前言
上一节中已经介绍了rtmp流监听，一旦有rtmp请求之后处理，单独为该链接开启一个线程等等，这一节我们来分析rtmp请求到来之后，除了创建线程之外，还要做哪些工作。同样，只是代码框架分析。
首先对Publish做的工作大概描述一下：
1、Rtmp协议几乎是从推送端过来的唯一协议（仅限于srs2.0及以下版本），因为Rtmp协议时效性。
2、建立Rtmp协议链接：握手、建立网络连接、建立流通道、Publish/Play。
3、接受Rtmp信息：Rtmp协议解析。
4、成功接受到Rtmp信息处理：缓存、抖动矫正、分发到消费者堆栈（Play消费者，Forward消费者）、进行HLS切片等等。
这一讲内容只讲述RTMP协议Publish过程，SRS源码相关其他总结：
SRS(simple-rtmp-server)流媒体服务器源码分析--系统启动
SRS(simple-rtmp-server)流媒体服务器源码分析--RTMP消息play
SRS(simple-rtmp-server)流媒体服务器源码分析--RTMP信息Publish
SRS(simple-rtmp-server)流媒体服务器源码分析--HLS切片
Rtmp信息接受框架如下：
一、进入循环
接着上一节rtmp链接分析
int SrsConnection::cycle()
{
int ret = ERROR_SUCCESS;
_srs_context->generate_id();
id = _srs_context->get_id();
ip = srs_get_peer_ip(st_netfd_fileno(stfd));
srs_trace("ip:%s", ip);
ret = do_cycle();
// if socket io error, set to closed.
if (srs_is_client_gracefully_close(ret)) {
ret = ERROR_SOCKET_CLOSED;
}
// success.
if (ret == ERROR_SUCCESS) {
srs_trace("client finished.");
}
// client close peer.
if (ret == ERROR_SOCKET_CLOSED) {
srs_warn("client disconnect peer. ret=%d", ret);
}
return ERROR_SUCCESS;
}进来之后获取客户端ip，进入do_cycle()函数，这里也要同样注意，在理解纯虚函数上是我的缺点。
二、rtmp协议链接
RTMP协议是应用层协议，是要靠底层可靠的传输层协议（通常是TCP）来保证信息传输的可靠性的。在基于传输层协议的链接建立完成后，RTMP协议也要客户端和服务器通过“握手”来建立基于传输层链接之上的RTMP Connection链接，在Connection链接上会传输一些控制信息，如SetChunkSize,SetACKWindowSize。其中CreateStream命令会创建一个Stream链接，用于传输具体的音视频数据和控制这些信息传输的命令信息。RTMP协议传输时会对数据做自己的格式化，这种格式的消息我们称之为RTMP Message，而实际传输的时候为了更好地实现多路复用、分包和信息的公平性，发送端会把Message划分为带有Message ID的Chunk，每个Chunk可能是一个单独的Message，也可能是Message的一部分，在接受端会根据chunk中包含的data的长度，messageid和message的长度把chunk还原成完整的Message，从而实现信息的收发。
Rtmp协议规定：
（1）、 握手：固定步骤，RTMP连接都是以握手作为开始。
（2）、 建立连接：用于建立客户端和服务端之间的“网络连接”
（3）、 建立流：用于建立客户端和服务端之间的“网络流”，即多媒体通道。
（4）、 Publish/play：推流或者播放
1、握手 建立网络连接
有关rtmp协议握手内容，可以参考rtmp协议博客：http://blog.csdn.net/ManagerUser/article/category/6996313
直接来看建立网络连接：
// TODO: return detail message when error for client.
int SrsRtmpConn::do_cycle()
{
int ret = ERROR_SUCCESS;
srs_trace("RTMP client ip=%s", ip.c_str());
rtmp->set_recv_timeout(SRS_CONSTS_RTMP_RECV_TIMEOUT_US);
rtmp->set_send_timeout(SRS_CONSTS_RTMP_SEND_TIMEOUT_US);
//1、rtmp握手
if ((ret = rtmp->handshake()) != ERROR_SUCCESS) {
srs_error("rtmp handshake failed. ret=%d", ret);
return ret;
}
srs_verbose("rtmp handshake success");
//2、rtmp建立网络连接
if ((ret = rtmp->connect_app(req)) != ERROR_SUCCESS) {
srs_error("rtmp connect vhost/app failed. ret=%d", ret);
return ret;
}
srs_verbose("rtmp connect app success");
// set client ip to request.
req->ip = ip;
// discovery vhost, resolve the vhost from config
SrsConfDirective* parsed_vhost = _srs_config->get_vhost(req->vhost);
if (parsed_vhost) {
req->vhost = parsed_vhost->arg0();
}
srs_info("discovery app success. schema=%s, vhost=%s, port=%s, app=%s",
req->schema.c_str(), req->vhost.c_str(), req->port.c_str(), req->app.c_str());
if (req->schema.empty() || req->vhost.empty() || req->port.empty() || req->app.empty()) {
ret = ERROR_RTMP_REQ_TCURL;
srs_error("discovery tcUrl failed. "
"tcUrl=%s, schema=%s, vhost=%s, port=%s, app=%s, ret=%d",
req->tcUrl.c_str(), req->schema.c_str(), req->vhost.c_str(), req->port.c_str(), req->app.c_str(), ret);
return ret;
}
// check vhost
if ((ret = check_vhost()) != ERROR_SUCCESS) {
srs_error("check vhost failed. ret=%d", ret);
return ret;
}
srs_verbose("check vhost success.");
srs_trace("connect app, "
"tcUrl=%s, pageUrl=%s, swfUrl=%s, schema=%s, vhost=%s, port=%s, app=%s, args=%s",
req->tcUrl.c_str(), req->pageUrl.c_str(), req->swfUrl.c_str(),
req->schema.c_str(), req->vhost.c_str(), req->port.c_str(),
req->app.c_str(), (req->args? "(obj)":"null"));
// show client identity
if(req->args) {
std::string srs_version;
std::string srs_server_ip;
int srs_pid = 0;
int srs_id = 0;
SrsAmf0Any* prop = NULL;
if ((prop = req->args->ensure_property_string("srs_version")) != NULL) {
srs_version = prop->to_str();
}
if ((prop = req->args->ensure_property_string("srs_server_ip")) != NULL) {
srs_server_ip = prop->to_str();
}
if ((prop = req->args->ensure_property_number("srs_pid")) != NULL) {
srs_pid = (int)prop->to_number();
}
if ((prop = req->args->ensure_property_number("srs_id")) != NULL) {
srs_id = (int)prop->to_number();
}
srs_info("edge-srs ip=%s, version=%s, pid=%d, id=%d",
srs_server_ip.c_str(), srs_version.c_str(), srs_pid, srs_id);
if (srs_pid > 0) {
srs_trace("edge-srs ip=%s, version=%s, pid=%d, id=%d",
srs_server_ip.c_str(), srs_version.c_str(), srs_pid, srs_id);
}
}
ret = service_cycle();
http_hooks_on_close();
return ret;
}在以上函数中，进行Rtmp握手，rtmp网络连接。建立网络连接之后要做一些确认工作。内容如下：
进入service_cycle()
int SrsRtmpConn::service_cycle()
{
int ret = ERROR_SUCCESS;
//1、服务器接收到连接命令消息后，发送确认窗口大小(Window Acknowledgement Size)协议消息到客户端，同时连接到连接命令中提到的应用程序。
if ((ret = rtmp->set_window_ack_size((int)(2.5 * 1000 * 1000))) != ERROR_SUCCESS) {
srs_error("set window acknowledgement size failed. ret=%d", ret);
return ret;
}
srs_verbose("set window acknowledgement size success");
//2、服务器发送设置带宽()协议消息到客户端。
if ((ret = rtmp->set_peer_bandwidth((int)(2.5 * 1000 * 1000), 2)) != ERROR_SUCCESS) {
srs_error("set peer bandwidth failed. ret=%d", ret);
return ret;
}
srs_verbose("set peer bandwidth success");
// get the ip which client connected.
std::string local_ip = srs_get_local_ip(st_netfd_fileno(stfd));
// do bandwidth test if connect to the vhost which is for bandwidth check.
if (_srs_config->get_bw_check_enabled(req->vhost)) {
return bandwidth->bandwidth_check(rtmp, skt, req, local_ip);
}
// do token traverse before serve it.
// @see https://github.com/ossrs/srs/pull/239
if (true) {
bool vhost_is_edge = _srs_config->get_vhost_is_edge(req->vhost);
bool edge_traverse = _srs_config->get_vhost_edge_token_traverse(req->vhost);
if (vhost_is_edge && edge_traverse) {
if ((ret = check_edge_token_traverse_auth()) != ERROR_SUCCESS) {
srs_warn("token auth failed, ret=%d", ret);
return ret;
}
}
}
// set chunk size to larger.
// set the chunk size before any larger response greater than 128,
// to make OBS happy, @see https://github.com/ossrs/srs/issues/454
int chunk_size = _srs_config->get_chunk_size(req->vhost);
if ((ret = rtmp->set_chunk_size(chunk_size)) != ERROR_SUCCESS) {
srs_error("set chunk_size=%d failed. ret=%d", chunk_size, ret);
return ret;
}
srs_info("set chunk_size=%d success", chunk_size);
// response the client connect ok.
if ((ret = rtmp->response_connect_app(req, local_ip.c_str())) != ERROR_SUCCESS) {
srs_error("response connect app failed. ret=%d", ret);
return ret;
}
srs_verbose("response connect app success");
if ((ret = rtmp->on_bw_done()) != ERROR_SUCCESS) {
srs_error("on_bw_done failed. ret=%d", ret);
return ret;
}
srs_verbose("on_bw_done success");
while (!disposed) {
ret = stream_service_cycle();
// stream service must terminated with error, never success.
// when terminated with success, it's user required to stop.
if (ret == ERROR_SUCCESS) {
continue;
}
// when not system control error, fatal error, return.
if (!srs_is_system_control_error(ret)) {
if (ret != ERROR_SOCKET_TIMEOUT && !srs_is_client_gracefully_close(ret)) {
srs_error("stream service cycle failed. ret=%d", ret);
}
return ret;
}
// for republish, continue service
if (ret == ERROR_CONTROL_REPUBLISH) {
// set timeout to a larger value, wait for encoder to republish.
rtmp->set_send_timeout(SRS_REPUBLISH_RECV_TIMEOUT_US);
rtmp->set_recv_timeout(SRS_REPUBLISH_SEND_TIMEOUT_US);
srs_trace("control message(unpublish) accept, retry stream service.");
continue;
}
// for "some" system control error,
// logical accept and retry stream service.
if (ret == ERROR_CONTROL_RTMP_CLOSE) {
// TODO: FIXME: use ping message to anti-death of socket.
// @see: https://github.com/ossrs/srs/issues/39
// set timeout to a larger value, for user paused.
rtmp->set_recv_timeout(SRS_PAUSED_RECV_TIMEOUT_US);
rtmp->set_send_timeout(SRS_PAUSED_SEND_TIMEOUT_US);
srs_trace("control message(close) accept, retry stream service.");
continue;
}
// for other system control message, fatal error.
srs_error("control message(%d) reject as error. ret=%d", ret, ret);
return ret;
}
return ret;
}
网络连接确认工作完成之后，进入客户身份识别代码中
2、client identify
进入stream_service_cycle()
int SrsRtmpConn::stream_service_cycle()
{
int ret = ERROR_SUCCESS;
SrsRtmpConnType type;
if ((ret = rtmp->identify_client(res->stream_id, type, req->stream, req->duration)) != ERROR_SUCCESS) {
if (!srs_is_client_gracefully_close(ret)) {
srs_error("identify client failed. ret=%d", ret);
}
return ret;
}
req->strip();
srs_trace("client identified, type=%s, stream_name=%s, duration=%.2f",
srs_client_type_string(type).c_str(), req->stream.c_str(), req->duration);
// security check
if ((ret = security->check(type, ip, req)) != ERROR_SUCCESS) {
srs_error("security check failed. ret=%d", ret);
return ret;
}
srs_info("security check ok");
// Never allow the empty stream name, for HLS may write to a file with empty name.
// @see https://github.com/ossrs/srs/issues/834
if (req->stream.empty()) {
ret = ERROR_RTMP_STREAM_NAME_EMPTY;
srs_error("RTMP: Empty stream name not allowed, ret=%d", ret);
return ret;
}
// client is identified, set the timeout to service timeout.
rtmp->set_recv_timeout(SRS_CONSTS_RTMP_RECV_TIMEOUT_US);
rtmp->set_send_timeout(SRS_CONSTS_RTMP_SEND_TIMEOUT_US);
// find a source to serve.
SrsSource* source = NULL;
if ((ret = SrsSource::fetch_or_create(req, server, &source)) != ERROR_SUCCESS) {
return ret;
}
srs_assert(source != NULL);
// update the statistic when source disconveried.
SrsStatistic* stat = SrsStatistic::instance();
if ((ret = stat->on_client(_srs_context->get_id(), req, this, type)) != ERROR_SUCCESS) {
srs_error("stat client failed. ret=%d", ret);
return ret;
}
bool vhost_is_edge = _srs_config->get_vhost_is_edge(req->vhost);
bool enabled_cache = _srs_config->get_gop_cache(req->vhost);
srs_trace("source url=%s, ip=%s, cache=%d, is_edge=%d, source_id=%d[%d]",
req->get_stream_url().c_str(), ip.c_str(), enabled_cache, vhost_is_edge,
source->source_id(), source->source_id());
source->set_cache(enabled_cache);
client_type = type;
switch (type) {
case SrsRtmpConnPlay: {
srs_verbose("start to play stream %s.", req->stream.c_str());
// response connection start play
if ((ret = rtmp->start_play(res->stream_id)) != ERROR_SUCCESS) {
srs_error("start to play stream failed. ret=%d", ret);
return ret;
}
if ((ret = http_hooks_on_play()) != ERROR_SUCCESS) {
srs_error("http hook on_play failed. ret=%d", ret);
return ret;
}
srs_info("start to play stream %s success", req->stream.c_str());
ret = playing(source);
http_hooks_on_stop();
return ret;
}
case SrsRtmpConnFMLEPublish: {
srs_verbose("FMLE start to publish stream %s.", req->stream.c_str());
if ((ret = rtmp->start_fmle_publish(res->stream_id)) != ERROR_SUCCESS) {
srs_error("start to publish stream failed. ret=%d", ret);
return ret;
}
return publishing(source);
}
case SrsRtmpConnHaivisionPublish: {
srs_verbose("Haivision start to publish stream %s.", req->stream.c_str());
if ((ret = rtmp->start_haivision_publish(res->stream_id)) != ERROR_SUCCESS) {
srs_error("start to publish stream failed. ret=%d", ret);
return ret;
}
return publishing(source);
}
case SrsRtmpConnFlashPublish: {
srs_verbose("flash start to publish stream %s.", req->stream.c_str());
if ((ret = rtmp->start_flash_publish(res->stream_id)) != ERROR_SUCCESS) {
srs_error("flash start to publish stream failed. ret=%d", ret);
return ret;
}
return publishing(source);
}
default: {
ret = ERROR_SYSTEM_CLIENT_INVALID;
srs_info("invalid client type=%d. ret=%d", type, ret);
return ret;
}
}
return ret;
}
通过客户身份识别identify_client()函数后，Srs就知道是publish还是play，进入相应的分支。首先这里分支有点多，SrsRtmpConnPlay播放rtmp流，SrsRtmpConnFMLEPublish，fmle发布rtmp流，我们接受rtmp流，就是从SrsRtmpConnFMLEPublish分支里面进去。
注意：使用vlc播放器播放从SrsRtmpConnPlay分支进入。
3、进入publish分支
进入publishing();
int SrsRtmpConn::publishing(SrsSource* source)
{
int ret = ERROR_SUCCESS;
if ((ret = refer->check(req->pageUrl, _srs_config->get_refer_publish(req->vhost))) != ERROR_SUCCESS) {
srs_error("check publish_refer failed. ret=%d", ret);
return ret;
}
srs_verbose("check publish_refer success.");
// http回调，通知vhost SRS发生了publish事件。
if ((ret = http_hooks_on_publish()) != ERROR_SUCCESS) {
srs_error("http hook on_publish failed. ret=%d", ret);
return ret;
}
bool vhost_is_edge = _srs_config->get_vhost_is_edge(req->vhost);
if ((ret = acquire_publish(source, vhost_is_edge)) == ERROR_SUCCESS) {
// use isolate thread to recv,
// @see: https://github.com/ossrs/srs/issues/237
// 创建了一个接受线程对象，这个在后面用，
SrsPublishRecvThread trd(rtmp, req,
st_netfd_fileno(stfd), 0, this, source,
client_type != SrsRtmpConnFlashPublish,
vhost_is_edge);
srs_info("start to publish stream %s success", req->stream.c_str());
ret = do_publishing(source, &trd);
// stop isolate recv thread
trd.stop();
}
// whatever the acquire publish, always release publish.
// when the acquire error in the midlle-way, the publish state changed,
// but failed, so we must cleanup it.
// @see https://github.com/ossrs/srs/issues/474
// @remark when stream is busy, should never release it.
if (ret != ERROR_SYSTEM_STREAM_BUSY) {
release_publish(source, vhost_is_edge);
}
http_hooks_on_unpublish();
return ret;
}
三、单独创建rtmp接收线程
进入do_publishing()，等待接受rtmp信息。
int SrsRtmpConn::do_publishing(SrsSource* source, SrsPublishRecvThread* trd)
{
int ret = ERROR_SUCCESS;
SrsPithyPrint* pprint = SrsPithyPrint::create_rtmp_publish();
SrsAutoFree(SrsPithyPrint, pprint);
// 创建一个独立的接收线程，专门用于Rtmp协议的接受，因此，下一步进入rtmp接受线程至关重要。
// start isolate recv thread.
if ((ret = trd->start()) != ERROR_SUCCESS) {
srs_error("start isolate recv thread failed. ret=%d", ret);
return ret;
}
// change the isolate recv thread context id,
// merge its log to current thread.
int receive_thread_cid = trd->get_cid();
trd->set_cid(_srs_context->get_id());
// initialize the publish timeout.
publish_1stpkt_timeout = _srs_config->get_publish_1stpkt_timeout(req->vhost);
publish_normal_timeout = _srs_config->get_publish_normal_timeout(req->vhost);
// set the sock options.
set_sock_options();
if (true) {
bool mr = _srs_config->get_mr_enabled(req->vhost);
int mr_sleep = _srs_config->get_mr_sleep_ms(req->vhost);
srs_trace("start publish mr=%d/%d, p1stpt=%d, pnt=%d, tcp_nodelay=%d, rtcid=%d",
mr, mr_sleep, publish_1stpkt_timeout, publish_normal_timeout, tcp_nodelay, receive_thread_cid);
}
int64_t nb_msgs = 0;
uint64_t nb_frames = 0;
while (!disposed) {
// 计算时间
pprint->elapse();
// 资源过期
// when source is set to expired, disconnect it.
if (expired) {
ret = ERROR_USER_DISCONNECT;
srs_trace("connection expired. ret=%d", ret);
return ret;
}
// 等待
// cond wait for timeout.
if (nb_msgs == 0) {
// when not got msgs, wait for a larger timeout.
// @see https://github.com/ossrs/srs/issues/441
trd->wait(publish_1stpkt_timeout);
} else {
trd->wait(publish_normal_timeout);
}
// 检测该线程出现问题
// check the thread error code.
if ((ret = trd->error_code()) != ERROR_SUCCESS) {
if (!srs_is_system_control_error(ret) && !srs_is_client_gracefully_close(ret)) {
srs_error("recv thread failed. ret=%d", ret);
}
return ret;
}
//上一次接受RTMP包数量如果等于现在接受的，说明本次没有接收到，publish超时
// when not got any messages, timeout.
if (trd->nb_msgs() <= nb_msgs) {
ret = ERROR_SOCKET_TIMEOUT;
srs_trace("publish timeout %dms, nb_msgs=%"PRId64", ret=%d",
nb_msgs? publish_normal_timeout : publish_1stpkt_timeout, nb_msgs, ret);
break;
}
nb_msgs = trd->nb_msgs();
// Update the stat for video fps.
// @remark https://github.com/ossrs/srs/issues/851
SrsStatistic* stat = SrsStatistic::instance();
if ((ret = stat->on_video_frames(req, (int)(trd->nb_video_frames() - nb_frames))) != ERROR_SUCCESS) {
return ret;
}
// 获取视频帧数量
nb_frames = trd->nb_video_frames();
// reportable
if (pprint->can_print()) {
kbps->sample();
bool mr = _srs_config->get_mr_enabled(req->vhost);
int mr_sleep = _srs_config->get_mr_sleep_ms(req->vhost);
srs_trace("<- "SRS_CONSTS_LOG_CLIENT_PUBLISH
" time=%"PRId64", okbps=%d,%d,%d, ikbps=%d,%d,%d, mr=%d/%d, p1stpt=%d, pnt=%d", pprint->age(),
kbps->get_send_kbps(), kbps->get_send_kbps_30s(), kbps->get_send_kbps_5m(),
kbps->get_recv_kbps(), kbps->get_recv_kbps_30s(), kbps->get_recv_kbps_5m(),
mr, mr_sleep, publish_1stpkt_timeout, publish_normal_timeout
);
}
}
return ret;
}
该while(!disposed)是一个检测rtmp连接循环，不是真正Rtmp接受线程。
四、rtmp接受
int SrsRecvThread::cycle()
{
int ret = ERROR_SUCCESS;
while (!trd->interrupted()) {
if (!handler->can_handle()) {
st_usleep(timeout * 1000);
continue;
}
SrsCommonMessage* msg = NULL;
srs_trace("come in isloate recv thread\n");
// recv and handle message
ret = rtmp->recv_message(&msg);
if (ret == ERROR_SUCCESS) {
ret = handler->handle(msg);
}
if (ret != ERROR_SUCCESS) {
if (!srs_is_client_gracefully_close(ret) && !srs_is_system_control_error(ret)) {
srs_error("thread process message failed. ret=%d", ret);
}
// we use no timeout to recv, should never got any error.
trd->interrupt();
// notice the handler got a recv error.
handler->on_recv_error(ret);
return ret;
}
srs_verbose("thread loop recv message. ret=%d", ret);
}
return ret;
ok，进入真正Rtmp接受线程，当然第一步是rtmp接受了，第二步就是rtmp信息处理了。我们来看rtmp接受。
1、rtmp接受
rtmp接受还是挺复杂的，绕了这么大的弯子，终于来到rtmp接受了，具体代码框架流程请看框架图。
int SrsProtocol::recv_message(SrsCommonMessage** pmsg)
{
*pmsg = NULL;
int ret = ERROR_SUCCESS;
while (true) {
SrsCommonMessage* msg = NULL;
//接受RTMP块流
if ((ret = recv_interlaced_message(&msg)) != ERROR_SUCCESS) {
if (ret != ERROR_SOCKET_TIMEOUT && !srs_is_client_gracefully_close(ret)) {
srs_error("recv interlaced message failed. ret=%d", ret);
}
srs_freep(msg);
return ret;
}
srs_trace("entire msg received");
if (!msg) {
srs_info("got empty message without error.");
continue;
}
if (msg->size <= 0 || msg->header.payload_length <= 0) {
srs_trace("ignore empty message(type=%d, size=%d, time=%"PRId64", sid=%d).",
msg->header.message_type, msg->header.payload_length,
msg->header.timestamp, msg->header.stream_id);
srs_freep(msg);
continue;
}
if ((ret = on_recv_message(msg)) != ERROR_SUCCESS) {
srs_error("hook the received msg failed. ret=%d", ret);
srs_freep(msg);
return ret;
}
srs_trace("got a msg, cid=%d, type=%d, size=%d, time=%"PRId64,
msg->header.perfer_cid, msg->header.message_type, msg->header.payload_length,
msg->header.timestamp);
*pmsg = msg;
break;
}
return ret;
}
进入接受recv_interlaced_message（）函数，
int SrsProtocol::recv_interlaced_message(SrsCommonMessage** pmsg)
{
int ret = ERROR_SUCCESS;
// chunk stream basic header.
char fmt = 0;
int cid = 0;
// 检查基本信息 1、CSID 2、Chunk type:0,1,2,3
if ((ret = read_basic_header(fmt, cid)) != ERROR_SUCCESS) {
if (ret != ERROR_SOCKET_TIMEOUT && !srs_is_client_gracefully_close(ret)) {
srs_error("read basic header failed. ret=%d", ret);
}
return ret;
}
srs_verbose("read basic header success. fmt=%d, cid=%d", fmt, cid);
// the cid must not negative.
srs_assert(cid >= 0);
// get the cached chunk stream.
SrsChunkStream* chunk = NULL;
// use chunk stream cache to get the chunk info.
// @see https://github.com/ossrs/srs/issues/249
if (cid < SRS_PERF_CHUNK_STREAM_CACHE) {
// chunk stream cache hit.
srs_verbose("cs-cache hit, cid=%d", cid);
// already init, use it direclty
chunk = cs_cache[cid];
srs_verbose("cached chunk stream: fmt=%d, cid=%d, size=%d, message(type=%d, size=%d, time=%"PRId64", sid=%d)",
chunk->fmt, chunk->cid, (chunk->msg? chunk->msg->size : 0), chunk->header.message_type, chunk->header.payload_length,
chunk->header.timestamp, chunk->header.stream_id);
} else {
// chunk stream cache miss, use map.
if (chunk_streams.find(cid) == chunk_streams.end()) {
chunk = chunk_streams[cid] = new SrsChunkStream(cid);
// set the perfer cid of chunk,
// which will copy to the message received.
chunk->header.perfer_cid = cid;
srs_verbose("cache new chunk stream: fmt=%d, cid=%d", fmt, cid);
} else {
chunk = chunk_streams[cid];
srs_verbose("cached chunk stream: fmt=%d, cid=%d, size=%d, message(type=%d, size=%d, time=%"PRId64", sid=%d)",
chunk->fmt, chunk->cid, (chunk->msg? chunk->msg->size : 0), chunk->header.message_type, chunk->header.payload_length,
chunk->header.timestamp, chunk->header.stream_id);
}
}
// 检查信息头信息
// chunk stream message header
if ((ret = read_message_header(chunk, fmt)) != ERROR_SUCCESS) {
if (ret != ERROR_SOCKET_TIMEOUT && !srs_is_client_gracefully_close(ret)) {
srs_error("read message header failed. ret=%d", ret);
}
return ret;
}
srs_verbose("read message header success. "
"fmt=%d, ext_time=%d, size=%d, message(type=%d, size=%d, time=%"PRId64", sid=%d)",
fmt, chunk->extended_timestamp, (chunk->msg? chunk->msg->size : 0), chunk->header.message_type,
chunk->header.payload_length, chunk->header.timestamp, chunk->header.stream_id);
//获取rtmp负载信息
// read msg payload from chunk stream.
SrsCommonMessage* msg = NULL;
if ((ret = read_message_payload(chunk, &msg)) != ERROR_SUCCESS) {
if (ret != ERROR_SOCKET_TIMEOUT && !srs_is_client_gracefully_close(ret)) {
srs_error("read message payload failed. ret=%d", ret);
}
return ret;
}
// 没有获得一个完整的RTMP信息，试着获取下一个块流
// not got an entire RTMP message, try next chunk.
if (!msg) {
srs_verbose("get partial message success. size=%d, message(type=%d, size=%d, time=%"PRId64", sid=%d)",
(msg? msg->size : (chunk->msg? chunk->msg->size : 0)), chunk->header.message_type, chunk->header.payload_length,
chunk->header.timestamp, chunk->header.stream_id);
return ret;
}
// 获取一个完整的RTMP信息
*pmsg = msg;
srs_info("get entire message success. size=%d, message(type=%d, size=%d, time=%"PRId64", sid=%d)",
(msg? msg->size : (chunk->msg? chunk->msg->size : 0)), chunk->header.message_type, chunk->header.payload_length,
chunk->header.timestamp, chunk->header.stream_id);
return ret;
}这里面就有一些相关Rtmp协议的东西，检车Basic Header、Message Header获取payload信息，ok，rtmp信息接收到了。
2、Rtmp信息处理
int SrsPublishRecvThread::handle(SrsCommonMessage* msg)
{
int ret = ERROR_SUCCESS;
// when cid changed, change it.
if (ncid != cid) {
_srs_context->set_id(ncid);
cid = ncid;
}
_nb_msgs++;
if (msg->header.is_video()) {
// 接受到的视频帧数量
video_frames++;
srs_trace("message:%d, video:%d\n", _nb_msgs, video_frames);
}
// log to show the time of recv thread.
srs_trace("recv thread now=%"PRId64"us, got msg time=%"PRId64"ms, size=%d",
srs_update_system_time_ms(), msg->header.timestamp, msg->size);
// the rtmp connection will handle this message
ret = _conn->handle_publish_message(_source, msg, _is_fmle, _is_edge);
// must always free it,
// the source will copy it if need to use.
srs_freep(msg);
return ret;
}首先做了一个统计，统计所有接受到的数据块和视频数据块。
进入process_publish_message（）函数
int SrsRtmpConn::process_publish_message(SrsSource* source, SrsCommonMessage* msg, bool vhost_is_edge)
{
int ret = ERROR_SUCCESS;
//发布到源站服务器
// for edge, directly proxy message to origin.
if (vhost_is_edge) {
if ((ret = source->on_edge_proxy_publish(msg)) != ERROR_SUCCESS) {
srs_error("edge publish proxy msg failed. ret=%d", ret);
return ret;
}
return ret;
}
// process audio packet
if (msg->header.is_audio()) {
if ((ret = source->on_audio(msg)) != ERROR_SUCCESS) {
srs_error("source process audio message failed. ret=%d", ret);
return ret;
}
return ret;
}
// process video packet
if (msg->header.is_video()) {
if ((ret = source->on_video(msg)) != ERROR_SUCCESS) {
srs_error("source process video message failed. ret=%d", ret);
return ret;
}
return ret;
}
// process aggregate packet
if (msg->header.is_aggregate()) {
if ((ret = source->on_aggregate(msg)) != ERROR_SUCCESS) {
srs_error("source process aggregate message failed. ret=%d", ret);
return ret;
}
return ret;
}
// process onMetaData
if (msg->header.is_amf0_data() || msg->header.is_amf3_data()) {
SrsPacket* pkt = NULL;
if ((ret = rtmp->decode_message(msg, &pkt)) != ERROR_SUCCESS) {
srs_error("decode onMetaData message failed. ret=%d", ret);
return ret;
}
SrsAutoFree(SrsPacket, pkt);
if (dynamic_cast<SrsOnMetaDataPacket*>(pkt)) {
SrsOnMetaDataPacket* metadata = dynamic_cast<SrsOnMetaDataPacket*>(pkt);
if ((ret = source->on_meta_data(msg, metadata)) != ERROR_SUCCESS) {
srs_error("source process onMetaData message failed. ret=%d", ret);
return ret;
}
srs_info("process onMetaData message success.");
return ret;
}
srs_info("ignore AMF0/AMF3 data message.");
return ret;
}
return ret;
}
该函数做了一下几点处理：
（1）发布到源站服务器（相关CDN知识请参考：http://blog.csdn.net/ManagerUser/article/details/73909997）
（2）处理音频包
（3）处理视频包
注意：这里对音视频数据包的处理差不多，所有只看一下视频数据包的处理即可。
进入on_video（）函数
int SrsSource::on_video(SrsCommonMessage* shared_video)
{
int ret = ERROR_SUCCESS;
// monotically increase detect.
if (!mix_correct && is_monotonically_increase) {
if (last_packet_time > 0 && shared_video->header.timestamp < last_packet_time) {
is_monotonically_increase = false;
srs_warn("VIDEO: stream not monotonically increase, please open mix_correct.");
}
}
last_packet_time = shared_video->header.timestamp;
// drop any unknown header video.
// @see https://github.com/ossrs/srs/issues/421
if (!SrsFlvCodec::video_is_acceptable(shared_video->payload, shared_video->size)) {
char b0 = 0x00;
if (shared_video->size > 0) {
b0 = shared_video->payload[0];
}
srs_warn("drop unknown header video, size=%d, bytes[0]=%#x", shared_video->size, b0);
return ret;
}
// convert shared_video to msg, user should not use shared_video again.
// the payload is transfer to msg, and set to NULL in shared_video.
//将SrsCommonMessage类的所有数据完全拷贝给SrsSharedPtrMessage
SrsSharedPtrMessage msg;
if ((ret = msg.create(shared_video)) != ERROR_SUCCESS) {
srs_error("initialize the video failed. ret=%d", ret);
return ret;
}
srs_info("Video dts=%"PRId64", size=%d", msg.timestamp, msg.size);
// directly process the audio message.
if (!mix_correct) {
return on_video_imp(&msg);
}
// msg.copy()返回msg自己指针
// insert msg to the queue.
mix_queue->push(msg.copy());
// fetch someone from mix queue.
SrsSharedPtrMessage* m = mix_queue->pop();
if (!m) {
return ret;
}
//消费单调递增消息
// consume the monotonically increase message.
if (m->is_audio()) {
ret = on_audio_imp(m);
} else {
ret = on_video_imp(m);
}
srs_freep(m);
return ret;
}该函数做了rtmp块数据的拷贝，不用太在意，只关系接受到的rtmp信息在SrsCommonMessage类中即可。
进入on_video_imp()
int SrsSource::on_video_imp(SrsSharedPtrMessage* msg)
{
int ret = ERROR_SUCCESS;
srs_info("Video dts=%"PRId64", size=%d", msg->timestamp, msg->size);
bool is_sequence_header = SrsFlvCodec::video_is_sequence_header(msg->payload, msg->size);
// whether consumer should drop for the duplicated sequence header.
bool drop_for_reduce = false;
if (is_sequence_header && cache_sh_video && _srs_config->get_reduce_sequence_header(_req->vhost)) {
if (cache_sh_video->size == msg->size) {
drop_for_reduce = srs_bytes_equals(cache_sh_video->payload, msg->payload, msg->size);
srs_warn("drop for reduce sh video, size=%d", msg->size);
}
}
// cache the sequence header if h264
// donot cache the sequence header to gop_cache, return here.
if (is_sequence_header) {
srs_freep(cache_sh_video);
// 缓存到shareMessage
cache_sh_video = msg->copy();
// parse detail audio codec
SrsAvcAacCodec codec;
// user can disable the sps parse to workaround when parse sps failed.
// @see https://github.com/ossrs/srs/issues/474
// 是否解析SPS
codec.avc_parse_sps = _srs_config->get_parse_sps(_req->vhost);
// AAC 高级音频编码
// AVC 高级视频编码
SrsCodecSample sample;
if ((ret = codec.video_avc_demux(msg->payload, msg->size, &sample)) != ERROR_SUCCESS) {
srs_error("source codec demux video failed. ret=%d", ret);
return ret;
}
// when got video stream info.
SrsStatistic* stat = SrsStatistic::instance();
if ((ret = stat->on_video_info(_req, SrsCodecVideoAVC, codec.avc_profile, codec.avc_level)) != ERROR_SUCCESS) {
return ret;
}
srs_trace("%dB video sh, codec(%d, profile=%s, level=%s, %dx%d, %dkbps, %dfps, %ds)",
msg->size, codec.video_codec_id,
srs_codec_avc_profile2str(codec.avc_profile).c_str(),
srs_codec_avc_level2str(codec.avc_level).c_str(), codec.width, codec.height,
codec.video_data_rate / 1000, codec.frame_rate, codec.duration);
}
#ifdef SRS_AUTO_HLS
if ((ret = hls->on_video(msg, is_sequence_header)) != ERROR_SUCCESS) {
// apply the error strategy for hls.
// @see https://github.com/ossrs/srs/issues/264
std::string hls_error_strategy = _srs_config->get_hls_on_error(_req->vhost);
if (srs_config_hls_is_on_error_ignore(hls_error_strategy)) {
srs_warn("hls process video message failed, ignore and disable hls. ret=%d", ret);
// unpublish, ignore ret.
hls->on_unpublish();
// ignore.
ret = ERROR_SUCCESS;
} else if (srs_config_hls_is_on_error_continue(hls_error_strategy)) {
if (srs_hls_can_continue(ret, cache_sh_video, msg)) {
ret = ERROR_SUCCESS;
} else {
srs_warn("hls continue video failed. ret=%d", ret);
return ret;
}
} else {
srs_warn("hls disconnect publisher for video error. ret=%d", ret);
return ret;
}
}
#endif
#ifdef SRS_AUTO_DVR
if ((ret = dvr->on_video(msg)) != ERROR_SUCCESS) {
srs_warn("dvr process video message failed, ignore and disable dvr. ret=%d", ret);
// unpublish, ignore ret.
dvr->on_unpublish();
// ignore.
ret = ERROR_SUCCESS;
}
#endif
#ifdef SRS_AUTO_HDS
if ((ret = hds->on_video(msg)) != ERROR_SUCCESS) {
srs_warn("hds process video message failed, ignore and disable dvr. ret=%d", ret);
// unpublish, ignore ret.
hds->on_unpublish();
// ignore.
ret = ERROR_SUCCESS;
}
#endif
// 将信息入到消费者堆栈里面
// copy to all consumer
if (!drop_for_reduce) {;
for (int i = 0; i < (int)consumers.size(); i++) {
SrsConsumer* consumer = consumers.at(i);
if ((ret = consumer->enqueue(msg, atc, jitter_algorithm)) != ERROR_SUCCESS) {
srs_error("dispatch the video failed. ret=%d", ret);
return ret;
}
}
srs_info("dispatch video success.");
}
// copy to all forwarders.
if (!forwarders.empty()) {
printf("forward to other server");
std::vector<SrsForwarder*>::iterator it;
for (it = forwarders.begin(); it != forwarders.end(); ++it) {
SrsForwarder* forwarder = *it;
if ((ret = forwarder->on_video(msg)) != ERROR_SUCCESS) {
srs_error("forwarder process video message failed. ret=%d", ret);
return ret;
}
}
}
// when sequence header, donot push to gop cache and adjust the timestamp.
if (is_sequence_header) {
return ret;
}
// cache the last gop packets
if ((ret = gop_cache->cache(msg)) != ERROR_SUCCESS) {
srs_error("gop cache msg failed. ret=%d", ret);
return ret;
}
srs_verbose("cache gop success.");
// if atc, update the sequence header to abs time.
if (atc) {
if (cache_sh_video) {
cache_sh_video->timestamp = msg->timestamp;
}
if (cache_metadata) {
cache_metadata->timestamp = msg->timestamp;
}
}
return ret;
}
该函数有点长，也非常重要，分为段来介绍；
前半段代码通过SrsAvcAacCodec类来检测视频编码，这里RTMP协议支持视频编码格式H264，音频AAC。
后半部分代码主要做了转发入栈操作：
（1）HLS转发（该部分内容在后面文章中介绍）
（2）将rtmp信息转发给消费者列表（用户列表，用户在play的时候直接到该列表中去数据并进行play）
（3）将rtmp信息forward到其他服务器（该部分内容在后面章节中介绍）
五、Publish总结
至此，rtmp信息publish过程就结束了，总结一下：
（1）rtmp协议链接。包括握手，创建网络连接，控制信息交互。
（2）创建单独的接受线程去处理Rtmp信息接受。
（3）缓存数据到SrsCommonMessage类。
（4）数据转发，转发给消费者，forward给其他服务器（边缘服务器或者源站服务器）
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