* MPEG Audio decoder
* Copyright (c) 2001, 2002 Fabrice Bellard.
*
- * This library is free software; you can redistribute it and/or
+ * 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 of the License, or (at your option) any later version.
+ * version 2.1 of the License, or (at your option) any later version.
*
- * This library is distributed in the hope that it will be useful,
+ * 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 this library; if not, write to the Free Software
+ * License along with FFmpeg; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#endif
#include "mpegaudio.h"
+#include "mpegaudiodecheader.h"
+
+#include "mathops.h"
-#define FRAC_ONE (1 << FRAC_BITS)
-
-#ifdef ARCH_X86
-# define MULL(ra, rb) \
- ({ int rt, dummy; asm (\
- "imull %3 \n\t"\
- "shrdl %4, %%edx, %%eax \n\t"\
- : "=a"(rt), "=d"(dummy)\
- : "a" (ra), "rm" (rb), "i"(FRAC_BITS));\
- rt; })
-# define MUL64(ra, rb) \
- ({ int64_t rt; asm ("imull %2\n\t" : "=A"(rt) : "a" (ra), "g" (rb)); rt; })
-# define MULH(ra, rb) \
- ({ int rt, dummy; asm ("imull %3\n\t" : "=d"(rt), "=a"(dummy): "a" (ra), "rm" (rb)); rt; })
-#elif defined(ARCH_ARMV4L)
-# define MULL(a, b) \
- ({ int lo, hi;\
- asm("smull %0, %1, %2, %3 \n\t"\
- "mov %0, %0, lsr %4\n\t"\
- "add %1, %0, %1, lsl %5\n\t"\
- : "=&r"(lo), "=&r"(hi)\
- : "r"(b), "r"(a), "i"(FRAC_BITS), "i"(32-FRAC_BITS));\
- hi; })
-# define MUL64(a,b) ((int64_t)(a) * (int64_t)(b))
-# define MULH(a, b) ({ int lo, hi; asm ("smull %0, %1, %2, %3" : "=&r"(lo), "=&r"(hi) : "r"(b), "r"(a)); hi; })
-#else
-# define MULL(a,b) (((int64_t)(a) * (int64_t)(b)) >> FRAC_BITS)
-# define MUL64(a,b) ((int64_t)(a) * (int64_t)(b))
-//#define MULH(a,b) (((int64_t)(a) * (int64_t)(b))>>32) //gcc 3.4 creates an incredibly bloated mess out of this
-static always_inline int MULH(int a, int b){
- return ((int64_t)(a) * (int64_t)(b))>>32;
-}
-#endif
-#define FIX(a) ((int)((a) * FRAC_ONE))
/* WARNING: only correct for posititive numbers */
#define FIXR(a) ((int)((a) * FRAC_ONE + 0.5))
#define FRAC_RND(a) (((a) + (FRAC_ONE/2)) >> FRAC_BITS)
/****************/
#define HEADER_SIZE 4
-#define BACKSTEP_SIZE 512
-#define EXTRABYTES 24
-
-struct GranuleDef;
-
-typedef struct MPADecodeContext {
- DECLARE_ALIGNED_8(uint8_t, last_buf[2*BACKSTEP_SIZE + EXTRABYTES]);
- int last_buf_size;
- int frame_size;
- /* next header (used in free format parsing) */
- uint32_t free_format_next_header;
- int error_protection;
- int layer;
- int sample_rate;
- int sample_rate_index; /* between 0 and 8 */
- int bit_rate;
- GetBitContext gb;
- GetBitContext in_gb;
- int nb_channels;
- int mode;
- int mode_ext;
- int lsf;
- MPA_INT synth_buf[MPA_MAX_CHANNELS][512 * 2] __attribute__((aligned(16)));
- int synth_buf_offset[MPA_MAX_CHANNELS];
- int32_t sb_samples[MPA_MAX_CHANNELS][36][SBLIMIT] __attribute__((aligned(16)));
- int32_t mdct_buf[MPA_MAX_CHANNELS][SBLIMIT * 18]; /* previous samples, for layer 3 MDCT */
-#ifdef DEBUG
- int frame_count;
-#endif
- void (*compute_antialias)(struct MPADecodeContext *s, struct GranuleDef *g);
- int adu_mode; ///< 0 for standard mp3, 1 for adu formatted mp3
- unsigned int dither_state;
-} MPADecodeContext;
/**
* Context for MP3On4 decoder
int32_t sb_hybrid[SBLIMIT * 18]; /* 576 samples */
} GranuleDef;
-#define MODE_EXT_MS_STEREO 2
-#define MODE_EXT_I_STEREO 1
-
-/* layer 3 huffman tables */
-typedef struct HuffTable {
- int xsize;
- const uint8_t *bits;
- const uint16_t *codes;
-} HuffTable;
-
+#include "mpegaudiodata.h"
#include "mpegaudiodectab.h"
static void compute_antialias_integer(MPADecodeContext *s, GranuleDef *g);
static uint16_t band_index_long[9][23];
/* XXX: free when all decoders are closed */
#define TABLE_4_3_SIZE (8191 + 16)*4
-static int8_t *table_4_3_exp;
-static uint32_t *table_4_3_value;
+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[512];
static uint32_t expval_table[512][16];
/* intensity stereo coef table */
SCALE_GEN(4.0 / 9.0), /* 9 steps */
};
-static MPA_INT window[512] __attribute__((aligned(16)));
+static DECLARE_ALIGNED_16(MPA_INT, window[512]);
+
+/**
+ * Convert region offsets to region sizes and truncate
+ * size to big_values.
+ */
+void ff_region_offset2size(GranuleDef *g){
+ int i, k, j=0;
+ g->region_size[2] = (576 / 2);
+ for(i=0;i<3;i++) {
+ k = FFMIN(g->region_size[i], g->big_values);
+ g->region_size[i] = k - j;
+ j = k;
+ }
+}
+
+void ff_init_short_region(MPADecodeContext *s, GranuleDef *g){
+ if (g->block_type == 2)
+ g->region_size[0] = (36 / 2);
+ else {
+ if (s->sample_rate_index <= 2)
+ g->region_size[0] = (36 / 2);
+ else if (s->sample_rate_index != 8)
+ g->region_size[0] = (54 / 2);
+ else
+ g->region_size[0] = (108 / 2);
+ }
+ g->region_size[1] = (576 / 2);
+}
+
+void ff_init_long_region(MPADecodeContext *s, GranuleDef *g, int ra1, int ra2){
+ int l;
+ g->region_size[0] =
+ band_index_long[s->sample_rate_index][ra1 + 1] >> 1;
+ /* should not overflow */
+ l = FFMIN(ra1 + ra2 + 2, 22);
+ g->region_size[1] =
+ band_index_long[s->sample_rate_index][l] >> 1;
+}
+
+void ff_compute_band_indexes(MPADecodeContext *s, GranuleDef *g){
+ if (g->block_type == 2) {
+ if (g->switch_point) {
+ /* if switched mode, we handle the 36 first samples as
+ long blocks. For 8000Hz, we handle the 48 first
+ exponents as long blocks (XXX: check this!) */
+ if (s->sample_rate_index <= 2)
+ g->long_end = 8;
+ else if (s->sample_rate_index != 8)
+ g->long_end = 6;
+ else
+ g->long_end = 4; /* 8000 Hz */
+
+ g->short_start = 2 + (s->sample_rate_index != 8);
+ } else {
+ g->long_end = 0;
+ g->short_start = 0;
+ }
+ } else {
+ g->short_start = 13;
+ g->long_end = 22;
+ }
+}
/* layer 1 unscaling */
/* n = number of bits of the mantissa minus 1 */
static int init=0;
int i, j, k;
+ s->avctx = avctx;
+
#if defined(USE_HIGHPRECISION) && defined(CONFIG_AUDIO_NONSHORT)
avctx->sample_fmt= SAMPLE_FMT_S32;
#else
avctx->sample_fmt= SAMPLE_FMT_S16;
#endif
+ s->error_resilience= avctx->error_resilience;
if(avctx->antialias_algo != FF_AA_FLOAT)
s->compute_antialias= compute_antialias_integer;
for(i=0;i<15;i++) {
int n, norm;
n = i + 2;
- norm = ((int64_t_C(1) << n) * FRAC_ONE) / ((1 << n) - 1);
+ norm = ((INT64_C(1) << n) * FRAC_ONE) / ((1 << n) - 1);
scale_factor_mult[i][0] = MULL(FIXR(1.0 * 2.0), norm);
scale_factor_mult[i][1] = MULL(FIXR(0.7937005259 * 2.0), norm);
scale_factor_mult[i][2] = MULL(FIXR(0.6299605249 * 2.0), norm);
- dprintf("%d: norm=%x s=%x %x %x\n",
+ dprintf(avctx, "%d: norm=%x s=%x %x %x\n",
i, norm,
scale_factor_mult[i][0],
scale_factor_mult[i][1],
}
/* compute n ^ (4/3) and store it in mantissa/exp format */
- table_4_3_exp= av_mallocz_static(TABLE_4_3_SIZE * sizeof(table_4_3_exp[0]));
- if(!table_4_3_exp)
- return -1;
- table_4_3_value= av_mallocz_static(TABLE_4_3_SIZE * sizeof(table_4_3_value[0]));
- if(!table_4_3_value)
- return -1;
int_pow_init();
for(i=1;i<TABLE_4_3_SIZE;i++) {
k = i & 1;
is_table_lsf[j][k ^ 1][i] = FIXR(f);
is_table_lsf[j][k][i] = FIXR(1.0);
- dprintf("is_table_lsf %d %d: %x %x\n",
+ dprintf(avctx, "is_table_lsf %d %d: %x %x\n",
i, j, is_table_lsf[j][0][i], is_table_lsf[j][1][i]);
}
}
return sum1;
}
-# if defined(ARCH_POWERPC_405)
- /* signed 16x16 -> 32 multiply add accumulate */
-# define MACS(rt, ra, rb) \
- asm ("maclhw %0, %2, %3" : "=r" (rt) : "0" (rt), "r" (ra), "r" (rb));
-
- /* signed 16x16 -> 32 multiply */
-# define MULS(ra, rb) \
- ({ int __rt; asm ("mullhw %0, %1, %2" : "=r" (__rt) : "r" (ra), "r" (rb)); __rt; })
-
-# elif defined(HAVE_ARMV5TE)
-
- /* signed 16x16 -> 32 multiply add accumulate */
-# define MACS(rt, ra, rb) \
- asm ("smlabb %0, %2, %3, %0" : "=r" (rt) : "0" (rt), "r" (ra), "r" (rb));
+/* signed 16x16 -> 32 multiply add accumulate */
+#define MACS(rt, ra, rb) MAC16(rt, ra, rb)
- /* signed 16x16 -> 32 multiply */
-# define MULS(ra, rb) \
- ({ int __rt; asm ("smulbb %0, %1, %2" : "=r" (__rt) : "r" (ra), "r" (rb)); __rt; })
+/* signed 16x16 -> 32 multiply */
+#define MULS(ra, rb) MUL16(ra, rb)
-# else
- /* signed 16x16 -> 32 multiply add accumulate */
-# define MACS(rt, ra, rb) rt += (ra) * (rb)
-
- /* signed 16x16 -> 32 multiply */
-# define MULS(ra, rb) ((ra) * (rb))
-# endif
#else
static inline int round_sample(int64_t *sum)
/* max = 18760, max sum over all 16 coefs : 44736 */
for(i=0;i<257;i++) {
int v;
- v = mpa_enwindow[i];
+ v = ff_mpa_enwindow[i];
#if WFRAC_BITS < 16
v = (v + (1 << (16 - WFRAC_BITS - 1))) >> (16 - WFRAC_BITS);
#endif
#if FRAC_BITS <= 15
/* NOTE: can cause a loss in precision if very high amplitude
sound */
- if (v > 32767)
- v = 32767;
- else if (v < -32768)
- v = -32768;
+ v = av_clip_int16(v);
#endif
synth_buf[j] = v;
}
buf[8 - 4] = MULH(t0, win[18 + 8 - 4]);
}
-/* 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 */
-static int decode_header(MPADecodeContext *s, uint32_t header)
-{
- int sample_rate, frame_size, mpeg25, padding;
- int sample_rate_index, bitrate_index;
- 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;
- sample_rate = 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 = 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)
- dprintf("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)
- dprintf("ms-");
- if (s->mode_ext & MODE_EXT_I_STEREO)
- dprintf("i-");
- }
- dprintf("stereo");
- } else {
- dprintf("mono");
- }
- dprintf("\n");
-#endif
- return 0;
-}
-
-/* useful helper to get mpeg audio stream infos. Return -1 if error in
- header, otherwise the coded frame size in bytes */
-int mpa_decode_header(AVCodecContext *avctx, uint32_t head)
-{
- MPADecodeContext s1, *s = &s1;
-
- if (ff_mpa_check_header(head) != 0)
- return -1;
-
- if (decode_header(s, head) != 0) {
- return -1;
- }
-
- switch(s->layer) {
- case 1:
- avctx->frame_size = 384;
- break;
- case 2:
- avctx->frame_size = 1152;
- break;
- default:
- case 3:
- if (s->lsf)
- avctx->frame_size = 576;
- else
- avctx->frame_size = 1152;
- break;
- }
-
- avctx->sample_rate = s->sample_rate;
- avctx->channels = s->nb_channels;
- avctx->bit_rate = s->bit_rate;
- avctx->sub_id = s->layer;
- return s->frame_size;
-}
-
/* return the number of decoded frames */
static int mp_decode_layer1(MPADecodeContext *s)
{
return 12;
}
-/* bitrate is in kb/s */
-int 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;
-}
-
static int mp_decode_layer2(MPADecodeContext *s)
{
int sblimit; /* number of used subbands */
int scale, qindex, bits, steps, k, l, m, b;
/* select decoding table */
- table = l2_select_table(s->bit_rate / 1000, s->nb_channels,
+ table = ff_mpa_l2_select_table(s->bit_rate / 1000, s->nb_channels,
s->sample_rate, s->lsf);
- sblimit = sblimit_table[table];
- alloc_table = alloc_tables[table];
+ sblimit = ff_mpa_sblimit_table[table];
+ alloc_table = ff_mpa_alloc_tables[table];
if (s->mode == MPA_JSTEREO)
bound = (s->mode_ext + 1) * 4;
else
bound = sblimit;
- dprintf("bound=%d sblimit=%d\n", bound, sblimit);
+ dprintf(s->avctx, "bound=%d sblimit=%d\n", bound, sblimit);
/* sanity check */
if( bound > sblimit ) bound = sblimit;
{
for(ch=0;ch<s->nb_channels;ch++) {
for(i=0;i<sblimit;i++)
- dprintf(" %d", bit_alloc[ch][i]);
- dprintf("\n");
+ dprintf(s->avctx, " %d", bit_alloc[ch][i]);
+ dprintf(s->avctx, "\n");
}
}
#endif
for(i=0;i<sblimit;i++) {
if (bit_alloc[ch][i]) {
sf = scale_factors[ch][i];
- dprintf(" %d %d %d", sf[0], sf[1], sf[2]);
+ dprintf(s->avctx, " %d %d %d", sf[0], sf[1], sf[2]);
} else {
- dprintf(" -");
+ dprintf(s->avctx, " -");
}
}
- dprintf("\n");
+ dprintf(s->avctx, "\n");
}
#endif
if (b) {
scale = scale_factors[ch][i][k];
qindex = alloc_table[j+b];
- bits = quant_bits[qindex];
+ bits = ff_mpa_quant_bits[qindex];
if (bits < 0) {
/* 3 values at the same time */
v = get_bits(&s->gb, -bits);
- steps = quant_steps[qindex];
+ steps = ff_mpa_quant_steps[qindex];
s->sb_samples[ch][k * 12 + l + 0][i] =
l2_unscale_group(steps, v % steps, scale);
v = v / steps;
scale0 = scale_factors[0][i][k];
scale1 = scale_factors[1][i][k];
qindex = alloc_table[j+b];
- bits = quant_bits[qindex];
+ bits = ff_mpa_quant_bits[qindex];
if (bits < 0) {
/* 3 values at the same time */
v = get_bits(&s->gb, -bits);
- steps = quant_steps[qindex];
+ steps = ff_mpa_quant_steps[qindex];
mant = v % steps;
v = v / steps;
s->sb_samples[0][k * 12 + l + 0][i] =
return get_bits(s, n);
}
+
+static void switch_buffer(MPADecodeContext *s, int *pos, int *end_pos, int *end_pos2){
+ if(s->in_gb.buffer && *pos >= s->gb.size_in_bits){
+ s->gb= s->in_gb;
+ s->in_gb.buffer=NULL;
+ assert((get_bits_count(&s->gb) & 7) == 0);
+ skip_bits_long(&s->gb, *pos - *end_pos);
+ *end_pos2=
+ *end_pos= *end_pos2 + get_bits_count(&s->gb) - *pos;
+ *pos= get_bits_count(&s->gb);
+ }
+}
+
static int huffman_decode(MPADecodeContext *s, GranuleDef *g,
int16_t *exponents, int end_pos2)
{
if (pos >= end_pos){
// av_log(NULL, AV_LOG_ERROR, "pos: %d %d %d %d\n", pos, end_pos, end_pos2, s_index);
- if(s->in_gb.buffer && pos >= s->gb.size_in_bits){
- s->gb= s->in_gb;
- s->in_gb.buffer=NULL;
- assert((get_bits_count(&s->gb) & 7) == 0);
- skip_bits_long(&s->gb, pos - end_pos);
- end_pos2=
- end_pos= end_pos2 + get_bits_count(&s->gb) - pos;
- pos= get_bits_count(&s->gb);
- }
+ switch_buffer(s, &pos, &end_pos, &end_pos2);
// av_log(NULL, AV_LOG_ERROR, "new pos: %d %d\n", pos, end_pos);
if(pos >= end_pos)
break;
exponent= exponents[s_index];
- dprintf("region=%d n=%d x=%d y=%d exp=%d\n",
+ dprintf(s->avctx, "region=%d n=%d x=%d y=%d exp=%d\n",
i, g->region_size[i] - j, x, y, exponent);
if(y&16){
x = y >> 5;
s_index -= 4;
skip_bits_long(&s->gb, last_pos - pos);
av_log(NULL, AV_LOG_INFO, "overread, skip %d enddists: %d %d\n", last_pos - pos, end_pos-pos, end_pos2-pos);
+ if(s->error_resilience >= FF_ER_COMPLIANT)
+ s_index=0;
break;
}
// av_log(NULL, AV_LOG_ERROR, "pos2: %d %d %d %d\n", pos, end_pos, end_pos2, s_index);
- if(s->in_gb.buffer && pos >= s->gb.size_in_bits){
- s->gb= s->in_gb;
- s->in_gb.buffer=NULL;
- assert((get_bits_count(&s->gb) & 7) == 0);
- skip_bits_long(&s->gb, pos - end_pos);
- end_pos2=
- end_pos= end_pos2 + get_bits_count(&s->gb) - pos;
- pos= get_bits_count(&s->gb);
- }
+ switch_buffer(s, &pos, &end_pos, &end_pos2);
// av_log(NULL, AV_LOG_ERROR, "new pos2: %d %d %d\n", pos, end_pos, s_index);
if(pos >= end_pos)
break;
last_pos= pos;
code = get_vlc2(&s->gb, vlc->table, vlc->bits, 1);
- dprintf("t=%d code=%d\n", g->count1table_select, code);
+ dprintf(s->avctx, "t=%d code=%d\n", g->count1table_select, code);
g->sb_hybrid[s_index+0]=
g->sb_hybrid[s_index+1]=
g->sb_hybrid[s_index+2]=
g->sb_hybrid[s_index+3]= 0;
while(code){
- const static int idxtab[16]={3,3,2,2,1,1,1,1,0,0,0,0,0,0,0,0};
+ static const int idxtab[16]={3,3,2,2,1,1,1,1,0,0,0,0,0,0,0,0};
int v;
int pos= s_index+idxtab[code];
code ^= 8>>idxtab[code];
}
s_index+=4;
}
- memset(&g->sb_hybrid[s_index], 0, sizeof(*g->sb_hybrid)*(576 - s_index));
-
/* skip extension bits */
- bits_left = end_pos - get_bits_count(&s->gb);
+ bits_left = end_pos2 - get_bits_count(&s->gb);
//av_log(NULL, AV_LOG_ERROR, "left:%d buf:%p\n", bits_left, s->in_gb.buffer);
- if (bits_left < 0) {
- dprintf("bits_left=%d\n", bits_left);
- return -1;
+ if (bits_left < 0/* || bits_left > 500*/) {
+ av_log(NULL, AV_LOG_ERROR, "bits_left=%d\n", bits_left);
+ s_index=0;
+ }else if(bits_left > 0 && s->error_resilience >= FF_ER_AGGRESSIVE){
+ av_log(NULL, AV_LOG_ERROR, "bits_left=%d\n", bits_left);
+ s_index=0;
}
+ memset(&g->sb_hybrid[s_index], 0, sizeof(*g->sb_hybrid)*(576 - s_index));
skip_bits_long(&s->gb, bits_left);
+ i= get_bits_count(&s->gb);
+ switch_buffer(s, &i, &end_pos, &end_pos2);
+
return 0;
}
for(gr=0;gr<nb_granules;gr++) {
for(ch=0;ch<s->nb_channels;ch++) {
- dprintf("gr=%d ch=%d: side_info\n", gr, ch);
+ dprintf(s->avctx, "gr=%d ch=%d: side_info\n", gr, ch);
g = &granules[ch][gr];
g->part2_3_length = get_bits(&s->gb, 12);
g->big_values = get_bits(&s->gb, 9);
+ if(g->big_values > 288){
+ av_log(s->avctx, AV_LOG_ERROR, "big_values too big\n");
+ return -1;
+ }
+
g->global_gain = get_bits(&s->gb, 8);
/* if MS stereo only is selected, we precompute the
1/sqrt(2) renormalization factor */
g->scalefac_compress = get_bits(&s->gb, 9);
else
g->scalefac_compress = get_bits(&s->gb, 4);
- blocksplit_flag = get_bits(&s->gb, 1);
+ blocksplit_flag = get_bits1(&s->gb);
if (blocksplit_flag) {
g->block_type = get_bits(&s->gb, 2);
- if (g->block_type == 0)
+ if (g->block_type == 0){
+ av_log(NULL, AV_LOG_ERROR, "invalid block type\n");
return -1;
- g->switch_point = get_bits(&s->gb, 1);
+ }
+ g->switch_point = get_bits1(&s->gb);
for(i=0;i<2;i++)
g->table_select[i] = get_bits(&s->gb, 5);
for(i=0;i<3;i++)
g->subblock_gain[i] = get_bits(&s->gb, 3);
- /* compute huffman coded region sizes */
- if (g->block_type == 2)
- g->region_size[0] = (36 / 2);
- else {
- if (s->sample_rate_index <= 2)
- g->region_size[0] = (36 / 2);
- else if (s->sample_rate_index != 8)
- g->region_size[0] = (54 / 2);
- else
- g->region_size[0] = (108 / 2);
- }
- g->region_size[1] = (576 / 2);
+ ff_init_short_region(s, g);
} else {
- int region_address1, region_address2, l;
+ int region_address1, region_address2;
g->block_type = 0;
g->switch_point = 0;
for(i=0;i<3;i++)
/* compute huffman coded region sizes */
region_address1 = get_bits(&s->gb, 4);
region_address2 = get_bits(&s->gb, 3);
- dprintf("region1=%d region2=%d\n",
+ dprintf(s->avctx, "region1=%d region2=%d\n",
region_address1, region_address2);
- g->region_size[0] =
- band_index_long[s->sample_rate_index][region_address1 + 1] >> 1;
- l = region_address1 + region_address2 + 2;
- /* should not overflow */
- if (l > 22)
- l = 22;
- g->region_size[1] =
- band_index_long[s->sample_rate_index][l] >> 1;
- }
- /* convert region offsets to region sizes and truncate
- size to big_values */
- g->region_size[2] = (576 / 2);
- j = 0;
- for(i=0;i<3;i++) {
- k = FFMIN(g->region_size[i], g->big_values);
- g->region_size[i] = k - j;
- j = k;
- }
-
- /* compute band indexes */
- if (g->block_type == 2) {
- if (g->switch_point) {
- /* if switched mode, we handle the 36 first samples as
- long blocks. For 8000Hz, we handle the 48 first
- exponents as long blocks (XXX: check this!) */
- if (s->sample_rate_index <= 2)
- g->long_end = 8;
- else if (s->sample_rate_index != 8)
- g->long_end = 6;
- else
- g->long_end = 4; /* 8000 Hz */
-
- g->short_start = 2 + (s->sample_rate_index != 8);
- } else {
- g->long_end = 0;
- g->short_start = 0;
- }
- } else {
- g->short_start = 13;
- g->long_end = 22;
+ ff_init_long_region(s, g, region_address1, region_address2);
}
+ ff_region_offset2size(g);
+ ff_compute_band_indexes(s, g);
g->preflag = 0;
if (!s->lsf)
- g->preflag = get_bits(&s->gb, 1);
- g->scalefac_scale = get_bits(&s->gb, 1);
- g->count1table_select = get_bits(&s->gb, 1);
- dprintf("block_type=%d switch_point=%d\n",
+ g->preflag = get_bits1(&s->gb);
+ g->scalefac_scale = get_bits1(&s->gb);
+ g->count1table_select = get_bits1(&s->gb);
+ dprintf(s->avctx, "block_type=%d switch_point=%d\n",
g->block_type, g->switch_point);
}
}
const uint8_t *ptr = s->gb.buffer + (get_bits_count(&s->gb)>>3);
assert((get_bits_count(&s->gb) & 7) == 0);
/* now we get bits from the main_data_begin offset */
- dprintf("seekback: %d\n", main_data_begin);
+ dprintf(s->avctx, "seekback: %d\n", main_data_begin);
//av_log(NULL, AV_LOG_ERROR, "backstep:%d, lastbuf:%d\n", main_data_begin, s->last_buf_size);
- if(main_data_begin > s->last_buf_size){
- av_log(NULL, AV_LOG_ERROR, "backstep:%d, lastbuf:%d\n", main_data_begin, s->last_buf_size);
- s->last_buf_size= main_data_begin;
- }
memcpy(s->last_buf + s->last_buf_size, ptr, EXTRABYTES);
s->in_gb= s->gb;
- init_get_bits(&s->gb, s->last_buf + s->last_buf_size - main_data_begin, main_data_begin*8);
+ init_get_bits(&s->gb, s->last_buf, s->last_buf_size*8);
+ skip_bits_long(&s->gb, 8*(s->last_buf_size - main_data_begin));
}
for(gr=0;gr<nb_granules;gr++) {
for(ch=0;ch<s->nb_channels;ch++) {
g = &granules[ch][gr];
+ if(get_bits_count(&s->gb)<0){
+ av_log(NULL, AV_LOG_ERROR, "mdb:%d, lastbuf:%d skipping granule %d\n",
+ main_data_begin, s->last_buf_size, gr);
+ skip_bits_long(&s->gb, g->part2_3_length);
+ memset(g->sb_hybrid, 0, sizeof(g->sb_hybrid));
+ if(get_bits_count(&s->gb) >= s->gb.size_in_bits && s->in_gb.buffer){
+ skip_bits_long(&s->in_gb, get_bits_count(&s->gb) - s->gb.size_in_bits);
+ s->gb= s->in_gb;
+ s->in_gb.buffer=NULL;
+ }
+ continue;
+ }
bits_pos = get_bits_count(&s->gb);
/* MPEG1 scale factors */
slen1 = slen_table[0][g->scalefac_compress];
slen2 = slen_table[1][g->scalefac_compress];
- dprintf("slen1=%d slen2=%d\n", slen1, slen2);
+ dprintf(s->avctx, "slen1=%d slen2=%d\n", slen1, slen2);
if (g->block_type == 2) {
n = g->switch_point ? 17 : 18;
j = 0;
}
#if defined(DEBUG)
{
- dprintf("scfsi=%x gr=%d ch=%d scale_factors:\n",
+ dprintf(s->avctx, "scfsi=%x gr=%d ch=%d scale_factors:\n",
g->scfsi, gr, ch);
for(i=0;i<j;i++)
- dprintf(" %d", g->scale_factors[i]);
- dprintf("\n");
+ dprintf(s->avctx, " %d", g->scale_factors[i]);
+ dprintf(s->avctx, "\n");
}
#endif
} else {
g->scale_factors[j] = 0;
#if defined(DEBUG)
{
- dprintf("gr=%d ch=%d scale_factors:\n",
+ dprintf(s->avctx, "gr=%d ch=%d scale_factors:\n",
gr, ch);
for(i=0;i<40;i++)
- dprintf(" %d", g->scale_factors[i]);
- dprintf("\n");
+ dprintf(s->avctx, " %d", g->scale_factors[i]);
+ dprintf(s->avctx, "\n");
}
#endif
}
exponents_from_scale_factors(s, g, exponents);
/* read Huffman coded residue */
- if (huffman_decode(s, g, exponents,
- bits_pos + g->part2_3_length) < 0)
- return -1;
+ huffman_decode(s, g, exponents, bits_pos + g->part2_3_length);
#if defined(DEBUG)
sample_dump(0, g->sb_hybrid, 576);
#endif
#endif
}
} /* gr */
+ if(get_bits_count(&s->gb)<0)
+ skip_bits_long(&s->gb, -get_bits_count(&s->gb));
return nb_granules * 18;
}
/* skip error protection field */
if (s->error_protection)
- get_bits(&s->gb, 16);
+ skip_bits(&s->gb, 16);
- dprintf("frame %d:\n", s->frame_count);
+ dprintf(s->avctx, "frame %d:\n", s->frame_count);
switch(s->layer) {
case 1:
nb_frames = mp_decode_layer1(s);
}else
av_log(NULL, AV_LOG_ERROR, "invalid old backstep %d\n", i);
s->gb= s->in_gb;
+ s->in_gb.buffer= NULL;
}
align_get_bits(&s->gb);
for(i=0;i<nb_frames;i++) {
for(ch=0;ch<s->nb_channels;ch++) {
int j;
- dprintf("%d-%d:", i, ch);
+ dprintf(s->avctx, "%d-%d:", i, ch);
for(j=0;j<SBLIMIT;j++)
- dprintf(" %0.6f", (double)s->sb_samples[ch][i][j] / FRAC_ONE);
- dprintf("\n");
+ dprintf(s->avctx, " %0.6f", (double)s->sb_samples[ch][i][j] / FRAC_ONE);
+ dprintf(s->avctx, "\n");
}
}
#endif
static int decode_frame(AVCodecContext * avctx,
void *data, int *data_size,
- uint8_t * buf, int buf_size)
+ const uint8_t * buf, int buf_size)
{
MPADecodeContext *s = avctx->priv_data;
uint32_t header;
if(buf_size < HEADER_SIZE)
return -1;
- header = (buf[0] << 24) | (buf[1] << 16) | (buf[2] << 8) | buf[3];
+ header = AV_RB32(buf);
if(ff_mpa_check_header(header) < 0){
buf++;
// buf_size--;
- av_log(avctx, AV_LOG_ERROR, "header missing skiping one byte\n");
+ av_log(avctx, AV_LOG_ERROR, "Header missing skipping one byte.\n");
goto retry;
}
- if (decode_header(s, header) == 1) {
+ if (ff_mpegaudio_decode_header(s, header) == 1) {
/* free format: prepare to compute frame size */
s->frame_size = -1;
return -1;
}
/* update codec info */
- avctx->sample_rate = s->sample_rate;
avctx->channels = s->nb_channels;
avctx->bit_rate = s->bit_rate;
avctx->sub_id = s->layer;
return -1;
}else if(s->frame_size < buf_size){
av_log(avctx, AV_LOG_ERROR, "incorrect frame size\n");
+ buf_size= s->frame_size;
}
out_size = mp_decode_frame(s, out_samples, buf, buf_size);
- if(out_size>=0)
+ if(out_size>=0){
*data_size = out_size;
- else
- av_log(avctx, AV_LOG_DEBUG, "Error while decoding mpeg audio frame\n"); //FIXME return -1 / but also return the number of bytes consumed
+ avctx->sample_rate = s->sample_rate;
+ //FIXME maybe move the other codec info stuff from above here too
+ }else
+ av_log(avctx, AV_LOG_DEBUG, "Error while decoding MPEG audio frame.\n"); //FIXME return -1 / but also return the number of bytes consumed
s->frame_size = 0;
return buf_size;
}
+static void flush(AVCodecContext *avctx){
+ MPADecodeContext *s = avctx->priv_data;
+ memset(s->synth_buf, 0, sizeof(s->synth_buf));
+ s->last_buf_size= 0;
+}
+#ifdef CONFIG_MP3ADU_DECODER
static int decode_frame_adu(AVCodecContext * avctx,
void *data, int *data_size,
- uint8_t * buf, int buf_size)
+ const uint8_t * buf, int buf_size)
{
MPADecodeContext *s = avctx->priv_data;
uint32_t header;
len = MPA_MAX_CODED_FRAME_SIZE;
// Get header and restore sync word
- header = (buf[0] << 24) | (buf[1] << 16) | (buf[2] << 8) | buf[3] | 0xffe00000;
+ header = AV_RB32(buf) | 0xffe00000;
if (ff_mpa_check_header(header) < 0) { // Bad header, discard frame
*data_size = 0;
return buf_size;
}
- decode_header(s, header);
+ ff_mpegaudio_decode_header(s, header);
/* update codec info */
avctx->sample_rate = s->sample_rate;
avctx->channels = s->nb_channels;
*data_size = out_size;
return buf_size;
}
+#endif /* CONFIG_MP3ADU_DECODER */
-
+#ifdef CONFIG_MP3ON4_DECODER
/* Next 3 arrays are indexed by channel config number (passed via codecdata) */
static int mp3Frames[16] = {0,1,1,2,3,3,4,5,2}; /* number of mp3 decoder instances */
static int mp3Channels[16] = {0,1,2,3,4,5,6,8,4}; /* total output channels */
/* Init the first mp3 decoder in standard way, so that all tables get builded
* We replace avctx->priv_data with the context of the first decoder so that
* decode_init() does not have to be changed.
- * Other decoders will be inited here copying data from the first context
+ * Other decoders will be initialized here copying data from the first context
*/
// Allocate zeroed memory for the first decoder context
s->mp3decctx[0] = av_mallocz(sizeof(MPADecodeContext));
s->mp3decctx[i] = av_mallocz(sizeof(MPADecodeContext));
s->mp3decctx[i]->compute_antialias = s->mp3decctx[0]->compute_antialias;
s->mp3decctx[i]->adu_mode = 1;
+ s->mp3decctx[i]->avctx = avctx;
}
return 0;
static int decode_frame_mp3on4(AVCodecContext * avctx,
void *data, int *data_size,
- uint8_t * buf, int buf_size)
+ const uint8_t * buf, int buf_size)
{
MP3On4DecodeContext *s = avctx->priv_data;
MPADecodeContext *m;
OUT_INT decoded_buf[MPA_FRAME_SIZE * MPA_MAX_CHANNELS];
OUT_INT *outptr, *bp;
int fsize;
- unsigned char *start2 = buf, *start;
+ const unsigned char *start2 = buf, *start;
int fr, i, j, n;
int off = avctx->channels;
int *coff = chan_offset[s->chan_cfg];
assert (m != NULL);
// Get header
- header = (start[0] << 24) | (start[1] << 16) | (start[2] << 8) | start[3] | 0xfff00000;
+ header = AV_RB32(start) | 0xfff00000;
if (ff_mpa_check_header(header) < 0) { // Bad header, discard block
*data_size = 0;
return buf_size;
}
- decode_header(m, header);
+ ff_mpegaudio_decode_header(m, header);
mp_decode_frame(m, decoded_buf, start, fsize);
n = MPA_FRAME_SIZE * m->nb_channels;
*data_size = out_size;
return buf_size;
}
+#endif /* CONFIG_MP3ON4_DECODER */
-
+#ifdef CONFIG_MP2_DECODER
AVCodec mp2_decoder =
{
"mp2",
NULL,
decode_frame,
CODEC_CAP_PARSE_ONLY,
+ .flush= flush,
};
-
+#endif
+#ifdef CONFIG_MP3_DECODER
AVCodec mp3_decoder =
{
"mp3",
NULL,
decode_frame,
CODEC_CAP_PARSE_ONLY,
+ .flush= flush,
};
-
+#endif
+#ifdef CONFIG_MP3ADU_DECODER
AVCodec mp3adu_decoder =
{
"mp3adu",
NULL,
decode_frame_adu,
CODEC_CAP_PARSE_ONLY,
+ .flush= flush,
};
-
+#endif
+#ifdef CONFIG_MP3ON4_DECODER
AVCodec mp3on4_decoder =
{
"mp3on4",
NULL,
decode_close_mp3on4,
decode_frame_mp3on4,
- 0
+ .flush= flush,
};
+#endif
projects / ffmpeg / blobdiff