/* * MP3 demuxer * Copyright (c) 2003 Fabrice Bellard * * 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 2.1 of the License, or (at your option) any later version. * * 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 FFmpeg; if not, write to the Free Software * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA */ #include "config.h" #include "libavutil/opt.h" #include "libavutil/avstring.h" #include "libavutil/intreadwrite.h" #include "libavutil/dict.h" #include "libavutil/mathematics.h" #include "avformat.h" #include "internal.h" #include "id3v2.h" #include "id3v1.h" #include "apetag.h" #include "libavcodec/mpegaudiodecheader.h" #define MP3_RESYNC_TOLERANCE_BYTES 65536 #define XING_FLAG_FRAMES 0x01 #define XING_FLAG_SIZE 0x02 #define XING_FLAG_TOC 0x04 #define XING_TOC_COUNT 100 typedef struct { AVClass *class; int64_t filesize; int64_t header_filesize; int xing_toc; int start_pad; int end_pad; int usetoc; int is_cbr; } MP3DecContext; /* mp3 read */ static int mp3_read_probe(AVProbeData *p) { int max_frames, first_frames = 0; int fsize, frames, sample_rate; uint32_t header; const uint8_t *buf, *buf0, *buf2, *end; AVCodecContext avctx; buf0 = p->buf; end = p->buf + p->buf_size - sizeof(uint32_t); while(buf0 < end && !*buf0) buf0++; max_frames = 0; buf = buf0; for(; buf < end; buf= buf2+1) { buf2 = buf; if(ff_mpa_check_header(AV_RB32(buf2))) continue; for(frames = 0; buf2 < end; frames++) { header = AV_RB32(buf2); fsize = avpriv_mpa_decode_header(&avctx, header, &sample_rate, &sample_rate, &sample_rate, &sample_rate); if(fsize < 0) break; buf2 += fsize; } max_frames = FFMAX(max_frames, frames); if(buf == buf0) first_frames= frames; } // keep this in sync with ac3 probe, both need to avoid // issues with MPEG-files! if (first_frames>=4) return AVPROBE_SCORE_EXTENSION + 1; else if(max_frames>200)return AVPROBE_SCORE_EXTENSION; else if(max_frames>=4) return AVPROBE_SCORE_EXTENSION / 2; else if(ff_id3v2_match(buf0, ID3v2_DEFAULT_MAGIC) && 2*ff_id3v2_tag_len(buf0) >= p->buf_size) return p->buf_size < PROBE_BUF_MAX ? AVPROBE_SCORE_EXTENSION / 4 : AVPROBE_SCORE_EXTENSION - 2; else if(max_frames>=1) return 1; else return 0; //mpegps_mp3_unrecognized_format.mpg has max_frames=3 } static void read_xing_toc(AVFormatContext *s, int64_t filesize, int64_t duration) { int i; MP3DecContext *mp3 = s->priv_data; int fill_index = mp3->usetoc && duration > 0; if (!filesize && !(filesize = avio_size(s->pb))) { av_log(s, AV_LOG_WARNING, "Cannot determine file size, skipping TOC table.\n"); fill_index = 0; } for (i = 0; i < XING_TOC_COUNT; i++) { uint8_t b = avio_r8(s->pb); if (fill_index) av_add_index_entry(s->streams[0], av_rescale(b, filesize, 256), av_rescale(i, duration, XING_TOC_COUNT), 0, 0, AVINDEX_KEYFRAME); } if (fill_index) mp3->xing_toc = 1; } /** * Try to find Xing/Info/VBRI tags and compute duration from info therein */ static int mp3_parse_vbr_tags(AVFormatContext *s, AVStream *st, int64_t base) { MP3DecContext *mp3 = s->priv_data; uint32_t v, spf; unsigned frames = 0; /* Total number of frames in file */ unsigned size = 0; /* Total number of bytes in the stream */ static const int64_t xing_offtbl[2][2] = {{32, 17}, {17,9}}; MPADecodeHeader c; int vbrtag_size = 0; int is_cbr; AVDictionaryEntry *de; uint64_t duration = 0; v = avio_rb32(s->pb); if(ff_mpa_check_header(v) < 0) return -1; if (avpriv_mpegaudio_decode_header(&c, v) == 0) vbrtag_size = c.frame_size; if(c.layer != 3) return -1; mp3->start_pad = 0; mp3->end_pad = 0; spf = c.lsf ? 576 : 1152; /* Samples per frame, layer 3 */ /* Check for Xing / Info tag */ avio_skip(s->pb, xing_offtbl[c.lsf == 1][c.nb_channels == 1]); v = avio_rb32(s->pb); is_cbr = v == MKBETAG('I', 'n', 'f', 'o'); if (v == MKBETAG('X', 'i', 'n', 'g') || is_cbr) { v = avio_rb32(s->pb); if(v & XING_FLAG_FRAMES) frames = avio_rb32(s->pb); if(v & XING_FLAG_SIZE) size = avio_rb32(s->pb); if (v & XING_FLAG_TOC) read_xing_toc(s, size, av_rescale_q(frames, (AVRational){spf, c.sample_rate}, st->time_base)); if(v & 8) avio_skip(s->pb, 4); v = avio_rb32(s->pb); if(v == MKBETAG('L', 'A', 'M', 'E') || v == MKBETAG('L', 'a', 'v', 'f')) { avio_skip(s->pb, 21-4); v= avio_rb24(s->pb); mp3->start_pad = v>>12; mp3-> end_pad = v&4095; if (mp3->end_pad >= 528 + 1) { mp3->end_pad = mp3->end_pad - (528 + 1); st->skip_samples = mp3->start_pad + 528 + 1; } else { st->skip_samples = mp3->start_pad; } if (!st->start_time) st->start_time = av_rescale_q(st->skip_samples, (AVRational){1, c.sample_rate}, st->time_base); av_log(s, AV_LOG_DEBUG, "pad %d %d\n", mp3->start_pad, mp3-> end_pad); } } /* Check for VBRI tag (always 32 bytes after end of mpegaudio header) */ avio_seek(s->pb, base + 4 + 32, SEEK_SET); v = avio_rb32(s->pb); if(v == MKBETAG('V', 'B', 'R', 'I')) { /* Check tag version */ if(avio_rb16(s->pb) == 1) { /* skip delay and quality */ avio_skip(s->pb, 4); size = avio_rb32(s->pb); frames = avio_rb32(s->pb); } } if (!frames) vbrtag_size = 0; if (s->metadata && (de = av_dict_get(s->metadata, "iTunSMPB", NULL, 0))) { uint32_t zero, start_pad, end_pad; uint64_t last_eight_frames_offset; if (sscanf(de->value, "%x %x %x %llx %x %llx", &zero, &start_pad, &end_pad, &duration, &zero, &last_eight_frames_offset) < 6) { duration = 0; } else { mp3->start_pad = start_pad; mp3->end_pad = end_pad; if (end_pad >= 528 + 1) mp3->end_pad = end_pad - (528 + 1); st->skip_samples = mp3->start_pad + 528 + 1; av_log(s, AV_LOG_DEBUG, "pad %d %d\n", mp3->start_pad, mp3->end_pad); if (s->pb->seekable) { int i; size = last_eight_frames_offset; avio_seek(s->pb, base + vbrtag_size + last_eight_frames_offset, SEEK_SET); for (i = 0; i < 8; ++i) { v = avio_rb32(s->pb); if (ff_mpa_check_header(v) < 0) return -1; if (avpriv_mpegaudio_decode_header(&c, v) != 0) break; size += c.frame_size; avio_skip(s->pb, c.frame_size - 4); } } } } if(!frames && !size && !duration) return -1; /* Skip the vbr tag frame */ avio_seek(s->pb, base + vbrtag_size, SEEK_SET); if (duration) st->duration = av_rescale_q(duration, (AVRational){1, c.sample_rate}, st->time_base); else if(frames) st->duration = av_rescale_q(frames, (AVRational){spf, c.sample_rate}, st->time_base) - av_rescale_q(mp3->end_pad, (AVRational){1, c.sample_rate}, st->time_base); if (size) { if (duration) st->codec->bit_rate = av_rescale(size, 8 * c.sample_rate, duration); else if (frames) st->codec->bit_rate = av_rescale(size, 8 * c.sample_rate, frames * (int64_t)spf); } mp3->is_cbr = is_cbr; mp3->header_filesize = size; return 0; } static int mp3_read_header(AVFormatContext *s) { MP3DecContext *mp3 = s->priv_data; AVStream *st; int64_t off; st = avformat_new_stream(s, NULL); if (!st) return AVERROR(ENOMEM); st->codec->codec_type = AVMEDIA_TYPE_AUDIO; st->codec->codec_id = AV_CODEC_ID_MP3; st->need_parsing = AVSTREAM_PARSE_FULL_RAW; st->start_time = 0; // lcm of all mp3 sample rates avpriv_set_pts_info(st, 64, 1, 14112000); s->pb->maxsize = -1; off = avio_tell(s->pb); if (!av_dict_get(s->metadata, "", NULL, AV_DICT_IGNORE_SUFFIX)) ff_id3v1_read(s); if(s->pb->seekable) mp3->filesize = avio_size(s->pb); if (mp3_parse_vbr_tags(s, st, off) < 0 && s->pb->seekable) { uint64_t duration = 0; uint8_t buf[8]; int sample_rate = 0; int retry_count; /* Time for a full parse! */ avio_seek(s->pb, mp3->filesize - 128, SEEK_SET); avio_read(s->pb, buf, 3); if (buf[0] == 'T' && buf[1] == 'A' && buf[2] == 'G') mp3->filesize -= 128; avio_seek(s->pb, mp3->filesize - APE_TAG_FOOTER_BYTES, SEEK_SET); avio_read(s->pb, buf, 8); if (memcmp(buf, APE_TAG_PREAMBLE, 8) == 0) { avio_seek(s->pb, 4, SEEK_CUR); mp3->filesize -= avio_rl32(s->pb) + APE_TAG_FOOTER_BYTES; } avio_seek(s->pb, off, SEEK_SET); retry_count = MP3_RESYNC_TOLERANCE_BYTES; while (avio_tell(s->pb) < mp3->filesize) { MPADecodeHeader c; uint32_t v, spf; v = avio_rb32(s->pb); if(ff_mpa_check_header(v) < 0) { if (--retry_count) { avio_seek(s->pb, -3, SEEK_CUR); continue; } else break; } retry_count = MP3_RESYNC_TOLERANCE_BYTES; if (avpriv_mpegaudio_decode_header(&c, v) != 0) break; if (!sample_rate) sample_rate = c.sample_rate; spf = c.lsf ? 576 : 1152; /* Samples per frame, layer 3 */ duration += spf; avio_skip(s->pb, c.frame_size - 4); } avio_seek(s->pb, off, SEEK_SET); st->duration = duration && sample_rate ? av_rescale_q(duration, (AVRational){1, sample_rate}, st->time_base) : 0; } /* the parameters will be extracted from the compressed bitstream */ return 0; } #define MP3_PACKET_SIZE 1024 static int mp3_read_packet(AVFormatContext *s, AVPacket *pkt) { MP3DecContext *mp3 = s->priv_data; int ret, size; int64_t pos; size= MP3_PACKET_SIZE; pos = avio_tell(s->pb); if(mp3->filesize > ID3v1_TAG_SIZE && pos < mp3->filesize) size= FFMIN(size, mp3->filesize - pos); ret= av_get_packet(s->pb, pkt, size); if (ret <= 0) { if(ret<0) return ret; return AVERROR_EOF; } pkt->flags &= ~AV_PKT_FLAG_CORRUPT; pkt->stream_index = 0; if (ret >= ID3v1_TAG_SIZE && memcmp(&pkt->data[ret - ID3v1_TAG_SIZE], "TAG", 3) == 0) ret -= ID3v1_TAG_SIZE; /* note: we need to modify the packet size here to handle the last packet */ pkt->size = ret; return ret; } static int check(AVFormatContext *s, int64_t pos) { int64_t ret = avio_seek(s->pb, pos, SEEK_SET); unsigned header; MPADecodeHeader sd; if (ret < 0) return ret; header = avio_rb32(s->pb); if (ff_mpa_check_header(header) < 0) return -1; if (avpriv_mpegaudio_decode_header(&sd, header) == 1) return -1; return sd.frame_size; } static int mp3_seek(AVFormatContext *s, int stream_index, int64_t timestamp, int flags) { MP3DecContext *mp3 = s->priv_data; AVIndexEntry *ie, ie1; AVStream *st = s->streams[0]; int64_t ret = av_index_search_timestamp(st, timestamp, flags); int i, j; int dir = (flags&AVSEEK_FLAG_BACKWARD) ? -1 : 1; if (mp3->is_cbr && st->duration > 0 && mp3->header_filesize > s->data_offset) { int64_t filesize = avio_size(s->pb); int64_t duration; if (filesize <= s->data_offset) filesize = mp3->header_filesize; filesize -= s->data_offset; duration = av_rescale(st->duration, filesize, mp3->header_filesize - s->data_offset); ie = &ie1; timestamp = av_clip64(timestamp, 0, duration); ie->timestamp = timestamp; ie->pos = av_rescale(timestamp, filesize, duration) + s->data_offset; } else if (mp3->xing_toc) { if (ret < 0) return ret; ie = &st->index_entries[ret]; } else { st->skip_samples = timestamp <= 0 ? mp3->start_pad + 528 + 1 : 0; return -1; } if (dir < 0) avio_seek(s->pb, FFMAX(ie->pos - 4096, 0), SEEK_SET); ret = avio_seek(s->pb, ie->pos, SEEK_SET); if (ret < 0) return ret; #define MIN_VALID 3 for(i=0; i<4096; i++) { int64_t pos = ie->pos + i*dir; for(j=0; jpb, ie->pos + i*dir, SEEK_SET); if (ret < 0) return ret; ff_update_cur_dts(s, st, ie->timestamp); st->skip_samples = ie->timestamp <= 0 ? mp3->start_pad + 528 + 1 : 0; return 0; } static const AVOption options[] = { { "usetoc", "use table of contents", offsetof(MP3DecContext, usetoc), AV_OPT_TYPE_INT, {.i64 = -1}, -1, 1, AV_OPT_FLAG_DECODING_PARAM}, { NULL }, }; static const AVClass demuxer_class = { .class_name = "mp3", .item_name = av_default_item_name, .option = options, .version = LIBAVUTIL_VERSION_INT, .category = AV_CLASS_CATEGORY_DEMUXER, }; AVInputFormat ff_mp3_demuxer = { .name = "mp3", .long_name = NULL_IF_CONFIG_SMALL("MP2/3 (MPEG audio layer 2/3)"), .read_probe = mp3_read_probe, .read_header = mp3_read_header, .read_packet = mp3_read_packet, .read_seek = mp3_seek, .priv_data_size = sizeof(MP3DecContext), .flags = AVFMT_GENERIC_INDEX, .extensions = "mp2,mp3,m2a,mpa", /* XXX: use probe */ .priv_class = &demuxer_class, };