SRS 流媒体服务器源码分析：Rtmp publish 流程
发布于: 2021 年 03 月 03 日
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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;}
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先看看listen_rtmp():
int SrsServer::listen_rtmp(){ int ret = ERROR_SUCCESS; // stream service port. std::vector<std::string> 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;}
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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);}
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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;}
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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; }
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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;}
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在上面根据type创建不同的SrsConnection，Rtmp创建了SrsRtmpConn，并且加入到std::vector<SrsConnection*> conns;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();}
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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;}
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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;}
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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;}
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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; } ...}
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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<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;}
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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;}
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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<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;}
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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."); }
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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;}
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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<int64_t, SrsSharedPtrMessage*> msgs。
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;}
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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); }}
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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。
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发布于: 2021 年 03 月 03 日阅读数: 12
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纸上得来终觉浅，绝知此事要躬行
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