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https://github.com/FFmpeg/FFmpeg.git
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9d76cf0b18
* qatar/master: rtpdec: Templatize the code for different g726 bitrate variants rv40: move loop filter to rv34dsp context lavf: make av_set_pts_info private. rtpdec: Add support for G726 audio rtpdec: Add an init function that can do custom codec context initialization avconv: make copy_tb on by default. matroskadec: don't set codec timebase. rmdec: don't set codec timebase. avconv: compute next_pts from input packet duration when possible. lavf: estimate frame duration from r_frame_rate. avconv: update InputStream.pts in the streamcopy case. Conflicts: avconv.c libavdevice/alsa-audio-dec.c libavdevice/bktr.c libavdevice/fbdev.c libavdevice/libdc1394.c libavdevice/oss_audio.c libavdevice/v4l.c libavdevice/v4l2.c libavdevice/vfwcap.c libavdevice/x11grab.c libavformat/au.c libavformat/eacdata.c libavformat/flvdec.c libavformat/mpegts.c libavformat/mxfenc.c libavformat/rtpdec_g726.c libavformat/wtv.c libavformat/xmv.c Merged-by: Michael Niedermayer <michaelni@gmx.at>
202 lines
5.7 KiB
C
202 lines
5.7 KiB
C
/*
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* MP3 demuxer
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* Copyright (c) 2003 Fabrice Bellard
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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#include "libavutil/avstring.h"
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#include "libavutil/intreadwrite.h"
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#include "libavutil/dict.h"
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#include "libavutil/mathematics.h"
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#include "avformat.h"
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#include "internal.h"
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#include "id3v2.h"
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#include "id3v1.h"
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#include "libavcodec/mpegaudiodecheader.h"
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/* mp3 read */
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static int mp3_read_probe(AVProbeData *p)
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{
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int max_frames, first_frames = 0;
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int fsize, frames, sample_rate;
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uint32_t header;
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uint8_t *buf, *buf0, *buf2, *end;
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AVCodecContext avctx;
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buf0 = p->buf;
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end = p->buf + p->buf_size - sizeof(uint32_t);
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while(buf0 < end && !*buf0)
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buf0++;
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max_frames = 0;
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buf = buf0;
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for(; buf < end; buf= buf2+1) {
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buf2 = buf;
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for(frames = 0; buf2 < end; frames++) {
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header = AV_RB32(buf2);
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fsize = avpriv_mpa_decode_header(&avctx, header, &sample_rate, &sample_rate, &sample_rate, &sample_rate);
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if(fsize < 0)
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break;
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buf2 += fsize;
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}
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max_frames = FFMAX(max_frames, frames);
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if(buf == buf0)
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first_frames= frames;
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}
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// keep this in sync with ac3 probe, both need to avoid
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// issues with MPEG-files!
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if (first_frames>=4) return AVPROBE_SCORE_MAX/2+1;
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else if(max_frames>200)return AVPROBE_SCORE_MAX/2;
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else if(max_frames>=4) return AVPROBE_SCORE_MAX/4;
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else if(max_frames>=1) return 1;
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else return 0;
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//mpegps_mp3_unrecognized_format.mpg has max_frames=3
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}
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/**
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* Try to find Xing/Info/VBRI tags and compute duration from info therein
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*/
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static int mp3_parse_vbr_tags(AVFormatContext *s, AVStream *st, int64_t base)
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{
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uint32_t v, spf;
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unsigned frames = 0; /* Total number of frames in file */
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unsigned size = 0; /* Total number of bytes in the stream */
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const int64_t xing_offtbl[2][2] = {{32, 17}, {17,9}};
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MPADecodeHeader c;
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int vbrtag_size = 0;
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v = avio_rb32(s->pb);
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if(ff_mpa_check_header(v) < 0)
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return -1;
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if (avpriv_mpegaudio_decode_header(&c, v) == 0)
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vbrtag_size = c.frame_size;
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if(c.layer != 3)
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return -1;
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/* Check for Xing / Info tag */
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avio_skip(s->pb, xing_offtbl[c.lsf == 1][c.nb_channels == 1]);
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v = avio_rb32(s->pb);
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if(v == MKBETAG('X', 'i', 'n', 'g') || v == MKBETAG('I', 'n', 'f', 'o')) {
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v = avio_rb32(s->pb);
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if(v & 0x1)
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frames = avio_rb32(s->pb);
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if(v & 0x2)
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size = avio_rb32(s->pb);
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}
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/* Check for VBRI tag (always 32 bytes after end of mpegaudio header) */
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avio_seek(s->pb, base + 4 + 32, SEEK_SET);
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v = avio_rb32(s->pb);
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if(v == MKBETAG('V', 'B', 'R', 'I')) {
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/* Check tag version */
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if(avio_rb16(s->pb) == 1) {
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/* skip delay and quality */
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avio_skip(s->pb, 4);
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size = avio_rb32(s->pb);
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frames = avio_rb32(s->pb);
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}
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}
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if(!frames && !size)
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return -1;
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/* Skip the vbr tag frame */
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avio_seek(s->pb, base + vbrtag_size, SEEK_SET);
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spf = c.lsf ? 576 : 1152; /* Samples per frame, layer 3 */
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if(frames)
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st->duration = av_rescale_q(frames, (AVRational){spf, c.sample_rate},
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st->time_base);
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if(size && frames)
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st->codec->bit_rate = av_rescale(size, 8 * c.sample_rate, frames * (int64_t)spf);
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return 0;
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}
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static int mp3_read_header(AVFormatContext *s,
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AVFormatParameters *ap)
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{
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AVStream *st;
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int64_t off;
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st = avformat_new_stream(s, NULL);
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if (!st)
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return AVERROR(ENOMEM);
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st->codec->codec_type = AVMEDIA_TYPE_AUDIO;
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st->codec->codec_id = CODEC_ID_MP3;
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st->need_parsing = AVSTREAM_PARSE_FULL;
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st->start_time = 0;
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// lcm of all mp3 sample rates
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avpriv_set_pts_info(st, 64, 1, 14112000);
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off = avio_tell(s->pb);
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if (!av_dict_get(s->metadata, "", NULL, AV_DICT_IGNORE_SUFFIX))
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ff_id3v1_read(s);
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if (mp3_parse_vbr_tags(s, st, off) < 0)
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avio_seek(s->pb, off, SEEK_SET);
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/* the parameters will be extracted from the compressed bitstream */
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return 0;
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}
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#define MP3_PACKET_SIZE 1024
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static int mp3_read_packet(AVFormatContext *s, AVPacket *pkt)
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{
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int ret, size;
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// AVStream *st = s->streams[0];
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size= MP3_PACKET_SIZE;
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ret= av_get_packet(s->pb, pkt, size);
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pkt->stream_index = 0;
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if (ret <= 0) {
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if(ret<0)
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return ret;
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return AVERROR_EOF;
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}
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if (ret > ID3v1_TAG_SIZE &&
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memcmp(&pkt->data[ret - ID3v1_TAG_SIZE], "TAG", 3) == 0)
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ret -= ID3v1_TAG_SIZE;
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/* note: we need to modify the packet size here to handle the last
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packet */
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pkt->size = ret;
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return ret;
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}
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AVInputFormat ff_mp3_demuxer = {
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.name = "mp3",
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.long_name = NULL_IF_CONFIG_SMALL("MPEG audio layer 2/3"),
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.read_probe = mp3_read_probe,
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.read_header = mp3_read_header,
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.read_packet = mp3_read_packet,
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.flags= AVFMT_GENERIC_INDEX,
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.extensions = "mp2,mp3,m2a", /* XXX: use probe */
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};
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