/*
* MPEG Audio decoder
- * Copyright (c) 2001, 2002 Fabrice Bellard.
+ * Copyright (c) 2001, 2002 Fabrice Bellard
*
* This file is part of FFmpeg.
*
*/
/**
- * @file mpegaudiodec.c
+ * @file libavcodec/mpegaudiodec.c
* MPEG Audio decoder.
*/
-//#define DEBUG
#include "avcodec.h"
-#include "bitstream.h"
+#include "get_bits.h"
#include "dsputil.h"
/*
* - test lsf / mpeg25 extensively.
*/
-/* define USE_HIGHPRECISION to have a bit exact (but slower) mpeg
- audio decoder */
-#ifdef CONFIG_MPEGAUDIO_HP
-# define USE_HIGHPRECISION
-#endif
-
#include "mpegaudio.h"
#include "mpegaudiodecheader.h"
#define HEADER_SIZE 4
-/**
- * Context for MP3On4 decoder
- */
-typedef struct MP3On4DecodeContext {
- int frames; ///< number of mp3 frames per block (number of mp3 decoder instances)
- int chan_cfg; ///< channel config number
- MPADecodeContext *mp3decctx[5]; ///< MPADecodeContext for every decoder instance
-} MP3On4DecodeContext;
-
/* layer 3 "granule" */
typedef struct GranuleDef {
uint8_t scfsi;
int32_t sb_hybrid[SBLIMIT * 18]; /* 576 samples */
} GranuleDef;
-#define MODE_EXT_MS_STEREO 2
-#define MODE_EXT_I_STEREO 1
-
#include "mpegaudiodata.h"
#include "mpegaudiodectab.h"
/* vlc structure for decoding layer 3 huffman tables */
static VLC huff_vlc[16];
+static VLC_TYPE huff_vlc_tables[
+ 0+128+128+128+130+128+154+166+
+ 142+204+190+170+542+460+662+414
+ ][2];
+static const int huff_vlc_tables_sizes[16] = {
+ 0, 128, 128, 128, 130, 128, 154, 166,
+ 142, 204, 190, 170, 542, 460, 662, 414
+};
static VLC huff_quad_vlc[2];
+static VLC_TYPE huff_quad_vlc_tables[128+16][2];
+static const int huff_quad_vlc_tables_sizes[2] = {
+ 128, 16
+};
/* computed from band_size_long */
static uint16_t band_index_long[9][23];
/* XXX: free when all decoders are closed */
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 inline int l1_unscale(int n, int mant, int scale_factor)
};
#endif
-static void int_pow_init(void)
+static av_cold void int_pow_init(void)
{
int i, a;
}
#endif
-static int decode_init(AVCodecContext * avctx)
+static av_cold int decode_init(AVCodecContext * avctx)
{
MPADecodeContext *s = avctx->priv_data;
static int init=0;
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;
+ avctx->sample_fmt= OUT_FMT;
+ s->error_recognition= avctx->error_recognition;
if(avctx->antialias_algo != FF_AA_FLOAT)
s->compute_antialias= compute_antialias_integer;
s->compute_antialias= compute_antialias_float;
if (!init && !avctx->parse_only) {
+ int offset;
+
/* scale factors table for layer 1/2 */
for(i=0;i<64;i++) {
int shift, mod;
int n, norm;
n = i + 2;
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);
+ scale_factor_mult[i][0] = MULL(FIXR(1.0 * 2.0), norm, FRAC_BITS);
+ scale_factor_mult[i][1] = MULL(FIXR(0.7937005259 * 2.0), norm, FRAC_BITS);
+ scale_factor_mult[i][2] = MULL(FIXR(0.6299605249 * 2.0), norm, FRAC_BITS);
dprintf(avctx, "%d: norm=%x s=%x %x %x\n",
i, norm,
scale_factor_mult[i][0],
ff_mpa_synth_init(window);
/* huffman decode tables */
+ offset = 0;
for(i=1;i<16;i++) {
const HuffTable *h = &mpa_huff_tables[i];
int xsize, x, y;
- unsigned int n;
uint8_t tmp_bits [512];
uint16_t tmp_codes[512];
memset(tmp_codes, 0, sizeof(tmp_codes));
xsize = h->xsize;
- n = xsize * xsize;
j = 0;
for(x=0;x<xsize;x++) {
}
/* XXX: fail test */
+ huff_vlc[i].table = huff_vlc_tables+offset;
+ huff_vlc[i].table_allocated = huff_vlc_tables_sizes[i];
init_vlc(&huff_vlc[i], 7, 512,
- tmp_bits, 1, 1, tmp_codes, 2, 2, 1);
+ tmp_bits, 1, 1, tmp_codes, 2, 2,
+ INIT_VLC_USE_NEW_STATIC);
+ offset += huff_vlc_tables_sizes[i];
}
+ assert(offset == FF_ARRAY_ELEMS(huff_vlc_tables));
+
+ offset = 0;
for(i=0;i<2;i++) {
+ huff_quad_vlc[i].table = huff_quad_vlc_tables+offset;
+ huff_quad_vlc[i].table_allocated = huff_quad_vlc_tables_sizes[i];
init_vlc(&huff_quad_vlc[i], i == 0 ? 7 : 4, 16,
- mpa_quad_bits[i], 1, 1, mpa_quad_codes[i], 1, 1, 1);
+ mpa_quad_bits[i], 1, 1, mpa_quad_codes[i], 1, 1,
+ INIT_VLC_USE_NEW_STATIC);
+ offset += huff_quad_vlc_tables_sizes[i];
}
+ assert(offset == FF_ARRAY_ELEMS(huff_quad_vlc_tables));
for(i=0;i<9;i++) {
k = 0;
/* normalized to FRAC_BITS */
table_4_3_value[i] = m;
-// av_log(NULL, AV_LOG_DEBUG, "%d %d %f\n", i, m, pow((double)i, 4.0 / 3.0));
table_4_3_exp[i] = -e;
}
for(i=0; i<512*16; i++){
csa_table_float[i][1] = ca;
csa_table_float[i][2] = ca + cs;
csa_table_float[i][3] = ca - cs;
-// printf("%d %d %d %d\n", FIX(cs), FIX(cs-1), FIX(ca), FIX(cs)-FIX(ca));
-// av_log(NULL, AV_LOG_DEBUG,"%f %f %f %f\n", cs, ca, ca+cs, ca-cs);
}
/* compute mdct windows */
mdct_win[j][i/3] = FIXHR((d / (1<<5)));
else
mdct_win[j][i ] = FIXHR((d / (1<<5)));
-// av_log(NULL, AV_LOG_DEBUG, "%2d %d %f\n", i,j,d / (1<<5));
}
}
}
}
-#if defined(DEBUG)
- for(j=0;j<8;j++) {
- av_log(avctx, AV_LOG_DEBUG, "win%d=\n", j);
- for(i=0;i<36;i++)
- av_log(avctx, AV_LOG_DEBUG, "%f, ", (double)mdct_win[j][i] / FRAC_ONE);
- av_log(avctx, AV_LOG_DEBUG, "\n");
- }
-#endif
init = 1;
}
-#ifdef DEBUG
- s->frame_count = 0;
-#endif
if (avctx->codec_id == CODEC_ID_MP3ADU)
s->adu_mode = 1;
return 0;
/* signed 16x16 -> 32 multiply */
#define MULS(ra, rb) MUL16(ra, rb)
+#define MLSS(rt, ra, rb) MLS16(rt, ra, rb)
+
#else
static inline int round_sample(int64_t *sum)
}
# define MULS(ra, rb) MUL64(ra, rb)
+# define MACS(rt, ra, rb) MAC64(rt, ra, rb)
+# define MLSS(rt, ra, rb) MLS64(rt, ra, rb)
#endif
-#define SUM8(sum, op, w, p) \
-{ \
- sum op MULS((w)[0 * 64], p[0 * 64]);\
- sum op MULS((w)[1 * 64], p[1 * 64]);\
- sum op MULS((w)[2 * 64], p[2 * 64]);\
- sum op MULS((w)[3 * 64], p[3 * 64]);\
- sum op MULS((w)[4 * 64], p[4 * 64]);\
- sum op MULS((w)[5 * 64], p[5 * 64]);\
- sum op MULS((w)[6 * 64], p[6 * 64]);\
- sum op MULS((w)[7 * 64], p[7 * 64]);\
+#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) \
{ \
int tmp;\
tmp = p[0 * 64];\
- sum1 op1 MULS((w1)[0 * 64], tmp);\
- sum2 op2 MULS((w2)[0 * 64], tmp);\
+ op1(sum1, (w1)[0 * 64], tmp);\
+ op2(sum2, (w2)[0 * 64], tmp);\
tmp = p[1 * 64];\
- sum1 op1 MULS((w1)[1 * 64], tmp);\
- sum2 op2 MULS((w2)[1 * 64], tmp);\
+ op1(sum1, (w1)[1 * 64], tmp);\
+ op2(sum2, (w2)[1 * 64], tmp);\
tmp = p[2 * 64];\
- sum1 op1 MULS((w1)[2 * 64], tmp);\
- sum2 op2 MULS((w2)[2 * 64], tmp);\
+ op1(sum1, (w1)[2 * 64], tmp);\
+ op2(sum2, (w2)[2 * 64], tmp);\
tmp = p[3 * 64];\
- sum1 op1 MULS((w1)[3 * 64], tmp);\
- sum2 op2 MULS((w2)[3 * 64], tmp);\
+ op1(sum1, (w1)[3 * 64], tmp);\
+ op2(sum2, (w2)[3 * 64], tmp);\
tmp = p[4 * 64];\
- sum1 op1 MULS((w1)[4 * 64], tmp);\
- sum2 op2 MULS((w2)[4 * 64], tmp);\
+ op1(sum1, (w1)[4 * 64], tmp);\
+ op2(sum2, (w2)[4 * 64], tmp);\
tmp = p[5 * 64];\
- sum1 op1 MULS((w1)[5 * 64], tmp);\
- sum2 op2 MULS((w2)[5 * 64], tmp);\
+ op1(sum1, (w1)[5 * 64], tmp);\
+ op2(sum2, (w2)[5 * 64], tmp);\
tmp = p[6 * 64];\
- sum1 op1 MULS((w1)[6 * 64], tmp);\
- sum2 op2 MULS((w2)[6 * 64], tmp);\
+ op1(sum1, (w1)[6 * 64], tmp);\
+ op2(sum2, (w2)[6 * 64], tmp);\
tmp = p[7 * 64];\
- sum1 op1 MULS((w1)[7 * 64], tmp);\
- sum2 op2 MULS((w2)[7 * 64], tmp);\
+ op1(sum1, (w1)[7 * 64], tmp);\
+ op2(sum2, (w2)[7 * 64], tmp);\
}
-void ff_mpa_synth_init(MPA_INT *window)
+void av_cold ff_mpa_synth_init(MPA_INT *window)
{
int i;
#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;
}
sum = *dither_state;
p = synth_buf + 16;
- SUM8(sum, +=, w, p);
+ SUM8(MACS, sum, w, p);
p = synth_buf + 48;
- SUM8(sum, -=, w + 32, p);
+ SUM8(MLSS, sum, w + 32, p);
*samples = round_sample(&sum);
samples += incr;
w++;
for(j=1;j<16;j++) {
sum2 = 0;
p = synth_buf + 16 + j;
- SUM8P2(sum, +=, sum2, -=, w, w2, p);
+ SUM8P2(sum, MACS, sum2, MLSS, w, w2, p);
p = synth_buf + 48 - j;
- SUM8P2(sum, -=, sum2, -=, w + 32, w2 + 32, p);
+ SUM8P2(sum, MLSS, sum2, MLSS, w + 32, w2 + 32, p);
*samples = round_sample(&sum);
samples += incr;
}
p = synth_buf + 32;
- SUM8(sum, -=, w + 32, p);
+ SUM8(MLSS, sum, w + 32, p);
*samples = round_sample(&sum);
*dither_state= sum;
t2 = tmp[i + 1];
t3 = tmp[i + 3];
s1 = MULH(2*(t3 + t2), icos36h[j]);
- s3 = MULL(t3 - t2, icos36[8 - j]);
+ s3 = MULL(t3 - t2, icos36[8 - j], FRAC_BITS);
t0 = s0 + s1;
t1 = s0 - s1;
j += 1 << bit_alloc_bits;
}
-#ifdef DEBUG
- {
- for(ch=0;ch<s->nb_channels;ch++) {
- for(i=0;i<sblimit;i++)
- dprintf(s->avctx, " %d", bit_alloc[ch][i]);
- dprintf(s->avctx, "\n");
- }
- }
-#endif
-
/* scale codes */
for(i=0;i<sblimit;i++) {
for(ch=0;ch<s->nb_channels;ch++) {
}
}
-#ifdef DEBUG
- for(ch=0;ch<s->nb_channels;ch++) {
- for(i=0;i<sblimit;i++) {
- if (bit_alloc[ch][i]) {
- sf = scale_factors[ch][i];
- dprintf(s->avctx, " %d %d %d", sf[0], sf[1], sf[2]);
- } else {
- dprintf(s->avctx, " -");
- }
- }
- dprintf(s->avctx, "\n");
- }
-#endif
-
/* samples */
for(k=0;k<3;k++) {
for(l=0;l<12;l+=3) {
part. We must go back into the data */
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)
+ av_log(s->avctx, AV_LOG_INFO, "overread, skip %d enddists: %d %d\n", last_pos - pos, end_pos-pos, end_pos2-pos);
+ if(s->error_recognition >= FF_ER_COMPLIANT)
s_index=0;
break;
}
/* skip extension bits */
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/* || bits_left > 500*/) {
- av_log(NULL, AV_LOG_ERROR, "bits_left=%d\n", bits_left);
+ if (bits_left < 0 && s->error_recognition >= FF_ER_COMPLIANT) {
+ av_log(s->avctx, 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);
+ }else if(bits_left > 0 && s->error_recognition >= FF_ER_AGGRESSIVE){
+ av_log(s->avctx, 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));
v2 = is_tab[1][sf];
for(j=0;j<len;j++) {
tmp0 = tab0[j];
- tab0[j] = MULL(tmp0, v1);
- tab1[j] = MULL(tmp0, v2);
+ tab0[j] = MULL(tmp0, v1, FRAC_BITS);
+ tab1[j] = MULL(tmp0, v2, FRAC_BITS);
}
} else {
found1:
for(j=0;j<len;j++) {
tmp0 = tab0[j];
tmp1 = tab1[j];
- tab0[j] = MULL(tmp0 + tmp1, ISQRT2);
- tab1[j] = MULL(tmp0 - tmp1, ISQRT2);
+ tab0[j] = MULL(tmp0 + tmp1, ISQRT2, FRAC_BITS);
+ tab1[j] = MULL(tmp0 - tmp1, ISQRT2, FRAC_BITS);
}
}
}
v2 = is_tab[1][sf];
for(j=0;j<len;j++) {
tmp0 = tab0[j];
- tab0[j] = MULL(tmp0, v1);
- tab1[j] = MULL(tmp0, v2);
+ tab0[j] = MULL(tmp0, v1, FRAC_BITS);
+ tab1[j] = MULL(tmp0, v2, FRAC_BITS);
}
} else {
found2:
for(j=0;j<len;j++) {
tmp0 = tab0[j];
tmp1 = tab1[j];
- tab0[j] = MULL(tmp0 + tmp1, ISQRT2);
- tab1[j] = MULL(tmp0 - tmp1, ISQRT2);
+ tab0[j] = MULL(tmp0 + tmp1, ISQRT2, FRAC_BITS);
+ tab1[j] = MULL(tmp0 - tmp1, ISQRT2, FRAC_BITS);
}
}
}
}
}
-#if defined(DEBUG)
-void sample_dump(int fnum, int32_t *tab, int n)
-{
- static FILE *files[16], *f;
- char buf[512];
- int i;
- int32_t v;
-
- f = files[fnum];
- if (!f) {
- snprintf(buf, sizeof(buf), "/tmp/out%d.%s.pcm",
- fnum,
-#ifdef USE_HIGHPRECISION
- "hp"
-#else
- "lp"
-#endif
- );
- f = fopen(buf, "w");
- if (!f)
- return;
- files[fnum] = f;
- }
-
- if (fnum == 0) {
- static int pos = 0;
- av_log(NULL, AV_LOG_DEBUG, "pos=%d\n", pos);
- for(i=0;i<n;i++) {
- av_log(NULL, AV_LOG_DEBUG, " %0.4f", (double)tab[i] / FRAC_ONE);
- if ((i % 18) == 17)
- av_log(NULL, AV_LOG_DEBUG, "\n");
- }
- pos += n;
- }
- for(i=0;i<n;i++) {
- /* normalize to 23 frac bits */
- v = tab[i] << (23 - FRAC_BITS);
- fwrite(&v, 1, sizeof(int32_t), f);
- }
-}
-#endif
-
-
/* main layer3 decoding function */
static int mp_decode_layer3(MPADecodeContext *s)
{
if (blocksplit_flag) {
g->block_type = get_bits(&s->gb, 2);
if (g->block_type == 0){
- av_log(NULL, AV_LOG_ERROR, "invalid block type\n");
+ av_log(s->avctx, AV_LOG_ERROR, "invalid block type\n");
return -1;
}
g->switch_point = get_bits1(&s->gb);
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++)
region_address2 = get_bits(&s->gb, 3);
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)
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",
+ av_log(s->avctx, 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));
}
g->scale_factors[j++] = 0;
}
-#if defined(DEBUG)
- {
- dprintf(s->avctx, "scfsi=%x gr=%d ch=%d scale_factors:\n",
- g->scfsi, gr, ch);
- for(i=0;i<j;i++)
- dprintf(s->avctx, " %d", g->scale_factors[i]);
- dprintf(s->avctx, "\n");
- }
-#endif
} else {
int tindex, tindex2, slen[4], sl, sf;
/* XXX: should compute exact size */
for(;j<40;j++)
g->scale_factors[j] = 0;
-#if defined(DEBUG)
- {
- dprintf(s->avctx, "gr=%d ch=%d scale_factors:\n",
- gr, ch);
- for(i=0;i<40;i++)
- dprintf(s->avctx, " %d", g->scale_factors[i]);
- dprintf(s->avctx, "\n");
- }
-#endif
}
exponents_from_scale_factors(s, g, exponents);
/* read Huffman coded residue */
huffman_decode(s, g, exponents, bits_pos + g->part2_3_length);
-#if defined(DEBUG)
- sample_dump(0, g->sb_hybrid, 576);
-#endif
} /* ch */
if (s->nb_channels == 2)
g = &granules[ch][gr];
reorder_block(s, g);
-#if defined(DEBUG)
- sample_dump(0, g->sb_hybrid, 576);
-#endif
s->compute_antialias(s, g);
-#if defined(DEBUG)
- sample_dump(1, g->sb_hybrid, 576);
-#endif
compute_imdct(s, g, &s->sb_samples[ch][18 * gr][0], s->mdct_buf[ch]);
-#if defined(DEBUG)
- sample_dump(2, &s->sb_samples[ch][18 * gr][0], 576);
-#endif
}
} /* gr */
if(get_bits_count(&s->gb)<0)
/* skip error protection field */
if (s->error_protection)
- get_bits(&s->gb, 16);
+ skip_bits(&s->gb, 16);
dprintf(s->avctx, "frame %d:\n", s->frame_count);
switch(s->layer) {
case 1:
+ s->avctx->frame_size = 384;
nb_frames = mp_decode_layer1(s);
break;
case 2:
+ s->avctx->frame_size = 1152;
nb_frames = mp_decode_layer2(s);
break;
case 3:
+ s->avctx->frame_size = s->lsf ? 576 : 1152;
default:
nb_frames = mp_decode_layer3(s);
memmove(s->last_buf, s->gb.buffer + (get_bits_count(&s->gb)>>3), i);
s->last_buf_size=i;
}else
- av_log(NULL, AV_LOG_ERROR, "invalid old backstep %d\n", i);
+ av_log(s->avctx, AV_LOG_ERROR, "invalid old backstep %d\n", i);
s->gb= s->in_gb;
s->in_gb.buffer= NULL;
}
i= (s->gb.size_in_bits - get_bits_count(&s->gb))>>3;
if(i<0 || i > BACKSTEP_SIZE || nb_frames<0){
- av_log(NULL, AV_LOG_ERROR, "invalid new backstep %d\n", i);
+ if(i<0)
+ av_log(s->avctx, AV_LOG_ERROR, "invalid new backstep %d\n", i);
i= FFMIN(BACKSTEP_SIZE, buf_size - HEADER_SIZE);
}
assert(i <= buf_size - HEADER_SIZE && i>= 0);
break;
}
-#if defined(DEBUG)
- for(i=0;i<nb_frames;i++) {
- for(ch=0;ch<s->nb_channels;ch++) {
- int j;
- dprintf(s->avctx, "%d-%d:", i, ch);
- for(j=0;j<SBLIMIT;j++)
- dprintf(s->avctx, " %0.6f", (double)s->sb_samples[ch][i][j] / FRAC_ONE);
- dprintf(s->avctx, "\n");
- }
- }
-#endif
+
/* apply the synthesis filter */
for(ch=0;ch<s->nb_channels;ch++) {
samples_ptr = samples + ch;
samples_ptr += 32 * s->nb_channels;
}
}
-#ifdef DEBUG
- s->frame_count++;
-#endif
+
return nb_frames * 32 * sizeof(OUT_INT) * s->nb_channels;
}
static int decode_frame(AVCodecContext * avctx,
void *data, int *data_size,
- uint8_t * buf, int buf_size)
+ AVPacket *avpkt)
{
+ const uint8_t *buf = avpkt->data;
+ int buf_size = avpkt->size;
MPADecodeContext *s = avctx->priv_data;
uint32_t header;
int out_size;
+ int skipped = 0;
OUT_INT *out_samples = data;
retry:
header = AV_RB32(buf);
if(ff_mpa_check_header(header) < 0){
buf++;
-// buf_size--;
+ buf_size--;
+ skipped++;
av_log(avctx, AV_LOG_ERROR, "Header missing skipping one byte.\n");
goto retry;
}
- if (ff_mpegaudio_decode_header(s, header) == 1) {
+ if (ff_mpegaudio_decode_header((MPADecodeHeader *)s, header) == 1) {
/* free format: prepare to compute frame size */
s->frame_size = -1;
return -1;
avctx->channels = s->nb_channels;
avctx->bit_rate = s->bit_rate;
avctx->sub_id = s->layer;
- switch(s->layer) {
- case 1:
- avctx->frame_size = 384;
- break;
- case 2:
- avctx->frame_size = 1152;
- break;
- case 3:
- if (s->lsf)
- avctx->frame_size = 576;
- else
- avctx->frame_size = 1152;
- break;
- }
if(s->frame_size<=0 || s->frame_size > buf_size){
av_log(avctx, AV_LOG_ERROR, "incomplete frame\n");
}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;
+ return buf_size + skipped;
}
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
+#if CONFIG_MP3ADU_DECODER
static int decode_frame_adu(AVCodecContext * avctx,
void *data, int *data_size,
- uint8_t * buf, int buf_size)
+ AVPacket *avpkt)
{
+ const uint8_t *buf = avpkt->data;
+ int buf_size = avpkt->size;
MPADecodeContext *s = avctx->priv_data;
uint32_t header;
int len, out_size;
return buf_size;
}
- ff_mpegaudio_decode_header(s, header);
+ ff_mpegaudio_decode_header((MPADecodeHeader *)s, header);
/* 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;
- avctx->frame_size=s->frame_size = len;
+ s->frame_size = len;
if (avctx->parse_only) {
out_size = buf_size;
}
#endif /* CONFIG_MP3ADU_DECODER */
-#ifdef CONFIG_MP3ON4_DECODER
+#if CONFIG_MP3ON4_DECODER
+
+/**
+ * Context for MP3On4 decoder
+ */
+typedef struct MP3On4DecodeContext {
+ int frames; ///< number of mp3 frames per block (number of mp3 decoder instances)
+ int syncword; ///< syncword patch
+ const uint8_t *coff; ///< channels offsets in output buffer
+ MPADecodeContext *mp3decctx[5]; ///< MPADecodeContext for every decoder instance
+} MP3On4DecodeContext;
+
+#include "mpeg4audio.h"
+
/* 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 */
+static const uint8_t mp3Frames[8] = {0,1,1,2,3,3,4,5}; /* number of mp3 decoder instances */
/* offsets into output buffer, assume output order is FL FR BL BR C LFE */
-static int chan_offset[9][5] = {
+static const uint8_t chan_offset[8][5] = {
{0},
{0}, // C
{0}, // FLR
{4,0,2}, // C FLR BLRS
{4,0,2,5}, // C FLR BLRS LFE
{4,0,2,6,5}, // C FLR BLRS BLR LFE
- {0,2} // FLR BLRS
};
static int decode_init_mp3on4(AVCodecContext * avctx)
{
MP3On4DecodeContext *s = avctx->priv_data;
+ MPEG4AudioConfig cfg;
int i;
if ((avctx->extradata_size < 2) || (avctx->extradata == NULL)) {
return -1;
}
- s->chan_cfg = (((unsigned char *)avctx->extradata)[1] >> 3) & 0x0f;
- s->frames = mp3Frames[s->chan_cfg];
- if(!s->frames) {
+ ff_mpeg4audio_get_config(&cfg, avctx->extradata, avctx->extradata_size);
+ if (!cfg.chan_config || cfg.chan_config > 7) {
av_log(avctx, AV_LOG_ERROR, "Invalid channel config number.\n");
return -1;
}
- avctx->channels = mp3Channels[s->chan_cfg];
+ s->frames = mp3Frames[cfg.chan_config];
+ s->coff = chan_offset[cfg.chan_config];
+ avctx->channels = ff_mpeg4audio_channels[cfg.chan_config];
+
+ if (cfg.sample_rate < 16000)
+ s->syncword = 0xffe00000;
+ else
+ s->syncword = 0xfff00000;
/* 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));
}
-static int decode_close_mp3on4(AVCodecContext * avctx)
+static av_cold int decode_close_mp3on4(AVCodecContext * avctx)
{
MP3On4DecodeContext *s = avctx->priv_data;
int i;
static int decode_frame_mp3on4(AVCodecContext * avctx,
void *data, int *data_size,
- uint8_t * buf, int buf_size)
+ AVPacket *avpkt)
{
+ const uint8_t *buf = avpkt->data;
+ int buf_size = avpkt->size;
MP3On4DecodeContext *s = avctx->priv_data;
MPADecodeContext *m;
- int len, out_size = 0;
+ int fsize, len = buf_size, out_size = 0;
uint32_t header;
OUT_INT *out_samples = data;
OUT_INT decoded_buf[MPA_FRAME_SIZE * MPA_MAX_CHANNELS];
OUT_INT *outptr, *bp;
- int fsize;
- unsigned char *start2 = buf, *start;
- int fr, i, j, n;
- int off = avctx->channels;
- int *coff = chan_offset[s->chan_cfg];
-
- len = buf_size;
+ int fr, j, n;
+ *data_size = 0;
// Discard too short frames
- if (buf_size < HEADER_SIZE) {
- *data_size = 0;
- return buf_size;
- }
+ if (buf_size < HEADER_SIZE)
+ return -1;
// If only one decoder interleave is not needed
outptr = s->frames == 1 ? out_samples : decoded_buf;
+ avctx->bit_rate = 0;
+
for (fr = 0; fr < s->frames; fr++) {
- start = start2;
- fsize = (start[0] << 4) | (start[1] >> 4);
- start2 += fsize;
- if (fsize > len)
- fsize = len;
- len -= fsize;
- if (fsize > MPA_MAX_CODED_FRAME_SIZE)
- fsize = MPA_MAX_CODED_FRAME_SIZE;
+ fsize = AV_RB16(buf) >> 4;
+ fsize = FFMIN3(fsize, len, MPA_MAX_CODED_FRAME_SIZE);
m = s->mp3decctx[fr];
assert (m != NULL);
- // Get header
- header = AV_RB32(start) | 0xfff00000;
+ header = (AV_RB32(buf) & 0x000fffff) | s->syncword; // patch header
- if (ff_mpa_check_header(header) < 0) { // Bad header, discard block
- *data_size = 0;
- return buf_size;
- }
+ if (ff_mpa_check_header(header) < 0) // Bad header, discard block
+ break;
- ff_mpegaudio_decode_header(m, header);
- mp_decode_frame(m, decoded_buf, start, fsize);
+ ff_mpegaudio_decode_header((MPADecodeHeader *)m, header);
+ out_size += mp_decode_frame(m, outptr, buf, fsize);
+ buf += fsize;
+ len -= fsize;
- n = MPA_FRAME_SIZE * m->nb_channels;
- out_size += n * sizeof(OUT_INT);
if(s->frames > 1) {
+ n = m->avctx->frame_size*m->nb_channels;
/* interleave output data */
- bp = out_samples + coff[fr];
+ bp = out_samples + s->coff[fr];
if(m->nb_channels == 1) {
for(j = 0; j < n; j++) {
*bp = decoded_buf[j];
- bp += off;
+ bp += avctx->channels;
}
} else {
for(j = 0; j < n; j++) {
bp[0] = decoded_buf[j++];
bp[1] = decoded_buf[j];
- bp += off;
+ bp += avctx->channels;
}
}
}
+ avctx->bit_rate += m->bit_rate;
}
/* update codec info */
avctx->sample_rate = s->mp3decctx[0]->sample_rate;
- avctx->frame_size= buf_size;
- avctx->bit_rate = 0;
- for (i = 0; i < s->frames; i++)
- avctx->bit_rate += s->mp3decctx[i]->bit_rate;
*data_size = out_size;
return buf_size;
}
#endif /* CONFIG_MP3ON4_DECODER */
-#ifdef CONFIG_MP2_DECODER
+#if CONFIG_MP1_DECODER
+AVCodec mp1_decoder =
+{
+ "mp1",
+ CODEC_TYPE_AUDIO,
+ CODEC_ID_MP1,
+ sizeof(MPADecodeContext),
+ decode_init,
+ NULL,
+ NULL,
+ decode_frame,
+ CODEC_CAP_PARSE_ONLY,
+ .flush= flush,
+ .long_name= NULL_IF_CONFIG_SMALL("MP1 (MPEG audio layer 1)"),
+};
+#endif
+#if CONFIG_MP2_DECODER
AVCodec mp2_decoder =
{
"mp2",
NULL,
decode_frame,
CODEC_CAP_PARSE_ONLY,
+ .flush= flush,
+ .long_name= NULL_IF_CONFIG_SMALL("MP2 (MPEG audio layer 2)"),
};
#endif
-#ifdef CONFIG_MP3_DECODER
+#if CONFIG_MP3_DECODER
AVCodec mp3_decoder =
{
"mp3",
decode_frame,
CODEC_CAP_PARSE_ONLY,
.flush= flush,
+ .long_name= NULL_IF_CONFIG_SMALL("MP3 (MPEG audio layer 3)"),
};
#endif
-#ifdef CONFIG_MP3ADU_DECODER
+#if CONFIG_MP3ADU_DECODER
AVCodec mp3adu_decoder =
{
"mp3adu",
decode_frame_adu,
CODEC_CAP_PARSE_ONLY,
.flush= flush,
+ .long_name= NULL_IF_CONFIG_SMALL("ADU (Application Data Unit) MP3 (MPEG audio layer 3)"),
};
#endif
-#ifdef CONFIG_MP3ON4_DECODER
+#if CONFIG_MP3ON4_DECODER
AVCodec mp3on4_decoder =
{
"mp3on4",
decode_close_mp3on4,
decode_frame_mp3on4,
.flush= flush,
+ .long_name= NULL_IF_CONFIG_SMALL("MP3onMP4"),
};
#endif
projects / frescor / ffmpeg.git / blobdiff