/** * Write a packet to an output media file. * * This function passes the packet directly to the muxer, without any buffering * or reordering. The caller is responsible for correctly interleaving the * packets if the format requires it. Callers that want libavformat to handle * the interleaving should call av_interleaved_write_frame() instead of this * function. * * @param s media file handle * @param pkt The packet containing the data to be written. Note that unlike * av_interleaved_write_frame(), this function does not take * ownership of the packet passed to it (though some muxers may make * an internal reference to the input packet). * <br> * This parameter can be NULL (at any time, not just at the end), in * order to immediately flush data buffered within the muxer, for * muxers that buffer up data internally before writing it to the * output. * <br> * Packet‘s @ref AVPacket.stream_index "stream_index" field must be * set to the index of the corresponding stream in @ref * AVFormatContext.streams "s->streams". It is very strongly * recommended that timing information (@ref AVPacket.pts "pts", @ref * AVPacket.dts "dts", @ref AVPacket.duration "duration") is set to * correct values. * @return < 0 on error, = 0 if OK, 1 if flushed and there is no more data to flush * * @see av_interleaved_write_frame() */ int av_write_frame(AVFormatContext *s, AVPacket *pkt);
s：用于输出的AVFormatContext。函数正常执行后返回值等于0。
pkt：等待输出的AVPacket。
这个函数最典型的例子可以参考：
最简单的基于FFMPEG的视频编码器（YUV编码为H.264）
int av_write_frame(AVFormatContext *s, AVPacket *pkt)
{
int ret;
ret = check_packet(s, pkt);
if (ret < 0)
return ret;
//Packet为NULL，Flush Encoder
if (!pkt) {
if (s->oformat->flags & AVFMT_ALLOW_FLUSH) {
ret = s->oformat->write_packet(s, NULL);
if (s->flush_packets && s->pb && s->pb->error >= 0 && s->flags & AVFMT_FLAG_FLUSH_PACKETS)
avio_flush(s->pb);
if (ret >= 0 && s->pb && s->pb->error < 0)
ret = s->pb->error;
return ret;
}
return 1;
}
ret = compute_pkt_fields2(s, s->streams[pkt->stream_index], pkt);
if (ret < 0 && !(s->oformat->flags & AVFMT_NOTIMESTAMPS))
return ret;
//写入
ret = write_packet(s, pkt);
if (ret >= 0 && s->pb && s->pb->error < 0)
ret = s->pb->error;
if (ret >= 0)
s->streams[pkt->stream_index]->nb_frames++;
return ret;
}
下面分别看一下这几个函数功能。
static int check_packet(AVFormatContext *s, AVPacket *pkt)
{
if (!pkt)
return 0;
if (pkt->stream_index < 0 || pkt->stream_index >= s->nb_streams) {
av_log(s, AV_LOG_ERROR, "Invalid packet stream index: %d\n",
pkt->stream_index);
return AVERROR(EINVAL);
}
if (s->streams[pkt->stream_index]->codec->codec_type == AVMEDIA_TYPE_ATTACHMENT) {
av_log(s, AV_LOG_ERROR, "Received a packet for an attachment stream.\n");
return AVERROR(EINVAL);
}
return 0;
}
//FIXME merge with compute_pkt_fields
static int compute_pkt_fields2(AVFormatContext *s, AVStream *st, AVPacket *pkt)
{
int delay = FFMAX(st->codec->has_b_frames, st->codec->max_b_frames > 0);
int num, den, i;
int frame_size;
av_dlog(s, "compute_pkt_fields2: pts:%s dts:%s cur_dts:%s b:%d size:%d st:%d\n",
av_ts2str(pkt->pts), av_ts2str(pkt->dts), av_ts2str(st->cur_dts), delay, pkt->size, pkt->stream_index);
if (pkt->duration < 0 && st->codec->codec_type != AVMEDIA_TYPE_SUBTITLE) {
av_log(s, AV_LOG_WARNING, "Packet with invalid duration %d in stream %d\n",
pkt->duration, pkt->stream_index);
pkt->duration = 0;
}
/* duration field */
if (pkt->duration == 0) {
ff_compute_frame_duration(s, &num, &den, st, NULL, pkt);
if (den && num) {
pkt->duration = av_rescale(1, num * (int64_t)st->time_base.den * st->codec->ticks_per_frame, den * (int64_t)st->time_base.num);
}
}
if (pkt->pts == AV_NOPTS_VALUE && pkt->dts != AV_NOPTS_VALUE && delay == 0)
pkt->pts = pkt->dts;
//XXX/FIXME this is a temporary hack until all encoders output pts
if ((pkt->pts == 0 || pkt->pts == AV_NOPTS_VALUE) && pkt->dts == AV_NOPTS_VALUE && !delay) {
static int warned;
if (!warned) {
av_log(s, AV_LOG_WARNING, "Encoder did not produce proper pts, making some up.\n");
warned = 1;
}
pkt->dts =
// pkt->pts= st->cur_dts;
pkt->pts = st->pts.val;
}
//calculate dts from pts
if (pkt->pts != AV_NOPTS_VALUE && pkt->dts == AV_NOPTS_VALUE && delay <= MAX_REORDER_DELAY) {
st->pts_buffer[0] = pkt->pts;
for (i = 1; i < delay + 1 && st->pts_buffer[i] == AV_NOPTS_VALUE; i++)
st->pts_buffer[i] = pkt->pts + (i - delay - 1) * pkt->duration;
for (i = 0; i<delay && st->pts_buffer[i] > st->pts_buffer[i + 1]; i++)
FFSWAP(int64_t, st->pts_buffer[i], st->pts_buffer[i + 1]);
pkt->dts = st->pts_buffer[0];
}
if (st->cur_dts && st->cur_dts != AV_NOPTS_VALUE &&
((!(s->oformat->flags & AVFMT_TS_NONSTRICT) &&
st->cur_dts >= pkt->dts) || st->cur_dts > pkt->dts)) {
av_log(s, AV_LOG_ERROR,
"Application provided invalid, non monotonically increasing dts to muxer in stream %d: %s >= %s\n",
st->index, av_ts2str(st->cur_dts), av_ts2str(pkt->dts));
return AVERROR(EINVAL);
}
if (pkt->dts != AV_NOPTS_VALUE && pkt->pts != AV_NOPTS_VALUE && pkt->pts < pkt->dts) {
av_log(s, AV_LOG_ERROR,
"pts (%s) < dts (%s) in stream %d\n",
av_ts2str(pkt->pts), av_ts2str(pkt->dts),
st->index);
return AVERROR(EINVAL);
}
av_dlog(s, "av_write_frame: pts2:%s dts2:%s\n",
av_ts2str(pkt->pts), av_ts2str(pkt->dts));
st->cur_dts = pkt->dts;
st->pts.val = pkt->dts;
/* update pts */
switch (st->codec->codec_type) {
case AVMEDIA_TYPE_AUDIO:
frame_size = (pkt->flags & AV_PKT_FLAG_UNCODED_FRAME) ?
((AVFrame *)pkt->data)->nb_samples :
av_get_audio_frame_duration(st->codec, pkt->size);
/* HACK/FIXME, we skip the initial 0 size packets as they are most
* likely equal to the encoder delay, but it would be better if we
* had the real timestamps from the encoder */
if (frame_size >= 0 && (pkt->size || st->pts.num != st->pts.den >> 1 || st->pts.val)) {
frac_add(&st->pts, (int64_t)st->time_base.den * frame_size);
}
break;
case AVMEDIA_TYPE_VIDEO:
frac_add(&st->pts, (int64_t)st->time_base.den * st->codec->time_base.num);
break;
}
return 0;
}
/**
* Make timestamps non negative, move side data from payload to internal struct, call muxer, and restore
* sidedata.
*
* FIXME: this function should NEVER get undefined pts/dts beside when the
* AVFMT_NOTIMESTAMPS is set.
* Those additional safety checks should be dropped once the correct checks
* are set in the callers.
*/
static int write_packet(AVFormatContext *s, AVPacket *pkt)
{
int ret, did_split;
if (s->output_ts_offset) {
AVStream *st = s->streams[pkt->stream_index];
int64_t offset = av_rescale_q(s->output_ts_offset, AV_TIME_BASE_Q, st->time_base);
if (pkt->dts != AV_NOPTS_VALUE)
pkt->dts += offset;
if (pkt->pts != AV_NOPTS_VALUE)
pkt->pts += offset;
}
if (s->avoid_negative_ts > 0) {
AVStream *st = s->streams[pkt->stream_index];
int64_t offset = st->mux_ts_offset;
if (s->offset == AV_NOPTS_VALUE && pkt->dts != AV_NOPTS_VALUE &&
(pkt->dts < 0 || s->avoid_negative_ts == AVFMT_AVOID_NEG_TS_MAKE_ZERO)) {
s->offset = -pkt->dts;
s->offset_timebase = st->time_base;
}
if (s->offset != AV_NOPTS_VALUE && !offset) {
offset = st->mux_ts_offset =
av_rescale_q_rnd(s->offset,
s->offset_timebase,
st->time_base,
AV_ROUND_UP);
}
if (pkt->dts != AV_NOPTS_VALUE)
pkt->dts += offset;
if (pkt->pts != AV_NOPTS_VALUE)
pkt->pts += offset;
av_assert2(pkt->dts == AV_NOPTS_VALUE || pkt->dts >= 0 || s->max_interleave_delta > 0);
if (pkt->dts != AV_NOPTS_VALUE && pkt->dts < 0) {
av_log(s, AV_LOG_WARNING,
"Packets poorly interleaved, failed to avoid negative "
"timestamp %s in stream %d.\n"
"Try -max_interleave_delta 0 as a possible workaround.\n",
av_ts2str(pkt->dts),
pkt->stream_index
);
}
}
did_split = av_packet_split_side_data(pkt);
if ((pkt->flags & AV_PKT_FLAG_UNCODED_FRAME)) {
AVFrame *frame = (AVFrame *)pkt->data;
av_assert0(pkt->size == UNCODED_FRAME_PACKET_SIZE);
ret = s->oformat->write_uncoded_frame(s, pkt->stream_index, &frame, 0);
av_frame_free(&frame);
} else {
//写入
ret = s->oformat->write_packet(s, pkt);
}
if (s->flush_packets && s->pb && ret >= 0 && s->flags & AVFMT_FLAG_FLUSH_PACKETS)
avio_flush(s->pb);
if (did_split)
av_packet_merge_side_data(pkt);
return ret;
}
AVOutputFormat ff_flv_muxer = {
.name = "flv",
.long_name = NULL_IF_CONFIG_SMALL("FLV (Flash Video)"),
.mime_type = "video/x-flv",
.extensions = "flv",
.priv_data_size = sizeof(FLVContext),
.audio_codec = CONFIG_LIBMP3LAME ? AV_CODEC_ID_MP3 : AV_CODEC_ID_ADPCM_SWF,
.video_codec = AV_CODEC_ID_FLV1,
.write_header = flv_write_header,
.write_packet = flv_write_packet,
.write_trailer = flv_write_trailer,
.codec_tag = (const AVCodecTag* const []) {
flv_video_codec_ids, flv_audio_codec_ids, 0
},
.flags = AVFMT_GLOBALHEADER | AVFMT_VARIABLE_FPS |
AVFMT_TS_NONSTRICT,
};从ff_flv_muxer的定义可以看出，write_packet()指向的是flv_write_packet()函数。在看flv_write_packet()函数的定义之前，我们先回顾一下FLV封装格式的结构。
FLV封装格式如下图所示。
PS：原图是网上找的，感觉画的很清晰，比官方的Video File Format Specification更加通俗易懂。但是图中有一个错误，就是TagHeader中的StreamID字段的长度写错了（查看了一下官方标准，应该是3字节，现在已经改过来了）。
0 = Linear PCM, platform endian第1个字节的第5-6位的数值表示采样率：0 = 5.5kHz，1 = 11KHz，2 = 22 kHz，3 = 44 kHz。
1 = ADPCM
2 = MP3
3 = Linear PCM, little endian
4 = Nellymoser 16-kHz mono
5 = Nellymoser 8-kHz mono
6 = Nellymoser
7 = G.711 A-law logarithmic PCM
8 = G.711 mu-law logarithmic PCM
9 = reserved
10 = AAC
14 = MP3 8-Khz
15 = Device-specific sound
1: keyframe (for AVC, a seekableframe)（关键帧）第1个字节的后4位的数值表示视频编码ID（CodecID）：
2: inter frame (for AVC, a nonseekableframe)
3: disposable inter frame (H.263only)
4: generated keyframe (reservedfor server use only)
5: video info/command frame
1: JPEG (currently unused)其中，当音频编码为AVC（H.264）的时候，第一个字节后面存储的是AVCVIDEOPACKET，格式如下所示。
2: Sorenson H.263
3: Screen video
4: On2 VP6
5: On2 VP6 with alpha channel
6: Screen video version 2
7: AVC
static int flv_write_packet(AVFormatContext *s, AVPacket *pkt)
{
AVIOContext *pb = s->pb;
AVCodecContext *enc = s->streams[pkt->stream_index]->codec;
FLVContext *flv = s->priv_data;
FLVStreamContext *sc = s->streams[pkt->stream_index]->priv_data;
unsigned ts;
int size = pkt->size;
uint8_t *data = NULL;
int flags = -1, flags_size, ret;
if (enc->codec_id == AV_CODEC_ID_VP6F || enc->codec_id == AV_CODEC_ID_VP6A ||
enc->codec_id == AV_CODEC_ID_VP6 || enc->codec_id == AV_CODEC_ID_AAC)
flags_size = 2;
else if (enc->codec_id == AV_CODEC_ID_H264 || enc->codec_id == AV_CODEC_ID_MPEG4)
flags_size = 5;
else
flags_size = 1;
if (flv->delay == AV_NOPTS_VALUE)
flv->delay = -pkt->dts;
if (pkt->dts < -flv->delay) {
av_log(s, AV_LOG_WARNING,
"Packets are not in the proper order with respect to DTS\n");
return AVERROR(EINVAL);
}
ts = pkt->dts + flv->delay; // add delay to force positive dts
if (s->event_flags & AVSTREAM_EVENT_FLAG_METADATA_UPDATED) {
write_metadata(s, ts);
s->event_flags &= ~AVSTREAM_EVENT_FLAG_METADATA_UPDATED;
}
//Tag Header
switch (enc->codec_type) {
case AVMEDIA_TYPE_VIDEO:
//Type
avio_w8(pb, FLV_TAG_TYPE_VIDEO);
flags = enc->codec_tag;
if (flags == 0) {
av_log(s, AV_LOG_ERROR,
"Video codec ‘%s‘ is not compatible with FLV\n",
avcodec_get_name(enc->codec_id));
return AVERROR(EINVAL);
}
//Key Frame?
flags |= pkt->flags & AV_PKT_FLAG_KEY ? FLV_FRAME_KEY : FLV_FRAME_INTER;
break;
case AVMEDIA_TYPE_AUDIO:
flags = get_audio_flags(s, enc);
av_assert0(size);
//Type
avio_w8(pb, FLV_TAG_TYPE_AUDIO);
break;
case AVMEDIA_TYPE_DATA:
//Type
avio_w8(pb, FLV_TAG_TYPE_META);
break;
default:
return AVERROR(EINVAL);
}
if (enc->codec_id == AV_CODEC_ID_H264 || enc->codec_id == AV_CODEC_ID_MPEG4) {
/* check if extradata looks like mp4 formated */
if (enc->extradata_size > 0 && *(uint8_t*)enc->extradata != 1)
if ((ret = ff_avc_parse_nal_units_buf(pkt->data, &data, &size)) < 0)
return ret;
} else if (enc->codec_id == AV_CODEC_ID_AAC && pkt->size > 2 &&
(AV_RB16(pkt->data) & 0xfff0) == 0xfff0) {
if (!s->streams[pkt->stream_index]->nb_frames) {
av_log(s, AV_LOG_ERROR, "Malformed AAC bitstream detected: "
"use the audio bitstream filter ‘aac_adtstoasc‘ to fix it "
"(‘-bsf:a aac_adtstoasc‘ option with ffmpeg)\n");
return AVERROR_INVALIDDATA;
}
av_log(s, AV_LOG_WARNING, "aac bitstream error\n");
}
/* check Speex packet duration */
if (enc->codec_id == AV_CODEC_ID_SPEEX && ts - sc->last_ts > 160)
av_log(s, AV_LOG_WARNING, "Warning: Speex stream has more than "
"8 frames per packet. Adobe Flash "
"Player cannot handle this!\n");
if (sc->last_ts < ts)
sc->last_ts = ts;
if (size + flags_size >= 1<<24) {
av_log(s, AV_LOG_ERROR, "Too large packet with size %u >= %u\n",
size + flags_size, 1<<24);
return AVERROR(EINVAL);
}
//Tag Header - Datasize
avio_wb24(pb, size + flags_size);
//Tag Header - Timestamp
avio_wb24(pb, ts & 0xFFFFFF);
avio_w8(pb, (ts >> 24) & 0x7F); // timestamps are 32 bits _signed_
//StreamID
avio_wb24(pb, flv->reserved);
if (enc->codec_type == AVMEDIA_TYPE_DATA) {
int data_size;
int64_t metadata_size_pos = avio_tell(pb);
if (enc->codec_id == AV_CODEC_ID_TEXT) {
// legacy FFmpeg magic?
avio_w8(pb, AMF_DATA_TYPE_STRING);
put_amf_string(pb, "onTextData");
avio_w8(pb, AMF_DATA_TYPE_MIXEDARRAY);
avio_wb32(pb, 2);
put_amf_string(pb, "type");
avio_w8(pb, AMF_DATA_TYPE_STRING);
put_amf_string(pb, "Text");
put_amf_string(pb, "text");
avio_w8(pb, AMF_DATA_TYPE_STRING);
put_amf_string(pb, pkt->data);
put_amf_string(pb, "");
avio_w8(pb, AMF_END_OF_OBJECT);
} else {
// just pass the metadata through
avio_write(pb, data ? data : pkt->data, size);
}
/* write total size of tag */
data_size = avio_tell(pb) - metadata_size_pos;
avio_seek(pb, metadata_size_pos - 10, SEEK_SET);
avio_wb24(pb, data_size);
avio_seek(pb, data_size + 10 - 3, SEEK_CUR);
avio_wb32(pb, data_size + 11);
} else {
av_assert1(flags>=0);
//First Byte of Tag Data
avio_w8(pb,flags);
if (enc->codec_id == AV_CODEC_ID_VP6)
avio_w8(pb,0);
if (enc->codec_id == AV_CODEC_ID_VP6F || enc->codec_id == AV_CODEC_ID_VP6A) {
if (enc->extradata_size)
avio_w8(pb, enc->extradata[0]);
else
avio_w8(pb, ((FFALIGN(enc->width, 16) - enc->width) << 4) |
(FFALIGN(enc->height, 16) - enc->height));
} else if (enc->codec_id == AV_CODEC_ID_AAC)
avio_w8(pb, 1); // AAC raw
else if (enc->codec_id == AV_CODEC_ID_H264 || enc->codec_id == AV_CODEC_ID_MPEG4) {
//AVCVIDEOPACKET-AVCPacketType
avio_w8(pb, 1); // AVC NALU
//AVCVIDEOPACKET-CompositionTime
avio_wb24(pb, pkt->pts - pkt->dts);
}
//Data
avio_write(pb, data ? data : pkt->data, size);
avio_wb32(pb, size + flags_size + 11); // previous tag size
flv->duration = FFMAX(flv->duration,
pkt->pts + flv->delay + pkt->duration);
}
av_free(data);
return pb->error;
}
avio_w8(pb, FLV_TAG_TYPE_VIDEO);如果是音频，代码如下：
avio_w8(pb, FLV_TAG_TYPE_AUDIO);（2）写入Tag Header的Datasize，Timestamp和StreamID（至此完成Tag Header）：
//Tag Header - Datasize
avio_wb24(pb, size + flags_size);
//Tag Header - Timestamp
avio_wb24(pb, ts & 0xFFFFFF);
avio_w8(pb, (ts >> 24) & 0x7F); // timestamps are 32 bits _signed_
//StreamID
avio_wb24(pb, flv->reserved);（3）写入Tag Data的第一字节（其中flag已经在前面的代码中设置完毕）： //First Byte of Tag Data
avio_w8(pb,flags);（4）如果编码格式VP6作相应的处理（不研究）；编码格式为AAC，写入AACAUDIODATA；编码格式为H.264，写入AVCVIDEOPACKET： if (enc->codec_id == AV_CODEC_ID_VP6F || enc->codec_id == AV_CODEC_ID_VP6A) {
if (enc->extradata_size)
avio_w8(pb, enc->extradata[0]);
else
avio_w8(pb, ((FFALIGN(enc->width, 16) - enc->width) << 4) |
(FFALIGN(enc->height, 16) - enc->height));
} else if (enc->codec_id == AV_CODEC_ID_AAC)
avio_w8(pb, 1); // AAC raw
else if (enc->codec_id == AV_CODEC_ID_H264 || enc->codec_id == AV_CODEC_ID_MPEG4) {
//AVCVIDEOPACKET-AVCPacketType
avio_w8(pb, 1); // AVC NALU
//AVCVIDEOPACKET-CompositionTime
avio_wb24(pb, pkt->pts - pkt->dts);
}（5）写入数据： //Data
avio_write(pb, data ? data : pkt->data, size);（6） 写入previous tag size：avio_wb32(pb, size + flags_size + 11); // previous tag size至此，flv_write_packet()就完成了一个Tag的写入。
雷霄骅
leixiaohua1020@126.com
http://blog.csdn.net/leixiaohua1020
FFmpeg源代码简单分析：av_write_frame()
原文地址：http://blog.csdn.net/leixiaohua1020/article/details/44199673
