULL;
*poutbuf_size = 0;
return next;
}
*poutbuf = buf;
*poutbuf_size = buf_size;
return next;
}
AVCodecParser ff_mpegaudio_parser = {
.codec_ids = { AV_CODEC_ID_MP1, AV_CODEC_ID_MP2, AV_CODEC_ID_MP3, AV_CODEC_ID_MP3ADU },
.priv_data_size = sizeof(MpegAudioParseContext),
.parser_parse = mpegaudio_parse,
.parser_close = ff_parse_close,
};

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/*
* Generate a header file for hardcoded mpegaudiodec tables
*
* Copyright (c) 2009 Reimar Döffinger <Reimar.Doeffinger@gmx.de>
*
* This file is part of FFmpeg.
*
* FFmpeg 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.
*
* FFmpeg 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 FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include <stdlib.h>
#define CONFIG_HARDCODED_TABLES 0
#include "libavutil/tablegen.h"
#include "mpegaudio_tablegen.h"
#include "tableprint.h"
int main(void)
{
mpegaudio_tableinit();
write_fileheader();
WRITE_ARRAY("static const", int8_t, table_4_3_exp);
WRITE_ARRAY("static const", uint32_t, table_4_3_value);
WRITE_ARRAY("static const", uint32_t, exp_table_fixed);
WRITE_ARRAY("static const", float, exp_table_float);
WRITE_2D_ARRAY("static const", uint32_t, expval_table_fixed);
WRITE_2D_ARRAY("static const", float, expval_table_float);
return 0;
}

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/*
* Header file for hardcoded mpegaudiodec tables
*
* Copyright (c) 2009 Reimar Döffinger <Reimar.Doeffinger@gmx.de>
*
* This file is part of FFmpeg.
*
* FFmpeg 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.
*
* FFmpeg 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 FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#ifndef AVCODEC_MPEGAUDIO_TABLEGEN_H
#define AVCODEC_MPEGAUDIO_TABLEGEN_H
#include <stdint.h>
#include <math.h>
#include "libavutil/attributes.h"
#define TABLE_4_3_SIZE (8191 + 16)*4
#if CONFIG_HARDCODED_TABLES
#define mpegaudio_tableinit()
#include "libavcodec/mpegaudio_tables.h"
#else
static int8_t table_4_3_exp[TABLE_4_3_SIZE];
static uint32_t table_4_3_value[TABLE_4_3_SIZE];
static uint32_t exp_table_fixed[512];
static uint32_t expval_table_fixed[512][16];
static float exp_table_float[512];
static float expval_table_float[512][16];
#define FRAC_BITS 23
#define IMDCT_SCALAR 1.759
static av_cold void mpegaudio_tableinit(void)
{
int i, value, exponent;
static const double exp2_lut[4] = {
1.00000000000000000000, /* 2 ^ (0 * 0.25) */
1.18920711500272106672, /* 2 ^ (1 * 0.25) */
M_SQRT2 , /* 2 ^ (2 * 0.25) */
1.68179283050742908606, /* 2 ^ (3 * 0.25) */
};
static double pow43_lut[16];
double exp2_base = 2.11758236813575084767080625169910490512847900390625e-22; // 2^(-72)
double exp2_val;
double pow43_val = 0;
for (i = 0; i < 16; ++i)
pow43_lut[i] = i * cbrt(i);
for (i = 1; i < TABLE_4_3_SIZE; i++) {
double f, fm;
int e, m;
double value = i / 4;
if ((i & 3) == 0)
pow43_val = value / IMDCT_SCALAR * cbrt(value);
f = pow43_val * exp2_lut[i & 3];
fm = frexp(f, &e);
m = llrint(fm * (1LL << 31));
e += FRAC_BITS - 31 + 5 - 100;
/* normalized to FRAC_BITS */
table_4_3_value[i] = m;
table_4_3_exp[i] = -e;
}
for (exponent = 0; exponent < 512; exponent++) {
if (exponent && (exponent & 3) == 0)
exp2_base *= 2;
exp2_val = exp2_base * exp2_lut[exponent & 3] / IMDCT_SCALAR;
for (value = 0; value < 16; value++) {
double f = pow43_lut[value] * exp2_val;
expval_table_fixed[exponent][value] = (f < 0xFFFFFFFF ? llrint(f) : 0xFFFFFFFF);
expval_table_float[exponent][value] = f;
}
exp_table_fixed[exponent] = expval_table_fixed[exponent][1];
exp_table_float[exponent] = expval_table_float[exponent][1];
}
}
#endif /* CONFIG_HARDCODED_TABLES */
#endif /* AVCODEC_MPEGAUDIO_TABLEGEN_H */

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/*
* MPEG Audio common tables
* copyright (c) 2002 Fabrice Bellard
*
* This file is part of FFmpeg.
*
* FFmpeg 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.
*
* FFmpeg 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 FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
* @file
* mpeg audio layer common tables.
*/
#include "mpegaudiodata.h"
const uint16_t avpriv_mpa_bitrate_tab[2][3][15] = {
{ {0, 32, 64, 96, 128, 160, 192, 224, 256, 288, 320, 352, 384, 416, 448 },
{0, 32, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256, 320, 384 },
{0, 32, 40, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256, 320 } },
{ {0, 32, 48, 56, 64, 80, 96, 112, 128, 144, 160, 176, 192, 224, 256},
{0, 8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128, 144, 160},
{0, 8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128, 144, 160}
}
};
const uint16_t avpriv_mpa_freq_tab[3] = { 44100, 48000, 32000 };
/*******************************************************/
/* layer 2 tables */
const int ff_mpa_sblimit_table[5] = { 27 , 30 , 8, 12 , 30 };
const int ff_mpa_quant_steps[17] = {
3, 5, 7, 9, 15,
31, 63, 127, 255, 511,
1023, 2047, 4095, 8191, 16383,
32767, 65535
};
/* we use a negative value if grouped */
const int ff_mpa_quant_bits[17] = {
-5, -7, 3, -10, 4,
5, 6, 7, 8, 9,
10, 11, 12, 13, 14,
15, 16
};
/* encoding tables which give the quantization index. Note how it is
possible to store them efficiently ! */
static const unsigned char alloc_table_1[] = {
4, 0, 2, 4, 5, 6, 7, 8, 9, 10, 11, 12, 13, 14, 15, 16,
4, 0, 2, 4, 5, 6, 7, 8, 9, 10, 11, 12, 13, 14, 15, 16,
4, 0, 2, 4, 5, 6, 7, 8, 9, 10, 11, 12, 13, 14, 15, 16,
4, 0, 1, 2, 3, 4, 5, 6, 7, 8, 9, 10, 11, 12, 13, 16,
4, 0, 1, 2, 3, 4, 5, 6, 7, 8, 9, 10, 11, 12, 13, 16,
4, 0, 1, 2, 3, 4, 5, 6, 7, 8, 9, 10, 11, 12, 13, 16,
4, 0, 1, 2, 3, 4, 5, 6, 7, 8, 9, 10, 11, 12, 13, 16,
4, 0, 1, 2, 3, 4, 5, 6, 7, 8, 9, 10, 11, 12, 13, 16,
4, 0, 1, 2, 3, 4, 5, 6, 7, 8, 9, 10, 11, 12, 13, 16,
4, 0, 1, 2, 3, 4, 5, 6, 7, 8, 9, 10, 11, 12, 13, 16,
4, 0, 1, 2, 3, 4, 5, 6, 7, 8, 9, 10, 11, 12, 13, 16,
3, 0, 1, 2, 3, 4, 5, 16,
3, 0, 1, 2, 3, 4, 5, 16,
3, 0, 1, 2, 3, 4, 5, 16,
3, 0, 1, 2, 3, 4, 5, 16,
3, 0, 1, 2, 3, 4, 5, 16,
3, 0, 1, 2, 3, 4, 5, 16,
3, 0, 1, 2, 3, 4, 5, 16,
3, 0, 1, 2, 3, 4, 5, 16,
3, 0, 1, 2, 3, 4, 5, 16,
3, 0, 1, 2, 3, 4, 5, 16,
3, 0, 1, 2, 3, 4, 5, 16,
3, 0, 1, 2, 3, 4, 5, 16,
2, 0, 1, 16,
2, 0, 1, 16,
2, 0, 1, 16,
2, 0, 1, 16,
2, 0, 1, 16,
2, 0, 1, 16,
2, 0, 1, 16,
};
static const unsigned char alloc_table_3[] = {
4, 0, 1, 3, 4, 5, 6, 7, 8, 9, 10, 11, 12, 13, 14, 15,
4, 0, 1, 3, 4, 5, 6, 7, 8, 9, 10, 11, 12, 13, 14, 15,
3, 0, 1, 3, 4, 5, 6, 7,
3, 0, 1, 3, 4, 5, 6, 7,
3, 0, 1, 3, 4, 5, 6, 7,
3, 0, 1, 3, 4, 5, 6, 7,
3, 0, 1, 3, 4, 5, 6, 7,
3, 0, 1, 3, 4, 5, 6, 7,
3, 0, 1, 3, 4, 5, 6, 7,
3, 0, 1, 3, 4, 5, 6, 7,
3, 0, 1, 3, 4, 5, 6, 7,
3, 0, 1, 3, 4, 5, 6, 7,
};
static const unsigned char alloc_table_4[] = {
4, 0, 1, 2, 3, 4, 5, 6, 7, 8, 9, 10, 11, 12, 13, 14,
4, 0, 1, 2, 3, 4, 5, 6, 7, 8, 9, 10, 11, 12, 13, 14,
4, 0, 1, 2, 3, 4, 5, 6, 7, 8, 9, 10, 11, 12, 13, 14,
4, 0, 1, 2, 3, 4, 5, 6, 7, 8, 9, 10, 11, 12, 13, 14,
3, 0, 1, 3, 4, 5, 6, 7,
3, 0, 1, 3, 4, 5, 6, 7,
3, 0, 1, 3, 4, 5, 6, 7,
3, 0, 1, 3, 4, 5, 6, 7,
3, 0, 1, 3, 4, 5, 6, 7,
3, 0, 1, 3, 4, 5, 6, 7,
3, 0, 1, 3, 4, 5, 6, 7,
2, 0, 1, 3,
2, 0, 1, 3,
2, 0, 1, 3,
2, 0, 1, 3,
2, 0, 1, 3,
2, 0, 1, 3,
2, 0, 1, 3,
2, 0, 1, 3,
2, 0, 1, 3,
2, 0, 1, 3,
2, 0, 1, 3,
2, 0, 1, 3,
2, 0, 1, 3,
2, 0, 1, 3,
2, 0, 1, 3,
2, 0, 1, 3,
2, 0, 1, 3,
2, 0, 1, 3,
2, 0, 1, 3,
};
const unsigned char * const ff_mpa_alloc_tables[5] =
{ alloc_table_1, alloc_table_1, alloc_table_3, alloc_table_3, alloc_table_4, };

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/*
* MPEG Audio common tables
* copyright (c) 2002 Fabrice Bellard
*
* This file is part of FFmpeg.
*
* FFmpeg 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.
*
* FFmpeg 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 FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
* @file
* mpeg audio layer common tables.
*/
#ifndef AVCODEC_MPEGAUDIODATA_H
#define AVCODEC_MPEGAUDIODATA_H
#include <stdint.h>
#include "internal.h"
#define MODE_EXT_MS_STEREO 2
#define MODE_EXT_I_STEREO 1
extern av_export_avcodec const uint16_t avpriv_mpa_bitrate_tab[2][3][15];
extern av_export_avcodec const uint16_t avpriv_mpa_freq_tab[3];
extern const int ff_mpa_sblimit_table[5];
extern const int ff_mpa_quant_steps[17];
extern const int ff_mpa_quant_bits[17];
extern const unsigned char * const ff_mpa_alloc_tables[5];
#endif /* AVCODEC_MPEGAUDIODATA_H */

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/*
* Fixed-point MPEG audio decoder
*
* This file is part of FFmpeg.
*
* FFmpeg 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.
*
* FFmpeg 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 FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include "config.h"
#include "libavutil/samplefmt.h"
#define USE_FLOATS 0
#include "mpegaudio.h"
#define SHR(a,b) (((int)(a))>>(b))
/* WARNING: only correct for positive numbers */
#define FIXR_OLD(a) ((int)((a) * FRAC_ONE + 0.5))
#define FIXR(a) ((int)((a) * FRAC_ONE + 0.5))
#define FIXHR(a) ((int)((a) * (1LL<<32) + 0.5))
#define MULH3(x, y, s) MULH((s)*(x), y)
#define MULLx(x, y, s) MULL((int)(x),(y),s)
#define RENAME(a) a ## _fixed
#define OUT_FMT AV_SAMPLE_FMT_S16
#define OUT_FMT_P AV_SAMPLE_FMT_S16P
#include "mpegaudiodec_template.c"
#if CONFIG_MP1_DECODER
AVCodec ff_mp1_decoder = {
.name = "mp1",
.long_name = NULL_IF_CONFIG_SMALL("MP1 (MPEG audio layer 1)"),
.type = AVMEDIA_TYPE_AUDIO,
.id = AV_CODEC_ID_MP1,
.priv_data_size = sizeof(MPADecodeContext),
.init = decode_init,
.decode = decode_frame,
.capabilities = AV_CODEC_CAP_DR1,
.flush = flush,
.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16P,
AV_SAMPLE_FMT_S16,
AV_SAMPLE_FMT_NONE },
};
#endif
#if CONFIG_MP2_DECODER
AVCodec ff_mp2_decoder = {
.name = "mp2",
.long_name = NULL_IF_CONFIG_SMALL("MP2 (MPEG audio layer 2)"),
.type = AVMEDIA_TYPE_AUDIO,
.id = AV_CODEC_ID_MP2,
.priv_data_size = sizeof(MPADecodeContext),
.init = decode_init,
.decode = decode_frame,
.capabilities = AV_CODEC_CAP_DR1,
.flush = flush,
.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16P,
AV_SAMPLE_FMT_S16,
AV_SAMPLE_FMT_NONE },
};
#endif
#if CONFIG_MP3_DECODER
AVCodec ff_mp3_decoder = {
.name = "mp3",
.long_name = NULL_IF_CONFIG_SMALL("MP3 (MPEG audio layer 3)"),
.type = AVMEDIA_TYPE_AUDIO,
.id = AV_CODEC_ID_MP3,
.priv_data_size = sizeof(MPADecodeContext),
.init = decode_init,
.decode = decode_frame,
.capabilities = AV_CODEC_CAP_DR1,
.flush = flush,
.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16P,
AV_SAMPLE_FMT_S16,
AV_SAMPLE_FMT_NONE },
};
#endif
#if CONFIG_MP3ADU_DECODER
AVCodec ff_mp3adu_decoder = {
.name = "mp3adu",
.long_name = NULL_IF_CONFIG_SMALL("ADU (Application Data Unit) MP3 (MPEG audio layer 3)"),
.type = AVMEDIA_TYPE_AUDIO,
.id = AV_CODEC_ID_MP3ADU,
.priv_data_size = sizeof(MPADecodeContext),
.init = decode_init,
.decode = decode_frame_adu,
.capabilities = AV_CODEC_CAP_DR1,
.flush = flush,
.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16P,
AV_SAMPLE_FMT_S16,
AV_SAMPLE_FMT_NONE },
};
#endif
#if CONFIG_MP3ON4_DECODER
AVCodec ff_mp3on4_decoder = {
.name = "mp3on4",
.long_name = NULL_IF_CONFIG_SMALL("MP3onMP4"),
.type = AVMEDIA_TYPE_AUDIO,
.id = AV_CODEC_ID_MP3ON4,
.priv_data_size = sizeof(MP3On4DecodeContext),
.init = decode_init_mp3on4,
.close = decode_close_mp3on4,
.decode = decode_frame_mp3on4,
.capabilities = AV_CODEC_CAP_DR1,
.flush = flush_mp3on4,
.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_S16P,
AV_SAMPLE_FMT_NONE },
};
#endif

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/*
* Float MPEG Audio decoder
* Copyright (c) 2010 Michael Niedermayer
*
* This file is part of FFmpeg.
*
* FFmpeg 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.
*
* FFmpeg 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 FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include "config.h"
#include "libavutil/samplefmt.h"
#define USE_FLOATS 1
#include "mpegaudio.h"
#define SHR(a,b) ((a)*(1.0f/(1<<(b))))
#define FIXR_OLD(a) ((int)((a) * FRAC_ONE + 0.5))
#define FIXR(x) ((float)(x))
#define FIXHR(x) ((float)(x))
#define MULH3(x, y, s) ((s)*(y)*(x))
#define MULLx(x, y, s) ((y)*(x))
#define RENAME(a) a ## _float
#define OUT_FMT AV_SAMPLE_FMT_FLT
#define OUT_FMT_P AV_SAMPLE_FMT_FLTP
#include "mpegaudiodec_template.c"
#if CONFIG_MP1FLOAT_DECODER
AVCodec ff_mp1float_decoder = {
.name = "mp1float",
.long_name = NULL_IF_CONFIG_SMALL("MP1 (MPEG audio layer 1)"),
.type = AVMEDIA_TYPE_AUDIO,
.id = AV_CODEC_ID_MP1,
.priv_data_size = sizeof(MPADecodeContext),
.init = decode_init,
.close = decode_close,
.decode = decode_frame,
.capabilities = AV_CODEC_CAP_DR1,
.flush = flush,
.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
AV_SAMPLE_FMT_FLT,
AV_SAMPLE_FMT_NONE },
};
#endif
#if CONFIG_MP2FLOAT_DECODER
AVCodec ff_mp2float_decoder = {
.name = "mp2float",
.long_name = NULL_IF_CONFIG_SMALL("MP2 (MPEG audio layer 2)"),
.type = AVMEDIA_TYPE_AUDIO,
.id = AV_CODEC_ID_MP2,
.priv_data_size = sizeof(MPADecodeContext),
.init = decode_init,
.decode = decode_frame,
.close = decode_close,
.capabilities = AV_CODEC_CAP_DR1,
.flush = flush,
.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
AV_SAMPLE_FMT_FLT,
AV_SAMPLE_FMT_NONE },
};
#endif
#if CONFIG_MP3FLOAT_DECODER
AVCodec ff_mp3float_decoder = {
.name = "mp3float",
.long_name = NULL_IF_CONFIG_SMALL("MP3 (MPEG audio layer 3)"),
.type = AVMEDIA_TYPE_AUDIO,
.id = AV_CODEC_ID_MP3,
.priv_data_size = sizeof(MPADecodeContext),
.init = decode_init,
.close = decode_close,
.decode = decode_frame,
.capabilities = AV_CODEC_CAP_DR1,
.flush = flush,
.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
AV_SAMPLE_FMT_FLT,
AV_SAMPLE_FMT_NONE },
};
#endif
#if CONFIG_MP3ADUFLOAT_DECODER
AVCodec ff_mp3adufloat_decoder = {
.name = "mp3adufloat",
.long_name = NULL_IF_CONFIG_SMALL("ADU (Application Data Unit) MP3 (MPEG audio layer 3)"),
.type = AVMEDIA_TYPE_AUDIO,
.id = AV_CODEC_ID_MP3ADU,
.priv_data_size = sizeof(MPADecodeContext),
.init = decode_init,
.close = decode_close,
.decode = decode_frame_adu,
.capabilities = AV_CODEC_CAP_DR1,
.flush = flush,
.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
AV_SAMPLE_FMT_FLT,
AV_SAMPLE_FMT_NONE },
};
#endif
#if CONFIG_MP3ON4FLOAT_DECODER
AVCodec ff_mp3on4float_decoder = {
.name = "mp3on4float",
.long_name = NULL_IF_CONFIG_SMALL("MP3onMP4"),
.type = AVMEDIA_TYPE_AUDIO,
.id = AV_CODEC_ID_MP3ON4,
.priv_data_size = sizeof(MP3On4DecodeContext),
.init = decode_init_mp3on4,
.close = decode_close_mp3on4,
.decode = decode_frame_mp3on4,
.capabilities = AV_CODEC_CAP_DR1,
.flush = flush_mp3on4,
.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLTP,
AV_SAMPLE_FMT_NONE },
};
#endif

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/*
* MPEG Audio header decoder
* Copyright (c) 2001, 2002 Fabrice Bellard
*
* This file is part of FFmpeg.
*
* FFmpeg 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.
*
* FFmpeg 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 FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
* @file
* MPEG Audio header decoder.
*/
#include "libavutil/common.h"
#include "avcodec.h"
#include "internal.h"
#include "mpegaudio.h"
#include "mpegaudiodata.h"
#include "mpegaudiodecheader.h"
int avpriv_mpegaudio_decode_header(MPADecodeHeader *s, uint32_t header)
{
int sample_rate, frame_size, mpeg25, padding;
int sample_rate_index, bitrate_index;
int ret;
ret = ff_mpa_check_header(header);
if (ret < 0)
return ret;
if (header & (1<<20)) {
s->lsf = (header & (1<<19)) ? 0 : 1;
mpeg25 = 0;
} else {
s->lsf = 1;
mpeg25 = 1;
}
s->layer = 4 - ((header >> 17) & 3);
/* extract frequency */
sample_rate_index = (header >> 10) & 3;
if (sample_rate_index >= FF_ARRAY_ELEMS(avpriv_mpa_freq_tab))
sample_rate_index = 0;
sample_rate = avpriv_mpa_freq_tab[sample_rate_index] >> (s->lsf + mpeg25);
sample_rate_index += 3 * (s->lsf + mpeg25);
s->sample_rate_index = sample_rate_index;
s->error_protection = ((header >> 16) & 1) ^ 1;
s->sample_rate = sample_rate;
bitrate_index = (header >> 12) & 0xf;
padding = (header >> 9) & 1;
//extension = (header >> 8) & 1;
s->mode = (header >> 6) & 3;
s->mode_ext = (header >> 4) & 3;
//copyright = (header >> 3) & 1;
//original = (header >> 2) & 1;
//emphasis = header & 3;
if (s->mode == MPA_MONO)
s->nb_channels = 1;
else
s->nb_channels = 2;
if (bitrate_index != 0) {
frame_size = avpriv_mpa_bitrate_tab[s->lsf][s->layer - 1][bitrate_index];
s->bit_rate = frame_size * 1000;
switch(s->layer) {
case 1:
frame_size = (frame_size * 12000) / sample_rate;
frame_size = (frame_size + padding) * 4;
break;
case 2:
frame_size = (frame_size * 144000) / sample_rate;
frame_size += padding;
break;
default:
case 3:
frame_size = (frame_size * 144000) / (sample_rate << s->lsf);
frame_size += padding;
break;
}
s->frame_size = frame_size;
} else {
/* if no frame size computed, signal it */
return 1;
}
#if defined(DEBUG)
ff_dlog(NULL, "layer%d, %d Hz, %d kbits/s, ",
s->layer, s->sample_rate, s->bit_rate);
if (s->nb_channels == 2) {
if (s->layer == 3) {
if (s->mode_ext & MODE_EXT_MS_STEREO)
ff_dlog(NULL, "ms-");
if (s->mode_ext & MODE_EXT_I_STEREO)
ff_dlog(NULL, "i-");
}
ff_dlog(NULL, "stereo");
} else {
ff_dlog(NULL, "mono");
}
ff_dlog(NULL, "\n");
#endif
return 0;
}
int ff_mpa_decode_header(uint32_t head, int *sample_rate, int *channels, int *frame_size, int *bit_rate, enum AVCodecID *codec_id)
{
MPADecodeHeader s1, *s = &s1;
if (avpriv_mpegaudio_decode_header(s, head) != 0) {
return -1;
}
switch(s->layer) {
case 1:
*codec_id = AV_CODEC_ID_MP1;
*frame_size = 384;
break;
case 2:
*codec_id = AV_CODEC_ID_MP2;
*frame_size = 1152;
break;
default:
case 3:
if (*codec_id != AV_CODEC_ID_MP3ADU)
*codec_id = AV_CODEC_ID_MP3;
if (s->lsf)
*frame_size = 576;
else
*frame_size = 1152;
break;
}
*sample_rate = s->sample_rate;
*channels = s->nb_channels;
*bit_rate = s->bit_rate;
return s->frame_size;
}

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/*
* MPEG Audio header decoder
* Copyright (c) 2001, 2002 Fabrice Bellard
*
* This file is part of FFmpeg.
*
* FFmpeg 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.
*
* FFmpeg 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 FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
* @file
* MPEG Audio header decoder.
*/
#ifndef AVCODEC_MPEGAUDIODECHEADER_H
#define AVCODEC_MPEGAUDIODECHEADER_H
#include "avcodec.h"
#define MP3_MASK 0xFFFE0CCF
#define MPA_DECODE_HEADER \
int frame_size; \
int error_protection; \
int layer; \
int sample_rate; \
int sample_rate_index; /* between 0 and 8 */ \
int bit_rate; \
int nb_channels; \
int mode; \
int mode_ext; \
int lsf;
typedef struct MPADecodeHeader {
MPA_DECODE_HEADER
} MPADecodeHeader;
/* header decoding. MUST check the header before because no
consistency check is done there. Return 1 if free format found and
that the frame size must be computed externally */
int avpriv_mpegaudio_decode_header(MPADecodeHeader *s, uint32_t header);
/* useful helper to get MPEG audio stream info. Return -1 if error in
header, otherwise the coded frame size in bytes */
int ff_mpa_decode_header(uint32_t head, int *sample_rate,
int *channels, int *frame_size, int *bitrate, enum AVCodecID *codec_id);
/* fast header check for resync */
static inline int ff_mpa_check_header(uint32_t header){
/* header */
if ((header & 0xffe00000) != 0xffe00000)
return -1;
/* version check */
if ((header & (3<<19)) == 1<<19)
return -1;
/* layer check */
if ((header & (3<<17)) == 0)
return -1;
/* bit rate */
if ((header & (0xf<<12)) == 0xf<<12)
return -1;
/* frequency */
if ((header & (3<<10)) == 3<<10)
return -1;
return 0;
}
#endif /* AVCODEC_MPEGAUDIODECHEADER_H */

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/*
* MPEG Audio decoder
* copyright (c) 2002 Fabrice Bellard
*
* This file is part of FFmpeg.
*
* FFmpeg 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.
*
* FFmpeg 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 FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
* @file
* mpeg audio layer decoder tables.
*/
#ifndef AVCODEC_MPEGAUDIODECTAB_H
#define AVCODEC_MPEGAUDIODECTAB_H
#include <stddef.h>
#include <stdint.h>
#include "mpegaudio.h"
/*******************************************************/
/* layer 3 tables */
/* layer 3 huffman tables */
typedef struct HuffTable {
int xsize;
const uint8_t *bits;
const uint16_t *codes;
} HuffTable;
/* layer3 scale factor size */
static const uint8_t slen_table[2][16] = {
{ 0, 0, 0, 0, 3, 1, 1, 1, 2, 2, 2, 3, 3, 3, 4, 4 },
{ 0, 1, 2, 3, 0, 1, 2, 3, 1, 2, 3, 1, 2, 3, 2, 3 },
};
/* number of lsf scale factors for a given size */
static const uint8_t lsf_nsf_table[6][3][4] = {
{ { 6, 5, 5, 5 }, { 9, 9, 9, 9 }, { 6, 9, 9, 9 } },
{ { 6, 5, 7, 3 }, { 9, 9, 12, 6 }, { 6, 9, 12, 6 } },
{ { 11, 10, 0, 0 }, { 18, 18, 0, 0 }, { 15, 18, 0, 0 } },
{ { 7, 7, 7, 0 }, { 12, 12, 12, 0 }, { 6, 15, 12, 0 } },
{ { 6, 6, 6, 3 }, { 12, 9, 9, 6 }, { 6, 12, 9, 6 } },
{ { 8, 8, 5, 0 }, { 15, 12, 9, 0 }, { 6, 18, 9, 0 } },
};
/* mpegaudio layer 3 huffman tables */
static const uint16_t mpa_huffcodes_1[4] = {
0x0001, 0x0001, 0x0001, 0x0000,
};
static const uint8_t mpa_huffbits_1[4] = {
1, 3, 2, 3,
};
static const uint16_t mpa_huffcodes_2[9] = {
0x0001, 0x0002, 0x0001, 0x0003, 0x0001, 0x0001, 0x0003, 0x0002,
0x0000,
};
static const uint8_t mpa_huffbits_2[9] = {
1, 3, 6, 3, 3, 5, 5, 5,
6,
};
static const uint16_t mpa_huffcodes_3[9] = {
0x0003, 0x0002, 0x0001, 0x0001, 0x0001, 0x0001, 0x0003, 0x0002,
0x0000,
};
static const uint8_t mpa_huffbits_3[9] = {
2, 2, 6, 3, 2, 5, 5, 5,
6,
};
static const uint16_t mpa_huffcodes_5[16] = {
0x0001, 0x0002, 0x0006, 0x0005, 0x0003, 0x0001, 0x0004, 0x0004,
0x0007, 0x0005, 0x0007, 0x0001, 0x0006, 0x0001, 0x0001, 0x0000,
};
static const uint8_t mpa_huffbits_5[16] = {
1, 3, 6, 7, 3, 3, 6, 7,
6, 6, 7, 8, 7, 6, 7, 8,
};
static const uint16_t mpa_huffcodes_6[16] = {
0x0007, 0x0003, 0x0005, 0x0001, 0x0006, 0x0002, 0x0003, 0x0002,
0x0005, 0x0004, 0x0004, 0x0001, 0x0003, 0x0003, 0x0002, 0x0000,
};
static const uint8_t mpa_huffbits_6[16] = {
3, 3, 5, 7, 3, 2, 4, 5,
4, 4, 5, 6, 6, 5, 6, 7,
};
static const uint16_t mpa_huffcodes_7[36] = {
0x0001, 0x0002, 0x000a, 0x0013, 0x0010, 0x000a, 0x0003, 0x0003,
0x0007, 0x000a, 0x0005, 0x0003, 0x000b, 0x0004, 0x000d, 0x0011,
0x0008, 0x0004, 0x000c, 0x000b, 0x0012, 0x000f, 0x000b, 0x0002,
0x0007, 0x0006, 0x0009, 0x000e, 0x0003, 0x0001, 0x0006, 0x0004,
0x0005, 0x0003, 0x0002, 0x0000,
};
static const uint8_t mpa_huffbits_7[36] = {
1, 3, 6, 8, 8, 9, 3, 4,
6, 7, 7, 8, 6, 5, 7, 8,
8, 9, 7, 7, 8, 9, 9, 9,
7, 7, 8, 9, 9, 10, 8, 8,
9, 10, 10, 10,
};
static const uint16_t mpa_huffcodes_8[36] = {
0x0003, 0x0004, 0x0006, 0x0012, 0x000c, 0x0005, 0x0005, 0x0001,
0x0002, 0x0010, 0x0009, 0x0003, 0x0007, 0x0003, 0x0005, 0x000e,
0x0007, 0x0003, 0x0013, 0x0011, 0x000f, 0x000d, 0x000a, 0x0004,
0x000d, 0x0005, 0x0008, 0x000b, 0x0005, 0x0001, 0x000c, 0x0004,
0x0004, 0x0001, 0x0001, 0x0000,
};
static const uint8_t mpa_huffbits_8[36] = {
2, 3, 6, 8, 8, 9, 3, 2,
4, 8, 8, 8, 6, 4, 6, 8,
8, 9, 8, 8, 8, 9, 9, 10,
8, 7, 8, 9, 10, 10, 9, 8,
9, 9, 11, 11,
};
static const uint16_t mpa_huffcodes_9[36] = {
0x0007, 0x0005, 0x0009, 0x000e, 0x000f, 0x0007, 0x0006, 0x0004,
0x0005, 0x0005, 0x0006, 0x0007, 0x0007, 0x0006, 0x0008, 0x0008,
0x0008, 0x0005, 0x000f, 0x0006, 0x0009, 0x000a, 0x0005, 0x0001,
0x000b, 0x0007, 0x0009, 0x0006, 0x0004, 0x0001, 0x000e, 0x0004,
0x0006, 0x0002, 0x0006, 0x0000,
};
static const uint8_t mpa_huffbits_9[36] = {
3, 3, 5, 6, 8, 9, 3, 3,
4, 5, 6, 8, 4, 4, 5, 6,
7, 8, 6, 5, 6, 7, 7, 8,
7, 6, 7, 7, 8, 9, 8, 7,
8, 8, 9, 9,
};
static const uint16_t mpa_huffcodes_10[64] = {
0x0001, 0x0002, 0x000a, 0x0017, 0x0023, 0x001e, 0x000c, 0x0011,
0x0003, 0x0003, 0x0008, 0x000c, 0x0012, 0x0015, 0x000c, 0x0007,
0x000b, 0x0009, 0x000f, 0x0015, 0x0020, 0x0028, 0x0013, 0x0006,
0x000e, 0x000d, 0x0016, 0x0022, 0x002e, 0x0017, 0x0012, 0x0007,
0x0014, 0x0013, 0x0021, 0x002f, 0x001b, 0x0016, 0x0009, 0x0003,
0x001f, 0x0016, 0x0029, 0x001a, 0x0015, 0x0014, 0x0005, 0x0003,
0x000e, 0x000d, 0x000a, 0x000b, 0x0010, 0x0006, 0x0005, 0x0001,
0x0009, 0x0008, 0x0007, 0x0008, 0x0004, 0x0004, 0x0002, 0x0000,
};
static const uint8_t mpa_huffbits_10[64] = {
1, 3, 6, 8, 9, 9, 9, 10,
3, 4, 6, 7, 8, 9, 8, 8,
6, 6, 7, 8, 9, 10, 9, 9,
7, 7, 8, 9, 10, 10, 9, 10,
8, 8, 9, 10, 10, 10, 10, 10,
9, 9, 10, 10, 11, 11, 10, 11,
8, 8, 9, 10, 10, 10, 11, 11,
9, 8, 9, 10, 10, 11, 11, 11,
};
static const uint16_t mpa_huffcodes_11[64] = {
0x0003, 0x0004, 0x000a, 0x0018, 0x0022, 0x0021, 0x0015, 0x000f,
0x0005, 0x0003, 0x0004, 0x000a, 0x0020, 0x0011, 0x000b, 0x000a,
0x000b, 0x0007, 0x000d, 0x0012, 0x001e, 0x001f, 0x0014, 0x0005,
0x0019, 0x000b, 0x0013, 0x003b, 0x001b, 0x0012, 0x000c, 0x0005,
0x0023, 0x0021, 0x001f, 0x003a, 0x001e, 0x0010, 0x0007, 0x0005,
0x001c, 0x001a, 0x0020, 0x0013, 0x0011, 0x000f, 0x0008, 0x000e,
0x000e, 0x000c, 0x0009, 0x000d, 0x000e, 0x0009, 0x0004, 0x0001,
0x000b, 0x0004, 0x0006, 0x0006, 0x0006, 0x0003, 0x0002, 0x0000,
};
static const uint8_t mpa_huffbits_11[64] = {
2, 3, 5, 7, 8, 9, 8, 9,
3, 3, 4, 6, 8, 8, 7, 8,
5, 5, 6, 7, 8, 9, 8, 8,
7, 6, 7, 9, 8, 10, 8, 9,
8, 8, 8, 9, 9, 10, 9, 10,
8, 8, 9, 10, 10, 11, 10, 11,
8, 7, 7, 8, 9, 10, 10, 10,
8, 7, 8, 9, 10, 10, 10, 10,
};
static const uint16_t mpa_huffcodes_12[64] = {
0x0009, 0x0006, 0x0010, 0x0021, 0x0029, 0x0027, 0x0026, 0x001a,
0x0007, 0x0005, 0x0006, 0x0009, 0x0017, 0x0010, 0x001a, 0x000b,
0x0011, 0x0007, 0x000b, 0x000e, 0x0015, 0x001e, 0x000a, 0x0007,
0x0011, 0x000a, 0x000f, 0x000c, 0x0012, 0x001c, 0x000e, 0x0005,
0x0020, 0x000d, 0x0016, 0x0013, 0x0012, 0x0010, 0x0009, 0x0005,
0x0028, 0x0011, 0x001f, 0x001d, 0x0011, 0x000d, 0x0004, 0x0002,
0x001b, 0x000c, 0x000b, 0x000f, 0x000a, 0x0007, 0x0004, 0x0001,
0x001b, 0x000c, 0x0008, 0x000c, 0x0006, 0x0003, 0x0001, 0x0000,
};
static const uint8_t mpa_huffbits_12[64] = {
4, 3, 5, 7, 8, 9, 9, 9,
3, 3, 4, 5, 7, 7, 8, 8,
5, 4, 5, 6, 7, 8, 7, 8,
6, 5, 6, 6, 7, 8, 8, 8,
7, 6, 7, 7, 8, 8, 8, 9,
8, 7, 8, 8, 8, 9, 8, 9,
8, 7, 7, 8, 8, 9, 9, 10,
9, 8, 8, 9, 9, 9, 9, 10,
};
static const uint16_t mpa_huffcodes_13[256] = {
0x0001, 0x0005, 0x000e, 0x0015, 0x0022, 0x0033, 0x002e, 0x0047,
0x002a, 0x0034, 0x0044, 0x0034, 0x0043, 0x002c, 0x002b, 0x0013,
0x0003, 0x0004, 0x000c, 0x0013, 0x001f, 0x001a, 0x002c, 0x0021,
0x001f, 0x0018, 0x0020, 0x0018, 0x001f, 0x0023, 0x0016, 0x000e,
0x000f, 0x000d, 0x0017, 0x0024, 0x003b, 0x0031, 0x004d, 0x0041,
0x001d, 0x0028, 0x001e, 0x0028, 0x001b, 0x0021, 0x002a, 0x0010,
0x0016, 0x0014, 0x0025, 0x003d, 0x0038, 0x004f, 0x0049, 0x0040,
0x002b, 0x004c, 0x0038, 0x0025, 0x001a, 0x001f, 0x0019, 0x000e,
0x0023, 0x0010, 0x003c, 0x0039, 0x0061, 0x004b, 0x0072, 0x005b,
0x0036, 0x0049, 0x0037, 0x0029, 0x0030, 0x0035, 0x0017, 0x0018,
0x003a, 0x001b, 0x0032, 0x0060, 0x004c, 0x0046, 0x005d, 0x0054,
0x004d, 0x003a, 0x004f, 0x001d, 0x004a, 0x0031, 0x0029, 0x0011,
0x002f, 0x002d, 0x004e, 0x004a, 0x0073, 0x005e, 0x005a, 0x004f,
0x0045, 0x0053, 0x0047, 0x0032, 0x003b, 0x0026, 0x0024, 0x000f,
0x0048, 0x0022, 0x0038, 0x005f, 0x005c, 0x0055, 0x005b, 0x005a,
0x0056, 0x0049, 0x004d, 0x0041, 0x0033, 0x002c, 0x002b, 0x002a,
0x002b, 0x0014, 0x001e, 0x002c, 0x0037, 0x004e, 0x0048, 0x0057,
0x004e, 0x003d, 0x002e, 0x0036, 0x0025, 0x001e, 0x0014, 0x0010,
0x0035, 0x0019, 0x0029, 0x0025, 0x002c, 0x003b, 0x0036, 0x0051,
0x0042, 0x004c, 0x0039, 0x0036, 0x0025, 0x0012, 0x0027, 0x000b,
0x0023, 0x0021, 0x001f, 0x0039, 0x002a, 0x0052, 0x0048, 0x0050,
0x002f, 0x003a, 0x0037, 0x0015, 0x0016, 0x001a, 0x0026, 0x0016,
0x0035, 0x0019, 0x0017, 0x0026, 0x0046, 0x003c, 0x0033, 0x0024,
0x0037, 0x001a, 0x0022, 0x0017, 0x001b, 0x000e, 0x0009, 0x0007,
0x0022, 0x0020, 0x001c, 0x0027, 0x0031, 0x004b, 0x001e, 0x0034,
0x0030, 0x0028, 0x0034, 0x001c, 0x0012, 0x0011, 0x0009, 0x0005,
0x002d, 0x0015, 0x0022, 0x0040, 0x0038, 0x0032, 0x0031, 0x002d,
0x001f, 0x0013, 0x000c, 0x000f, 0x000a, 0x0007, 0x0006, 0x0003,
0x0030, 0x0017, 0x0014, 0x0027, 0x0024, 0x0023, 0x0035, 0x0015,
0x0010, 0x0017, 0x000d, 0x000a, 0x0006, 0x0001, 0x0004, 0x0002,
0x0010, 0x000f, 0x0011, 0x001b, 0x0019, 0x0014, 0x001d, 0x000b,
0x0011, 0x000c, 0x0010, 0x0008, 0x0001, 0x0001, 0x0000, 0x0001,
};
static const uint8_t mpa_huffbits_13[256] = {
1, 4, 6, 7, 8, 9, 9, 10,
9, 10, 11, 11, 12, 12, 13, 13,
3, 4, 6, 7, 8, 8, 9, 9,
9, 9, 10, 10, 11, 12, 12, 12,
6, 6, 7, 8, 9, 9, 10, 10,
9, 10, 10, 11, 11, 12, 13, 13,
7, 7, 8, 9, 9, 10, 10, 10,
10, 11, 11, 11, 11, 12, 13, 13,
8, 7, 9, 9, 10, 10, 11, 11,
10, 11, 11, 12, 12, 13, 13, 14,
9, 8, 9, 10, 10, 10, 11, 11,
11, 11, 12, 11, 13, 13, 14, 14,
9, 9, 10, 10, 11, 11, 11, 11,
11, 12, 12, 12, 13, 13, 14, 14,
10, 9, 10, 11, 11, 11, 12, 12,
12, 12, 13, 13, 13, 14, 16, 16,
9, 8, 9, 10, 10, 11, 11, 12,
12, 12, 12, 13, 13, 14, 15, 15,
10, 9, 10, 10, 11, 11, 11, 13,
12, 13, 13, 14, 14, 14, 16, 15,
10, 10, 10, 11, 11, 12, 12, 13,
12, 13, 14, 13, 14, 15, 16, 17,
11, 10, 10, 11, 12, 12, 12, 12,
13, 13, 13, 14, 15, 15, 15, 16,
11, 11, 11, 12, 12, 13, 12, 13,
14, 14, 15, 15, 15, 16, 16, 16,
12, 11, 12, 13, 13, 13, 14, 14,
14, 14, 14, 15, 16, 15, 16, 16,
13, 12, 12, 13, 13, 13, 15, 14,
14, 17, 15, 15, 15, 17, 16, 16,
12, 12, 13, 14, 14, 14, 15, 14,
15, 15, 16, 16, 19, 18, 19, 16,
};
static const uint16_t mpa_huffcodes_15[256] = {
0x0007, 0x000c, 0x0012, 0x0035, 0x002f, 0x004c, 0x007c, 0x006c,
0x0059, 0x007b, 0x006c, 0x0077, 0x006b, 0x0051, 0x007a, 0x003f,
0x000d, 0x0005, 0x0010, 0x001b, 0x002e, 0x0024, 0x003d, 0x0033,
0x002a, 0x0046, 0x0034, 0x0053, 0x0041, 0x0029, 0x003b, 0x0024,
0x0013, 0x0011, 0x000f, 0x0018, 0x0029, 0x0022, 0x003b, 0x0030,
0x0028, 0x0040, 0x0032, 0x004e, 0x003e, 0x0050, 0x0038, 0x0021,
0x001d, 0x001c, 0x0019, 0x002b, 0x0027, 0x003f, 0x0037, 0x005d,
0x004c, 0x003b, 0x005d, 0x0048, 0x0036, 0x004b, 0x0032, 0x001d,
0x0034, 0x0016, 0x002a, 0x0028, 0x0043, 0x0039, 0x005f, 0x004f,
0x0048, 0x0039, 0x0059, 0x0045, 0x0031, 0x0042, 0x002e, 0x001b,
0x004d, 0x0025, 0x0023, 0x0042, 0x003a, 0x0034, 0x005b, 0x004a,
0x003e, 0x0030, 0x004f, 0x003f, 0x005a, 0x003e, 0x0028, 0x0026,
0x007d, 0x0020, 0x003c, 0x0038, 0x0032, 0x005c, 0x004e, 0x0041,
0x0037, 0x0057, 0x0047, 0x0033, 0x0049, 0x0033, 0x0046, 0x001e,
0x006d, 0x0035, 0x0031, 0x005e, 0x0058, 0x004b, 0x0042, 0x007a,
0x005b, 0x0049, 0x0038, 0x002a, 0x0040, 0x002c, 0x0015, 0x0019,
0x005a, 0x002b, 0x0029, 0x004d, 0x0049, 0x003f, 0x0038, 0x005c,
0x004d, 0x0042, 0x002f, 0x0043, 0x0030, 0x0035, 0x0024, 0x0014,
0x0047, 0x0022, 0x0043, 0x003c, 0x003a, 0x0031, 0x0058, 0x004c,
0x0043, 0x006a, 0x0047, 0x0036, 0x0026, 0x0027, 0x0017, 0x000f,
0x006d, 0x0035, 0x0033, 0x002f, 0x005a, 0x0052, 0x003a, 0x0039,
0x0030, 0x0048, 0x0039, 0x0029, 0x0017, 0x001b, 0x003e, 0x0009,
0x0056, 0x002a, 0x0028, 0x0025, 0x0046, 0x0040, 0x0034, 0x002b,
0x0046, 0x0037, 0x002a, 0x0019, 0x001d, 0x0012, 0x000b, 0x000b,
0x0076, 0x0044, 0x001e, 0x0037, 0x0032, 0x002e, 0x004a, 0x0041,
0x0031, 0x0027, 0x0018, 0x0010, 0x0016, 0x000d, 0x000e, 0x0007,
0x005b, 0x002c, 0x0027, 0x0026, 0x0022, 0x003f, 0x0034, 0x002d,
0x001f, 0x0034, 0x001c, 0x0013, 0x000e, 0x0008, 0x0009, 0x0003,
0x007b, 0x003c, 0x003a, 0x0035, 0x002f, 0x002b, 0x0020, 0x0016,
0x0025, 0x0018, 0x0011, 0x000c, 0x000f, 0x000a, 0x0002, 0x0001,
0x0047, 0x0025, 0x0022, 0x001e, 0x001c, 0x0014, 0x0011, 0x001a,
0x0015, 0x0010, 0x000a, 0x0006, 0x0008, 0x0006, 0x0002, 0x0000,
};
static const uint8_t mpa_huffbits_15[256] = {
3, 4, 5, 7, 7, 8, 9, 9,
9, 10, 10, 11, 11, 11, 12, 13,
4, 3, 5, 6, 7, 7, 8, 8,
8, 9, 9, 10, 10, 10, 11, 11,
5, 5, 5, 6, 7, 7, 8, 8,
8, 9, 9, 10, 10, 11, 11, 11,
6, 6, 6, 7, 7, 8, 8, 9,
9, 9, 10, 10, 10, 11, 11, 11,
7, 6, 7, 7, 8, 8, 9, 9,
9, 9, 10, 10, 10, 11, 11, 11,
8, 7, 7, 8, 8, 8, 9, 9,
9, 9, 10, 10, 11, 11, 11, 12,
9, 7, 8, 8, 8, 9, 9, 9,
9, 10, 10, 10, 11, 11, 12, 12,
9, 8, 8, 9, 9, 9, 9, 10,
10, 10, 10, 10, 11, 11, 11, 12,
9, 8, 8, 9, 9, 9, 9, 10,
10, 10, 10, 11, 11, 12, 12, 12,
9, 8, 9, 9, 9, 9, 10, 10,
10, 11, 11, 11, 11, 12, 12, 12,
10, 9, 9, 9, 10, 10, 10, 10,
10, 11, 11, 11, 11, 12, 13, 12,
10, 9, 9, 9, 10, 10, 10, 10,
11, 11, 11, 11, 12, 12, 12, 13,
11, 10, 9, 10, 10, 10, 11, 11,
11, 11, 11, 11, 12, 12, 13, 13,
11, 10, 10, 10, 10, 11, 11, 11,
11, 12, 12, 12, 12, 12, 13, 13,
12, 11, 11, 11, 11, 11, 11, 11,
12, 12, 12, 12, 13, 13, 12, 13,
12, 11, 11, 11, 11, 11, 11, 12,
12, 12, 12, 12, 13, 13, 13, 13,
};
static const uint16_t mpa_huffcodes_16[256] = {
0x0001, 0x0005, 0x000e, 0x002c, 0x004a, 0x003f, 0x006e, 0x005d,
0x00ac, 0x0095, 0x008a, 0x00f2, 0x00e1, 0x00c3, 0x0178, 0x0011,
0x0003, 0x0004, 0x000c, 0x0014, 0x0023, 0x003e, 0x0035, 0x002f,
0x0053, 0x004b, 0x0044, 0x0077, 0x00c9, 0x006b, 0x00cf, 0x0009,
0x000f, 0x000d, 0x0017, 0x0026, 0x0043, 0x003a, 0x0067, 0x005a,
0x00a1, 0x0048, 0x007f, 0x0075, 0x006e, 0x00d1, 0x00ce, 0x0010,
0x002d, 0x0015, 0x0027, 0x0045, 0x0040, 0x0072, 0x0063, 0x0057,
0x009e, 0x008c, 0x00fc, 0x00d4, 0x00c7, 0x0183, 0x016d, 0x001a,
0x004b, 0x0024, 0x0044, 0x0041, 0x0073, 0x0065, 0x00b3, 0x00a4,
0x009b, 0x0108, 0x00f6, 0x00e2, 0x018b, 0x017e, 0x016a, 0x0009,
0x0042, 0x001e, 0x003b, 0x0038, 0x0066, 0x00b9, 0x00ad, 0x0109,
0x008e, 0x00fd, 0x00e8, 0x0190, 0x0184, 0x017a, 0x01bd, 0x0010,
0x006f, 0x0036, 0x0034, 0x0064, 0x00b8, 0x00b2, 0x00a0, 0x0085,
0x0101, 0x00f4, 0x00e4, 0x00d9, 0x0181, 0x016e, 0x02cb, 0x000a,
0x0062, 0x0030, 0x005b, 0x0058, 0x00a5, 0x009d, 0x0094, 0x0105,
0x00f8, 0x0197, 0x018d, 0x0174, 0x017c, 0x0379, 0x0374, 0x0008,
0x0055, 0x0054, 0x0051, 0x009f, 0x009c, 0x008f, 0x0104, 0x00f9,
0x01ab, 0x0191, 0x0188, 0x017f, 0x02d7, 0x02c9, 0x02c4, 0x0007,
0x009a, 0x004c, 0x0049, 0x008d, 0x0083, 0x0100, 0x00f5, 0x01aa,
0x0196, 0x018a, 0x0180, 0x02df, 0x0167, 0x02c6, 0x0160, 0x000b,
0x008b, 0x0081, 0x0043, 0x007d, 0x00f7, 0x00e9, 0x00e5, 0x00db,
0x0189, 0x02e7, 0x02e1, 0x02d0, 0x0375, 0x0372, 0x01b7, 0x0004,
0x00f3, 0x0078, 0x0076, 0x0073, 0x00e3, 0x00df, 0x018c, 0x02ea,
0x02e6, 0x02e0, 0x02d1, 0x02c8, 0x02c2, 0x00df, 0x01b4, 0x0006,
0x00ca, 0x00e0, 0x00de, 0x00da, 0x00d8, 0x0185, 0x0182, 0x017d,
0x016c, 0x0378, 0x01bb, 0x02c3, 0x01b8, 0x01b5, 0x06c0, 0x0004,
0x02eb, 0x00d3, 0x00d2, 0x00d0, 0x0172, 0x017b, 0x02de, 0x02d3,
0x02ca, 0x06c7, 0x0373, 0x036d, 0x036c, 0x0d83, 0x0361, 0x0002,
0x0179, 0x0171, 0x0066, 0x00bb, 0x02d6, 0x02d2, 0x0166, 0x02c7,
0x02c5, 0x0362, 0x06c6, 0x0367, 0x0d82, 0x0366, 0x01b2, 0x0000,
0x000c, 0x000a, 0x0007, 0x000b, 0x000a, 0x0011, 0x000b, 0x0009,
0x000d, 0x000c, 0x000a, 0x0007, 0x0005, 0x0003, 0x0001, 0x0003,
};
static const uint8_t mpa_huffbits_16[256] = {
1, 4, 6, 8, 9, 9, 10, 10,
11, 11, 11, 12, 12, 12, 13, 9,
3, 4, 6, 7, 8, 9, 9, 9,
10, 10, 10, 11, 12, 11, 12, 8,
6, 6, 7, 8, 9, 9, 10, 10,
11, 10, 11, 11, 11, 12, 12, 9,
8, 7, 8, 9, 9, 10, 10, 10,
11, 11, 12, 12, 12, 13, 13, 10,
9, 8, 9, 9, 10, 10, 11, 11,
11, 12, 12, 12, 13, 13, 13, 9,
9, 8, 9, 9, 10, 11, 11, 12,
11, 12, 12, 13, 13, 13, 14, 10,
10, 9, 9, 10, 11, 11, 11, 11,
12, 12, 12, 12, 13, 13, 14, 10,
10, 9, 10, 10, 11, 11, 11, 12,
12, 13, 13, 13, 13, 15, 15, 10,
10, 10, 10, 11, 11, 11, 12, 12,
13, 13, 13, 13, 14, 14, 14, 10,
11, 10, 10, 11, 11, 12, 12, 13,
13, 13, 13, 14, 13, 14, 13, 11,
11, 11, 10, 11, 12, 12, 12, 12,
13, 14, 14, 14, 15, 15, 14, 10,
12, 11, 11, 11, 12, 12, 13, 14,
14, 14, 14, 14, 14, 13, 14, 11,
12, 12, 12, 12, 12, 13, 13, 13,
13, 15, 14, 14, 14, 14, 16, 11,
14, 12, 12, 12, 13, 13, 14, 14,
14, 16, 15, 15, 15, 17, 15, 11,
13, 13, 11, 12, 14, 14, 13, 14,
14, 15, 16, 15, 17, 15, 14, 11,
9, 8, 8, 9, 9, 10, 10, 10,
11, 11, 11, 11, 11, 11, 11, 8,
};
static const uint16_t mpa_huffcodes_24[256] = {
0x000f, 0x000d, 0x002e, 0x0050, 0x0092, 0x0106, 0x00f8, 0x01b2,
0x01aa, 0x029d, 0x028d, 0x0289, 0x026d, 0x0205, 0x0408, 0x0058,
0x000e, 0x000c, 0x0015, 0x0026, 0x0047, 0x0082, 0x007a, 0x00d8,
0x00d1, 0x00c6, 0x0147, 0x0159, 0x013f, 0x0129, 0x0117, 0x002a,
0x002f, 0x0016, 0x0029, 0x004a, 0x0044, 0x0080, 0x0078, 0x00dd,
0x00cf, 0x00c2, 0x00b6, 0x0154, 0x013b, 0x0127, 0x021d, 0x0012,
0x0051, 0x0027, 0x004b, 0x0046, 0x0086, 0x007d, 0x0074, 0x00dc,
0x00cc, 0x00be, 0x00b2, 0x0145, 0x0137, 0x0125, 0x010f, 0x0010,
0x0093, 0x0048, 0x0045, 0x0087, 0x007f, 0x0076, 0x0070, 0x00d2,
0x00c8, 0x00bc, 0x0160, 0x0143, 0x0132, 0x011d, 0x021c, 0x000e,
0x0107, 0x0042, 0x0081, 0x007e, 0x0077, 0x0072, 0x00d6, 0x00ca,
0x00c0, 0x00b4, 0x0155, 0x013d, 0x012d, 0x0119, 0x0106, 0x000c,
0x00f9, 0x007b, 0x0079, 0x0075, 0x0071, 0x00d7, 0x00ce, 0x00c3,
0x00b9, 0x015b, 0x014a, 0x0134, 0x0123, 0x0110, 0x0208, 0x000a,
0x01b3, 0x0073, 0x006f, 0x006d, 0x00d3, 0x00cb, 0x00c4, 0x00bb,
0x0161, 0x014c, 0x0139, 0x012a, 0x011b, 0x0213, 0x017d, 0x0011,
0x01ab, 0x00d4, 0x00d0, 0x00cd, 0x00c9, 0x00c1, 0x00ba, 0x00b1,
0x00a9, 0x0140, 0x012f, 0x011e, 0x010c, 0x0202, 0x0179, 0x0010,
0x014f, 0x00c7, 0x00c5, 0x00bf, 0x00bd, 0x00b5, 0x00ae, 0x014d,
0x0141, 0x0131, 0x0121, 0x0113, 0x0209, 0x017b, 0x0173, 0x000b,
0x029c, 0x00b8, 0x00b7, 0x00b3, 0x00af, 0x0158, 0x014b, 0x013a,
0x0130, 0x0122, 0x0115, 0x0212, 0x017f, 0x0175, 0x016e, 0x000a,
0x028c, 0x015a, 0x00ab, 0x00a8, 0x00a4, 0x013e, 0x0135, 0x012b,
0x011f, 0x0114, 0x0107, 0x0201, 0x0177, 0x0170, 0x016a, 0x0006,
0x0288, 0x0142, 0x013c, 0x0138, 0x0133, 0x012e, 0x0124, 0x011c,
0x010d, 0x0105, 0x0200, 0x0178, 0x0172, 0x016c, 0x0167, 0x0004,
0x026c, 0x012c, 0x0128, 0x0126, 0x0120, 0x011a, 0x0111, 0x010a,
0x0203, 0x017c, 0x0176, 0x0171, 0x016d, 0x0169, 0x0165, 0x0002,
0x0409, 0x0118, 0x0116, 0x0112, 0x010b, 0x0108, 0x0103, 0x017e,
0x017a, 0x0174, 0x016f, 0x016b, 0x0168, 0x0166, 0x0164, 0x0000,
0x002b, 0x0014, 0x0013, 0x0011, 0x000f, 0x000d, 0x000b, 0x0009,
0x0007, 0x0006, 0x0004, 0x0007, 0x0005, 0x0003, 0x0001, 0x0003,
};
static const uint8_t mpa_huffbits_24[256] = {
4, 4, 6, 7, 8, 9, 9, 10,
10, 11, 11, 11, 11, 11, 12, 9,
4, 4, 5, 6, 7, 8, 8, 9,
9, 9, 10, 10, 10, 10, 10, 8,
6, 5, 6, 7, 7, 8, 8, 9,
9, 9, 9, 10, 10, 10, 11, 7,
7, 6, 7, 7, 8, 8, 8, 9,
9, 9, 9, 10, 10, 10, 10, 7,
8, 7, 7, 8, 8, 8, 8, 9,
9, 9, 10, 10, 10, 10, 11, 7,
9, 7, 8, 8, 8, 8, 9, 9,
9, 9, 10, 10, 10, 10, 10, 7,
9, 8, 8, 8, 8, 9, 9, 9,
9, 10, 10, 10, 10, 10, 11, 7,
10, 8, 8, 8, 9, 9, 9, 9,
10, 10, 10, 10, 10, 11, 11, 8,
10, 9, 9, 9, 9, 9, 9, 9,
9, 10, 10, 10, 10, 11, 11, 8,
10, 9, 9, 9, 9, 9, 9, 10,
10, 10, 10, 10, 11, 11, 11, 8,
11, 9, 9, 9, 9, 10, 10, 10,
10, 10, 10, 11, 11, 11, 11, 8,
11, 10, 9, 9, 9, 10, 10, 10,
10, 10, 10, 11, 11, 11, 11, 8,
11, 10, 10, 10, 10, 10, 10, 10,
10, 10, 11, 11, 11, 11, 11, 8,
11, 10, 10, 10, 10, 10, 10, 10,
11, 11, 11, 11, 11, 11, 11, 8,
12, 10, 10, 10, 10, 10, 10, 11,
11, 11, 11, 11, 11, 11, 11, 8,
8, 7, 7, 7, 7, 7, 7, 7,
7, 7, 7, 8, 8, 8, 8, 4,
};
static const HuffTable mpa_huff_tables[16] = {
{ 1, NULL, NULL },
{ 2, mpa_huffbits_1, mpa_huffcodes_1 },
{ 3, mpa_huffbits_2, mpa_huffcodes_2 },
{ 3, mpa_huffbits_3, mpa_huffcodes_3 },
{ 4, mpa_huffbits_5, mpa_huffcodes_5 },
{ 4, mpa_huffbits_6, mpa_huffcodes_6 },
{ 6, mpa_huffbits_7, mpa_huffcodes_7 },
{ 6, mpa_huffbits_8, mpa_huffcodes_8 },
{ 6, mpa_huffbits_9, mpa_huffcodes_9 },
{ 8, mpa_huffbits_10, mpa_huffcodes_10 },
{ 8, mpa_huffbits_11, mpa_huffcodes_11 },
{ 8, mpa_huffbits_12, mpa_huffcodes_12 },
{ 16, mpa_huffbits_13, mpa_huffcodes_13 },
{ 16, mpa_huffbits_15, mpa_huffcodes_15 },
{ 16, mpa_huffbits_16, mpa_huffcodes_16 },
{ 16, mpa_huffbits_24, mpa_huffcodes_24 },
};
static const uint8_t mpa_huff_data[32][2] = {
{ 0, 0 },
{ 1, 0 },
{ 2, 0 },
{ 3, 0 },
{ 0, 0 },
{ 4, 0 },
{ 5, 0 },
{ 6, 0 },
{ 7, 0 },
{ 8, 0 },
{ 9, 0 },
{ 10, 0 },
{ 11, 0 },
{ 12, 0 },
{ 0, 0 },
{ 13, 0 },
{ 14, 1 },
{ 14, 2 },
{ 14, 3 },
{ 14, 4 },
{ 14, 6 },
{ 14, 8 },
{ 14, 10 },
{ 14, 13 },
{ 15, 4 },
{ 15, 5 },
{ 15, 6 },
{ 15, 7 },
{ 15, 8 },
{ 15, 9 },
{ 15, 11 },
{ 15, 13 },
};
/* huffman tables for quadrules */
static const uint8_t mpa_quad_codes[2][16] = {
{ 1, 5, 4, 5, 6, 5, 4, 4, 7, 3, 6, 0, 7, 2, 3, 1, },
{ 15, 14, 13, 12, 11, 10, 9, 8, 7, 6, 5, 4, 3, 2, 1, 0, },
};
static const uint8_t mpa_quad_bits[2][16] = {
{ 1, 4, 4, 5, 4, 6, 5, 6, 4, 5, 5, 6, 5, 6, 6, 6, },
{ 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, 4, },
};
/* band size tables */
static const uint8_t band_size_long[9][22] = {
{ 4, 4, 4, 4, 4, 4, 6, 6, 8, 8, 10,
12, 16, 20, 24, 28, 34, 42, 50, 54, 76, 158, }, /* 44100 */
{ 4, 4, 4, 4, 4, 4, 6, 6, 6, 8, 10,
12, 16, 18, 22, 28, 34, 40, 46, 54, 54, 192, }, /* 48000 */
{ 4, 4, 4, 4, 4, 4, 6, 6, 8, 10, 12,
16, 20, 24, 30, 38, 46, 56, 68, 84, 102, 26, }, /* 32000 */
{ 6, 6, 6, 6, 6, 6, 8, 10, 12, 14, 16,
20, 24, 28, 32, 38, 46, 52, 60, 68, 58, 54, }, /* 22050 */
{ 6, 6, 6, 6, 6, 6, 8, 10, 12, 14, 16,
18, 22, 26, 32, 38, 46, 52, 64, 70, 76, 36, }, /* 24000 */
{ 6, 6, 6, 6, 6, 6, 8, 10, 12, 14, 16,
20, 24, 28, 32, 38, 46, 52, 60, 68, 58, 54, }, /* 16000 */
{ 6, 6, 6, 6, 6, 6, 8, 10, 12, 14, 16,
20, 24, 28, 32, 38, 46, 52, 60, 68, 58, 54, }, /* 11025 */
{ 6, 6, 6, 6, 6, 6, 8, 10, 12, 14, 16,
20, 24, 28, 32, 38, 46, 52, 60, 68, 58, 54, }, /* 12000 */
{ 12, 12, 12, 12, 12, 12, 16, 20, 24, 28, 32,
40, 48, 56, 64, 76, 90, 2, 2, 2, 2, 2, }, /* 8000 */
};
static const uint8_t band_size_short[9][13] = {
{ 4, 4, 4, 4, 6, 8, 10, 12, 14, 18, 22, 30, 56, }, /* 44100 */
{ 4, 4, 4, 4, 6, 6, 10, 12, 14, 16, 20, 26, 66, }, /* 48000 */
{ 4, 4, 4, 4, 6, 8, 12, 16, 20, 26, 34, 42, 12, }, /* 32000 */
{ 4, 4, 4, 6, 6, 8, 10, 14, 18, 26, 32, 42, 18, }, /* 22050 */
{ 4, 4, 4, 6, 8, 10, 12, 14, 18, 24, 32, 44, 12, }, /* 24000 */
{ 4, 4, 4, 6, 8, 10, 12, 14, 18, 24, 30, 40, 18, }, /* 16000 */
{ 4, 4, 4, 6, 8, 10, 12, 14, 18, 24, 30, 40, 18, }, /* 11025 */
{ 4, 4, 4, 6, 8, 10, 12, 14, 18, 24, 30, 40, 18, }, /* 12000 */
{ 8, 8, 8, 12, 16, 20, 24, 28, 36, 2, 2, 2, 26, }, /* 8000 */
};
static const uint8_t mpa_pretab[2][22] = {
{ 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0 },
{ 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 1, 1, 1, 1, 2, 2, 3, 3, 3, 2, 0 },
};
/* table for alias reduction (XXX: store it as integer !) */
static const float ci_table[8] = {
-0.6, -0.535, -0.33, -0.185, -0.095, -0.041, -0.0142, -0.0037,
};
#endif /* AVCODEC_MPEGAUDIODECTAB_H */

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/*
* Copyright (c) 2011 Mans Rullgard
*
* This file is part of FFmpeg.
*
* FFmpeg 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.
*
* FFmpeg 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 FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include "config.h"
#include "libavutil/attributes.h"
#include "libavutil/thread.h"
#include "mpegaudiodsp.h"
#include "dct.h"
#include "dct32.h"
static AVOnce mpadsp_float_table_init = AV_ONCE_INIT;
static AVOnce mpadsp_fixed_table_init = AV_ONCE_INIT;
av_cold void ff_mpadsp_init(MPADSPContext *s)
{
DCTContext dct;
ff_dct_init(&dct, 5, DCT_II);
ff_thread_once(&mpadsp_float_table_init, &ff_init_mpadsp_tabs_float);
ff_thread_once(&mpadsp_fixed_table_init, &ff_init_mpadsp_tabs_fixed);
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;
s->imdct36_blocks_float = ff_imdct36_blocks_float;
s->imdct36_blocks_fixed = ff_imdct36_blocks_fixed;
if (ARCH_AARCH64) ff_mpadsp_init_aarch64(s);
if (ARCH_ARM) ff_mpadsp_init_arm(s);
if (ARCH_PPC) ff_mpadsp_init_ppc(s);
if (ARCH_X86) ff_mpadsp_init_x86(s);
if (HAVE_MIPSFPU) ff_mpadsp_init_mipsfpu(s);
if (HAVE_MIPSDSP) ff_mpadsp_init_mipsdsp(s);
}

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/*
* This file is part of FFmpeg.
*
* FFmpeg 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.
*
* FFmpeg 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 FFmpeg; 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 <stddef.h>
#include <stdint.h>
#include "libavutil/common.h"
typedef struct MPADSPContext {
void (*apply_window_float)(float *synth_buf, float *window,
int *dither_state, float *samples,
ptrdiff_t incr);
void (*apply_window_fixed)(int32_t *synth_buf, int32_t *window,
int *dither_state, int16_t *samples,
ptrdiff_t incr);
void (*dct32_float)(float *dst, const float *src);
void (*dct32_fixed)(int *dst, const int *src);
void (*imdct36_blocks_float)(float *out, float *buf, float *in,
int count, int switch_point, int block_type);
void (*imdct36_blocks_fixed)(int *out, int *buf, int *in,
int count, int switch_point, int block_type);
} MPADSPContext;
void ff_mpadsp_init(MPADSPContext *s);
extern int32_t ff_mpa_synth_window_fixed[];
extern float ff_mpa_synth_window_float[];
extern const int32_t ff_mpa_enwindow[257];
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, ptrdiff_t incr,
int32_t *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, ptrdiff_t incr,
float *sb_samples);
void ff_mpadsp_init_aarch64(MPADSPContext *s);
void ff_mpadsp_init_arm(MPADSPContext *s);
void ff_mpadsp_init_ppc(MPADSPContext *s);
void ff_mpadsp_init_x86(MPADSPContext *s);
void ff_mpadsp_init_mipsfpu(MPADSPContext *s);
void ff_mpadsp_init_mipsdsp(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,
ptrdiff_t incr);
void ff_mpadsp_apply_window_fixed(int32_t *synth_buf, int32_t *window,
int *dither_state, int16_t *samples,
ptrdiff_t incr);
void ff_imdct36_blocks_float(float *out, float *buf, float *in,
int count, int switch_point, int block_type);
void ff_imdct36_blocks_fixed(int *out, int *buf, int *in,
int count, int switch_point, int block_type);
void ff_init_mpadsp_tabs_float(void);
void ff_init_mpadsp_tabs_fixed(void);
/** For SSE implementation, MDCT_BUF_SIZE/2 should be 128-bit aligned */
#define MDCT_BUF_SIZE FFALIGN(36, 2*4)
extern int ff_mdct_win_fixed[8][MDCT_BUF_SIZE];
extern float ff_mdct_win_float[8][MDCT_BUF_SIZE];
#endif /* AVCODEC_MPEGAUDIODSP_H */

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/*
* This file is part of FFmpeg.
*
* FFmpeg 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.
*
* FFmpeg 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 FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include "mpegaudiodsp.h"
/* half mpeg encoding window (full precision) */
const int32_t ff_mpa_enwindow[257] = {
0, -1, -1, -1, -1, -1, -1, -2,
-2, -2, -2, -3, -3, -4, -4, -5,
-5, -6, -7, -7, -8, -9, -10, -11,
-13, -14, -16, -17, -19, -21, -24, -26,
-29, -31, -35, -38, -41, -45, -49, -53,
-58, -63, -68, -73, -79, -85, -91, -97,
-104, -111, -117, -125, -132, -139, -147, -154,
-161, -169, -176, -183, -190, -196, -202, -208,
213, 218, 222, 225, 227, 228, 228, 227,
224, 221, 215, 208, 200, 189, 177, 163,
146, 127, 106, 83, 57, 29, -2, -36,
-72, -111, -153, -197, -244, -294, -347, -401,
-459, -519, -581, -645, -711, -779, -848, -919,
-991, -1064, -1137, -1210, -1283, -1356, -1428, -1498,
-1567, -1634, -1698, -1759, -1817, -1870, -1919, -1962,
-2001, -2032, -2057, -2075, -2085, -2087, -2080, -2063,
2037, 2000, 1952, 1893, 1822, 1739, 1644, 1535,
1414, 1280, 1131, 970, 794, 605, 402, 185,
-45, -288, -545, -814, -1095, -1388, -1692, -2006,
-2330, -2663, -3004, -3351, -3705, -4063, -4425, -4788,
-5153, -5517, -5879, -6237, -6589, -6935, -7271, -7597,
-7910, -8209, -8491, -8755, -8998, -9219, -9416, -9585,
-9727, -9838, -9916, -9959, -9966, -9935, -9863, -9750,
-9592, -9389, -9139, -8840, -8492, -8092, -7640, -7134,
6574, 5959, 5288, 4561, 3776, 2935, 2037, 1082,
70, -998, -2122, -3300, -4533, -5818, -7154, -8540,
-9975,-11455,-12980,-14548,-16155,-17799,-19478,-21189,
-22929,-24694,-26482,-28289,-30112,-31947,-33791,-35640,
-37489,-39336,-41176,-43006,-44821,-46617,-48390,-50137,
-51853,-53534,-55178,-56778,-58333,-59838,-61289,-62684,
-64019,-65290,-66494,-67629,-68692,-69679,-70590,-71420,
-72169,-72835,-73415,-73908,-74313,-74630,-74856,-74992,
75038,
};

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/*
* This file is part of FFmpeg.
*
* FFmpeg 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.
*
* FFmpeg 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 FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#define USE_FLOATS 0
#include "mpegaudiodsp_template.c"

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/*
* This file is part of FFmpeg.
*
* FFmpeg 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.
*
* FFmpeg 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 FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#define USE_FLOATS 1
#include "mpegaudiodsp_template.c"

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/*
* Copyright (c) 2001, 2002 Fabrice Bellard
*
* This file is part of FFmpeg.
*
* FFmpeg 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.
*
* FFmpeg 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 FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include <stdint.h>
#include "libavutil/attributes.h"
#include "libavutil/mem.h"
#include "dct32.h"
#include "mathops.h"
#include "mpegaudiodsp.h"
#include "mpegaudio.h"
#if USE_FLOATS
#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 MULH3(x, y, s) ((s)*(y)*(x))
#define MLSS(rt, ra, rb) rt-=(ra)*(rb)
#define MULLx(x, y, s) ((y)*(x))
#define FIXHR(x) ((float)(x))
#define FIXR(x) ((float)(x))
#define SHR(a,b) ((a)*(1.0f/(1<<(b))))
#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)
# define MULH3(x, y, s) MULH((s)*(x), y)
# define MULLx(x, y, s) MULL((int)(x),(y),s)
# define SHR(a,b) (((int)(a))>>(b))
# define FIXR(a) ((int)((a) * FRAC_ONE + 0.5))
# define FIXHR(a) ((int)((a) * (1LL<<32) + 0.5))
#endif
/** Window for MDCT. Actually only the elements in [0,17] and
[MDCT_BUF_SIZE/2, MDCT_BUF_SIZE/2 + 17] are actually used. The rest
is just to preserve alignment for SIMD implementations.
*/
DECLARE_ALIGNED(16, INTFLOAT, RENAME(ff_mdct_win))[8][MDCT_BUF_SIZE];
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,
ptrdiff_t incr)
{
register const MPA_INT *w, *w2, *p;
int j;
OUT_INT *samples2;
#if USE_FLOATS
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, ptrdiff_t 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;
}
av_cold void 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 USE_FLOATS
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];
}
av_cold void RENAME(ff_init_mpadsp_tabs)(void)
{
int i, j;
/* compute mdct windows */
for (i = 0; i < 36; i++) {
for (j = 0; j < 4; j++) {
double d;
if (j == 2 && i % 3 != 1)
continue;
d = sin(M_PI * (i + 0.5) / 36.0);
if (j == 1) {
if (i >= 30) d = 0;
else if (i >= 24) d = sin(M_PI * (i - 18 + 0.5) / 12.0);
else if (i >= 18) d = 1;
} else if (j == 3) {
if (i < 6) d = 0;
else if (i < 12) d = sin(M_PI * (i - 6 + 0.5) / 12.0);
else if (i < 18) d = 1;
}
//merge last stage of imdct into the window coefficients
d *= 0.5 * IMDCT_SCALAR / cos(M_PI * (2 * i + 19) / 72);
if (j == 2)
RENAME(ff_mdct_win)[j][i/3] = FIXHR((d / (1<<5)));
else {
int idx = i < 18 ? i : i + (MDCT_BUF_SIZE/2 - 18);
RENAME(ff_mdct_win)[j][idx] = FIXHR((d / (1<<5)));
}
}
}
/* NOTE: we do frequency inversion adter the MDCT by changing
the sign of the right window coefs */
for (j = 0; j < 4; j++) {
for (i = 0; i < MDCT_BUF_SIZE; i += 2) {
RENAME(ff_mdct_win)[j + 4][i ] = RENAME(ff_mdct_win)[j][i ];
RENAME(ff_mdct_win)[j + 4][i + 1] = -RENAME(ff_mdct_win)[j][i + 1];
}
}
}
/* cos(pi*i/18) */
#define C1 FIXHR(0.98480775301220805936/2)
#define C2 FIXHR(0.93969262078590838405/2)
#define C3 FIXHR(0.86602540378443864676/2)
#define C4 FIXHR(0.76604444311897803520/2)
#define C5 FIXHR(0.64278760968653932632/2)
#define C6 FIXHR(0.5/2)
#define C7 FIXHR(0.34202014332566873304/2)
#define C8 FIXHR(0.17364817766693034885/2)
/* 0.5 / cos(pi*(2*i+1)/36) */
static const INTFLOAT icos36[9] = {
FIXR(0.50190991877167369479),
FIXR(0.51763809020504152469), //0
FIXR(0.55168895948124587824),
FIXR(0.61038729438072803416),
FIXR(0.70710678118654752439), //1
FIXR(0.87172339781054900991),
FIXR(1.18310079157624925896),
FIXR(1.93185165257813657349), //2
FIXR(5.73685662283492756461),
};
/* 0.5 / cos(pi*(2*i+1)/36) */
static const INTFLOAT icos36h[9] = {
FIXHR(0.50190991877167369479/2),
FIXHR(0.51763809020504152469/2), //0
FIXHR(0.55168895948124587824/2),
FIXHR(0.61038729438072803416/2),
FIXHR(0.70710678118654752439/2), //1
FIXHR(0.87172339781054900991/2),
FIXHR(1.18310079157624925896/4),
FIXHR(1.93185165257813657349/4), //2
// FIXHR(5.73685662283492756461),
};
/* using Lee like decomposition followed by hand coded 9 points DCT */
static void imdct36(INTFLOAT *out, INTFLOAT *buf, SUINTFLOAT *in, INTFLOAT *win)
{
int i, j;
SUINTFLOAT t0, t1, t2, t3, s0, s1, s2, s3;
SUINTFLOAT tmp[18], *tmp1, *in1;
for (i = 17; i >= 1; i--)
in[i] += in[i-1];
for (i = 17; i >= 3; i -= 2)
in[i] += in[i-2];
for (j = 0; j < 2; j++) {
tmp1 = tmp + j;
in1 = in + j;
t2 = in1[2*4] + in1[2*8] - in1[2*2];
t3 = in1[2*0] + SHR(in1[2*6],1);
t1 = in1[2*0] - in1[2*6];
tmp1[ 6] = t1 - SHR(t2,1);
tmp1[16] = t1 + t2;
t0 = MULH3(in1[2*2] + in1[2*4] , C2, 2);
t1 = MULH3(in1[2*4] - in1[2*8] , -2*C8, 1);
t2 = MULH3(in1[2*2] + in1[2*8] , -C4, 2);
tmp1[10] = t3 - t0 - t2;
tmp1[ 2] = t3 + t0 + t1;
tmp1[14] = t3 + t2 - t1;
tmp1[ 4] = MULH3(in1[2*5] + in1[2*7] - in1[2*1], -C3, 2);
t2 = MULH3(in1[2*1] + in1[2*5], C1, 2);
t3 = MULH3(in1[2*5] - in1[2*7], -2*C7, 1);
t0 = MULH3(in1[2*3], C3, 2);
t1 = MULH3(in1[2*1] + in1[2*7], -C5, 2);
tmp1[ 0] = t2 + t3 + t0;
tmp1[12] = t2 + t1 - t0;
tmp1[ 8] = t3 - t1 - t0;
}
i = 0;
for (j = 0; j < 4; j++) {
t0 = tmp[i];
t1 = tmp[i + 2];
s0 = t1 + t0;
s2 = t1 - t0;
t2 = tmp[i + 1];
t3 = tmp[i + 3];
s1 = MULH3(t3 + t2, icos36h[ j], 2);
s3 = MULLx(t3 - t2, icos36 [8 - j], FRAC_BITS);
t0 = s0 + s1;
t1 = s0 - s1;
out[(9 + j) * SBLIMIT] = MULH3(t1, win[ 9 + j], 1) + buf[4*(9 + j)];
out[(8 - j) * SBLIMIT] = MULH3(t1, win[ 8 - j], 1) + buf[4*(8 - j)];
buf[4 * ( 9 + j )] = MULH3(t0, win[MDCT_BUF_SIZE/2 + 9 + j], 1);
buf[4 * ( 8 - j )] = MULH3(t0, win[MDCT_BUF_SIZE/2 + 8 - j], 1);
t0 = s2 + s3;
t1 = s2 - s3;
out[(9 + 8 - j) * SBLIMIT] = MULH3(t1, win[ 9 + 8 - j], 1) + buf[4*(9 + 8 - j)];
out[ j * SBLIMIT] = MULH3(t1, win[ j], 1) + buf[4*( j)];
buf[4 * ( 9 + 8 - j )] = MULH3(t0, win[MDCT_BUF_SIZE/2 + 9 + 8 - j], 1);
buf[4 * ( j )] = MULH3(t0, win[MDCT_BUF_SIZE/2 + j], 1);
i += 4;
}
s0 = tmp[16];
s1 = MULH3(tmp[17], icos36h[4], 2);
t0 = s0 + s1;
t1 = s0 - s1;
out[(9 + 4) * SBLIMIT] = MULH3(t1, win[ 9 + 4], 1) + buf[4*(9 + 4)];
out[(8 - 4) * SBLIMIT] = MULH3(t1, win[ 8 - 4], 1) + buf[4*(8 - 4)];
buf[4 * ( 9 + 4 )] = MULH3(t0, win[MDCT_BUF_SIZE/2 + 9 + 4], 1);
buf[4 * ( 8 - 4 )] = MULH3(t0, win[MDCT_BUF_SIZE/2 + 8 - 4], 1);
}
void RENAME(ff_imdct36_blocks)(INTFLOAT *out, INTFLOAT *buf, INTFLOAT *in,
int count, int switch_point, int block_type)
{
int j;
for (j=0 ; j < count; j++) {
/* apply window & overlap with previous buffer */
/* select window */
int win_idx = (switch_point && j < 2) ? 0 : block_type;
INTFLOAT *win = RENAME(ff_mdct_win)[win_idx + (4 & -(j & 1))];
imdct36(out, buf, in, win);
in += 18;
buf += ((j&3) != 3 ? 1 : (72-3));
out++;
}
}

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/*
* The simplest mpeg audio layer 2 encoder
* Copyright (c) 2000, 2001 Fabrice Bellard
*
* This file is part of FFmpeg.
*
* FFmpeg 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.
*
* FFmpeg 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 FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include "mpegaudioenc_template.c"
AVCodec ff_mp2fixed_encoder = {
.name = "mp2fixed",
.long_name = NULL_IF_CONFIG_SMALL("MP2 fixed point (MPEG audio layer 2)"),
.type = AVMEDIA_TYPE_AUDIO,
.id = AV_CODEC_ID_MP2,
.priv_data_size = sizeof(MpegAudioContext),
.init = MPA_encode_init,
.encode2 = MPA_encode_frame,
.sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16,
AV_SAMPLE_FMT_NONE },
.supported_samplerates = (const int[]){
44100, 48000, 32000, 22050, 24000, 16000, 0
},
.channel_layouts = (const uint64_t[]){ AV_CH_LAYOUT_MONO,
AV_CH_LAYOUT_STEREO,
0 },
.defaults = mp2_defaults,
};

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/*
* The simplest mpeg audio layer 2 encoder
* Copyright (c) 2000, 2001 Fabrice Bellard
*
* This file is part of FFmpeg.
*
* FFmpeg 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.
*
* FFmpeg 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 FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#define USE_FLOATS 1
#include "mpegaudioenc_template.c"
AVCodec ff_mp2_encoder = {
.name = "mp2",
.long_name = NULL_IF_CONFIG_SMALL("MP2 (MPEG audio layer 2)"),
.type = AVMEDIA_TYPE_AUDIO,
.id = AV_CODEC_ID_MP2,
.priv_data_size = sizeof(MpegAudioContext),
.init = MPA_encode_init,
.encode2 = MPA_encode_frame,
.sample_fmts = (const enum AVSampleFormat[]){ AV_SAMPLE_FMT_S16,
AV_SAMPLE_FMT_NONE },
.supported_samplerates = (const int[]){
44100, 48000, 32000, 22050, 24000, 16000, 0
},
.channel_layouts = (const uint64_t[]){ AV_CH_LAYOUT_MONO,
AV_CH_LAYOUT_STEREO,
0 },
.defaults = mp2_defaults,
};

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/*
* The simplest mpeg audio layer 2 encoder
* Copyright (c) 2000, 2001 Fabrice Bellard
*
* This file is part of FFmpeg.
*
* FFmpeg 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.
*
* FFmpeg 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 FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
* @file
* The simplest mpeg audio layer 2 encoder.
*/
#include "libavutil/channel_layout.h"
#include "avcodec.h"
#include "internal.h"
#include "put_bits.h"
#define FRAC_BITS 15 /* fractional bits for sb_samples and dct */
#define WFRAC_BITS 14 /* fractional bits for window */
#include "mpegaudio.h"
#include "mpegaudiodsp.h"
#include "mpegaudiodata.h"
#include "mpegaudiotab.h"
/* currently, cannot change these constants (need to modify
quantization stage) */
#define MUL(a,b) (((int64_t)(a) * (int64_t)(b)) >> FRAC_BITS)
#define SAMPLES_BUF_SIZE 4096
typedef struct MpegAudioContext {
PutBitContext pb;
int nb_channels;
int lsf; /* 1 if mpeg2 low bitrate selected */
int bitrate_index; /* bit rate */
int freq_index;
int frame_size; /* frame size, in bits, without padding */
/* padding computation */
int frame_frac, frame_frac_incr, do_padding;
short samples_buf[MPA_MAX_CHANNELS][SAMPLES_BUF_SIZE]; /* buffer for filter */
int samples_offset[MPA_MAX_CHANNELS]; /* offset in samples_buf */
int sb_samples[MPA_MAX_CHANNELS][3][12][SBLIMIT];
unsigned char scale_factors[MPA_MAX_CHANNELS][SBLIMIT][3]; /* scale factors */
/* code to group 3 scale factors */
unsigned char scale_code[MPA_MAX_CHANNELS][SBLIMIT];
int sblimit; /* number of used subbands */
const unsigned char *alloc_table;
int16_t filter_bank[512];
int scale_factor_table[64];
unsigned char scale_diff_table[128];
#if USE_FLOATS
float scale_factor_inv_table[64];
#else
int8_t scale_factor_shift[64];
unsigned short scale_factor_mult[64];
#endif
unsigned short total_quant_bits[17]; /* total number of bits per allocation group */
} MpegAudioContext;
static av_cold int MPA_encode_init(AVCodecContext *avctx)
{
MpegAudioContext *s = avctx->priv_data;
int freq = avctx->sample_rate;
int bitrate = avctx->bit_rate;
int channels = avctx->channels;
int i, v, table;
float a;
if (channels <= 0 || channels > 2){
av_log(avctx, AV_LOG_ERROR, "encoding %d channel(s) is not allowed in mp2\n", channels);
return AVERROR(EINVAL);
}
bitrate = bitrate / 1000;
s->nb_channels = channels;
avctx->frame_size = MPA_FRAME_SIZE;
avctx->initial_padding = 512 - 32 + 1;
/* encoding freq */
s->lsf = 0;
for(i=0;i<3;i++) {
if (avpriv_mpa_freq_tab[i] == freq)
break;
if ((avpriv_mpa_freq_tab[i] / 2) == freq) {
s->lsf = 1;
break;
}
}
if (i == 3){
av_log(avctx, AV_LOG_ERROR, "Sampling rate %d is not allowed in mp2\n", freq);
return AVERROR(EINVAL);
}
s->freq_index = i;
/* encoding bitrate & frequency */
for(i=1;i<15;i++) {
if (avpriv_mpa_bitrate_tab[s->lsf][1][i] == bitrate)
break;
}
if (i == 15 && !avctx->bit_rate) {
i = 14;
bitrate = avpriv_mpa_bitrate_tab[s->lsf][1][i];
avctx->bit_rate = bitrate * 1000;
}
if (i == 15){
av_log(avctx, AV_LOG_ERROR, "bitrate %d is not allowed in mp2\n", bitrate);
return AVERROR(EINVAL);
}
s->bitrate_index = i;
/* compute total header size & pad bit */
a = (float)(bitrate * 1000 * MPA_FRAME_SIZE) / (freq * 8.0);
s->frame_size = ((int)a) * 8;
/* frame fractional size to compute padding */
s->frame_frac = 0;
s->frame_frac_incr = (int)((a - floor(a)) * 65536.0);
/* select the right allocation table */
table = ff_mpa_l2_select_table(bitrate, s->nb_channels, freq, s->lsf);
/* number of used subbands */
s->sblimit = ff_mpa_sblimit_table[table];
s->alloc_table = ff_mpa_alloc_tables[table];
ff_dlog(avctx, "%d kb/s, %d Hz, frame_size=%d bits, table=%d, padincr=%x\n",
bitrate, freq, s->frame_size, table, s->frame_frac_incr);
for(i=0;i<s->nb_channels;i++)
s->samples_offset[i] = 0;
for(i=0;i<257;i++) {
int v;
v = ff_mpa_enwindow[i];
#if WFRAC_BITS != 16
v = (v + (1 << (16 - WFRAC_BITS - 1))) >> (16 - WFRAC_BITS);
#endif
s->filter_bank[i] = v;
if ((i & 63) != 0)
v = -v;
if (i != 0)
s->filter_bank[512 - i] = v;
}
for(i=0;i<64;i++) {
v = (int)(exp2((3 - i) / 3.0) * (1 << 20));
if (v <= 0)
v = 1;
s->scale_factor_table[i] = v;
#if USE_FLOATS
s->scale_factor_inv_table[i] = exp2(-(3 - i) / 3.0) / (float)(1 << 20);
#else
#define P 15
s->scale_factor_shift[i] = 21 - P - (i / 3);
s->scale_factor_mult[i] = (1 << P) * exp2((i % 3) / 3.0);
#endif
}
for(i=0;i<128;i++) {
v = i - 64;
if (v <= -3)
v = 0;
else if (v < 0)
v = 1;
else if (v == 0)
v = 2;
else if (v < 3)
v = 3;
else
v = 4;
s->scale_diff_table[i] = v;
}
for(i=0;i<17;i++) {
v = ff_mpa_quant_bits[i];
if (v < 0)
v = -v;
else
v = v * 3;
s->total_quant_bits[i] = 12 * v;
}
return 0;
}
/* 32 point floating point IDCT without 1/sqrt(2) coef zero scaling */
static void idct32(int *out, int *tab)
{
int i, j;
int *t, *t1, xr;
const int *xp = costab32;
for(j=31;j>=3;j-=2) tab[j] += tab[j - 2];
t = tab + 30;
t1 = tab + 2;
do {
t[0] += t[-4];
t[1] += t[1 - 4];
t -= 4;
} while (t != t1);
t = tab + 28;
t1 = tab + 4;
do {
t[0] += t[-8];
t[1] += t[1-8];
t[2] += t[2-8];
t[3] += t[3-8];
t -= 8;
} while (t != t1);
t = tab;
t1 = tab + 32;
do {
t[ 3] = -t[ 3];
t[ 6] = -t[ 6];
t[11] = -t[11];
t[12] = -t[12];
t[13] = -t[13];
t[15] = -t[15];
t += 16;
} while (t != t1);
t = tab;
t1 = tab + 8;
do {
int x1, x2, x3, x4;
x3 = MUL(t[16], FIX(M_SQRT2*0.5));
x4 = t[0] - x3;
x3 = t[0] + x3;
x2 = MUL(-(t[24] + t[8]), FIX(M_SQRT2*0.5));
x1 = MUL((t[8] - x2), xp[0]);
x2 = MUL((t[8] + x2), xp[1]);
t[ 0] = x3 + x1;
t[ 8] = x4 - x2;
t[16] = x4 + x2;
t[24] = x3 - x1;
t++;
} while (t != t1);
xp += 2;
t = tab;
t1 = tab + 4;
do {
xr = MUL(t[28],xp[0]);
t[28] = (t[0] - xr);
t[0] = (t[0] + xr);
xr = MUL(t[4],xp[1]);
t[ 4] = (t[24] - xr);
t[24] = (t[24] + xr);
xr = MUL(t[20],xp[2]);
t[20] = (t[8] - xr);
t[ 8] = (t[8] + xr);
xr = MUL(t[12],xp[3]);
t[12] = (t[16] - xr);
t[16] = (t[16] + xr);
t++;
} while (t != t1);
xp += 4;
for (i = 0; i < 4; i++) {
xr = MUL(tab[30-i*4],xp[0]);
tab[30-i*4] = (tab[i*4] - xr);
tab[ i*4] = (tab[i*4] + xr);
xr = MUL(tab[ 2+i*4],xp[1]);
tab[ 2+i*4] = (tab[28-i*4] - xr);
tab[28-i*4] = (tab[28-i*4] + xr);
xr = MUL(tab[31-i*4],xp[0]);
tab[31-i*4] = (tab[1+i*4] - xr);
tab[ 1+i*4] = (tab[1+i*4] + xr);
xr = MUL(tab[ 3+i*4],xp[1]);
tab[ 3+i*4] = (tab[29-i*4] - xr);
tab[29-i*4] = (tab[29-i*4] + xr);
xp += 2;
}
t = tab + 30;
t1 = tab + 1;
do {
xr = MUL(t1[0], *xp);
t1[0] = (t[0] - xr);
t[0] = (t[0] + xr);
t -= 2;
t1 += 2;
xp++;
} while (t >= tab);
for(i=0;i<32;i++) {
out[i] = tab[bitinv32[i]];
}
}
#define WSHIFT (WFRAC_BITS + 15 - FRAC_BITS)
static void filter(MpegAudioContext *s, int ch, const short *samples, int incr)
{
short *p, *q;
int sum, offset, i, j;
int tmp[64];
int tmp1[32];
int *out;
offset = s->samples_offset[ch];
out = &s->sb_samples[ch][0][0][0];
for(j=0;j<36;j++) {
/* 32 samples at once */
for(i=0;i<32;i++) {
s->samples_buf[ch][offset + (31 - i)] = samples[0];
samples += incr;
}
/* filter */
p = s->samples_buf[ch] + offset;
q = s->filter_bank;
/* maxsum = 23169 */
for(i=0;i<64;i++) {
sum = p[0*64] * q[0*64];
sum += p[1*64] * q[1*64];
sum += p[2*64] * q[2*64];
sum += p[3*64] * q[3*64];
sum += p[4*64] * q[4*64];
sum += p[5*64] * q[5*64];
sum += p[6*64] * q[6*64];
sum += p[7*64] * q[7*64];
tmp[i] = sum;
p++;
q++;
}
tmp1[0] = tmp[16] >> WSHIFT;
for( i=1; i<=16; i++ ) tmp1[i] = (tmp[i+16]+tmp[16-i]) >> WSHIFT;
for( i=17; i<=31; i++ ) tmp1[i] = (tmp[i+16]-tmp[80-i]) >> WSHIFT;
idct32(out, tmp1);
/* advance of 32 samples */
offset -= 32;
out += 32;
/* handle the wrap around */
if (offset < 0) {
memmove(s->samples_buf[ch] + SAMPLES_BUF_SIZE - (512 - 32),
s->samples_buf[ch], (512 - 32) * 2);
offset = SAMPLES_BUF_SIZE - 512;
}
}
s->samples_offset[ch] = offset;
}
static void compute_scale_factors(MpegAudioContext *s,
unsigned char scale_code[SBLIMIT],
unsigned char scale_factors[SBLIMIT][3],
int sb_samples[3][12][SBLIMIT],
int sblimit)
{
int *p, vmax, v, n, i, j, k, code;
int index, d1, d2;
unsigned char *sf = &scale_factors[0][0];
for(j=0;j<sblimit;j++) {
for(i=0;i<3;i++) {
/* find the max absolute value */
p = &sb_samples[i][0][j];
vmax = abs(*p);
for(k=1;k<12;k++) {
p += SBLIMIT;
v = abs(*p);
if (v > vmax)
vmax = v;
}
/* compute the scale factor index using log 2 computations */
if (vmax > 1) {
n = av_log2(vmax);
/* n is the position of the MSB of vmax. now
use at most 2 compares to find the index */
index = (21 - n) * 3 - 3;
if (index >= 0) {
while (vmax <= s->scale_factor_table[index+1])
index++;
} else {
index = 0; /* very unlikely case of overflow */
}
} else {
index = 62; /* value 63 is not allowed */
}
ff_dlog(NULL, "%2d:%d in=%x %x %d\n",
j, i, vmax, s->scale_factor_table[index], index);
/* store the scale factor */
av_assert2(index >=0 && index <= 63);
sf[i] = index;
}
/* compute the transmission factor : look if the scale factors
are close enough to each other */
d1 = s->scale_diff_table[sf[0] - sf[1] + 64];
d2 = s->scale_diff_table[sf[1] - sf[2] + 64];
/* handle the 25 cases */
switch(d1 * 5 + d2) {
case 0*5+0:
case 0*5+4:
case 3*5+4:
case 4*5+0:
case 4*5+4:
code = 0;
break;
case 0*5+1:
case 0*5+2:
case 4*5+1:
case 4*5+2:
code = 3;
sf[2] = sf[1];
break;
case 0*5+3:
case 4*5+3:
code = 3;
sf[1] = sf[2];
break;
case 1*5+0:
case 1*5+4:
case 2*5+4:
code = 1;
sf[1] = sf[0];
break;
case 1*5+1:
case 1*5+2:
case 2*5+0:
case 2*5+1:
case 2*5+2:
code = 2;
sf[1] = sf[2] = sf[0];
break;
case 2*5+3:
case 3*5+3:
code = 2;
sf[0] = sf[1] = sf[2];
break;
case 3*5+0:
case 3*5+1:
case 3*5+2:
code = 2;
sf[0] = sf[2] = sf[1];
break;
case 1*5+3:
code = 2;
if (sf[0] > sf[2])
sf[0] = sf[2];
sf[1] = sf[2] = sf[0];
break;
default:
av_assert2(0); //cannot happen
code = 0; /* kill warning */
}
ff_dlog(NULL, "%d: %2d %2d %2d %d %d -> %d\n", j,
sf[0], sf[1], sf[2], d1, d2, code);
scale_code[j] = code;
sf += 3;
}
}
/* The most important function : psycho acoustic module. In this
encoder there is basically none, so this is the worst you can do,
but also this is the simpler. */
static void psycho_acoustic_model(MpegAudioContext *s, short smr[SBLIMIT])
{
int i;
for(i=0;i<s->sblimit;i++) {
smr[i] = (int)(fixed_smr[i] * 10);
}
}
#define SB_NOTALLOCATED 0
#define SB_ALLOCATED 1
#define SB_NOMORE 2
/* Try to maximize the smr while using a number of bits inferior to
the frame size. I tried to make the code simpler, faster and
smaller than other encoders :-) */
static void compute_bit_allocation(MpegAudioContext *s,
short smr1[MPA_MAX_CHANNELS][SBLIMIT],
unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT],
int *padding)
{
int i, ch, b, max_smr, max_ch, max_sb, current_frame_size, max_frame_size;
int incr;
short smr[MPA_MAX_CHANNELS][SBLIMIT];
unsigned char subband_status[MPA_MAX_CHANNELS][SBLIMIT];
const unsigned char *alloc;
memcpy(smr, smr1, s->nb_channels * sizeof(short) * SBLIMIT);
memset(subband_status, SB_NOTALLOCATED, s->nb_channels * SBLIMIT);
memset(bit_alloc, 0, s->nb_channels * SBLIMIT);
/* compute frame size and padding */
max_frame_size = s->frame_size;
s->frame_frac += s->frame_frac_incr;
if (s->frame_frac >= 65536) {
s->frame_frac -= 65536;
s->do_padding = 1;
max_frame_size += 8;
} else {
s->do_padding = 0;
}
/* compute the header + bit alloc size */
current_frame_size = 32;
alloc = s->alloc_table;
for(i=0;i<s->sblimit;i++) {
incr = alloc[0];
current_frame_size += incr * s->nb_channels;
alloc += 1 << incr;
}
for(;;) {
/* look for the subband with the largest signal to mask ratio */
max_sb = -1;
max_ch = -1;
max_smr = INT_MIN;
for(ch=0;ch<s->nb_channels;ch++) {
for(i=0;i<s->sblimit;i++) {
if (smr[ch][i] > max_smr && subband_status[ch][i] != SB_NOMORE) {
max_smr = smr[ch][i];
max_sb = i;
max_ch = ch;
}
}
}
if (max_sb < 0)
break;
ff_dlog(NULL, "current=%d max=%d max_sb=%d max_ch=%d alloc=%d\n",
current_frame_size, max_frame_size, max_sb, max_ch,
bit_alloc[max_ch][max_sb]);
/* find alloc table entry (XXX: not optimal, should use
pointer table) */
alloc = s->alloc_table;
for(i=0;i<max_sb;i++) {
alloc += 1 << alloc[0];
}
if (subband_status[max_ch][max_sb] == SB_NOTALLOCATED) {
/* nothing was coded for this band: add the necessary bits */
incr = 2 + nb_scale_factors[s->scale_code[max_ch][max_sb]] * 6;
incr += s->total_quant_bits[alloc[1]];
} else {
/* increments bit allocation */
b = bit_alloc[max_ch][max_sb];
incr = s->total_quant_bits[alloc[b + 1]] -
s->total_quant_bits[alloc[b]];
}
if (current_frame_size + incr <= max_frame_size) {
/* can increase size */
b = ++bit_alloc[max_ch][max_sb];
current_frame_size += incr;
/* decrease smr by the resolution we added */
smr[max_ch][max_sb] = smr1[max_ch][max_sb] - quant_snr[alloc[b]];
/* max allocation size reached ? */
if (b == ((1 << alloc[0]) - 1))
subband_status[max_ch][max_sb] = SB_NOMORE;
else
subband_status[max_ch][max_sb] = SB_ALLOCATED;
} else {
/* cannot increase the size of this subband */
subband_status[max_ch][max_sb] = SB_NOMORE;
}
}
*padding = max_frame_size - current_frame_size;
av_assert0(*padding >= 0);
}
/*
* Output the MPEG audio layer 2 frame. Note how the code is small
* compared to other encoders :-)
*/
static void encode_frame(MpegAudioContext *s,
unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT],
int padding)
{
int i, j, k, l, bit_alloc_bits, b, ch;
unsigned char *sf;
int q[3];
PutBitContext *p = &s->pb;
/* header */
put_bits(p, 12, 0xfff);
put_bits(p, 1, 1 - s->lsf); /* 1 = MPEG-1 ID, 0 = MPEG-2 lsf ID */
put_bits(p, 2, 4-2); /* layer 2 */
put_bits(p, 1, 1); /* no error protection */
put_bits(p, 4, s->bitrate_index);
put_bits(p, 2, s->freq_index);
put_bits(p, 1, s->do_padding); /* use padding */
put_bits(p, 1, 0); /* private_bit */
put_bits(p, 2, s->nb_channels == 2 ? MPA_STEREO : MPA_MONO);
put_bits(p, 2, 0); /* mode_ext */
put_bits(p, 1, 0); /* no copyright */
put_bits(p, 1, 1); /* original */
put_bits(p, 2, 0); /* no emphasis */
/* bit allocation */
j = 0;
for(i=0;i<s->sblimit;i++) {
bit_alloc_bits = s->alloc_table[j];
for(ch=0;ch<s->nb_channels;ch++) {
put_bits(p, bit_alloc_bits, bit_alloc[ch][i]);
}
j += 1 << bit_alloc_bits;
}
/* scale codes */
for(i=0;i<s->sblimit;i++) {
for(ch=0;ch<s->nb_channels;ch++) {
if (bit_alloc[ch][i])
put_bits(p, 2, s->scale_code[ch][i]);
}
}
/* scale factors */
for(i=0;i<s->sblimit;i++) {
for(ch=0;ch<s->nb_channels;ch++) {
if (bit_alloc[ch][i]) {
sf = &s->scale_factors[ch][i][0];
switch(s->scale_code[ch][i]) {
case 0:
put_bits(p, 6, sf[0]);
put_bits(p, 6, sf[1]);
put_bits(p, 6, sf[2]);
break;
case 3:
case 1:
put_bits(p, 6, sf[0]);
put_bits(p, 6, sf[2]);
break;
case 2:
put_bits(p, 6, sf[0]);
break;
}
}
}
}
/* quantization & write sub band samples */
for(k=0;k<3;k++) {
for(l=0;l<12;l+=3) {
j = 0;
for(i=0;i<s->sblimit;i++) {
bit_alloc_bits = s->alloc_table[j];
for(ch=0;ch<s->nb_channels;ch++) {
b = bit_alloc[ch][i];
if (b) {
int qindex, steps, m, sample, bits;
/* we encode 3 sub band samples of the same sub band at a time */
qindex = s->alloc_table[j+b];
steps = ff_mpa_quant_steps[qindex];
for(m=0;m<3;m++) {
sample = s->sb_samples[ch][k][l + m][i];
/* divide by scale factor */
#if USE_FLOATS
{
float a;
a = (float)sample * s->scale_factor_inv_table[s->scale_factors[ch][i][k]];
q[m] = (int)((a + 1.0) * steps * 0.5);
}
#else
{
int q1, e, shift, mult;
e = s->scale_factors[ch][i][k];
shift = s->scale_factor_shift[e];
mult = s->scale_factor_mult[e];
/* normalize to P bits */
if (shift < 0)
q1 = sample << (-shift);
else
q1 = sample >> shift;
q1 = (q1 * mult) >> P;
q1 += 1 << P;
if (q1 < 0)
q1 = 0;
q[m] = (q1 * (unsigned)steps) >> (P + 1);
}
#endif
if (q[m] >= steps)
q[m] = steps - 1;
av_assert2(q[m] >= 0 && q[m] < steps);
}
bits = ff_mpa_quant_bits[qindex];
if (bits < 0) {
/* group the 3 values to save bits */
put_bits(p, -bits,
q[0] + steps * (q[1] + steps * q[2]));
} else {
put_bits(p, bits, q[0]);
put_bits(p, bits, q[1]);
put_bits(p, bits, q[2]);
}
}
}
/* next subband in alloc table */
j += 1 << bit_alloc_bits;
}
}
}
/* padding */
for(i=0;i<padding;i++)
put_bits(p, 1, 0);
/* flush */
flush_put_bits(p);
}
static int MPA_encode_frame(AVCodecContext *avctx, AVPacket *avpkt,
const AVFrame *frame, int *got_packet_ptr)
{
MpegAudioContext *s = avctx->priv_data;
const int16_t *samples = (const int16_t *)frame->data[0];
short smr[MPA_MAX_CHANNELS][SBLIMIT];
unsigned char bit_alloc[MPA_MAX_CHANNELS][SBLIMIT];
int padding, i, ret;
for(i=0;i<s->nb_channels;i++) {
filter(s, i, samples + i, s->nb_channels);
}
for(i=0;i<s->nb_channels;i++) {
compute_scale_factors(s, s->scale_code[i], s->scale_factors[i],
s->sb_samples[i], s->sblimit);
}
for(i=0;i<s->nb_channels;i++) {
psycho_acoustic_model(s, smr[i]);
}
compute_bit_allocation(s, smr, bit_alloc, &padding);
if ((ret = ff_alloc_packet2(avctx, avpkt, MPA_MAX_CODED_FRAME_SIZE, 0)) < 0)
return ret;
init_put_bits(&s->pb, avpkt->data, avpkt->size);
encode_frame(s, bit_alloc, padding);
if (frame->pts != AV_NOPTS_VALUE)
avpkt->pts = frame->pts - ff_samples_to_time_base(avctx, avctx->initial_padding);
avpkt->size = put_bits_count(&s->pb) / 8;
*got_packet_ptr = 1;
return 0;
}
static const AVCodecDefault mp2_defaults[] = {
{ "b", "0" },
{ NULL },
};

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/*
* mpeg audio layer 2 tables. Most of them come from the mpeg audio
* specification.
*
* Copyright (c) 2000, 2001 Fabrice Bellard
*
* This file is part of FFmpeg.
*
* FFmpeg 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.
*
* FFmpeg 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 FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
* @file
* mpeg audio layer 2 tables.
* Most of them come from the mpeg audio specification.
*/
#ifndef AVCODEC_MPEGAUDIOTAB_H
#define AVCODEC_MPEGAUDIOTAB_H
#include <stdint.h>
#include "mpegaudio.h"
static const int costab32[30] = {
FIX(0.54119610014619701222),
FIX(1.3065629648763763537),
FIX(0.50979557910415917998),
FIX(2.5629154477415054814),
FIX(0.89997622313641556513),
FIX(0.60134488693504528634),
FIX(0.5024192861881556782),
FIX(5.1011486186891552563),
FIX(0.78815462345125020249),
FIX(0.64682178335999007679),
FIX(0.56694403481635768927),
FIX(1.0606776859903470633),
FIX(1.7224470982383341955),
FIX(0.52249861493968885462),
FIX(10.19000812354803287),
FIX(0.674808341455005678),
FIX(1.1694399334328846596),
FIX(0.53104259108978413284),
FIX(2.0577810099534108446),
FIX(0.58293496820613388554),
FIX(0.83934964541552681272),
FIX(0.50547095989754364798),
FIX(3.4076084184687189804),
FIX(0.62250412303566482475),
FIX(0.97256823786196078263),
FIX(0.51544730992262455249),
FIX(1.4841646163141661852),
FIX(0.5531038960344445421),
FIX(0.74453627100229857749),
FIX(0.5006029982351962726),
};
static const int bitinv32[32] = {
0, 16, 8, 24, 4, 20, 12, 28,
2, 18, 10, 26, 6, 22, 14, 30,
1, 17, 9, 25, 5, 21, 13, 29,
3, 19, 11, 27, 7, 23, 15, 31
};
/* signal to noise ratio of each quantification step (could be
computed from quant_steps[]). The values are dB multiplied by 10
*/
static const unsigned short quant_snr[17] = {
70, 110, 160, 208,
253, 316, 378, 439,
499, 559, 620, 680,
740, 800, 861, 920,
980
};
/* fixed psycho acoustic model. Values of SNR taken from the 'toolame'
project */
static const float fixed_smr[SBLIMIT] = {
30, 17, 16, 10, 3, 12, 8, 2.5,
5, 5, 6, 6, 5, 6, 10, 6,
-4, -10, -21, -30, -42, -55, -68, -75,
-75, -75, -75, -75, -91, -107, -110, -108
};
static const unsigned char nb_scale_factors[4] = { 3, 2, 1, 2 };
#endif /* AVCODEC_MPEGAUDIOTAB_H */

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/*
* (I)RDFT transforms
* Copyright (c) 2009 Alex Converse <alex dot converse at gmail dot com>
*
* This file is part of FFmpeg.
*
* FFmpeg 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.
*
* FFmpeg 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 FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include <stdlib.h>
#include <math.h>
#include "libavutil/mathematics.h"
#include "rdft.h"
/**
* @file
* (Inverse) Real Discrete Fourier Transforms.
*/
/** Map one real FFT into two parallel real even and odd FFTs. Then interleave
* the two real FFTs into one complex FFT. Unmangle the results.
* ref: http://www.engineeringproductivitytools.com/stuff/T0001/PT10.HTM
*/
static void rdft_calc_c(RDFTContext *s, FFTSample *data)
{
int i, i1, i2;
FFTComplex ev, od, odsum;
const int n = 1 << s->nbits;
const float k1 = 0.5;
const float k2 = 0.5 - s->inverse;
const FFTSample *tcos = s->tcos;
const FFTSample *tsin = s->tsin;
if (!s->inverse) {
s->fft.fft_permute(&s->fft, (FFTComplex*)data);
s->fft.fft_calc(&s->fft, (FFTComplex*)data);
}
/* i=0 is a special case because of packing, the DC term is real, so we
are going to throw the N/2 term (also real) in with it. */
ev.re = data[0];
data[0] = ev.re+data[1];
data[1] = ev.re-data[1];
#define RDFT_UNMANGLE(sign0, sign1) \
for (i = 1; i < (n>>2); i++) { \
i1 = 2*i; \
i2 = n-i1; \
/* Separate even and odd FFTs */ \
ev.re = k1*(data[i1 ]+data[i2 ]); \
od.im = k2*(data[i2 ]-data[i1 ]); \
ev.im = k1*(data[i1+1]-data[i2+1]); \
od.re = k2*(data[i1+1]+data[i2+1]); \
/* Apply twiddle factors to the odd FFT and add to the even FFT */ \
odsum.re = od.re*tcos[i] sign0 od.im*tsin[i]; \
odsum.im = od.im*tcos[i] sign1 od.re*tsin[i]; \
data[i1 ] = ev.re + odsum.re; \
data[i1+1] = ev.im + odsum.im; \
data[i2 ] = ev.re - odsum.re; \
data[i2+1] = odsum.im - ev.im; \
}
if (s->negative_sin) {
RDFT_UNMANGLE(+,-)
} else {
RDFT_UNMANGLE(-,+)
}
data[2*i+1]=s->sign_convention*data[2*i+1];
if (s->inverse) {
data[0] *= k1;
data[1] *= k1;
s->fft.fft_permute(&s->fft, (FFTComplex*)data);
s->fft.fft_calc(&s->fft, (FFTComplex*)data);
}
}
av_cold int ff_rdft_init(RDFTContext *s, int nbits, enum RDFTransformType trans)
{
int n = 1 << nbits;
int ret;
s->nbits = nbits;
s->inverse = trans == IDFT_C2R || trans == DFT_C2R;
s->sign_convention = trans == IDFT_R2C || trans == DFT_C2R ? 1 : -1;
s->negative_sin = trans == DFT_C2R || trans == DFT_R2C;
if (nbits < 4 || nbits > 16)
return AVERROR(EINVAL);
if ((ret = ff_fft_init(&s->fft, nbits-1, trans == IDFT_C2R || trans == IDFT_R2C)) < 0)
return ret;
ff_init_ff_cos_tabs(nbits);
s->tcos = ff_cos_tabs[nbits];
s->tsin = ff_cos_tabs[nbits] + (n >> 2);
s->rdft_calc = rdft_calc_c;
if (ARCH_ARM) ff_rdft_init_arm(s);
return 0;
}
av_cold void ff_rdft_end(RDFTContext *s)
{
ff_fft_end(&s->fft);
}