1.引言
大家好，距离上篇文章已经过去有一段时间了，主要是最近太忙了，一直没有更新。今天总算能抽出一点时间，说说ffmpeg源码级别的优化了，这块应该会连载，请大家持续关注。废话不多说，接下来就进入正题，说说怎么样进行优化。
2.问题引出及解决
使用ffmpeg接口去推流，当网络不稳定时，av_interleaved_write_frame有时候会出现长期无法返回，也就是卡死在里面，造成无法退出或不能进行下一步处理。下面我们就来分析下该接口究竟干了什么，揭开其神秘的面纱。
int av_interleaved_write_frame(AVFormatContext *s, AVPacket *pkt)
{
int ret, flush = 0;
ret = prepare_input_packet(s, pkt);
if (ret < 0)
goto fail;
if (pkt) {
AVStream *st = s->streams[pkt->stream_index];
ret = do_packet_auto_bsf(s, pkt);
if (ret == 0)
return 0;
else if (ret < 0)
goto fail;
if (s->debug & FF_FDEBUG_TS)
av_log(s, AV_LOG_TRACE, "av_interleaved_write_frame size:%d dts:%s pts:%s\n",
pkt->size, av_ts2str(pkt->dts), av_ts2str(pkt->pts));
#if FF_API_COMPUTE_PKT_FIELDS2 && FF_API_LAVF_AVCTX
if ((ret = compute_muxer_pkt_fields(s, st, pkt)) < 0 && !(s->oformat->flags & AVFMT_NOTIMESTAMPS))
goto fail;
#endif
if (pkt->dts == AV_NOPTS_VALUE && !(s->oformat->flags & AVFMT_NOTIMESTAMPS)) {
ret = AVERROR(EINVAL);
goto fail;
}
} else {
av_log(s, AV_LOG_TRACE, "av_interleaved_write_frame FLUSH\n");
flush = 1;
}
for (;; ) {
AVPacket opkt;
int ret = interleave_packet(s, &opkt, pkt, flush);
if (pkt) {
memset(pkt, 0, sizeof(*pkt));
av_init_packet(pkt);
pkt = NULL;
}
if (ret <= 0) //FIXME cleanup needed for ret<0 ?
return ret;
ret = write_packet(s, &opkt);
if (ret >= 0)
s->streams[opkt.stream_index]->nb_frames++;
av_packet_unref(&opkt);
if (ret < 0)
return ret;
if(s->pb && s->pb->error)
return s->pb->error;
}
fail:
av_packet_unref(pkt);
return ret;
}
上面就是发送封装包的源代码，下面来一一分析。
static int prepare_input_packet(AVFormatContext *s, AVPacket *pkt)
{
int ret;
/*数据包是否合格*/
ret = check_packet(s, pkt);
if (ret < 0)
return ret;
#if !FF_API_COMPUTE_PKT_FIELDS2 || !FF_API_LAVF_AVCTX
/* 清除时间戳 */
if (!(s->oformat->flags & AVFMT_NOTIMESTAMPS)) {
AVStream *st = s->streams[pkt->stream_index];
/* 当没有重新排序时，也就是pts等于dts，则需要把pts和dts分别进行设置*/
if (!st->internal->reorder) {
if (pkt->pts == AV_NOPTS_VALUE && pkt->dts != AV_NOPTS_VALUE)
pkt->pts = pkt->dts;
if (pkt->dts == AV_NOPTS_VALUE && pkt->pts != AV_NOPTS_VALUE)
pkt->dts = pkt->pts;
}
/* 检查是否设置了时间戳 */
if (pkt->pts == AV_NOPTS_VALUE || pkt->dts == AV_NOPTS_VALUE) {
av_log(s, AV_LOG_ERROR,
"Timestamps are unset in a packet for stream %d\n", st->index);
return AVERROR(EINVAL);
}
/*检查dts是否增加，如果dts没有增加，则需要去检查是否在封包时有其它异常*/
if (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: %" PRId64 " >= %" PRId64 "\n",
st->index, st->cur_dts, pkt->dts);
return AVERROR(EINVAL);
}
/*
如果pts小于dts，则视为异常，因为dts是解码时间戳，pts是显示时间戳
正常情况是，先解码，后显示，所以pts应该是要大于dts，这样才会合理。
*/
if (pkt->pts < pkt->dts) {
av_log(s, AV_LOG_ERROR, "pts %" PRId64 " < dts %" PRId64 " in stream %d\n",
pkt->pts, pkt->dts, st->index);
return AVERROR(EINVAL);
}
}
#endif
return 0;
}
static int do_packet_auto_bsf(AVFormatContext *s, AVPacket *pkt) {
AVStream *st = s->streams[pkt->stream_index];
int i, ret;
if (!(s->flags & AVFMT_FLAG_AUTO_BSF))
return 1;
/*检查是否设置了码流过滤，以及额外数据提取*/
if (s->oformat->check_bitstream) {
if (!st->internal->bitstream_checked) {
if ((ret = s->oformat->check_bitstream(s, pkt)) < 0)
return ret;
else if (ret == 1)
st->internal->bitstream_checked = 1;
}
}
#if FF_API_LAVF_MERGE_SD
FF_DISABLE_DEPRECATION_WARNINGS
if (st->internal->nb_bsfcs) {
/*分离出头部数据*/
ret = av_packet_split_side_data(pkt);
if (ret < 0)
av_log(s, AV_LOG_WARNING, "Failed to split side data before bitstream filter\n");
}
FF_ENABLE_DEPRECATION_WARNINGS
#endif
for (i = 0; i < st->internal->nb_bsfcs; i++) {
AVBSFContext *ctx = st->internal->bsfcs[i];
/*当任何过滤器需要在流结束时刷新，则我们需要在刷新整个BSF数据链表*/
if ((ret = av_bsf_send_packet(ctx, pkt)) < 0) {
av_log(ctx, AV_LOG_ERROR,
"Failed to send packet to filter %s for stream %d\n",
ctx->filter->name, pkt->stream_index);
return ret;
}
/*当有多个码流过滤器，需要循环去生成，并调用*/
if ((ret = av_bsf_receive_packet(ctx, pkt)) < 0) {
if (ret == AVERROR(EAGAIN) || ret == AVERROR_EOF)
return 0;
av_log(ctx, AV_LOG_ERROR,
"Failed to send packet to filter %s for stream %d\n",
ctx->filter->name, pkt->stream_index);
return ret;
}
}
return 1;
}
下面就是开始循环发包的过程，详细分析如下:
/*
功能:间隔发送数据包，实际上是缓存了。这也是av_interleaved_write_frame与av_write_frame的区别之处
AVPacket *out：表述输出数据包
AVPacket *in:表示输入数据包
flush:如果为1表示输入没有更多数据包，应该输出剩余数据包
返回值:1表示输出数据包。0:表示没有数据包
<0，表示发生错误
*/
static int interleave_packet(AVFormatContext *s, AVPacket *out, AVPacket *in, int flush)
{
if (s->oformat->interleave_packet) {
int ret = s->oformat->interleave_packet(s, out, in, flush);
if (in)
av_packet_unref(in);
return ret;
} else
/*把数据包缓存到队列中，那什么时候发送？也就是flush什么时候为1呢?*/
return ff_interleave_packet_per_dts(s, out, in, flush);
}
int ff_interleave_packet_per_dts(AVFormatContext *s, AVPacket *out,
AVPacket *pkt, int flush)
{
AVPacketList *pktl;
int stream_count = 0;
int noninterleaved_count = 0;
int i, ret;
int eof = flush;
if (pkt) {
/*比较数据包的dts添加数据包到想要队列中*/
if ((ret = ff_interleave_add_packet(s, pkt, interleave_compare_dts)) < 0)
return ret;
}
for (i = 0; i < s->nb_streams; i++) {
if (s->streams[i]->last_in_packet_buffer) {
/*记录数据包个数*/
++stream_count;
} else if (s->streams[i]->codecpar->codec_type != AVMEDIA_TYPE_ATTACHMENT &&
s->streams[i]->codecpar->codec_id != AV_CODEC_ID_VP8 &&
s->streams[i]->codecpar->codec_id != AV_CODEC_ID_VP9) {
++noninterleaved_count;
}
}
/*表示缓冲区已满，需要输出*/
if (s->internal->nb_interleaved_streams == stream_count)
flush = 1;
if (s->max_interleave_delta > 0 &&
s->internal->packet_buffer &&
!flush &&
s->internal->nb_interleaved_streams == stream_count+noninterleaved_count
) {
AVPacket *top_pkt = &s->internal->packet_buffer->pkt;
int64_t delta_dts = INT64_MIN;
int64_t top_dts = av_rescale_q(top_pkt->dts,
s->streams[top_pkt->stream_index]->time_base,
AV_TIME_BASE_Q);
for (i = 0; i < s->nb_streams; i++) {
int64_t last_dts;
const AVPacketList *last = s->streams[i]->last_in_packet_buffer;
if (!last)
continue;
last_dts = av_rescale_q(last->pkt.dts,
s->streams[i]->time_base,
AV_TIME_BASE_Q);
delta_dts = FFMAX(delta_dts, last_dts - top_dts);
}
/*缓冲区中最老与最新数据包的dts间隔，大于了最大间隔，则需要输出数据*/
if (delta_dts > s->max_interleave_delta) {
av_log(s, AV_LOG_DEBUG,
"Delay between the first packet and last packet in the "
"muxing queue is %"PRId64" > %"PRId64": forcing output\n",
delta_dts, s->max_interleave_delta);
flush = 1;
}
}
if (s->internal->packet_buffer &&
eof &&
(s->flags & AVFMT_FLAG_SHORTEST) &&
s->internal->shortest_end == AV_NOPTS_VALUE) {
AVPacket *top_pkt = &s->internal->packet_buffer->pkt;
s->internal->shortest_end = av_rescale_q(top_pkt->dts,
s->streams[top_pkt->stream_index]->time_base,
AV_TIME_BASE_Q);
}
if (s->internal->shortest_end != AV_NOPTS_VALUE) {
while (s->internal->packet_buffer) {
AVPacket *top_pkt = &s->internal->packet_buffer->pkt;
AVStream *st;
int64_t top_dts = av_rescale_q(top_pkt->dts,
s->streams[top_pkt->stream_index]->time_base,
AV_TIME_BASE_Q);
if (s->internal->shortest_end + 1 >= top_dts)
break;
pktl = s->internal->packet_buffer;
st = s->streams[pktl->pkt.stream_index];
s->internal->packet_buffer = pktl->next;
if (!s->internal->packet_buffer)
s->internal->packet_buffer_end = NULL;
if (st->last_in_packet_buffer == pktl)
st->last_in_packet_buffer = NULL;
av_packet_unref(&pktl->pkt);
av_freep(&pktl);
flush = 0;
}
}
if (stream_count && flush) {
AVStream *st;
pktl = s->internal->packet_buffer;
*out = pktl->pkt;
st = s->streams[out->stream_index];
s->internal->packet_buffer = pktl->next;
if (!s->internal->packet_buffer)
s->internal->packet_buffer_end = NULL;
if (st->last_in_packet_buffer == pktl)
st->last_in_packet_buffer = NULL;
av_freep(&pktl);
return 1;
} else {
av_init_packet(out);
return 0;
}
}
前面做了数据包的合格性检查，pts与dts的检查，缓冲到队列中，并在合适的时机，去发送数据包。接下来就该主角上场了。
/**
功能:发送数据包
AVFormatContext *s:封装上下文
AVPacket *pkt:需要发送的数据包
*/
static int write_packet(AVFormatContext *s, AVPacket *pkt)
{
int ret, did_split;
int64_t pts_backup, dts_backup;
pts_backup = pkt->pts;
dts_backup = pkt->dts;
/*调整pts与dts时间偏移，这个偏移值是可以灵活调整*/
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;
}
/*根据时间戳的不同情况，调整时间戳，并进行时间基的转化*/
if (s->avoid_negative_ts > 0) {
AVStream *st = s->streams[pkt->stream_index];
int64_t offset = st->mux_ts_offset;
int64_t ts = s->internal->avoid_negative_ts_use_pts ? pkt->pts : pkt->dts;
if (s->internal->offset == AV_NOPTS_VALUE && ts != AV_NOPTS_VALUE &&
(ts < 0 || s->avoid_negative_ts == AVFMT_AVOID_NEG_TS_MAKE_ZERO)) {
s->internal->offset = -ts;
s->internal->offset_timebase = st->time_base;
}
if (s->internal->offset != AV_NOPTS_VALUE && !offset) {
offset = st->mux_ts_offset =
av_rescale_q_rnd(s->internal->offset,
s->internal->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;
if (s->internal->avoid_negative_ts_use_pts) {
if (pkt->pts != AV_NOPTS_VALUE && pkt->pts < 0) {
av_log(s, AV_LOG_WARNING, "failed to avoid negative "
"pts %s in stream %d.\n"
"Try -avoid_negative_ts 1 as a possible workaround.\n",
av_ts2str(pkt->pts),
pkt->stream_index
);
}
} else {
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
);
}
}
}
#if FF_API_LAVF_MERGE_SD
FF_DISABLE_DEPRECATION_WARNINGS
did_split = av_packet_split_side_data(pkt);
FF_ENABLE_DEPRECATION_WARNINGS
#endif
/*发送数据包的头部数据*/
if (!s->internal->header_written) {
ret = s->internal->write_header_ret ? s->internal->write_header_ret : write_header_internal(s);
if (ret < 0)
goto fail;
}
if ((pkt->flags & AV_PKT_FLAG_UNCODED_FRAME)) {
AVFrame *frame = (AVFrame *)pkt->data;
av_assert0(pkt->size == UNCODED_FRAME_PACKET_SIZE);
/*发送没有编码的帧*/
ret = s->oformat->write_uncoded_frame(s, pkt->stream_index, &frame, 0);
av_frame_free(&frame);
} else {
/*根据不同的封装格式，调用不同发送数据包的函数
这里是一个函数指针，把封装协议，注册到封装上下文，实际的调用就在Flvenc.c里面
*/
ret = s->oformat->write_packet(s, pkt);
}
if (s->pb && ret >= 0) {
flush_if_needed(s);
if (s->pb->error < 0)
ret = s->pb->error;
}
fail:
#if FF_API_LAVF_MERGE_SD
FF_DISABLE_DEPRECATION_WARNINGS
if (did_split)
av_packet_merge_side_data(pkt);
FF_ENABLE_DEPRECATION_WARNINGS
#endif
if (ret < 0) {
pkt->pts = pts_backup;
pkt->dts = dts_backup;
}
return ret;
}
/*这里就是把具体实现的各类函数，进行注册的地方，如发送头部，发送数据包，发送数据包尾部*/
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,
.priv_class = &flv_muxer_class,
};
接下来就看看发送数据包的真正实现
static int flv_write_packet(AVFormatContext *s, AVPacket *pkt)
{
AVIOContext *pb = s->pb;
AVCodecParameters *par = s->streams[pkt->stream_index]->codecpar;
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;
int64_t cur_offset = avio_tell(pb);
if (par->codec_id == AV_CODEC_ID_VP6F || par->codec_id == AV_CODEC_ID_VP6A ||
par->codec_id == AV_CODEC_ID_VP6 || par->codec_id == AV_CODEC_ID_AAC)
flags_size = 2;
else if (par->codec_id == AV_CODEC_ID_H264 || par->codec_id == AV_CODEC_ID_MPEG4)
flags_size = 5;
else
flags_size = 1;
if (par->codec_id == AV_CODEC_ID_AAC || par->codec_id == AV_CODEC_ID_H264
|| par->codec_id == AV_CODEC_ID_MPEG4) {
int side_size = 0;
/*获取头部数据*/
uint8_t *side = av_packet_get_side_data(pkt, AV_PKT_DATA_NEW_EXTRADATA, &side_size);
if (side && side_size > 0 && (side_size != par->extradata_size || memcmp(side, par->extradata, side_size))) {
av_free(par->extradata);
par->extradata = av_mallocz(side_size + AV_INPUT_BUFFER_PADDING_SIZE);
if (!par->extradata) {
par->extradata_size = 0;
return AVERROR(ENOMEM);
}
memcpy(par->extradata, side, side_size);
par->extradata_size = side_size;
/*通过网络IO发送头部数据*/
flv_write_codec_header(s, par);
}
}
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;
if (s->event_flags & AVSTREAM_EVENT_FLAG_METADATA_UPDATED) {
write_metadata(s, ts);
s->event_flags &= ~AVSTREAM_EVENT_FLAG_METADATA_UPDATED;
}
/*刷新以前的数据，并标记新的数据开始，实际上就是以一组GOP为主去发送，使用关键帧去做判断*/
avio_write_marker(pb, av_rescale(ts, AV_TIME_BASE, 1000),
pkt->flags & AV_PKT_FLAG_KEY && (flv->video_par ? par->codec_type == AVMEDIA_TYPE_VIDEO : 1) ? AVIO_DATA_MARKER_SYNC_POINT : AVIO_DATA_MARKER_BOUNDARY_POINT);
switch (par->codec_type) {
case AVMEDIA_TYPE_VIDEO:
avio_w8(pb, FLV_TAG_TYPE_VIDEO);
flags = ff_codec_get_tag(flv_video_codec_ids, par->codec_id);
flags |= pkt->flags & AV_PKT_FLAG_KEY ? FLV_FRAME_KEY : FLV_FRAME_INTER;
break;
case AVMEDIA_TYPE_AUDIO:
flags = get_audio_flags(s, par);
av_assert0(size);
avio_w8(pb, FLV_TAG_TYPE_AUDIO);
break;
case AVMEDIA_TYPE_SUBTITLE:
case AVMEDIA_TYPE_DATA:
avio_w8(pb, FLV_TAG_TYPE_META);
break;
default:
return AVERROR(EINVAL);
}
if (par->codec_id == AV_CODEC_ID_H264 || par->codec_id == AV_CODEC_ID_MPEG4) {
/* check if extradata looks like mp4 formatted */
if (par->extradata_size > 0 && *(uint8_t*)par->extradata != 1)
/*寻找nal单元的起始码，对于nal单元不理解的，可以看看我前面的文章*/
if ((ret = ff_avc_parse_nal_units_buf(pkt->data, &data, &size)) < 0)
return ret;
} else if (par->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 (par->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);
}
avio_wb24(pb, size + flags_size);
avio_wb24(pb, ts & 0xFFFFFF);
avio_w8(pb, (ts >> 24) & 0x7F); // timestamps are 32 bits _signed_
avio_wb24(pb, flv->reserved);
/*拼接字幕*/
if (par->codec_type == AVMEDIA_TYPE_DATA ||
par->codec_type == AVMEDIA_TYPE_SUBTITLE ) {
int data_size;
int64_t metadata_size_pos = avio_tell(pb);
if (par->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);
avio_w8(pb,flags);
if (par->codec_id == AV_CODEC_ID_VP6)
avio_w8(pb,0);
if (par->codec_id == AV_CODEC_ID_VP6F || par->codec_id == AV_CODEC_ID_VP6A) {
if (par->extradata_size)
avio_w8(pb, par->extradata[0]);
else
avio_w8(pb, ((FFALIGN(par->width, 16) - par->width) << 4) |
(FFALIGN(par->height, 16) - par->height));
} else if (par->codec_id == AV_CODEC_ID_AAC)
//拼接音频数据
avio_w8(pb, 1); // AAC raw
else if (par->codec_id == AV_CODEC_ID_H264 || par->codec_id == AV_CODEC_ID_MPEG4) {
//拼接AVC的NALU单元
avio_w8(pb, 1); // AVC NALU
avio_wb24(pb, pkt->pts - pkt->dts);
}
/*通过网络发送正真的音视频数据*/
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);
}
if (flv->flags & FLV_ADD_KEYFRAME_INDEX) {
switch (par->codec_type) {
case AVMEDIA_TYPE_VIDEO:
flv->videosize += (avio_tell(pb) - cur_offset);
flv->lasttimestamp = flv->acurframeindex / flv->framerate;
if (pkt->flags & AV_PKT_FLAG_KEY) {
double ts = flv->acurframeindex / flv->framerate;
int64_t pos = cur_offset;
flv->lastkeyframetimestamp = flv->acurframeindex / flv->framerate;
flv->lastkeyframelocation = pos;
flv_append_keyframe_info(s, flv, ts, pos);
}
flv->acurframeindex++;
break;
case AVMEDIA_TYPE_AUDIO:
flv->audiosize += (avio_tell(pb) - cur_offset);
break;
default:
av_log(s, AV_LOG_WARNING, "par->codec_type is type = [%d]\n", par->codec_type);
break;
}
}
av_free(data);
return pb->error;
}
通过上面分析，通过网络发送数据到服务器，主要就是通过avio_write(pb, data ? data : pkt->data, size);所以下面继续详细剖析这个网络发送函数。
void avio_write(AVIOContext *s, const unsigned char *buf, int size)
{
/*不通过缓冲区,直接发送数据*/
if (s->direct && !s->update_checksum) {
avio_flush(s);
/*发数据*/
writeout(s, buf, size);
return;
}
/*通过缓冲区发数据*/
while (size > 0) {
int len = FFMIN(s->buf_end - s->buf_ptr, size);
memcpy(s->buf_ptr, buf, len);
s->buf_ptr += len;
if (s->buf_ptr >= s->buf_end)
/*本质还是调用writeout*/
flush_buffer(s);
buf += len;
size -= len;
}
}
static void writeout(AVIOContext *s, const uint8_t *data, int len)
{
if (!s->error) {
int ret = 0;
if (s->write_data_type)
ret = s->write_data_type(s->opaque, (uint8_t *)data,
len,
s->current_type,
s->last_time);
else if (s->write_packet)
/*使用不同协议的接口发送数据*/
ret = s->write_packet(s->opaque, (uint8_t *)data, len);
if (ret < 0) {
s->error = ret;
} else {
if (s->pos + len > s->written)
s->written = s->pos + len;
}
}
if (s->current_type == AVIO_DATA_MARKER_SYNC_POINT ||
s->current_type == AVIO_DATA_MARKER_BOUNDARY_POINT) {
s->current_type = AVIO_DATA_MARKER_UNKNOWN;
}
s->last_time = AV_NOPTS_VALUE;
s->writeout_count ++;
s->pos += len;
}
s->write_packet(s->opaque, (uint8_t *)data, len)，实际上这也是一个函数指针，使用不同的协议来发送数据，我这里就以Rtmp协议为例子。
int ffurl_write(URLContext *h, const unsigned char *buf, int size)
{
if (!(h->flags & AVIO_FLAG_WRITE))
return AVERROR(EIO);
/* 避免发送太大包 */
if (h->max_packet_size && size > h->max_packet_size)
return AVERROR(EIO);
/*这里会使用具体的协议去发送，如Rtmp发送数据*/
return retry_transfer_wrapper(h, (unsigned char *)buf, size, size,
(int (*)(struct URLContext *, uint8_t *, int))
h->prot->url_write);
}
static inline int retry_transfer_wrapper(URLContext *h, uint8_t *buf,
int size, int size_min,
int (*transfer_func)(URLContext *h,
uint8_t *buf,
int size))
{
int ret, len;
int fast_retries = 5;
int64_t wait_since = 0;
len = 0;
while (len < size_min) {
if (ff_check_interrupt(&h->interrupt_callback))
return AVERROR_EXIT;
/*回调Rtmp协议发送*/
ret = transfer_func(h, buf + len, size - len);
if (ret == AVERROR(EINTR))
continue;
if (h->flags & AVIO_FLAG_NONBLOCK)
return ret;
if (ret == AVERROR(EAGAIN)) {
ret = 0;
if (fast_retries) {
fast_retries--;
} else {
/*默认没有设置超时时间，默认是一直阻塞，所以当网络不好就会一直堵住，导致发送
数据包接口，一直无法返回。在哪里去设置超时时间？
*/
if (h->rw_timeout) {
if (!wait_since)
wait_since = av_gettime_relative();
else if (av_gettime_relative() > wait_since + h->rw_timeout)
return AVERROR(EIO);
}
av_usleep(1000);
}
} else if (ret < 1)
return (ret < 0 && ret != AVERROR_EOF) ? ret : len;
if (ret) {
fast_retries = FFMAX(fast_retries, 2);
wait_since = 0;
}
len += ret;
}
return len;
}
防止一直阻塞堵住发送数据包接口，使用超时时间，需要在各类协议初始化这里设置超时时间。
static int url_alloc_for_protocol(URLContext **puc, const URLProtocol *up,
const char *filename, int flags,
const AVIOInterruptCB *int_cb)
{
URLContext *uc;
int err;
#if CONFIG_NETWORK
if (up->flags & URL_PROTOCOL_FLAG_NETWORK && !ff_network_init())
return AVERROR(EIO);
#endif
if ((flags & AVIO_FLAG_READ) && !up->url_read) {
av_log(NULL, AV_LOG_ERROR,
"Impossible to open the '%s' protocol for reading\n", up->name);
return AVERROR(EIO);
}
if ((flags & AVIO_FLAG_WRITE) && !up->url_write) {
av_log(NULL, AV_LOG_ERROR,
"Impossible to open the '%s' protocol for writing\n", up->name);
return AVERROR(EIO);
}
uc = av_mallocz(sizeof(URLContext) + strlen(filename) + 1);
if (!uc) {
err = AVERROR(ENOMEM);
goto fail;
}
uc->av_class = &ffurl_context_class;
uc->filename = (char *)&uc[1];
strcpy(uc->filename, filename);
uc->prot = up;
uc->flags = flags;
uc->is_streamed = 0; /* default = not streamed */
uc->max_packet_size = 0; /* default: stream file */
/*这里就是设置超时时间，防止卡死的办法*/
uc->rw_timeout = 3000;//3s
if (up->priv_data_size) {
uc->priv_data = av_mallocz(up->priv_data_size);
if (!uc->priv_data) {
err = AVERROR(ENOMEM);
goto fail;
}
if (up->priv_data_class) {
int proto_len= strlen(up->name);
char *start = strchr(uc->filename, ',');
*(const AVClass **)uc->priv_data = up->priv_data_class;
av_opt_set_defaults(uc->priv_data);
if(!strncmp(up->name, uc->filename, proto_len) && uc->filename + proto_len == start){
int ret= 0;
char *p= start;
char sep= *++p;
char *key, *val;
p++;
if (strcmp(up->name, "subfile"))
ret = AVERROR(EINVAL);
while(ret >= 0 && (key= strchr(p, sep)) && p<key && (val = strchr(key+1, sep))){
*val= *key= 0;
if (strcmp(p, "start") && strcmp(p, "end")) {
ret = AVERROR_OPTION_NOT_FOUND;
} else
ret= av_opt_set(uc->priv_data, p, key+1, 0);
if (ret == AVERROR_OPTION_NOT_FOUND)
av_log(uc, AV_LOG_ERROR, "Key '%s' not found.\n", p);
*val= *key= sep;
p= val+1;
}
if(ret<0 || p!=key){
av_log(uc, AV_LOG_ERROR, "Error parsing options string %s\n", start);
av_freep(&uc->priv_data);
av_freep(&uc);
err = AVERROR(EINVAL);
goto fail;
}
memmove(start, key+1, strlen(key));
}
}
}
if (int_cb)
uc->interrupt_callback = *int_cb;
*puc = uc;
return 0;
fail:
*puc = NULL;
if (uc)
av_freep(&uc->priv_data);
av_freep(&uc);
#if CONFIG_NETWORK
if (up->flags & URL_PROTOCOL_FLAG_NETWORK)
ff_network_close();
#endif
return err;
}
这个h->prot->url_write是怎样注册呢?实际就是在里注册，并调用rtmp_write函数。
const URLProtocol ff_##flavor##_protocol = { \
.name = #flavor, \
.url_open2 = rtmp_open, \
.url_read = rtmp_read, \
.url_read_seek = rtmp_seek, \
.url_read_pause = rtmp_pause, \
.url_write = rtmp_write, \
.url_close = rtmp_close, \
.priv_data_size = sizeof(RTMPContext), \
.flags = URL_PROTOCOL_FLAG_NETWORK, \
.priv_data_class= &flavor##_class, \
};
static int rtmp_write(URLContext *s, const uint8_t *buf, int size)
{
RTMPContext *rt = s->priv_data;
int size_temp = size;
int pktsize, pkttype, copy;
uint32_t ts;
const uint8_t *buf_temp = buf;
uint8_t c;
int ret;
do {
if (rt->skip_bytes) {
int skip = FFMIN(rt->skip_bytes, size_temp);
buf_temp += skip;
size_temp -= skip;
rt->skip_bytes -= skip;
continue;
}
/*拼接RTMP头*/
if (rt->flv_header_bytes < RTMP_HEADER) {
const uint8_t *header = rt->flv_header;
int channel = RTMP_AUDIO_CHANNEL;
copy = FFMIN(RTMP_HEADER - rt->flv_header_bytes, size_temp);
bytestream_get_buffer(&buf_temp, rt->flv_header + rt->flv_header_bytes, copy);
rt->flv_header_bytes += copy;
size_temp -= copy;
if (rt->flv_header_bytes < RTMP_HEADER)
break;
pkttype = bytestream_get_byte(&header);
pktsize = bytestream_get_be24(&header);
ts = bytestream_get_be24(&header);
ts |= bytestream_get_byte(&header) << 24;
bytestream_get_be24(&header);
rt->flv_size = pktsize;
if (pkttype == RTMP_PT_VIDEO)
channel = RTMP_VIDEO_CHANNEL;
if (((pkttype == RTMP_PT_VIDEO || pkttype == RTMP_PT_AUDIO) && ts == 0) ||
pkttype == RTMP_PT_NOTIFY) {
if ((ret = ff_rtmp_check_alloc_array(&rt->prev_pkt[1],
&rt->nb_prev_pkt[1],
channel)) < 0)
return ret;
/*强制发送12字节*/
rt->prev_pkt[1][channel].channel_id = 0;
}
............
/*Rtmp发送数据*/
if ((ret = rtmp_send_packet(rt, &rt->out_pkt, 0)) < 0)
return ret;
rt->flv_size = 0;
rt->flv_off = 0;
rt->flv_header_bytes = 0;
rt->flv_nb_packets++;
}
} while (buf_temp - buf < size);
if (rt->flv_nb_packets < rt->flush_interval)
return size;
rt->flv_nb_packets = 0;
/* set stream into nonblocking mode */
rt->stream->flags |= AVIO_FLAG_NONBLOCK;
/* try to read one byte from the stream */
ret = ffurl_read(rt->stream, &c, 1);
/* switch the stream back into blocking mode */
rt->stream->flags &= ~AVIO_FLAG_NONBLOCK;
if (ret == AVERROR(EAGAIN)) {
/* no incoming data to handle */
return size;
} else if (ret < 0) {
return ret;
} else if (ret == 1) {
RTMPPacket rpkt = { 0 };
if ((ret = ff_rtmp_packet_read_internal(rt->stream, &rpkt,
rt->in_chunk_size,
&rt->prev_pkt[0],
&rt->nb_prev_pkt[0], c)) <= 0)
return ret;
if ((ret = rtmp_parse_result(s, rt, &rpkt)) < 0)
return ret;
ff_rtmp_packet_destroy(&rpkt);
}
return size;
}
int ff_rtmp_packet_write(URLContext *h, RTMPPacket *pkt,
int chunk_size, RTMPPacket **prev_pkt_ptr,
int *nb_prev_pkt)
{
uint8_t pkt_hdr[16], *p = pkt_hdr;
int mode = RTMP_PS_TWELVEBYTES;
int off = 0;
int written = 0;
int ret;
RTMPPacket *prev_pkt;
int use_delta; // flag if using timestamp delta, not RTMP_PS_TWELVEBYTES
uint32_t timestamp; // full 32-bit timestamp or delta value
if ((ret = ff_rtmp_check_alloc_array(prev_pkt_ptr, nb_prev_pkt,
pkt->channel_id)) < 0)
return ret;
prev_pkt = *prev_pkt_ptr;
//if channel_id = 0, this is first presentation of prev_pkt, send full hdr.
use_delta = prev_pkt[pkt->channel_id].channel_id &&
pkt->extra == prev_pkt[pkt->channel_id].extra &&
pkt->timestamp >= prev_pkt[pkt->channel_id].timestamp;
timestamp = pkt->timestamp;
if (use_delta) {
timestamp -= prev_pkt[pkt->channel_id].timestamp;
}
if (timestamp >= 0xFFFFFF) {
pkt->ts_field = 0xFFFFFF;
} else {
pkt->ts_field = timestamp;
}
if (use_delta) {
if (pkt->type == prev_pkt[pkt->channel_id].type &&
pkt->size == prev_pkt[pkt->channel_id].size) {
mode = RTMP_PS_FOURBYTES;
if (pkt->ts_field == prev_pkt[pkt->channel_id].ts_field)
mode = RTMP_PS_ONEBYTE;
} else {
mode = RTMP_PS_EIGHTBYTES;
}
}
if (pkt->channel_id < 64) {
bytestream_put_byte(&p, pkt->channel_id | (mode << 6));
} else if (pkt->channel_id < 64 + 256) {
bytestream_put_byte(&p, 0 | (mode << 6));
bytestream_put_byte(&p, pkt->channel_id - 64);
} else {
bytestream_put_byte(&p, 1 | (mode << 6));
bytestream_put_le16(&p, pkt->channel_id - 64);
}
if (mode != RTMP_PS_ONEBYTE) {
bytestream_put_be24(&p, pkt->ts_field);
if (mode != RTMP_PS_FOURBYTES) {
bytestream_put_be24(&p, pkt->size);
bytestream_put_byte(&p, pkt->type);
if (mode == RTMP_PS_TWELVEBYTES)
bytestream_put_le32(&p, pkt->extra);
}
}
if (pkt->ts_field == 0xFFFFFF)
bytestream_put_be32(&p, timestamp);
// save history
prev_pkt[pkt->channel_id].channel_id = pkt->channel_id;
prev_pkt[pkt->channel_id].type = pkt->type;
prev_pkt[pkt->channel_id].size = pkt->size;
prev_pkt[pkt->channel_id].timestamp = pkt->timestamp;
prev_pkt[pkt->channel_id].ts_field = pkt->ts_field;
prev_pkt[pkt->channel_id].extra = pkt->extra;
if ((ret = ffurl_write(h, pkt_hdr, p - pkt_hdr)) < 0)
return ret;
written = p - pkt_hdr + pkt->size;
while (off < pkt->size) {
int towrite = FFMIN(chunk_size, pkt->size - off);
/*这里又回到之前AVIO.c的发送，就是如此往复的去发送数据*/
if ((ret = ffurl_write(h, pkt->data + off, towrite)) < 0)
return ret;
off += towrite;
if (off < pkt->size) {
uint8_t marker = 0xC0 | pkt->channel_id;
if ((ret = ffurl_write(h, &marker, 1)) < 0)
return ret;
written++;
if (pkt->ts_field == 0xFFFFFF) {
uint8_t ts_header[4];
AV_WB32(ts_header, timestamp);
if ((ret = ffurl_write(h, ts_header, 4)) < 0)
return ret;
written += 4;
}
}
}
return written;
}
由于Rtmp协议也是基于Tcp协议，所以在rtmp_open这个函数里，就已经使用Tcp去连接服务器。实际上最终的收发数据，都是通过Tcp。
static int rtmp_open(URLContext *s, const char *uri, int flags, AVDictionary **opts)
{
RTMPContext *rt = s->priv_data;
char proto[8], hostname[256], path[1024], auth[100], *fname;
char *old_app, *qmark, *n, fname_buffer[1024];
uint8_t buf[2048];
int port;
int ret;
if (rt->listen_timeout > 0)
rt->listen = 1;
rt->is_input = !(flags & AVIO_FLAG_WRITE);
av_url_split(proto, sizeof(proto), auth, sizeof(auth),
hostname, sizeof(hostname), &port,
path, sizeof(path), s->filename);
n = strchr(path, ' ');
if (n) {
av_log(s, AV_LOG_WARNING,
"Detected librtmp style URL parameters, these aren't supported "
"by the libavformat internal RTMP handler currently enabled. "
"See the documentation for the correct way to pass parameters.\n");
*n = '\0'; // Trim not supported part
}
if (auth[0]) {
char *ptr = strchr(auth, ':');
if (ptr) {
*ptr = '\0';
av_strlcpy(rt->username, auth, sizeof(rt->username));
av_strlcpy(rt->password, ptr + 1, sizeof(rt->password));
}
}
if (rt->listen && strcmp(proto, "rtmp")) {
av_log(s, AV_LOG_ERROR, "rtmp_listen not available for %s\n",
proto);
return AVERROR(EINVAL);
}
if (!strcmp(proto, "rtmpt") || !strcmp(proto, "rtmpts")) {
if (!strcmp(proto, "rtmpts"))
av_dict_set(opts, "ffrtmphttp_tls", "1", 1);
/* open the http tunneling connection */
ff_url_join(buf, sizeof(buf), "ffrtmphttp", NULL, hostname, port, NULL);
} else if (!strcmp(proto, "rtmps")) {
/* open the tls connection */
if (port < 0)
port = RTMPS_DEFAULT_PORT;
ff_url_join(buf, sizeof(buf), "tls", NULL, hostname, port, NULL);
} else if (!strcmp(proto, "rtmpe") || (!strcmp(proto, "rtmpte"))) {
if (!strcmp(proto, "rtmpte"))
av_dict_set(opts, "ffrtmpcrypt_tunneling", "1", 1);
/* open the encrypted connection */
ff_url_join(buf, sizeof(buf), "ffrtmpcrypt", NULL, hostname, port, NULL);
rt->encrypted = 1;
} else {
/* 打开Tcp连接 */
if (port < 0)
port = RTMP_DEFAULT_PORT;
if (rt->listen)
//一直等待这个连接线程结束
ff_url_join(buf, sizeof(buf), "tcp", NULL, hostname, port,
"?listen&listen_timeout=%d",
rt->listen_timeout * 1000);
else
ff_url_join(buf, sizeof(buf), "tcp", NULL, hostname, port, NULL);
}
reconnect:
if ((ret = ffurl_open_whitelist(&rt->stream, buf, AVIO_FLAG_READ_WRITE,
&s->interrupt_callback, opts,
s->protocol_whitelist, s->protocol_blacklist, s)) < 0) {
av_log(s , AV_LOG_ERROR, "Cannot open connection %s\n", buf);
goto fail;
}
if (rt->swfverify) {
if ((ret = rtmp_calc_swfhash(s)) < 0)
goto fail;
}
rt->state = STATE_START;
if (!rt->listen && (ret = rtmp_handshake(s, rt)) < 0)
goto fail;
if (rt->listen && (ret = rtmp_server_handshake(s, rt)) < 0)
goto fail;
rt->out_chunk_size = 128;
rt->in_chunk_size = 128; // Probably overwritten later
rt->state = STATE_HANDSHAKED;
// Keep the application name when it has been defined by the user.
old_app = rt->app;
rt->app = av_malloc(APP_MAX_LENGTH);
if (!rt->app) {
ret = AVERROR(ENOMEM);
goto fail;
}
//extract "app" part from path
qmark = strchr(path, '?');
if (qmark && strstr(qmark, "slist=")) {
char* amp;
// After slist we have the playpath, the full path is used as app
av_strlcpy(rt->app, path + 1, APP_MAX_LENGTH);
fname = strstr(path, "slist=") + 6;
// Strip any further query parameters from fname
amp = strchr(fname, '&');
if (amp) {
av_strlcpy(fname_buffer, fname, FFMIN(amp - fname + 1,
sizeof(fname_buffer)));
fname = fname_buffer;
}
} else if (!strncmp(path, "/ondemand/", 10)) {
fname = path + 10;
memcpy(rt->app, "ondemand", 9);
} else {
char *next = *path ? path + 1 : path;
char *p = strchr(next, '/');
if (!p) {
if (old_app) {
// If name of application has been defined by the user, assume that
// playpath is provided in the URL
fname = next;
} else {
fname = NULL;
av_strlcpy(rt->app, next, APP_MAX_LENGTH);
}
} else {
// make sure we do not mismatch a playpath for an application instance
char *c = strchr(p + 1, ':');
fname = strchr(p + 1, '/');
if (!fname || (c && c < fname)) {
fname = p + 1;
av_strlcpy(rt->app, path + 1, FFMIN(p - path, APP_MAX_LENGTH));
} else {
fname++;
av_strlcpy(rt->app, path + 1, FFMIN(fname - path - 1, APP_MAX_LENGTH));
}
}
}
if (old_app) {
// The name of application has been defined by the user, override it.
if (strlen(old_app) >= APP_MAX_LENGTH) {
ret = AVERROR(EINVAL);
goto fail;
}
av_free(rt->app);
rt->app = old_app;
}
if (!rt->playpath) {
rt->playpath = av_malloc(PLAYPATH_MAX_LENGTH);
if (!rt->playpath) {
ret = AVERROR(ENOMEM);
goto fail;
}
if (fname) {
int len = strlen(fname);
if (!strchr(fname, ':') && len >= 4 &&
(!strcmp(fname + len - 4, ".f4v") ||
!strcmp(fname + len - 4, ".mp4"))) {
memcpy(rt->playpath, "mp4:", 5);
} else {
if (len >= 4 && !strcmp(fname + len - 4, ".flv"))
fname[len - 4] = '\0';
rt->playpath[0] = 0;
}
av_strlcat(rt->playpath, fname, PLAYPATH_MAX_LENGTH);
} else {
rt->playpath[0] = '\0';
}
}
if (!rt->tcurl) {
rt->tcurl = av_malloc(TCURL_MAX_LENGTH);
if (!rt->tcurl) {
ret = AVERROR(ENOMEM);
goto fail;
}
ff_url_join(rt->tcurl, TCURL_MAX_LENGTH, proto, NULL, hostname,
port, "/%s", rt->app);
}
if (!rt->flashver) {
rt->flashver = av_malloc(FLASHVER_MAX_LENGTH);
if (!rt->flashver) {
ret = AVERROR(ENOMEM);
goto fail;
}
if (rt->is_input) {
snprintf(rt->flashver, FLASHVER_MAX_LENGTH, "%s %d,%d,%d,%d",
RTMP_CLIENT_PLATFORM, RTMP_CLIENT_VER1, RTMP_CLIENT_VER2,
RTMP_CLIENT_VER3, RTMP_CLIENT_VER4);
} else {
snprintf(rt->flashver, FLASHVER_MAX_LENGTH,
"FMLE/3.0 (compatible; %s)", LIBAVFORMAT_IDENT);
}
}
rt->receive_report_size = 1048576;
rt->bytes_read = 0;
rt->has_audio = 0;
rt->has_video = 0;
rt->received_metadata = 0;
rt->last_bytes_read = 0;
rt->max_sent_unacked = 2500000;
rt->duration = 0;
av_log(s, AV_LOG_DEBUG, "Proto = %s, path = %s, app = %s, fname = %s\n",
proto, path, rt->app, rt->playpath);
if (!rt->listen) {
if ((ret = gen_connect(s, rt)) < 0)
goto fail;
} else {
if ((ret = read_connect(s, s->priv_data)) < 0)
goto fail;
}
do {
ret = get_packet(s, 1);
} while (ret == AVERROR(EAGAIN));
if (ret < 0)
goto fail;
if (rt->do_reconnect) {
int i;
ffurl_close(rt->stream);
rt->stream = NULL;
rt->do_reconnect = 0;
rt->nb_invokes = 0;
for (i = 0; i < 2; i++)
memset(rt->prev_pkt[i], 0,
sizeof(**rt->prev_pkt) * rt->nb_prev_pkt[i]);
free_tracked_methods(rt);
goto reconnect;
}
if (rt->is_input) {
// generate FLV header for demuxer
rt->flv_size = 13;
if ((ret = av_reallocp(&rt->flv_data, rt->flv_size)) < 0)
goto fail;
rt->flv_off = 0;
memcpy(rt->flv_data, "FLV\1\0\0\0\0\011\0\0\0\0", rt->flv_size);
// Read packets until we reach the first A/V packet or read metadata.
// If there was a metadata package in front of the A/V packets, we can
// build the FLV header from this. If we do not receive any metadata,
// the FLV decoder will allocate the needed streams when their first
// audio or video packet arrives.
while (!rt->has_audio && !rt->has_video && !rt->received_metadata) {
if ((ret = get_packet(s, 0)) < 0)
goto fail;
}
// Either after we have read the metadata or (if there is none) the
// first packet of an A/V stream, we have a better knowledge about the
// streams, so set the FLV header accordingly.
if (rt->has_audio) {
rt->flv_data[4] |= FLV_HEADER_FLAG_HASAUDIO;
}
if (rt->has_video) {
rt->flv_data[4] |= FLV_HEADER_FLAG_HASVIDEO;
}
// If we received the first packet of an A/V stream and no metadata but
// the server returned a valid duration, create a fake metadata packet
// to inform the FLV decoder about the duration.
if (!rt->received_metadata && rt->duration > 0) {
if ((ret = inject_fake_duration_metadata(rt)) < 0)
goto fail;
}
} else {
rt->flv_size = 0;
rt->flv_data = NULL;
rt->flv_off = 0;
rt->skip_bytes = 13;
}
s->max_packet_size = rt->stream->max_packet_size;
s->is_streamed = 1;
return 0;
fail:
av_dict_free(opts);
rtmp_close(s);
return ret;
}
我们之前设置了超时时间，解决了网络不好一直阻塞的办法，那我们经过这么久的分析，还没找到在哪里堵塞？那现在就到Tcp.c去找找。
/* return non zero if error */
static int tcp_open(URLContext *h, const char *uri, int flags)
{
struct addrinfo hints = { 0 }, *ai, *cur_ai;
int port, fd = -1;
TCPContext *s = h->priv_data;
const char *p;
char buf[256];
int ret;
char hostname[1024],proto[1024],path[1024];
char portstr[10];
/*默认TCp连接时间*/
s->open_timeout = 5000000;
av_url_split(proto, sizeof(proto), NULL, 0, hostname, sizeof(hostname),
&port, path, sizeof(path), uri);
if (strcmp(proto, "tcp"))
return AVERROR(EINVAL);
if (port <= 0 || port >= 65536) {
av_log(h, AV_LOG_ERROR, "Port missing in uri\n");
return AVERROR(EINVAL);
}
p = strchr(uri, '?');
if (p) {
if (av_find_info_tag(buf, sizeof(buf), "listen", p)) {
char *endptr = NULL;
s->listen = strtol(buf, &endptr, 10);
/* assume if no digits were found it is a request to enable it */
if (buf == endptr)
s->listen = 1;
}
if (av_find_info_tag(buf, sizeof(buf), "timeout", p)) {
s->rw_timeout = strtol(buf, NULL, 10);
}
if (av_find_info_tag(buf, sizeof(buf), "listen_timeout", p)) {
s->listen_timeout = strtol(buf, NULL, 10);
}
}
if (s->rw_timeout >= 0) {
//这里就是把之前的超时时间设置进来了
s->open_timeout =
h->rw_timeout = s->rw_timeout;
}
hints.ai_family = AF_UNSPEC;
hints.ai_socktype = SOCK_STREAM;
snprintf(portstr, sizeof(portstr), "%d", port);
if (s->listen)
hints.ai_flags |= AI_PASSIVE;
if (!hostname[0])
ret = getaddrinfo(NULL, portstr, &hints, &ai);
else
ret = getaddrinfo(hostname, portstr, &hints, &ai);
if (ret) {
av_log(h, AV_LOG_ERROR,
"Failed to resolve hostname %s: %s\n",
hostname, gai_strerror(ret));
return AVERROR(EIO);
}
cur_ai = ai;
restart:
#if HAVE_STRUCT_SOCKADDR_IN6
// workaround for IOS9 getaddrinfo in IPv6 only network use hardcode IPv4 address can not resolve port number.
if (cur_ai->ai_family == AF_INET6){
struct sockaddr_in6 * sockaddr_v6 = (struct sockaddr_in6 *)cur_ai->ai_addr;
if (!sockaddr_v6->sin6_port){
sockaddr_v6->sin6_port = htons(port);
}
}
#endif
fd = ff_socket(cur_ai->ai_family,
cur_ai->ai_socktype,
cur_ai->ai_protocol);
if (fd < 0) {
ret = ff_neterrno();
goto fail;
}
/* Set the socket's send or receive buffer sizes, if specified.
If unspecified or setting fails, system default is used. */
//默认收发一次数据是8.5K,频繁收发，网络开销会很大
//这里最好自己去设置网络发送和接收的缓冲区，否则使用默认的协议栈缓冲区，因为频繁发送数据会导致网络开销大。
if (s->recv_buffer_size > 0) {
setsockopt (fd, SOL_SOCKET, SO_RCVBUF, &s->recv_buffer_size, sizeof (s->recv_buffer_size));
}
if (s->send_buffer_size > 0) {
setsockopt (fd, SOL_SOCKET, SO_SNDBUF, &s->send_buffer_size, sizeof (s->send_buffer_size));
}
if (s->listen == 2) {
// multi-client
if ((ret = ff_listen(fd, cur_ai->ai_addr, cur_ai->ai_addrlen)) < 0)
goto fail1;
} else if (s->listen == 1) {
// single client
if ((ret = ff_listen_bind(fd, cur_ai->ai_addr, cur_ai->ai_addrlen,
s->listen_timeout, h)) < 0)
goto fail1;
// Socket descriptor already closed here. Safe to overwrite to client one.
fd = ret;
} else {
if ((ret = ff_listen_connect(fd, cur_ai->ai_addr, cur_ai->ai_addrlen,
s->open_timeout / 1000, h, !!cur_ai->ai_next)) < 0) {
if (ret == AVERROR_EXIT)
goto fail1;
else
goto fail;
}
}
h->is_streamed = 1;
s->fd = fd;
freeaddrinfo(ai);
return 0;
fail:
if (cur_ai->ai_next) {
/* Retry with the next sockaddr */
cur_ai = cur_ai->ai_next;
if (fd >= 0)
closesocket(fd);
ret = 0;
goto restart;
}
fail1:
if (fd >= 0)
closesocket(fd);
freeaddrinfo(ai);
return ret;
}
tcp发送数据干了什么呢?
static int tcp_write(URLContext *h, const uint8_t *buf, int size)
{
TCPContext *s = h->priv_data;
int ret;
if (!(h->flags & AVIO_FLAG_NONBLOCK)) {
//阻塞或超时会调用这个方法
ret = ff_network_wait_fd_timeout(s->fd, 1, h->rw_timeout, &h->interrupt_callback);
if (ret)
return ret;
}
//是不是很熟悉啊，这就是Tcp协议的原生接口，不管上面封装怎么变，本质还是没有变
ret = send(s->fd, buf, size, MSG_NOSIGNAL);
return ret < 0 ? ff_neterrno() : ret;
}
int ff_network_wait_fd_timeout(int fd, int write, int64_t timeout, AVIOInterruptCB *int_cb)
{
int ret;
int64_t wait_start = 0;
while (1) {
if (ff_check_interrupt(int_cb))
return AVERROR_EXIT;
ret = ff_network_wait_fd(fd, write);
if (ret != AVERROR(EAGAIN))
return ret;
/*如果没有超时时间，这里就不会进来，也就不会返回超时*/
if (timeout > 0) {
if (!wait_start)
wait_start = av_gettime_relative();
else if (av_gettime_relative() - wait_start > timeout)
return AVERROR(ETIMEDOUT);
}
}
}
int ff_network_wait_fd(int fd, int write)
{
//输入还是输出呢
int ev = write ? POLLOUT : POLLIN;
struct pollfd p = { .fd = fd, .events = ev, .revents = 0 };
int ret;
//使用的轮询办法
ret = poll(&p, 1, POLLING_TIME);
return ret < 0 ? ff_neterrno() : p.revents & (ev | POLLERR | POLLHUP) ? 0 : AVERROR(EAGAIN);
}
/**多么熟悉的方法*/
int ff_poll(struct pollfd *fds, nfds_t numfds, int timeout)
{
fd_set read_set;
fd_set write_set;
fd_set exception_set;
nfds_t i;
int n;
int rc;
#if HAVE_WINSOCK2_H
if (numfds >= FD_SETSIZE) {
errno = EINVAL;
return -1;
}
#endif /* HAVE_WINSOCK2_H */
FD_ZERO(&read_set);
FD_ZERO(&write_set);
FD_ZERO(&exception_set);
n = 0;
for (i = 0; i < numfds; i++) {
if (fds[i].fd < 0)
continue;
#if !HAVE_WINSOCK2_H
if (fds[i].fd >= FD_SETSIZE) {
errno = EINVAL;
return -1;
}
#endif /* !HAVE_WINSOCK2_H */
if (fds[i].events & POLLIN)
FD_SET(fds[i].fd, &read_set);
if (fds[i].events & POLLOUT)
FD_SET(fds[i].fd, &write_set);
if (fds[i].events & POLLERR)
FD_SET(fds[i].fd, &exception_set);
if (fds[i].fd >= n)
n = fds[i].fd + 1;
}
if (n == 0)
/* Hey!? Nothing to poll, in fact!!! */
return 0;
if (timeout < 0) {
/*如果网络不好，这里的写事件就不会触发，也就一直会堵塞在这里,这就是堵塞的本质*/
rc = select(n, &read_set, &write_set, &exception_set, NULL);
} else {
struct timeval tv;
tv.tv_sec = timeout / 1000;
tv.tv_usec = 1000 * (timeout % 1000);
/*设置了超时时间，就不会堵塞,如果使用epoll，这里也是可以优化，提高效率。特别是针对海量事件*/
rc = select(n, &read_set, &write_set, &exception_set, &tv);
}
if (rc < 0)
return rc;
for (i = 0; i < numfds; i++) {
fds[i].revents = 0;
if (FD_ISSET(fds[i].fd, &read_set))
fds[i].revents |= POLLIN;
if (FD_ISSET(fds[i].fd, &write_set))
fds[i].revents |= POLLOUT;
if (FD_ISSET(fds[i].fd, &exception_set))
fds[i].revents |= POLLERR;
}
return rc;
}
#endif /* !HAVE_POLL_H */
3.总结
经过长篇大论的分析，终于优化了网络阻塞开始的问题，找到了问题的本质。如果你能够看到这里，说明你也能够看懂了。最后总结下，关于发送数据，优化了3处，第一个是增加发送数据大小，第二个增加超时时间，解决阻塞，第三个降低网络负载，提高效率。如果要看懂这里面的关系，需要结合源码，反复多看几次，就会看透本质，实际还是linux最原始的那一套。希望这篇文章能够帮助到各位，谢谢!
原文链接：ffmpeg源码优化之推流发送篇_antonio@qaa的博客-CSDN博客
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