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ffmpeg.wasm-core
Commit 2456e28d authored by Fabrice Bellard's avatar Fabrice Bellard
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merged code and tables between encoder and decoder 


Originally committed as revision 119 to svn://svn.ffmpeg.org/ffmpeg/trunk
parent 9a9b2b5d
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Showing with 69 additions and 278 deletions
libavcodec/mpegaudio.c
View file @ 2456e28d
......@@ -20,14 +20,36 @@
#include <math.h>
#include "mpegaudio.h"
#define DCT_BITS 14 /* number of bits for the DCT */
#define MUL(a,b) (((a) * (b)) >> DCT_BITS)
#define FIX(a) ((int)((a) * (1 << DCT_BITS)))
#define SAMPLES_BUF_SIZE 4096
typedef struct MpegAudioContext {
PutBitContext pb;
int nb_channels;
int freq, bit_rate;
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 */
INT64 nb_samples; /* total number of samples encoded */
/* 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;
} MpegAudioContext;
/* define it to use floats in quantization (I don't like floats !) */
//#define USE_FLOATS
#define MPA_STEREO 0
#define MPA_JSTEREO 1
#define MPA_DUAL 2
#define MPA_MONO 3
#include "mpegaudiotab.h"
int MPA_encode_init(AVCodecContext *avctx)
......@@ -36,7 +58,7 @@ int MPA_encode_init(AVCodecContext *avctx)
int freq = avctx->sample_rate;
int bitrate = avctx->bit_rate;
int channels = avctx->channels;
int i, v, table, ch_bitrate;
int i, v, table;
float a;
if (channels > 2)
......@@ -51,9 +73,9 @@ int MPA_encode_init(AVCodecContext *avctx)
/* encoding freq */
s->lsf = 0;
for(i=0;i<3;i++) {
if (freq_tab[i] == freq)
if (mpa_freq_tab[i] == freq)
break;
if ((freq_tab[i] / 2) == freq) {
if ((mpa_freq_tab[i] / 2) == freq) {
s->lsf = 1;
break;
}
......@@ -64,7 +86,7 @@ int MPA_encode_init(AVCodecContext *avctx)
/* encoding bitrate & frequency */
for(i=0;i<15;i++) {
if (bitrate_tab[1-s->lsf][i] == bitrate)
if (mpa_bitrate_tab[s->lsf][1][i] == bitrate)
break;
}
if (i == 15)
......@@ -81,20 +103,8 @@ int MPA_encode_init(AVCodecContext *avctx)
s->frame_frac_incr = (int)((a - floor(a)) * 65536.0);
/* select the right allocation table */
ch_bitrate = bitrate / s->nb_channels;
if (!s->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;
}
table = l2_select_table(bitrate, s->nb_channels, freq, s->lsf);
/* number of used subbands */
s->sblimit = sblimit_table[table];
s->alloc_table = alloc_tables[table];
......@@ -107,10 +117,16 @@ int MPA_encode_init(AVCodecContext *avctx)
for(i=0;i<s->nb_channels;i++)
s->samples_offset[i] = 0;
for(i=0;i<512;i++) {
float a = enwindow[i] * 32768.0 * 16.0;
filter_bank[i] = (int)(a);
for(i=0;i<257;i++) {
int v;
v = (mpa_enwindow[i] + 2) >> 2;
filter_bank[i] = v;
if ((i & 63) != 0)
v = -v;
if (i != 0)
filter_bank[512 - i] = v;
}
for(i=0;i<64;i++) {
v = (int)(pow(2.0, (3 - i) / 3.0) * (1 << 20));
if (v <= 0)
......@@ -151,7 +167,7 @@ int MPA_encode_init(AVCodecContext *avctx)
return 0;
}
/* 32 point floating point IDCT */
/* 32 point floating point IDCT without 1/sqrt(2) coef zero scaling */
static void idct32(int *out, int *tab, int sblimit, int left_shift)
{
int i, j;
......
libavcodec/mpegaudio.h
View file @ 2456e28d
/* mpeg audio declarations for both encoder and decoder */
/* max frame size, in samples */
#define MPA_FRAME_SIZE 1152
/* max compressed frame size */
#define MPA_MAX_CODED_FRAME_SIZE 1200
#define MPA_MAX_CODED_FRAME_SIZE 1792
#define MPA_FRAME_SIZE 1152
#define MPA_MAX_CHANNELS 2
#define SAMPLES_BUF_SIZE 4096
#define SBLIMIT 32 /* number of subbands */
#define DCT_BITS 14 /* number of bits for the DCT */
#define MUL(a,b) (((a) * (b)) >> DCT_BITS)
#define FIX(a) ((int)((a) * (1 << DCT_BITS)))
typedef struct MpegAudioContext {
PutBitContext pb;
int nb_channels;
int freq, bit_rate;
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 */
INT64 nb_samples; /* total number of samples encoded */
/* 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;
} MpegAudioContext;
#define MPA_STEREO 0
#define MPA_JSTEREO 1
#define MPA_DUAL 2
#define MPA_MONO 3
int l2_select_table(int bitrate, int nb_channels, int freq, int lsf);
const UINT16 mpa_bitrate_tab[2][3][15];
const UINT16 mpa_freq_tab[3];
extern const unsigned char *alloc_tables[5];
extern const double enwindow[512];
extern const int sblimit_table[5];
extern const int quant_steps[17];
extern const int quant_bits[17];
extern const INT32 mpa_enwindow[257];
libavcodec/mpegaudiotab.h
View file @ 2456e28d
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