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![Michael Niedermayer](/assets/img/avatar_default.png)
* qatar/master: (58 commits) amrnbdec: check frame size before decoding. cscd: use negative error values to indicate decode_init() failures. h264: prevent overreads in intra PCM decoding. FATE: do not decode audio in the nuv test. dxa: set audio stream time base using the sample rate psx-str: do not allow seeking by bytes asfdec: Do not set AVCodecContext.frame_size vqf: set packet parameters after av_new_packet() mpegaudiodec: use DSPUtil.butterflies_float(). FATE: add mp3 test for sample that exhibited false overreads fate: add cdxl test for bit line plane arrangement vmnc: return error on decode_init() failure. libvorbis: add/update error messages libvorbis: use AVFifoBuffer for output packet buffer libvorbis: remove unneeded e_o_s check libvorbis: check return values for functions that can return errors libvorbis: use float input instead of s16 libvorbis: do not flush libvorbis analysis if dsp state was not initialized libvorbis: use VBR by default, with default quality of 3 libvorbis: fix use of minrate/maxrate AVOptions ... Conflicts: Changelog doc/APIchanges libavcodec/avcodec.h libavcodec/dpxenc.c libavcodec/libvorbis.c libavcodec/vmnc.c libavformat/asfdec.c libavformat/id3v2enc.c libavformat/internal.h libavformat/mp3enc.c libavformat/utils.c libavformat/version.h libswscale/utils.c tests/fate/video.mak tests/ref/fate/nuv tests/ref/fate/prores-alpha tests/ref/lavf/ffm tests/ref/vsynth1/prores tests/ref/vsynth2/prores Merged-by: Michael Niedermayer <michaelni@gmx.at>
491 lines
14 KiB
C
491 lines
14 KiB
C
/*
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* MP3 muxer
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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 "avformat.h"
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#include "avio_internal.h"
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#include "id3v1.h"
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#include "id3v2.h"
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#include "rawenc.h"
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#include "libavutil/avstring.h"
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#include "libavcodec/mpegaudio.h"
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#include "libavcodec/mpegaudiodata.h"
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#include "libavcodec/mpegaudiodecheader.h"
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#include "libavutil/intreadwrite.h"
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#include "libavutil/opt.h"
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#include "libavcodec/mpegaudio.h"
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#include "libavcodec/mpegaudiodata.h"
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#include "libavcodec/mpegaudiodecheader.h"
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#include "libavformat/avio_internal.h"
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#include "libavutil/dict.h"
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#include "libavutil/avassert.h"
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static int id3v1_set_string(AVFormatContext *s, const char *key,
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uint8_t *buf, int buf_size)
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{
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AVDictionaryEntry *tag;
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if ((tag = av_dict_get(s->metadata, key, NULL, 0)))
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av_strlcpy(buf, tag->value, buf_size);
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return !!tag;
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}
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static int id3v1_create_tag(AVFormatContext *s, uint8_t *buf)
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{
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AVDictionaryEntry *tag;
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int i, count = 0;
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memset(buf, 0, ID3v1_TAG_SIZE); /* fail safe */
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buf[0] = 'T';
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buf[1] = 'A';
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buf[2] = 'G';
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/* we knowingly overspecify each tag length by one byte to compensate for the mandatory null byte added by av_strlcpy */
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count += id3v1_set_string(s, "TIT2", buf + 3, 30 + 1); //title
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count += id3v1_set_string(s, "TPE1", buf + 33, 30 + 1); //author|artist
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count += id3v1_set_string(s, "TALB", buf + 63, 30 + 1); //album
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count += id3v1_set_string(s, "TDRL", buf + 93, 4 + 1); //date
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count += id3v1_set_string(s, "comment", buf + 97, 30 + 1);
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if ((tag = av_dict_get(s->metadata, "TRCK", NULL, 0))) { //track
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buf[125] = 0;
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buf[126] = atoi(tag->value);
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count++;
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}
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buf[127] = 0xFF; /* default to unknown genre */
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if ((tag = av_dict_get(s->metadata, "TCON", NULL, 0))) { //genre
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for(i = 0; i <= ID3v1_GENRE_MAX; i++) {
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if (!av_strcasecmp(tag->value, ff_id3v1_genre_str[i])) {
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buf[127] = i;
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count++;
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break;
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}
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}
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}
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return count;
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}
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#define VBR_NUM_BAGS 400
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#define VBR_TOC_SIZE 100
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typedef struct MP3Context {
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const AVClass *class;
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ID3v2EncContext id3;
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int id3v2_version;
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int write_id3v1;
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int64_t frames_offset;
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int32_t frames;
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int32_t size;
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uint32_t want;
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uint32_t seen;
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uint32_t pos;
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uint64_t bag[VBR_NUM_BAGS];
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/* index of the audio stream */
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int audio_stream_idx;
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/* number of attached pictures we still need to write */
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int pics_to_write;
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/* audio packets are queued here until we get all the attached pictures */
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AVPacketList *queue, *queue_end;
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} MP3Context;
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static const int64_t xing_offtbl[2][2] = {{32, 17}, {17,9}};
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/*
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* Write an empty XING header and initialize respective data.
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*/
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static int mp3_write_xing(AVFormatContext *s)
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{
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MP3Context *mp3 = s->priv_data;
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AVCodecContext *codec = s->streams[mp3->audio_stream_idx]->codec;
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int bitrate_idx;
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int best_bitrate_idx = -1;
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int best_bitrate_error= INT_MAX;
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int64_t xing_offset;
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int32_t header, mask;
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MPADecodeHeader c;
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int srate_idx, i, channels;
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int needed;
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if (!s->pb->seekable)
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return 0;
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for (i = 0; i < FF_ARRAY_ELEMS(avpriv_mpa_freq_tab); i++)
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if (avpriv_mpa_freq_tab[i] == codec->sample_rate) {
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srate_idx = i;
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break;
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}
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if (i == FF_ARRAY_ELEMS(avpriv_mpa_freq_tab)) {
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av_log(s, AV_LOG_ERROR, "Unsupported sample rate.\n");
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return -1;
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}
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switch (codec->channels) {
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case 1: channels = MPA_MONO; break;
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case 2: channels = MPA_STEREO; break;
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default: av_log(s, AV_LOG_ERROR, "Unsupported number of channels.\n"); return -1;
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}
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/* dummy MPEG audio header */
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header = 0xff << 24; // sync
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header |= (0x7 << 5 | 0x3 << 3 | 0x1 << 1 | 0x1) << 16; // sync/mpeg-1/layer 3/no crc*/
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header |= (srate_idx << 2) << 8;
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header |= channels << 6;
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for (bitrate_idx=1; bitrate_idx<15; bitrate_idx++) {
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int error;
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avpriv_mpegaudio_decode_header(&c, header | (bitrate_idx << (4+8)));
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error= FFABS(c.bit_rate - codec->bit_rate);
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if(error < best_bitrate_error){
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best_bitrate_error= error;
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best_bitrate_idx = bitrate_idx;
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}
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}
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av_assert0(best_bitrate_idx >= 0);
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for (bitrate_idx= best_bitrate_idx;; bitrate_idx++) {
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if (15 == bitrate_idx)
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return -1;
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mask = bitrate_idx << (4+8);
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header |= mask;
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avpriv_mpegaudio_decode_header(&c, header);
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xing_offset=xing_offtbl[c.lsf == 1][c.nb_channels == 1];
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needed = 4 // header
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+ xing_offset
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+ 4 // xing tag
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+ 4 // frames/size/toc flags
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+ 4 // frames
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+ 4 // size
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+ VBR_TOC_SIZE; // toc
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if (needed <= c.frame_size)
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break;
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header &= ~mask;
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}
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avio_wb32(s->pb, header);
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ffio_fill(s->pb, 0, xing_offset);
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avio_wb32(s->pb, MKBETAG('X', 'i', 'n', 'g'));
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avio_wb32(s->pb, 0x01 | 0x02 | 0x04); // frames/size/toc
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mp3->frames_offset = avio_tell(s->pb);
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mp3->size = c.frame_size;
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mp3->want=1;
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mp3->seen=0;
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mp3->pos=0;
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avio_wb32(s->pb, 0); // frames
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avio_wb32(s->pb, 0); // size
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// toc
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for (i = 0; i < VBR_TOC_SIZE; ++i)
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avio_w8(s->pb, (uint8_t)(255 * i / VBR_TOC_SIZE));
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ffio_fill(s->pb, 0, c.frame_size - needed);
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avio_flush(s->pb);
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return 0;
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}
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/*
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* Add a frame to XING data.
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* Following lame's "VbrTag.c".
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*/
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static void mp3_xing_add_frame(AVFormatContext *s, AVPacket *pkt)
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{
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MP3Context *mp3 = s->priv_data;
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int i;
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++mp3->frames;
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mp3->size += pkt->size;
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if (mp3->want == ++mp3->seen) {
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mp3->bag[mp3->pos] = mp3->size;
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if (VBR_NUM_BAGS == ++mp3->pos) {
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/* shrink table to half size by throwing away each second bag. */
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for (i = 1; i < VBR_NUM_BAGS; i += 2)
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mp3->bag[i >> 1] = mp3->bag[i];
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/* double wanted amount per bag. */
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mp3->want <<= 1;
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/* adjust current position to half of table size. */
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mp3->pos >>= 1;
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}
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mp3->seen = 0;
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}
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}
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static void mp3_fix_xing(AVFormatContext *s)
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{
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MP3Context *mp3 = s->priv_data;
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int i;
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avio_flush(s->pb);
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avio_seek(s->pb, mp3->frames_offset, SEEK_SET);
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avio_wb32(s->pb, mp3->frames);
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avio_wb32(s->pb, mp3->size);
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avio_w8(s->pb, 0); // first toc entry has to be zero.
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for (i = 1; i < VBR_TOC_SIZE; ++i) {
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int j = i * mp3->pos / VBR_TOC_SIZE;
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int seek_point = 256LL * mp3->bag[j] / mp3->size;
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avio_w8(s->pb, FFMIN(seek_point, 255));
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}
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avio_flush(s->pb);
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avio_seek(s->pb, 0, SEEK_END);
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}
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static int mp3_write_packet_internal(AVFormatContext *s, AVPacket *pkt)
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{
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if (! pkt || ! pkt->data || pkt->size < 4)
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return ff_raw_write_packet(s, pkt);
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else {
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MP3Context *mp3 = s->priv_data;
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#ifdef FILTER_VBR_HEADERS
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MPADecodeHeader c;
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int base;
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ff_mpegaudio_decode_header(&c, AV_RB32(pkt->data));
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/* filter out XING and INFO headers. */
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base = 4 + xing_offtbl[c.lsf == 1][c.nb_channels == 1];
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if (base + 4 <= pkt->size) {
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uint32_t v = AV_RB32(pkt->data + base);
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if (MKBETAG('X','i','n','g') == v || MKBETAG('I','n','f','o') == v)
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return 0;
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}
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/* filter out VBRI headers. */
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base = 4 + 32;
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if (base + 4 <= pkt->size && MKBETAG('V','B','R','I') == AV_RB32(pkt->data + base))
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return 0;
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#endif
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if (mp3->frames_offset)
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mp3_xing_add_frame(s, pkt);
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return ff_raw_write_packet(s, pkt);
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}
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}
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static int mp3_queue_flush(AVFormatContext *s)
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{
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MP3Context *mp3 = s->priv_data;
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AVPacketList *pktl;
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int ret = 0, write = 1;
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ff_id3v2_finish(&mp3->id3, s->pb);
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mp3_write_xing(s);
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while ((pktl = mp3->queue)) {
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if (write && (ret = mp3_write_packet_internal(s, &pktl->pkt)) < 0)
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write = 0;
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av_free_packet(&pktl->pkt);
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mp3->queue = pktl->next;
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av_freep(&pktl);
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}
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mp3->queue_end = NULL;
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return ret;
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}
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static int mp2_write_trailer(struct AVFormatContext *s)
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{
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uint8_t buf[ID3v1_TAG_SIZE];
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MP3Context *mp3 = s->priv_data;
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if (mp3 && mp3->pics_to_write) {
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av_log(s, AV_LOG_WARNING, "No packets were sent for some of the "
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"attached pictures.\n");
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mp3_queue_flush(s);
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}
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/* write the id3v1 tag */
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if (mp3 && mp3->write_id3v1 && id3v1_create_tag(s, buf) > 0) {
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avio_write(s->pb, buf, ID3v1_TAG_SIZE);
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}
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/* write number of frames */
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if (mp3 && mp3->frames_offset) {
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avio_seek(s->pb, mp3->frames_offset, SEEK_SET);
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avio_wb32(s->pb, s->streams[mp3->audio_stream_idx]->nb_frames);
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avio_seek(s->pb, 0, SEEK_END);
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}
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avio_flush(s->pb);
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return 0;
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}
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#if CONFIG_MP2_MUXER
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AVOutputFormat ff_mp2_muxer = {
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.name = "mp2",
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.long_name = NULL_IF_CONFIG_SMALL("MPEG audio layer 2"),
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.mime_type = "audio/x-mpeg",
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.extensions = "mp2,m2a",
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.audio_codec = CODEC_ID_MP2,
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.video_codec = CODEC_ID_NONE,
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.write_packet = ff_raw_write_packet,
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.write_trailer = mp2_write_trailer,
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.flags = AVFMT_NOTIMESTAMPS,
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};
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#endif
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#if CONFIG_MP3_MUXER
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static const AVOption options[] = {
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{ "id3v2_version", "Select ID3v2 version to write. Currently 3 and 4 are supported.",
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offsetof(MP3Context, id3v2_version), AV_OPT_TYPE_INT, {.dbl = 4}, 3, 4, AV_OPT_FLAG_ENCODING_PARAM},
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{ "write_id3v1", "Enable ID3v1 writing. ID3v1 tags are written in UTF-8 which may not be supported by most software.",
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offsetof(MP3Context, write_id3v1), AV_OPT_TYPE_INT, {.dbl = 0}, 0, 1, AV_OPT_FLAG_ENCODING_PARAM},
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{ NULL },
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};
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static const AVClass mp3_muxer_class = {
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.class_name = "MP3 muxer",
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.item_name = av_default_item_name,
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.option = options,
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.version = LIBAVUTIL_VERSION_INT,
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};
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static int mp3_write_packet(AVFormatContext *s, AVPacket *pkt)
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{
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MP3Context *mp3 = s->priv_data;
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if (pkt->stream_index == mp3->audio_stream_idx) {
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if (mp3->pics_to_write) {
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/* buffer audio packets until we get all the pictures */
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AVPacketList *pktl = av_mallocz(sizeof(*pktl));
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if (!pktl)
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return AVERROR(ENOMEM);
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pktl->pkt = *pkt;
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pkt->destruct = NULL;
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if (mp3->queue_end)
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mp3->queue_end->next = pktl;
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else
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mp3->queue = pktl;
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mp3->queue_end = pktl;
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} else
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return mp3_write_packet_internal(s, pkt);
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} else {
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int ret;
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/* warn only once for each stream */
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if (s->streams[pkt->stream_index]->nb_frames == 1) {
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av_log(s, AV_LOG_WARNING, "Got more than one picture in stream %d,"
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" ignoring.\n", pkt->stream_index);
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}
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if (!mp3->pics_to_write || s->streams[pkt->stream_index]->nb_frames >= 1)
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return 0;
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if ((ret = ff_id3v2_write_apic(s, &mp3->id3, pkt)) < 0)
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return ret;
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mp3->pics_to_write--;
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/* flush the buffered audio packets */
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if (!mp3->pics_to_write &&
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(ret = mp3_queue_flush(s)) < 0)
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return ret;
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}
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return 0;
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}
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/**
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* Write an ID3v2 header at beginning of stream
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*/
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static int mp3_write_header(struct AVFormatContext *s)
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{
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MP3Context *mp3 = s->priv_data;
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int ret, i;
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/* check the streams -- we want exactly one audio and arbitrary number of
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* video (attached pictures) */
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mp3->audio_stream_idx = -1;
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for (i = 0; i < s->nb_streams; i++) {
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AVStream *st = s->streams[i];
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if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
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if (mp3->audio_stream_idx >= 0 || st->codec->codec_id != CODEC_ID_MP3) {
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av_log(s, AV_LOG_ERROR, "Invalid audio stream. Exactly one MP3 "
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"audio stream is required.\n");
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return AVERROR(EINVAL);
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}
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mp3->audio_stream_idx = i;
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} else if (st->codec->codec_type != AVMEDIA_TYPE_VIDEO) {
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av_log(s, AV_LOG_ERROR, "Only audio streams and pictures are allowed in MP3.\n");
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return AVERROR(EINVAL);
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}
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}
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if (mp3->audio_stream_idx < 0) {
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av_log(s, AV_LOG_ERROR, "No audio stream present.\n");
|
|
return AVERROR(EINVAL);
|
|
}
|
|
mp3->pics_to_write = s->nb_streams - 1;
|
|
|
|
ff_id3v2_start(&mp3->id3, s->pb, mp3->id3v2_version, ID3v2_DEFAULT_MAGIC);
|
|
ret = ff_id3v2_write_metadata(s, &mp3->id3);
|
|
if (ret < 0)
|
|
return ret;
|
|
|
|
if (!mp3->pics_to_write) {
|
|
ff_id3v2_finish(&mp3->id3, s->pb);
|
|
mp3_write_xing(s);
|
|
}
|
|
|
|
return 0;
|
|
}
|
|
|
|
static int mp3_write_trailer(AVFormatContext *s)
|
|
{
|
|
MP3Context *mp3 = s->priv_data;
|
|
int ret=mp2_write_trailer(s);
|
|
|
|
if (ret < 0)
|
|
return ret;
|
|
|
|
if (mp3->frames_offset)
|
|
mp3_fix_xing(s);
|
|
|
|
return 0;
|
|
}
|
|
|
|
AVOutputFormat ff_mp3_muxer = {
|
|
.name = "mp3",
|
|
.long_name = NULL_IF_CONFIG_SMALL("MPEG audio layer 3"),
|
|
.mime_type = "audio/x-mpeg",
|
|
.extensions = "mp3",
|
|
.priv_data_size = sizeof(MP3Context),
|
|
.audio_codec = CODEC_ID_MP3,
|
|
.video_codec = CODEC_ID_PNG,
|
|
.write_header = mp3_write_header,
|
|
.write_packet = mp3_write_packet,
|
|
.write_trailer = mp3_write_trailer,
|
|
.flags = AVFMT_NOTIMESTAMPS,
|
|
.priv_class = &mp3_muxer_class,
|
|
};
|
|
#endif
|