线程模型
srs使用了state-threads协程库，是单线程多协程模型。
这个协程的概念类似于lua的协程，都是单线程中可以创建多个协程。而golang中的goroutine协程是多线程并发的，goroutine有可能运行在同一个线程也可能在不同线程，这样就有了线程安全问题，所以需要chan通信或者mutex加锁共享资源。
而srs因为是单线程多协程所以不用考虑线程安全，数据不用加锁。
主流程分析
撇掉程序启动的一些初始化和设置，直接进入：
int SrsServer::listen()
{
int ret = ERROR_SUCCESS;
if ((ret = listen_rtmp()) != ERROR_SUCCESS) {
return ret;
}
if ((ret = listen_http_api()) != ERROR_SUCCESS) {
return ret;
}
if ((ret = listen_http_stream()) != ERROR_SUCCESS) {
return ret;
}
if ((ret = listen_stream_caster()) != ERROR_SUCCESS) {
return ret;
}
return ret;
}
先看看listen_rtmp():
int SrsServer::listen_rtmp()
{
int ret = ERROR_SUCCESS;
// stream service port.
std::vector ip_ports = _srs_config->get_listens();
srs_assert((int)ip_ports.size() > 0);
close_listeners(SrsListenerRtmpStream);
for (int i = 0; i < (int)ip_ports.size(); i++) {
SrsListener* listener = new SrsStreamListener(this, SrsListenerRtmpStream);
listeners.push_back(listener);
std::string ip;
int port;
srs_parse_endpoint(ip_ports[i], ip, port);
if ((ret = listener->listen(ip, port)) != ERROR_SUCCESS) {
srs_error("RTMP stream listen at %s:%d failed. ret=%d", ip.c_str(), port, ret);
return ret;
}
}
return ret;
}
创建了SrsStreamListener，在SrsStreamListener::listen中又创建了SrsTcpListener进行listen。
SrsTcpListener::SrsTcpListener(ISrsTcpHandler* h, string i, int p)
{
handler = h;
ip = i;
port = p;
_fd = -1;
_stfd = NULL;
pthread = new SrsReusableThread("tcp", this);
}
在SrsTcpListener中创建了pthread: SrsReusableThread。
在int SrsTcpListener::listen()中调用了pthread->start()，协程会回调到int SrsTcpListener::cycle()。
int SrsTcpListener::cycle()
{
int ret = ERROR_SUCCESS;
st_netfd_t client_stfd = st_accept(_stfd, NULL, NULL, ST_UTIME_NO_TIMEOUT);
if(client_stfd == NULL){
// ignore error.
if (errno != EINTR) {
srs_error("ignore accept thread stoppped for accept client error");
}
return ret;
}
srs_verbose("get a client. fd=%d", st_netfd_fileno(client_stfd));
if ((ret = handler->on_tcp_client(client_stfd)) != ERROR_SUCCESS) {
srs_warn("accept client error. ret=%d", ret);
return ret;
}
return ret;
}
accept连接后，回调到on_tcp_client。
也就是SrsStreamListener::on_tcp_client：
int SrsStreamListener::on_tcp_client(st_netfd_t stfd)
{
int ret = ERROR_SUCCESS;
if ((ret = server->accept_client(type, stfd)) != ERROR_SUCCESS) {
srs_warn("accept client error. ret=%d", ret);
return ret;
}
return ret;
}
int SrsServer::accept_client(SrsListenerType type, st_netfd_t client_stfd)
{
...
SrsConnection* conn = NULL;
if (type == SrsListenerRtmpStream) {
conn = new SrsRtmpConn(this, client_stfd);
} else if (type == SrsListenerHttpApi) {
#ifdef SRS_AUTO_HTTP_API
conn = new SrsHttpApi(this, client_stfd, http_api_mux);
#else
srs_warn("close http client for server not support http-api");
srs_close_stfd(client_stfd);
return ret;
#endif
} else if (type == SrsListenerHttpStream) {
#ifdef SRS_AUTO_HTTP_SERVER
conn = new SrsResponseOnlyHttpConn(this, client_stfd, http_server);
#else
srs_warn("close http client for server not support http-server");
srs_close_stfd(client_stfd);
return ret;
#endif
} else {
// TODO: FIXME: handler others
}
srs_assert(conn);
// directly enqueue, the cycle thread will remove the client.
conns.push_back(conn);
srs_verbose("add conn to vector.");
// cycle will start process thread and when finished remove the client.
// @remark never use the conn, for it maybe destroyed.
if ((ret = conn->start()) != ERROR_SUCCESS) {
return ret;
}
srs_verbose("conn started success.");
srs_verbose("accept client finished. conns=%d, ret=%d", (int)conns.size(), ret);
return ret;
}
在上面根据type创建不同的SrsConnection，Rtmp创建了SrsRtmpConn，并且加入到std::vector中，然后执行conn->start()。
SrsConnection基类创建了一个协程pthread: SrsOneCycleThread，上面的conn->start()，实际上是pthread->start()：
SrsConnection::SrsConnection(IConnectionManager* cm, st_netfd_t c)
{
id = 0;
manager = cm;
stfd = c;
disposed = false;
expired = false;
// the client thread should reap itself,
// so we never use joinable.
// TODO: FIXME: maybe other thread need to stop it.
// @see: https://github.com/ossrs/srs/issues/78
pthread = new SrsOneCycleThread("conn", this);
}
int SrsConnection::start()
{
return pthread->start();
}
在int SrsConnection::cycle()调用了do_cycle()，派生类实现了这个方法。
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);
//正式进入rtmp握手。
if ((ret = rtmp->handshake()) != ERROR_SUCCESS) {
srs_error("rtmp handshake failed. ret=%d", ret);
return ret;
}
srs_verbose("rtmp handshake success");
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;
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;
}
在这儿正式进入rtmp协议处理阶段。先进行握手：rtmp->handshake()等操作，然后进入service_cycle();。
int SrsRtmpConn::service_cycle()
{
...
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;
}
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;
}
srs_discovery_tc_url(req->tcUrl, req->schema, req->host, req->vhost, req->app, req->stream, req->port, req->param);
req->strip();
srs_trace("client identified, type=%s, stream_name=%s, duration=%.2f, param=%s",
srs_client_type_string(type).c_str(), req->stream.c_str(), req->duration, req->param.c_str());
// discovery vhost, resolve the vhost from config
SrsConfDirective* parsed_vhost = _srs_config->get_vhost(req->vhost);
if (parsed_vhost) {
req->vhost = parsed_vhost->arg0();
}
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;
}
if ((ret = check_vhost()) != ERROR_SUCCESS) {
srs_error("check vhost failed. ret=%d", ret);
return ret;
}
srs_trace("connected stream, tcUrl=%s, pageUrl=%s, swfUrl=%s, schema=%s, vhost=%s, port=%s, app=%s, stream=%s, param=%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->stream.c_str(), req->param.c_str(), (req->args? "(obj)":"null"));
// 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;
}
}
}
// 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;
//根据客户端类型进入不同分支
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;
}
先进行tmp->identify_client客户端身份识别。
然后根据根据客户端类型(type)进入不同分支。
SrsRtmpConnPlay 客户端播流。
SrsRtmpConnFMLEPublish Rtmp推流到服务器。
SrsRtmpConnHaivisionPublish 应该是海康威视推流到服务器？
SrsRtmpConnFlashPublish Flash推流到服务器。
这儿只看SrsRtmpConnFMLEPublish：
进入int SrsRtmpConn::publishing(SrsSource* source)，然后int SrsRtmpConn::do_publishing(SrsSource* source, SrsPublishRecvThread* trd)，
int SrsRtmpConn::do_publishing(SrsSource* source, SrsPublishRecvThread* trd)
{
...
// start isolate recv thread.
if ((ret = trd->start()) != ERROR_SUCCESS) {
srs_error("start isolate recv thread failed. ret=%d", ret);
return ret;
}
...
}
trd协程运行，协程循环：执行rtmp->recv_message(&msg)后调用int SrsPublishRecvThread::handle(SrsCommonMessage* msg)。
再回调到int SrsRtmpConn::handle_publish_message(SrsSource* source, SrsCommonMessage* msg, bool is_fmle, bool vhost_is_edge)。
之后处理收到的数据：
int SrsRtmpConn::process_publish_message(SrsSource* source, SrsCommonMessage* msg, bool vhost_is_edge)
{
int ret = ERROR_SUCCESS;
// 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(pkt)) {
SrsOnMetaDataPacket* metadata = dynamic_cast(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;
}
如果本服务器是edge边缘服务器（vhost_is_edge）直接推流回源到源服务器。
audio和video分开处理。
这儿只看一下video的处理：
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.
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);
}
// insert msg to the queue.
mix_queue->push(msg.copy());
// fetch someone from mix queue.
SrsSharedPtrMessage* m = mix_queue->pop();
if (!m) {
return ret;
}
// 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;
}
把shared_video转换为SrsSharedPtrMessage。
调用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);
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
codec.avc_parse_sps = _srs_config->get_parse_sps(_req->vhost);
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
// 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()) {
std::vector::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;
}
以上进行了缓存h264 sequence header，hls分发，客户端消费者分发，forwarders推流等等。
这里主要看一下消费者分发：
// 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.");
}
int SrsConsumer::enqueue(SrsSharedPtrMessage* shared_msg, bool atc, SrsRtmpJitterAlgorithm ag)
{
int ret = ERROR_SUCCESS;
//这儿的copy操作只是增加引用计数，没有实际的内存拷贝。
SrsSharedPtrMessage* msg = shared_msg->copy();
if (!atc) {
if ((ret = jitter->correct(msg, ag)) != ERROR_SUCCESS) {
srs_freep(msg);
return ret;
}
}
if ((ret = queue->enqueue(msg, NULL)) != ERROR_SUCCESS) {
return ret;
}
#ifdef SRS_PERF_QUEUE_COND_WAIT
srs_verbose("enqueue msg, time=%"PRId64", size=%d, duration=%d, waiting=%d, min_msg=%d",
msg->timestamp, msg->size, queue->duration(), mw_waiting, mw_min_msgs);
// fire the mw when msgs is enough.
if (mw_waiting) {
int duration_ms = queue->duration();
bool match_min_msgs = queue->size() > mw_min_msgs;
// For ATC, maybe the SH timestamp bigger than A/V packet,
// when encoder republish or overflow.
// @see https://github.com/ossrs/srs/pull/749
if (atc && duration_ms < 0) {
st_cond_signal(mw_wait);
mw_waiting = false;
return ret;
}
// when duration ok, signal to flush.
if (match_min_msgs && duration_ms > mw_duration) {
st_cond_signal(mw_wait);
mw_waiting = false;
return ret;
}
}
#endif
return ret;
}
每个SrsConsumer消费者拥有独立的SrsMessageQueue* queue队列。内部队列实现实际上是std::multimap。
SrsMessageQueue有数量大小限制，当队列满的时候删除丢弃旧的messages：
队列大小限制queue_size设置为配置文件中的"queue_length"。如果没设置则默认#define SRS_PERF_PLAY_QUEUE 30。
queue_size_ms = (int)(queue_size * 1000);
int SrsMessageQueue::enqueue(SrsSharedPtrMessage* msg, bool* is_overflow)
{
int ret = ERROR_SUCCESS;
if (msg->is_av()) {
if (av_start_time == -1) {
av_start_time = msg->timestamp;
}
av_end_time = msg->timestamp;
}
msgs.push_back(msg);
while (av_end_time - av_start_time > queue_size_ms) {
// notice the caller queue already overflow and shrinked.
if (is_overflow) {
*is_overflow = true;
}
shrink();
}
return ret;
}
void SrsMessageQueue::shrink()
{
SrsSharedPtrMessage* video_sh = NULL;
SrsSharedPtrMessage* audio_sh = NULL;
int msgs_size = (int)msgs.size();
// remove all msg
// igone the sequence header
for (int i = 0; i < (int)msgs.size(); i++) {
SrsSharedPtrMessage* msg = msgs.at(i);
if (msg->is_video() && SrsFlvCodec::video_is_sequence_header(msg->payload, msg->size)) {
srs_freep(video_sh);
video_sh = msg;
continue;
}
else if (msg->is_audio() && SrsFlvCodec::audio_is_sequence_header(msg->payload, msg->size)) {
srs_freep(audio_sh);
audio_sh = msg;
continue;
}
srs_freep(msg);
}
msgs.clear();
// update av_start_time
av_start_time = av_end_time;
//push_back secquence header and update timestamp
if (video_sh) {
video_sh->timestamp = av_end_time;
msgs.push_back(video_sh);
}
if (audio_sh) {
audio_sh->timestamp = av_end_time;
msgs.push_back(audio_sh);
}
if (_ignore_shrink) {
srs_info("shrink the cache queue, size=%d, removed=%d, max=%.2f",
(int)msgs.size(), msgs_size - (int)msgs.size(), queue_size_ms / 1000.0);
} else {
srs_trace("shrink the cache queue, size=%d, removed=%d, max=%.2f",
(int)msgs.size(), msgs_size - (int)msgs.size(), queue_size_ms / 1000.0);
}
}
保存最近的sequence_header，然后清除其他messages。
AVC sequence header
AAC sequence header
这两个header非常重要，是客户端解码的必需部分，所以不能删除。
这个丢包策略没有根据整个GOP进行丢包，而是直接丢掉除sequence_header的包，有可能会造成客户端花屏。
总结
客户端Rtmp推流到服务器，服务器将消息缓存到各个客户端消费者自己的队列中，数据使用引用计数没有内存拷贝操作。过期数据将被清除。
客户端消费者是SrsRtmpConnPlay类型，消费者播放流的流程在下一篇文章中介绍。