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FFmpeg/libavformat/mp3enc.c

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/*
* MP3 muxer
* 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 "avformat.h"
#include "avio_internal.h"
#include "id3v1.h"
#include "id3v2.h"
#include "rawenc.h"
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#include "libavutil/avstring.h"
#include "libavcodec/mpegaudio.h"
#include "libavcodec/mpegaudiodata.h"
#include "libavcodec/mpegaudiodecheader.h"
#include "libavutil/intreadwrite.h"
#include "libavutil/opt.h"
#include "libavutil/dict.h"
#include "libavutil/avassert.h"
#include "libavutil/crc.h"
#include "libavutil/mathematics.h"
#include "libavutil/replaygain.h"
static int id3v1_set_string(AVFormatContext *s, const char *key,
uint8_t *buf, int buf_size)
{
AVDictionaryEntry *tag;
if ((tag = av_dict_get(s->metadata, key, NULL, 0)))
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av_strlcpy(buf, tag->value, buf_size);
return !!tag;
}
static int id3v1_create_tag(AVFormatContext *s, uint8_t *buf)
{
AVDictionaryEntry *tag;
int i, count = 0;
memset(buf, 0, ID3v1_TAG_SIZE); /* fail safe */
buf[0] = 'T';
buf[1] = 'A';
buf[2] = 'G';
/* we knowingly overspecify each tag length by one byte to compensate for the mandatory null byte added by av_strlcpy */
count += id3v1_set_string(s, "TIT2", buf + 3, 30 + 1); //title
count += id3v1_set_string(s, "TPE1", buf + 33, 30 + 1); //author|artist
count += id3v1_set_string(s, "TALB", buf + 63, 30 + 1); //album
count += id3v1_set_string(s, "TDRC", buf + 93, 4 + 1); //date
count += id3v1_set_string(s, "comment", buf + 97, 30 + 1);
if ((tag = av_dict_get(s->metadata, "TRCK", NULL, 0))) { //track
buf[125] = 0;
buf[126] = atoi(tag->value);
count++;
}
buf[127] = 0xFF; /* default to unknown genre */
if ((tag = av_dict_get(s->metadata, "TCON", NULL, 0))) { //genre
for(i = 0; i <= ID3v1_GENRE_MAX; i++) {
if (!av_strcasecmp(tag->value, ff_id3v1_genre_str[i])) {
buf[127] = i;
count++;
break;
}
}
}
return count;
}
#define XING_NUM_BAGS 400
#define XING_TOC_SIZE 100
// size of the XING/LAME data, starting from the Xing tag
#define XING_SIZE 156
typedef struct MP3Context {
const AVClass *class;
ID3v2EncContext id3;
int id3v2_version;
int write_id3v1;
int write_xing;
/* xing header */
// a buffer containing the whole XING/LAME frame
uint8_t *xing_frame;
int xing_frame_size;
AVCRC audio_crc; // CRC of the audio data
uint32_t audio_size; // total size of the audio data
// offset of the XING/LAME frame in the file
int64_t xing_frame_offset;
// offset of the XING/INFO tag in the frame
int xing_offset;
int32_t frames;
int32_t size;
uint32_t want;
uint32_t seen;
uint32_t pos;
uint64_t bag[XING_NUM_BAGS];
int initial_bitrate;
int has_variable_bitrate;
/* index of the audio stream */
int audio_stream_idx;
/* number of attached pictures we still need to write */
int pics_to_write;
/* audio packets are queued here until we get all the attached pictures */
AVPacketList *queue, *queue_end;
} MP3Context;
static const uint8_t xing_offtbl[2][2] = {{32, 17}, {17, 9}};
/*
* Write an empty XING header and initialize respective data.
*/
static int mp3_write_xing(AVFormatContext *s)
{
MP3Context *mp3 = s->priv_data;
AVCodecContext *codec = s->streams[mp3->audio_stream_idx]->codec;
AVDictionaryEntry *enc = av_dict_get(s->streams[mp3->audio_stream_idx]->metadata, "encoder", NULL, 0);
AVIOContext *dyn_ctx;
int32_t header;
MPADecodeHeader mpah;
int srate_idx, i, channels;
int bitrate_idx;
int best_bitrate_idx = -1;
int best_bitrate_error = INT_MAX;
int ret;
int ver = 0;
int bytes_needed;
if (!s->pb->seekable || !mp3->write_xing)
Merge remote-tracking branch 'qatar/master' * 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>
2012-03-01 03:13:16 +03:00
return 0;
for (i = 0; i < FF_ARRAY_ELEMS(avpriv_mpa_freq_tab); i++) {
const uint16_t base_freq = avpriv_mpa_freq_tab[i];
if (codec->sample_rate == base_freq) ver = 0x3; // MPEG 1
else if (codec->sample_rate == base_freq / 2) ver = 0x2; // MPEG 2
else if (codec->sample_rate == base_freq / 4) ver = 0x0; // MPEG 2.5
else continue;
srate_idx = i;
break;
}
if (i == FF_ARRAY_ELEMS(avpriv_mpa_freq_tab)) {
av_log(s, AV_LOG_WARNING, "Unsupported sample rate, not writing Xing header.\n");
return -1;
}
switch (codec->channels) {
case 1: channels = MPA_MONO; break;
case 2: channels = MPA_STEREO; break;
default: av_log(s, AV_LOG_WARNING, "Unsupported number of channels, "
"not writing Xing header.\n");
return -1;
}
/* dummy MPEG audio header */
header = 0xffU << 24; // sync
header |= (0x7 << 5 | ver << 3 | 0x1 << 1 | 0x1) << 16; // sync/audio-version/layer 3/no crc*/
header |= (srate_idx << 2) << 8;
header |= channels << 6;
for (bitrate_idx = 1; bitrate_idx < 15; bitrate_idx++) {
int bit_rate = 1000 * avpriv_mpa_bitrate_tab[ver != 3][3 - 1][bitrate_idx];
int error = FFABS(bit_rate - codec->bit_rate);
if (error < best_bitrate_error) {
best_bitrate_error = error;
best_bitrate_idx = bitrate_idx;
}
}
av_assert0(best_bitrate_idx >= 0);
for (bitrate_idx = best_bitrate_idx; ; bitrate_idx++) {
int32_t mask = bitrate_idx << (4 + 8);
if (15 == bitrate_idx)
return -1;
header |= mask;
avpriv_mpegaudio_decode_header(&mpah, header);
mp3->xing_offset = xing_offtbl[mpah.lsf == 1][mpah.nb_channels == 1] + 4;
bytes_needed = mp3->xing_offset + XING_SIZE;
if (bytes_needed <= mpah.frame_size)
break;
header &= ~mask;
}
ret = avio_open_dyn_buf(&dyn_ctx);
if (ret < 0)
return ret;
avio_wb32(dyn_ctx, header);
ffio_fill(dyn_ctx, 0, mp3->xing_offset - 4);
ffio_wfourcc(dyn_ctx, "Xing");
avio_wb32(dyn_ctx, 0x01 | 0x02 | 0x04 | 0x08); // frames / size / TOC / vbr scale
mp3->size = mpah.frame_size;
mp3->want=1;
mp3->seen=0;
mp3->pos=0;
avio_wb32(dyn_ctx, 0); // frames
avio_wb32(dyn_ctx, 0); // size
// TOC
for (i = 0; i < XING_TOC_SIZE; i++)
avio_w8(dyn_ctx, (uint8_t)(255 * i / XING_TOC_SIZE));
// vbr quality
// we write it, because some (broken) tools always expect it to be present
avio_wb32(dyn_ctx, 0);
// encoder short version string
if (enc) {
uint8_t encoder_str[9] = { 0 };
if ( strlen(enc->value) > sizeof(encoder_str)
&& !strcmp("Lavc libmp3lame", enc->value)) {
memcpy(encoder_str, "Lavf lame", 9);
} else
memcpy(encoder_str, enc->value, FFMIN(strlen(enc->value), sizeof(encoder_str)));
avio_write(dyn_ctx, encoder_str, sizeof(encoder_str));
} else
avio_write(dyn_ctx, "Lavf\0\0\0\0\0", 9);
avio_w8(dyn_ctx, 0); // tag revision 0 / unknown vbr method
avio_w8(dyn_ctx, 0); // unknown lowpass filter value
ffio_fill(dyn_ctx, 0, 8); // empty replaygain fields
avio_w8(dyn_ctx, 0); // unknown encoding flags
avio_w8(dyn_ctx, 0); // unknown abr/minimal bitrate
// encoder delay
if (codec->initial_padding - 528 - 1 >= 1 << 12) {
av_log(s, AV_LOG_WARNING, "Too many samples of initial padding.\n");
}
avio_wb24(dyn_ctx, FFMAX(codec->initial_padding - 528 - 1, 0)<<12);
avio_w8(dyn_ctx, 0); // misc
avio_w8(dyn_ctx, 0); // mp3gain
avio_wb16(dyn_ctx, 0); // preset
// audio length and CRCs (will be updated later)
avio_wb32(dyn_ctx, 0); // music length
avio_wb16(dyn_ctx, 0); // music crc
avio_wb16(dyn_ctx, 0); // tag crc
ffio_fill(dyn_ctx, 0, mpah.frame_size - bytes_needed);
mp3->xing_frame_size = avio_close_dyn_buf(dyn_ctx, &mp3->xing_frame);
mp3->xing_frame_offset = avio_tell(s->pb);
avio_write(s->pb, mp3->xing_frame, mp3->xing_frame_size);
mp3->audio_size = mp3->xing_frame_size;
return 0;
}
/*
* Add a frame to XING data.
* Following lame's "VbrTag.c".
*/
static void mp3_xing_add_frame(MP3Context *mp3, AVPacket *pkt)
{
int i;
mp3->frames++;
mp3->seen++;
mp3->size += pkt->size;
if (mp3->want == mp3->seen) {
mp3->bag[mp3->pos] = mp3->size;
if (XING_NUM_BAGS == ++mp3->pos) {
/* shrink table to half size by throwing away each second bag. */
for (i = 1; i < XING_NUM_BAGS; i += 2)
mp3->bag[i >> 1] = mp3->bag[i];
/* double wanted amount per bag. */
mp3->want *= 2;
/* adjust current position to half of table size. */
mp3->pos = XING_NUM_BAGS / 2;
}
mp3->seen = 0;
}
}
static int mp3_write_audio_packet(AVFormatContext *s, AVPacket *pkt)
{
MP3Context *mp3 = s->priv_data;
if (pkt->data && pkt->size >= 4) {
MPADecodeHeader mpah;
int av_unused base;
uint32_t h;
h = AV_RB32(pkt->data);
if (ff_mpa_check_header(h) == 0) {
avpriv_mpegaudio_decode_header(&mpah, h);
if (!mp3->initial_bitrate)
mp3->initial_bitrate = mpah.bit_rate;
if ((mpah.bit_rate == 0) || (mp3->initial_bitrate != mpah.bit_rate))
mp3->has_variable_bitrate = 1;
} else {
av_log(s, AV_LOG_WARNING, "Audio packet of size %d (starting with %08X...) "
"is invalid, writing it anyway.\n", pkt->size, h);
}
#ifdef FILTER_VBR_HEADERS
/* filter out XING and INFO headers. */
base = 4 + xing_offtbl[mpah.lsf == 1][mpah.nb_channels == 1];
if (base + 4 <= pkt->size) {
uint32_t v = AV_RB32(pkt->data + base);
if (MKBETAG('X','i','n','g') == v || MKBETAG('I','n','f','o') == v)
return 0;
}
/* filter out VBRI headers. */
base = 4 + 32;
if (base + 4 <= pkt->size && MKBETAG('V','B','R','I') == AV_RB32(pkt->data + base))
return 0;
#endif
if (mp3->xing_offset) {
mp3_xing_add_frame(mp3, pkt);
mp3->audio_size += pkt->size;
mp3->audio_crc = av_crc(av_crc_get_table(AV_CRC_16_ANSI_LE),
mp3->audio_crc, pkt->data, pkt->size);
}
}
return ff_raw_write_packet(s, pkt);
}
static int mp3_queue_flush(AVFormatContext *s)
{
MP3Context *mp3 = s->priv_data;
AVPacketList *pktl;
int ret = 0, write = 1;
ff_id3v2_finish(&mp3->id3, s->pb, s->metadata_header_padding);
mp3_write_xing(s);
while ((pktl = mp3->queue)) {
if (write && (ret = mp3_write_audio_packet(s, &pktl->pkt)) < 0)
write = 0;
av_free_packet(&pktl->pkt);
mp3->queue = pktl->next;
av_freep(&pktl);
}
mp3->queue_end = NULL;
return ret;
}
static void mp3_update_xing(AVFormatContext *s)
{
MP3Context *mp3 = s->priv_data;
AVReplayGain *rg;
uint16_t tag_crc;
uint8_t *toc;
int i, rg_size;
/* replace "Xing" identification string with "Info" for CBR files. */
if (!mp3->has_variable_bitrate)
AV_WL32(mp3->xing_frame + mp3->xing_offset, MKTAG('I', 'n', 'f', 'o'));
AV_WB32(mp3->xing_frame + mp3->xing_offset + 8, mp3->frames);
AV_WB32(mp3->xing_frame + mp3->xing_offset + 12, mp3->size);
toc = mp3->xing_frame + mp3->xing_offset + 16;
toc[0] = 0; // first toc entry has to be zero.
for (i = 1; i < XING_TOC_SIZE; ++i) {
int j = i * mp3->pos / XING_TOC_SIZE;
int seek_point = 256LL * mp3->bag[j] / mp3->size;
toc[i] = FFMIN(seek_point, 255);
}
/* write replaygain */
rg = (AVReplayGain*)av_stream_get_side_data(s->streams[0], AV_PKT_DATA_REPLAYGAIN,
&rg_size);
if (rg && rg_size >= sizeof(*rg)) {
uint16_t val;
AV_WB32(mp3->xing_frame + mp3->xing_offset + 131,
av_rescale(rg->track_peak, 1 << 23, 100000));
if (rg->track_gain != INT32_MIN) {
val = FFABS(rg->track_gain / 10000) & ((1 << 9) - 1);
val |= (rg->track_gain < 0) << 9;
val |= 1 << 13;
AV_WB16(mp3->xing_frame + mp3->xing_offset + 135, val);
}
if (rg->album_gain != INT32_MIN) {
val = FFABS(rg->album_gain / 10000) & ((1 << 9) - 1);
val |= (rg->album_gain < 0) << 9;
val |= 1 << 14;
AV_WB16(mp3->xing_frame + mp3->xing_offset + 137, val);
}
}
AV_WB32(mp3->xing_frame + mp3->xing_offset + XING_SIZE - 8, mp3->audio_size);
AV_WB16(mp3->xing_frame + mp3->xing_offset + XING_SIZE - 4, mp3->audio_crc);
tag_crc = av_crc(av_crc_get_table(AV_CRC_16_ANSI_LE), 0, mp3->xing_frame, 190);
AV_WB16(mp3->xing_frame + mp3->xing_offset + XING_SIZE - 2, tag_crc);
avio_seek(s->pb, mp3->xing_frame_offset, SEEK_SET);
avio_write(s->pb, mp3->xing_frame, mp3->xing_frame_size);
avio_seek(s->pb, 0, SEEK_END);
}
static int mp3_write_trailer(struct AVFormatContext *s)
{
uint8_t buf[ID3v1_TAG_SIZE];
MP3Context *mp3 = s->priv_data;
if (mp3->pics_to_write) {
av_log(s, AV_LOG_WARNING, "No packets were sent for some of the "
"attached pictures.\n");
mp3_queue_flush(s);
}
/* write the id3v1 tag */
if (mp3->write_id3v1 && id3v1_create_tag(s, buf) > 0) {
avio_write(s->pb, buf, ID3v1_TAG_SIZE);
}
if (mp3->xing_offset)
mp3_update_xing(s);
av_freep(&mp3->xing_frame);
return 0;
}
static int query_codec(enum AVCodecID id, int std_compliance)
{
const CodecMime *cm= ff_id3v2_mime_tags;
while(cm->id != AV_CODEC_ID_NONE) {
if(id == cm->id)
return MKTAG('A', 'P', 'I', 'C');
cm++;
}
return -1;
}
#if CONFIG_MP2_MUXER
AVOutputFormat ff_mp2_muxer = {
.name = "mp2",
.long_name = NULL_IF_CONFIG_SMALL("MP2 (MPEG audio layer 2)"),
.mime_type = "audio/mpeg",
.extensions = "mp2,m2a,mpa",
.audio_codec = AV_CODEC_ID_MP2,
.video_codec = AV_CODEC_ID_NONE,
.write_packet = ff_raw_write_packet,
Merge remote-tracking branch 'qatar/master' * 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>
2012-03-01 03:13:16 +03:00
.flags = AVFMT_NOTIMESTAMPS,
};
#endif
#if CONFIG_MP3_MUXER
static const AVOption options[] = {
{ "id3v2_version", "Select ID3v2 version to write. Currently 3 and 4 are supported.",
offsetof(MP3Context, id3v2_version), AV_OPT_TYPE_INT, {.i64 = 4}, 0, 4, AV_OPT_FLAG_ENCODING_PARAM},
{ "write_id3v1", "Enable ID3v1 writing. ID3v1 tags are written in UTF-8 which may not be supported by most software.",
offsetof(MP3Context, write_id3v1), AV_OPT_TYPE_INT, {.i64 = 0}, 0, 1, AV_OPT_FLAG_ENCODING_PARAM},
{ "write_xing", "Write the Xing header containing file duration.",
offsetof(MP3Context, write_xing), AV_OPT_TYPE_INT, {.i64 = 1}, 0, 1, AV_OPT_FLAG_ENCODING_PARAM},
{ NULL },
};
static const AVClass mp3_muxer_class = {
.class_name = "MP3 muxer",
.item_name = av_default_item_name,
.option = options,
.version = LIBAVUTIL_VERSION_INT,
};
static int mp3_write_packet(AVFormatContext *s, AVPacket *pkt)
{
MP3Context *mp3 = s->priv_data;
if (pkt->stream_index == mp3->audio_stream_idx) {
if (mp3->pics_to_write) {
/* buffer audio packets until we get all the pictures */
AVPacketList *pktl = av_mallocz(sizeof(*pktl));
int ret;
if (!pktl) {
av_log(s, AV_LOG_WARNING, "Not enough memory to buffer audio. Skipping picture streams\n");
mp3->pics_to_write = 0;
mp3_queue_flush(s);
return mp3_write_audio_packet(s, pkt);
}
ret = av_copy_packet(&pktl->pkt, pkt);
if (ret < 0) {
av_freep(&pktl);
return ret;
}
if (mp3->queue_end)
mp3->queue_end->next = pktl;
else
mp3->queue = pktl;
mp3->queue_end = pktl;
} else
return mp3_write_audio_packet(s, pkt);
} else {
int ret;
/* warn only once for each stream */
if (s->streams[pkt->stream_index]->nb_frames == 1) {
av_log(s, AV_LOG_WARNING, "Got more than one picture in stream %d,"
" ignoring.\n", pkt->stream_index);
}
if (!mp3->pics_to_write || s->streams[pkt->stream_index]->nb_frames >= 1)
return 0;
if ((ret = ff_id3v2_write_apic(s, &mp3->id3, pkt)) < 0)
return ret;
mp3->pics_to_write--;
/* flush the buffered audio packets */
if (!mp3->pics_to_write &&
(ret = mp3_queue_flush(s)) < 0)
return ret;
}
return 0;
}
/**
* Write an ID3v2 header at beginning of stream
*/
static int mp3_write_header(struct AVFormatContext *s)
{
MP3Context *mp3 = s->priv_data;
int ret, i;
if (mp3->id3v2_version &&
mp3->id3v2_version != 3 &&
mp3->id3v2_version != 4) {
av_log(s, AV_LOG_ERROR, "Invalid ID3v2 version requested: %d. Only "
"3, 4 or 0 (disabled) are allowed.\n", mp3->id3v2_version);
return AVERROR(EINVAL);
}
/* check the streams -- we want exactly one audio and arbitrary number of
* video (attached pictures) */
mp3->audio_stream_idx = -1;
for (i = 0; i < s->nb_streams; i++) {
AVStream *st = s->streams[i];
if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
if (mp3->audio_stream_idx >= 0 || st->codec->codec_id != AV_CODEC_ID_MP3) {
av_log(s, AV_LOG_ERROR, "Invalid audio stream. Exactly one MP3 "
"audio stream is required.\n");
return AVERROR(EINVAL);
}
mp3->audio_stream_idx = i;
} else if (st->codec->codec_type != AVMEDIA_TYPE_VIDEO) {
av_log(s, AV_LOG_ERROR, "Only audio streams and pictures are allowed in MP3.\n");
return AVERROR(EINVAL);
}
}
if (mp3->audio_stream_idx < 0) {
av_log(s, AV_LOG_ERROR, "No audio stream present.\n");
return AVERROR(EINVAL);
}
mp3->pics_to_write = s->nb_streams - 1;
if (mp3->pics_to_write && !mp3->id3v2_version) {
av_log(s, AV_LOG_ERROR, "Attached pictures were requested, but the "
"ID3v2 header is disabled.\n");
return AVERROR(EINVAL);
}
if (mp3->id3v2_version) {
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) {
if (mp3->id3v2_version)
ff_id3v2_finish(&mp3->id3, s->pb, s->metadata_header_padding);
mp3_write_xing(s);
}
return 0;
}
AVOutputFormat ff_mp3_muxer = {
.name = "mp3",
.long_name = NULL_IF_CONFIG_SMALL("MP3 (MPEG audio layer 3)"),
.mime_type = "audio/mpeg",
.extensions = "mp3",
.priv_data_size = sizeof(MP3Context),
.audio_codec = AV_CODEC_ID_MP3,
.video_codec = AV_CODEC_ID_PNG,
.write_header = mp3_write_header,
.write_packet = mp3_write_packet,
.write_trailer = mp3_write_trailer,
.query_codec = query_codec,
.flags = AVFMT_NOTIMESTAMPS,
.priv_class = &mp3_muxer_class,
};
#endif