X-Git-Url: https://git.sesse.net/?a=blobdiff_plain;f=libavcodec%2Fmpegaudiodec.c;h=c3efd8657ab7757eea6e10da682e2e6563b2eab8;hb=8d872e8ac9d17dd004af60787cd69eac69ef8891;hp=3c4b757d6ad207fc268dff18dc2c1c5c79ea9479;hpb=7ae7300ee3bc830d9ece8c4c1fa27330b76c8218;p=ffmpeg diff --git a/libavcodec/mpegaudiodec.c b/libavcodec/mpegaudiodec.c index 3c4b757d6ad..c3efd8657ab 100644 --- a/libavcodec/mpegaudiodec.c +++ b/libavcodec/mpegaudiodec.c @@ -1,6 +1,6 @@ /* * MPEG Audio decoder - * Copyright (c) 2001, 2002 Fabrice Bellard. + * Copyright (c) 2001, 2002 Fabrice Bellard * * This file is part of FFmpeg. * @@ -20,13 +20,12 @@ */ /** - * @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" /* @@ -35,12 +34,6 @@ * - 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" @@ -56,38 +49,6 @@ #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; - int part2_3_length; - int big_values; - int global_gain; - int scalefac_compress; - uint8_t block_type; - uint8_t switch_point; - int table_select[3]; - int subblock_gain[3]; - uint8_t scalefac_scale; - uint8_t count1table_select; - int region_size[3]; /* number of huffman codes in each region */ - int preflag; - int short_start, long_end; /* long/short band indexes */ - uint8_t scale_factors[40]; - 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" @@ -96,15 +57,22 @@ static void compute_antialias_float(MPADecodeContext *s, GranuleDef *g); /* 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 */ -#define TABLE_4_3_SIZE (8191 + 16)*4 -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]; +#include "mpegaudio_tablegen.h" /* intensity stereo coef table */ static int32_t is_table[2][16]; static int32_t is_table_lsf[2][2][16]; @@ -127,7 +95,69 @@ static const int32_t scale_factor_mult2[3][3] = { SCALE_GEN(4.0 / 9.0), /* 9 steps */ }; -static DECLARE_ALIGNED_16(MPA_INT, window[512]); +DECLARE_ALIGNED_16(MPA_INT, ff_mpa_synth_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 */ @@ -195,7 +225,7 @@ static int pow_mult3[3] = { }; #endif -static void int_pow_init(void) +static av_cold void int_pow_init(void) { int i, a; @@ -253,7 +283,7 @@ static int int_pow(int i, int *exp_ptr) } #endif -static int decode_init(AVCodecContext * avctx) +static av_cold int decode_init(AVCodecContext * avctx) { MPADecodeContext *s = avctx->priv_data; static int init=0; @@ -261,12 +291,8 @@ static int decode_init(AVCodecContext * avctx) 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; @@ -274,6 +300,8 @@ static int decode_init(AVCodecContext * avctx) 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; @@ -288,9 +316,9 @@ static int decode_init(AVCodecContext * avctx) 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], @@ -298,13 +326,13 @@ static int decode_init(AVCodecContext * avctx) scale_factor_mult[i][2]); } - ff_mpa_synth_init(window); + ff_mpa_synth_init(ff_mpa_synth_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]; @@ -312,7 +340,6 @@ static int decode_init(AVCodecContext * avctx) memset(tmp_codes, 0, sizeof(tmp_codes)); xsize = h->xsize; - n = xsize * xsize; j = 0; for(x=0;x>4); - double f= pow(i&15, 4.0 / 3.0) * pow(2, (exponent-400)*0.25 + FRAC_BITS + 5); - expval_table[exponent][i&15]= llrint(f); - if((i&15)==1) - exp_table[exponent]= llrint(f); - } + mpegaudio_tableinit(); for(i=0;i<7;i++) { float f; @@ -408,8 +428,6 @@ static int decode_init(AVCodecContext * avctx) 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 */ @@ -437,7 +455,6 @@ static int decode_init(AVCodecContext * avctx) 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)); } } @@ -450,20 +467,9 @@ static int decode_init(AVCodecContext * avctx) } } -#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; @@ -703,11 +709,7 @@ static inline int round_sample(int *sum) int sum1; sum1 = (*sum) >> OUT_SHIFT; *sum &= (1< OUT_MAX) - sum1 = OUT_MAX; - return sum1; + return av_clip(sum1, OUT_MIN, OUT_MAX); } /* signed 16x16 -> 32 multiply add accumulate */ @@ -716,6 +718,8 @@ static inline int round_sample(int *sum) /* 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) @@ -723,58 +727,56 @@ static inline int round_sample(int64_t *sum) int sum1; sum1 = (int)((*sum) >> OUT_SHIFT); *sum &= (1< OUT_MAX) - sum1 = OUT_MAX; - return sum1; + return av_clip(sum1, OUT_MIN, OUT_MAX); } # 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; @@ -801,34 +803,31 @@ void ff_mpa_synth_filter(MPA_INT *synth_buf_ptr, int *synth_buf_offset, OUT_INT *samples, int incr, int32_t sb_samples[SBLIMIT]) { - int32_t tmp[32]; register MPA_INT *synth_buf; register const MPA_INT *w, *w2, *p; - int j, offset, v; + int j, offset; OUT_INT *samples2; #if FRAC_BITS <= 15 + int32_t tmp[32]; int sum, sum2; #else int64_t sum, sum2; #endif - dct32(tmp, sb_samples); - offset = *synth_buf_offset; synth_buf = synth_buf_ptr + offset; - for(j=0;j<32;j++) { - v = tmp[j]; #if FRAC_BITS <= 15 + dct32(tmp, sb_samples); + for(j=0;j<32;j++) { /* NOTE: can cause a loss in precision if very high amplitude sound */ - if (v > 32767) - v = 32767; - else if (v < -32768) - v = -32768; -#endif - synth_buf[j] = v; + synth_buf[j] = av_clip_int16(tmp[j]); } +#else + dct32(synth_buf, sb_samples); +#endif + /* copy to avoid wrap */ memcpy(synth_buf + 512, synth_buf, 32 * sizeof(MPA_INT)); @@ -838,9 +837,9 @@ void ff_mpa_synth_filter(MPA_INT *synth_buf_ptr, int *synth_buf_offset, 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++; @@ -850,9 +849,9 @@ void ff_mpa_synth_filter(MPA_INT *synth_buf_ptr, int *synth_buf_offset, 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; @@ -864,7 +863,7 @@ void ff_mpa_synth_filter(MPA_INT *synth_buf_ptr, int *synth_buf_offset, } p = synth_buf + 32; - SUM8(sum, -=, w + 32, p); + SUM8(MLSS, sum, w + 32, p); *samples = round_sample(&sum); *dither_state= sum; @@ -1035,7 +1034,7 @@ static void imdct36(int *out, int *buf, int *in, int *win) 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; @@ -1173,16 +1172,6 @@ static int mp_decode_layer2(MPADecodeContext *s) j += 1 << bit_alloc_bits; } -#ifdef DEBUG - { - for(ch=0;chnb_channels;ch++) { - for(i=0;iavctx, " %d", bit_alloc[ch][i]); - dprintf(s->avctx, "\n"); - } - } -#endif - /* scale codes */ for(i=0;inb_channels;ch++) { @@ -1223,20 +1212,6 @@ static int mp_decode_layer2(MPADecodeContext *s) } } -#ifdef DEBUG - for(ch=0;chnb_channels;ch++) { - for(i=0;iavctx, " %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) { @@ -1526,8 +1501,8 @@ static int huffman_decode(MPADecodeContext *s, GranuleDef *g, 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; } @@ -1561,11 +1536,11 @@ static int huffman_decode(MPADecodeContext *s, GranuleDef *g, /* 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)); @@ -1667,8 +1642,8 @@ static void compute_stereo(MPADecodeContext *s, v2 = is_tab[1][sf]; for(j=0;jgb, 5); nb_granules = 2; for(ch=0;chnb_channels;ch++) { - granules[ch][0].scfsi = 0; /* all scale factors are transmitted */ - granules[ch][1].scfsi = get_bits(&s->gb, 4); + s->granules[ch][0].scfsi = 0;/* all scale factors are transmitted */ + s->granules[ch][1].scfsi = get_bits(&s->gb, 4); } } for(gr=0;grnb_channels;ch++) { dprintf(s->avctx, "gr=%d ch=%d: side_info\n", gr, ch); - g = &granules[ch][gr]; + g = &s->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){ @@ -2006,7 +1938,7 @@ 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); @@ -2014,20 +1946,9 @@ static int mp_decode_layer3(MPADecodeContext *s) 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++) @@ -2037,47 +1958,10 @@ static int mp_decode_layer3(MPADecodeContext *s) 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) @@ -2104,9 +1988,9 @@ static int mp_decode_layer3(MPADecodeContext *s) for(gr=0;grnb_channels;ch++) { - g = &granules[ch][gr]; + g = &s->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_DEBUG, "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)); @@ -2148,7 +2032,7 @@ static int mp_decode_layer3(MPADecodeContext *s) g->scale_factors[j++] = 0; } } else { - sc = granules[ch][0].scale_factors; + sc = s->granules[ch][0].scale_factors; j = 0; for(k=0;k<4;k++) { n = (k == 0 ? 6 : 5); @@ -2171,15 +2055,6 @@ static int mp_decode_layer3(MPADecodeContext *s) } 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;iavctx, " %d", g->scale_factors[i]); - dprintf(s->avctx, "\n"); - } -#endif } else { int tindex, tindex2, slen[4], sl, sf; @@ -2233,44 +2108,23 @@ static int mp_decode_layer3(MPADecodeContext *s) /* 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) - compute_stereo(s, &granules[0][gr], &granules[1][gr]); + compute_stereo(s, &s->granules[0][gr], &s->granules[1][gr]); for(ch=0;chnb_channels;ch++) { - g = &granules[ch][gr]; + g = &s->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) @@ -2293,34 +2147,38 @@ static int mp_decode_frame(MPADecodeContext *s, 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); s->last_buf_size=0; if(s->in_gb.buffer){ align_get_bits(&s->gb); - i= (s->gb.size_in_bits - get_bits_count(&s->gb))>>3; + i= get_bits_left(&s->gb)>>3; if(i >= 0 && i <= BACKSTEP_SIZE){ 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; } align_get_bits(&s->gb); assert((get_bits_count(&s->gb) & 7) == 0); - i= (s->gb.size_in_bits - get_bits_count(&s->gb))>>3; + i= get_bits_left(&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); @@ -2329,56 +2187,43 @@ static int mp_decode_frame(MPADecodeContext *s, break; } -#if defined(DEBUG) - for(i=0;inb_channels;ch++) { - int j; - dprintf(s->avctx, "%d-%d:", i, ch); - for(j=0;javctx, " %0.6f", (double)s->sb_samples[ch][i][j] / FRAC_ONE); - dprintf(s->avctx, "\n"); - } - } -#endif + /* apply the synthesis filter */ for(ch=0;chnb_channels;ch++) { samples_ptr = samples + ch; for(i=0;isynth_buf[ch], &(s->synth_buf_offset[ch]), - window, &s->dither_state, + ff_mpa_synth_window, &s->dither_state, samples_ptr, s->nb_channels, s->sb_samples[ch][i]); 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; OUT_INT *out_samples = data; -retry: if(buf_size < HEADER_SIZE) return -1; header = AV_RB32(buf); if(ff_mpa_check_header(header) < 0){ - buf++; -// buf_size--; - av_log(avctx, AV_LOG_ERROR, "Header missing skipping one byte.\n"); - goto retry; + av_log(avctx, AV_LOG_ERROR, "Header missing\n"); + return -1; } - 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; @@ -2387,20 +2232,10 @@ retry: 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(*data_size < 1152*avctx->channels*sizeof(OUT_INT)) + return -1; + *data_size = 0; if(s->frame_size<=0 || s->frame_size > buf_size){ av_log(avctx, AV_LOG_ERROR, "incomplete frame\n"); @@ -2423,14 +2258,17 @@ retry: 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; @@ -2456,14 +2294,14 @@ static int decode_frame_adu(AVCodecContext * avctx, 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; @@ -2476,12 +2314,24 @@ static int decode_frame_adu(AVCodecContext * avctx, } #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 @@ -2490,13 +2340,13 @@ static int chan_offset[9][5] = { {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)) { @@ -2504,18 +2354,24 @@ static int decode_init_mp3on4(AVCodecContext * avctx) 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)); @@ -2540,7 +2396,7 @@ static int decode_init_mp3on4(AVCodecContext * avctx) } -static int decode_close_mp3on4(AVCodecContext * avctx) +static av_cold int decode_close_mp3on4(AVCodecContext * avctx) { MP3On4DecodeContext *s = avctx->priv_data; int i; @@ -2555,88 +2411,93 @@ static int decode_close_mp3on4(AVCodecContext * avctx) 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]; + int fr, j, n; - len = buf_size; + if(*data_size < MPA_FRAME_SIZE * MPA_MAX_CHANNELS * s->frames * sizeof(OUT_INT)) + return -1; + *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", @@ -2648,9 +2509,11 @@ AVCodec mp2_decoder = 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", @@ -2663,9 +2526,10 @@ AVCodec mp3_decoder = 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", @@ -2678,9 +2542,10 @@ AVCodec mp3adu_decoder = 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", @@ -2692,5 +2557,6 @@ AVCodec mp3on4_decoder = decode_close_mp3on4, decode_frame_mp3on4, .flush= flush, + .long_name= NULL_IF_CONFIG_SMALL("MP3onMP4"), }; #endif