X-Git-Url: https://git.sesse.net/?a=blobdiff_plain;f=libavcodec%2Fmpegaudiodec.c;h=221941abcc4218110597d1f6c05b7b7aa4c5ccbe;hb=99e5a9d1ea2a61ac9429427431e5b9c2fefb76a5;hp=b7fe2e9219c418a113b22de371ad673eadec7754;hpb=62bb489b13c1f7967946bf538492dce0af1150fe;p=ffmpeg diff --git a/libavcodec/mpegaudiodec.c b/libavcodec/mpegaudiodec.c index b7fe2e9219c..221941abcc4 100644 --- a/libavcodec/mpegaudiodec.c +++ b/libavcodec/mpegaudiodec.c @@ -1,30 +1,31 @@ /* * MPEG Audio decoder - * Copyright (c) 2001, 2002 Fabrice Bellard. + * 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 */ /** - * @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" /* @@ -33,19 +34,11 @@ * - 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" #include "mathops.h" -#define FRAC_ONE (1 << FRAC_BITS) - -#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) @@ -55,48 +48,6 @@ /****************/ #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 - int dither_state; -} MPADecodeContext; - -/** - * 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 { @@ -118,16 +69,7 @@ typedef struct GranuleDef { 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); @@ -135,13 +77,25 @@ 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; -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 */ @@ -166,7 +120,69 @@ static const int32_t scale_factor_mult2[3][3] = { 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 */ @@ -234,7 +250,7 @@ static int pow_mult3[3] = { }; #endif -static void int_pow_init(void) +static av_cold void int_pow_init(void) { int i, a; @@ -292,17 +308,16 @@ 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; int i, j, k; -#if defined(USE_HIGHPRECISION) && defined(CONFIG_AUDIO_NONSHORT) - avctx->sample_fmt= SAMPLE_FMT_S32; -#else - avctx->sample_fmt= SAMPLE_FMT_S16; -#endif + s->avctx = avctx; + + avctx->sample_fmt= OUT_FMT; + s->error_recognition= avctx->error_recognition; if(avctx->antialias_algo != FF_AA_FLOAT) s->compute_antialias= compute_antialias_integer; @@ -310,6 +325,8 @@ 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; @@ -323,11 +340,11 @@ static int decode_init(AVCodecContext * avctx) for(i=0;i<15;i++) { int n, norm; n = i + 2; - norm = ((int64_t_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", + norm = ((INT64_C(1) << n) * FRAC_ONE) / ((1 << n) - 1); + 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], scale_factor_mult[i][1], @@ -337,10 +354,10 @@ static int decode_init(AVCodecContext * avctx) 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]; @@ -348,7 +365,6 @@ static int decode_init(AVCodecContext * avctx) memset(tmp_codes, 0, sizeof(tmp_codes)); xsize = h->xsize; - n = xsize * xsize; j = 0; for(x=0;xframe_count = 0; -#endif if (avctx->codec_id == CODEC_ID_MP3ADU) s->adu_mode = 1; return 0; @@ -745,11 +752,7 @@ 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 */ @@ -758,6 +761,8 @@ 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) @@ -765,65 +770,63 @@ 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; /* 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 @@ -843,34 +846,31 @@ 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)); @@ -880,9 +880,9 @@ 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++; @@ -892,9 +892,9 @@ 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; @@ -906,7 +906,7 @@ 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; @@ -1077,7 +1077,7 @@ 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; @@ -1105,123 +1105,6 @@ static void imdct36(int *out, int *buf, int *in, int *win) 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) { @@ -1289,28 +1172,6 @@ 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 */ @@ -1322,17 +1183,17 @@ static int mp_decode_layer2(MPADecodeContext *s) 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; @@ -1354,16 +1215,6 @@ static int mp_decode_layer2(MPADecodeContext *s) j += 1 << bit_alloc_bits; } -#ifdef DEBUG - { - for(ch=0;chnb_channels;ch++) { - for(i=0;inb_channels;ch++) { @@ -1404,20 +1255,6 @@ static int mp_decode_layer2(MPADecodeContext *s) } } -#ifdef DEBUG - for(ch=0;chnb_channels;ch++) { - for(i=0;igb, -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; @@ -1467,11 +1304,11 @@ static int mp_decode_layer2(MPADecodeContext *s) 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] = @@ -1588,6 +1425,19 @@ static inline int get_bitsz(GetBitContext *s, int n) 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) { @@ -1623,15 +1473,7 @@ static int huffman_decode(MPADecodeContext *s, GranuleDef *g, 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; @@ -1647,7 +1489,7 @@ static int huffman_decode(MPADecodeContext *s, GranuleDef *g, 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; @@ -1702,19 +1544,13 @@ 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); + 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; } // 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; @@ -1722,13 +1558,13 @@ static int huffman_decode(MPADecodeContext *s, GranuleDef *g, 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]; @@ -1740,17 +1576,22 @@ static int huffman_decode(MPADecodeContext *s, GranuleDef *g, } 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 && 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_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)); skip_bits_long(&s->gb, bits_left); + i= get_bits_count(&s->gb); + switch_buffer(s, &i, &end_pos, &end_pos2); + return 0; } @@ -1844,8 +1685,8 @@ static void compute_stereo(MPADecodeContext *s, v2 = is_tab[1][sf]; for(j=0;jnb_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 */ @@ -2174,30 +1977,21 @@ static int mp_decode_layer3(MPADecodeContext *s) 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(s->avctx, 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++) @@ -2205,56 +1999,19 @@ static int mp_decode_layer3(MPADecodeContext *s) /* 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); } } @@ -2263,21 +2020,30 @@ static int mp_decode_layer3(MPADecodeContext *s) 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;grnb_channels;ch++) { g = &granules[ch][gr]; + if(get_bits_count(&s->gb)<0){ + 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)); + 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); @@ -2288,7 +2054,7 @@ static int mp_decode_layer3(MPADecodeContext *s) /* 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; @@ -2332,15 +2098,6 @@ static int mp_decode_layer3(MPADecodeContext *s) } g->scale_factors[j++] = 0; } -#if defined(DEBUG) - { - dprintf("scfsi=%x gr=%d ch=%d scale_factors:\n", - g->scfsi, gr, ch); - for(i=0;iscale_factors[i]); - dprintf("\n"); - } -#endif } else { int tindex, tindex2, slen[4], sl, sf; @@ -2394,26 +2151,12 @@ 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("gr=%d ch=%d scale_factors:\n", - gr, ch); - for(i=0;i<40;i++) - dprintf(" %d", g->scale_factors[i]); - dprintf("\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; -#if defined(DEBUG) - sample_dump(0, g->sb_hybrid, 576); -#endif + huffman_decode(s, g, exponents, bits_pos + g->part2_3_length); } /* ch */ if (s->nb_channels == 2) @@ -2423,19 +2166,12 @@ static int mp_decode_layer3(MPADecodeContext *s) 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_bits_long(&s->gb, -get_bits_count(&s->gb)); return nb_granules * 18; } @@ -2449,17 +2185,20 @@ static int mp_decode_frame(MPADecodeContext *s, /* 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: + 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); @@ -2471,8 +2210,9 @@ static int mp_decode_frame(MPADecodeContext *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; } align_get_bits(&s->gb); @@ -2480,7 +2220,8 @@ static int mp_decode_frame(MPADecodeContext *s, 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); @@ -2489,17 +2230,7 @@ static int mp_decode_frame(MPADecodeContext *s, break; } -#if defined(DEBUG) - for(i=0;inb_channels;ch++) { - int j; - dprintf("%d-%d:", i, ch); - for(j=0;jsb_samples[ch][i][j] / FRAC_ONE); - dprintf("\n"); - } - } -#endif + /* apply the synthesis filter */ for(ch=0;chnb_channels;ch++) { samples_ptr = samples + ch; @@ -2511,79 +2242,72 @@ static int mp_decode_frame(MPADecodeContext *s, 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 = (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"); - goto retry; + av_log(avctx, AV_LOG_ERROR, "Header missing\n"); + return -1; } - if (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; } /* 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; - 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"); 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; } -#ifdef CONFIG_MP3ADU_DECODER +static void flush(AVCodecContext *avctx){ + MPADecodeContext *s = avctx->priv_data; + memset(s->synth_buf, 0, sizeof(s->synth_buf)); + s->last_buf_size= 0; +} + +#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; @@ -2602,21 +2326,21 @@ static int decode_frame_adu(AVCodecContext * avctx, 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((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; @@ -2629,12 +2353,24 @@ 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 @@ -2643,13 +2379,13 @@ 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)) { @@ -2657,18 +2393,24 @@ 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)); @@ -2686,13 +2428,14 @@ static int decode_init_mp3on4(AVCodecContext * avctx) 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_close_mp3on4(AVCodecContext * avctx) +static av_cold int decode_close_mp3on4(AVCodecContext * avctx) { MP3On4DecodeContext *s = avctx->priv_data; int i; @@ -2707,88 +2450,90 @@ 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]; - - 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 = (start[0] << 24) | (start[1] << 16) | (start[2] << 8) | start[3] | 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; - 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", @@ -2800,9 +2545,11 @@ 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", @@ -2814,9 +2561,11 @@ AVCodec mp3_decoder = NULL, 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", @@ -2828,9 +2577,11 @@ AVCodec mp3adu_decoder = NULL, 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", @@ -2841,6 +2592,7 @@ AVCodec mp3on4_decoder = NULL, decode_close_mp3on4, decode_frame_mp3on4, - 0 + .flush= flush, + .long_name= NULL_IF_CONFIG_SMALL("MP3onMP4"), }; #endif