Merge branch 'master' into dev

.clang-format

@@ -50,8 +50,8 @@ BraceWrapping:
 BreakConstructorInitializers: BeforeComma
 # 继承过长需要换行时也在 `,` 前
 BreakInheritanceList: BeforeComma
-# 列宽 120
+# 列宽 160
-ColumnLimit: 120
+ColumnLimit: 160
 # c++11 括号内起始/结束无空格, false 会加上
 Cpp11BracedListStyle: false
 # 命名空间后的注释会修正为: // namespace_name

3rdpart/CMakeLists.txt

@@ -153,7 +153,7 @@ file(GLOB ToolKit_SRC_LIST
   ${ToolKit_ROOT}/src/*/*.c)
 if(IOS)
   list(APPEND ToolKit_SRC_LIST
-    ${ToolKit_ROOT}/Network/Socket_ios.mm)
+    ${ToolKit_ROOT}/src/Network/Socket_ios.mm)
 endif()
 
 ###################################################################

3rdpart/ZLToolKit

@@ -1 +1 @@
-Subproject commit ca26e43a5f62986bb8a007226e0bad148d154abc
+Subproject commit 7e40c751659d5c1ec623699732284c12e0a4feb8

3rdpart/media-server

@@ -1 +1 @@
-Subproject commit 5aa9884660df1c193d730a90835af36ee411668c
+Subproject commit cdbb3d6b9ea254f454c6e466c5962af5ace01199

api/source/mk_common.cpp

@@ -308,7 +308,7 @@ API_EXPORT void API_CALL mk_webrtc_get_answer_sdp2(void *user_data, on_user_data
     WebRtcPluginManager::Instance().getAnswerSdp(*session, type, WebRtcArgsUrl(url),
                                                  [offer_str, session, ptr, cb](const WebRtcInterface &exchanger) mutable {
         try {
-            auto sdp_answer = const_cast<WebRtcInterface &>(exchanger).getAnswerSdp(offer_str);
+            auto sdp_answer = exchangeSdp(exchanger, offer_str);
             cb(ptr.get(), sdp_answer.data(), nullptr);
         } catch (std::exception &ex) {
             cb(ptr.get(), nullptr, ex.what());

api/source/mk_media.cpp

@@ -27,13 +27,9 @@ public:
 
     ~MediaHelper() = default;
 
-    void attachEvent(){
+    void attachEvent() { _channel->setMediaListener(shared_from_this()); }
-        _channel->setMediaListener(shared_from_this());
-    }
 
-    DevChannel::Ptr &getChannel(){
+    DevChannel::Ptr &getChannel() { return _channel; }
-        return _channel;
-    }
 
     void setOnClose(on_mk_media_close cb, std::shared_ptr<void> user_data) {
         _on_close = cb;
@@ -103,9 +99,7 @@ protected:
         }
     }
 
-    toolkit::EventPoller::Ptr getOwnerPoller(MediaSource &sender) {
+    toolkit::EventPoller::Ptr getOwnerPoller(MediaSource &sender) override { return _poller; }
-        return _poller;
-    }
 
 private:
     EventPoller::Ptr _poller;
@@ -290,7 +284,8 @@ API_EXPORT void API_CALL mk_media_start_send_rtp(mk_media ctx, const char *dst_u
     mk_media_start_send_rtp2(ctx, dst_url, dst_port, ssrc, is_udp, cb, user_data, nullptr);
 }
 
-API_EXPORT void API_CALL mk_media_start_send_rtp2(mk_media ctx, const char *dst_url, uint16_t dst_port, const char *ssrc, int is_udp, on_mk_media_send_rtp_result cb, void *user_data, on_user_data_free user_data_free){
+API_EXPORT void API_CALL mk_media_start_send_rtp2(mk_media ctx, const char *dst_url, uint16_t dst_port, const char *ssrc, int is_udp, on_mk_media_send_rtp_result cb, void *user_data,
+    on_user_data_free user_data_free) {
     assert(ctx && dst_url && ssrc);
     MediaHelper::Ptr* obj = (MediaHelper::Ptr*) ctx;
 

api/source/mk_player.cpp

@@ -24,7 +24,7 @@ public:
     MediaPlayerForC(){
         _player = std::make_shared<MediaPlayer>();
     }
-    ~MediaPlayerForC(){}
+    ~MediaPlayerForC() = default;
 
     MediaPlayer *operator->(){
         return _player.get();

api/source/mk_tcp.cpp

@@ -115,7 +115,7 @@ API_EXPORT uint16_t API_CALL mk_sock_info_local_port(const mk_sock_info ctx){
 API_EXPORT mk_sock_info API_CALL mk_tcp_session_get_sock_info(const mk_tcp_session ctx) {
     assert(ctx);
     SessionForC *session = (SessionForC *)ctx;
-    return (mk_sock_info)session;
+    return reinterpret_cast<mk_sock_info>(static_cast<SockInfo *>(session));
 }
 
 API_EXPORT void API_CALL mk_tcp_session_shutdown(const mk_tcp_session ctx,int err,const char *err_msg){
@@ -320,7 +320,7 @@ TcpClientForC::Ptr *mk_tcp_client_create_l(mk_tcp_client_events *events, mk_tcp_
 API_EXPORT mk_sock_info API_CALL mk_tcp_client_get_sock_info(const mk_tcp_client ctx){
     assert(ctx);
     TcpClientForC::Ptr *client = (TcpClientForC::Ptr *)ctx;
-    return (mk_sock_info)(SockInfo *)client->get();
+    return reinterpret_cast<mk_sock_info>(static_cast<SockInfo *>(client->get()));
 }
 
 API_EXPORT mk_tcp_client API_CALL mk_tcp_client_create(mk_tcp_client_events *events, mk_tcp_type type){

dockerfile

@@ -8,6 +8,7 @@ EXPOSE 443/tcp
 EXPOSE 10000/udp
 EXPOSE 10000/tcp
 EXPOSE 8000/udp
+EXPOSE 8000/tcp
 EXPOSE 9000/udp
 
 # ADD sources.list /etc/apt/sources.list
@@ -82,4 +83,4 @@ COPY --from=build /opt/media/ZLMediaKit/release/linux/${MODEL}/MediaServer /opt/
 COPY --from=build /opt/media/ZLMediaKit/release/linux/${MODEL}/config.ini /opt/media/conf/
 COPY --from=build /opt/media/ZLMediaKit/www/ /opt/media/bin/www/
 ENV PATH /opt/media/bin:$PATH
-CMD ["sh","-c","./MediaServer -s default.pem -c ../conf/config.ini"]
+CMD ["sh","-c","./MediaServer -s default.pem -c ../conf/config.ini -l 0"]

postman/ZLMediaKit.postman_collection.json

@@ -1488,6 +1488,42 @@
 			},
 			"response": []
 		},
+		{
+			"name": "更新RTP服务器过滤SSRC(updateRtpServerSSRC)",
+			"request": {
+				"method": "GET",
+				"header": [],
+				"url": {
+					"raw": "{{ZLMediaKit_URL}}/index/api/updateRtpServerSSRC?secret={{ZLMediaKit_secret}}&stream_id=test&ssrc=123456",
+					"host": [
+						"{{ZLMediaKit_URL}}"
+					],
+					"path": [
+						"index",
+						"api",
+						"updateRtpServerSSRC"
+					],
+					"query": [
+						{
+							"key": "secret",
+							"value": "{{ZLMediaKit_secret}}",
+							"description": "api操作密钥(配置文件配置)，如果操作ip是127.0.0.1，则不需要此参数"
+						},
+						{
+							"key": "stream_id",
+							"value": "test",
+							"description": "该端口绑定的流id"
+						},
+						{
+							"key": "ssrc",
+							"value": "123456",
+							"description": "十进制ssrc"
+						}
+					]
+				}
+			},
+			"response": []
+		},
 		{
 			"name": "暂停RTP超时检查(pauseRtpCheck)",
 			"request": {

server/WebApi.cpp

@@ -247,7 +247,7 @@ static inline void addHttpListener(){
                     size = body->remainSize();
                 }
 
-                LogContextCapture log(getLogger(), LDebug, __FILE__, "http api debug", __LINE__);
+                LogContextCapture log(getLogger(), toolkit::LTrace, __FILE__, "http api debug", __LINE__);
                 log << "\r\n# request:\r\n" << parser.Method() << " " << parser.FullUrl() << "\r\n";
                 log << "# header:\r\n";
 
@@ -966,7 +966,7 @@ void installWebApi() {
         //开始推流，如果推流失败或者推流中止，将会自动重试若干次，默认一直重试
         pusher->setPushCallbackOnce([cb, key, url](const SockException &ex) {
             if (ex) {
-                WarnL << "Push " << url << " failed, key: " << key << ", err: " << ex.what();
+                WarnL << "Push " << url << " failed, key: " << key << ", err: " << ex;
                 lock_guard<recursive_mutex> lck(s_proxyPusherMapMtx);
                 s_proxyPusherMap.erase(key);
             }
@@ -975,7 +975,7 @@ void installWebApi() {
 
         //被主动关闭推流
         pusher->setOnClose([key, url](const SockException &ex) {
-            WarnL << "Push " << url << " failed, key: " << key << ", err: " << ex.what();
+            WarnL << "Push " << url << " failed, key: " << key << ", err: " << ex;
             lock_guard<recursive_mutex> lck(s_proxyPusherMapMtx);
             s_proxyPusherMap.erase(key);
         });
@@ -1182,6 +1182,18 @@ void installWebApi() {
         val["hit"] = 1;
     });
 
+    api_regist("/index/api/updateRtpServerSSRC",[](API_ARGS_MAP){
+        CHECK_SECRET();
+        CHECK_ARGS("stream_id", "ssrc");
+
+        lock_guard<recursive_mutex> lck(s_rtpServerMapMtx);
+        auto it = s_rtpServerMap.find(allArgs["stream_id"]);
+        if (it == s_rtpServerMap.end()) {
+            throw ApiRetException("RtpServer not found by stream_id", API::NotFound);
+        }
+        it->second->updateSSRC(allArgs["ssrc"]);
+    });
+
     api_regist("/index/api/listRtpServer",[](API_ARGS_MAP){
         CHECK_SECRET();
 
@@ -1595,7 +1607,7 @@ void installWebApi() {
         auto offer = allArgs.getArgs();
         CHECK(!offer.empty(), "http body(webrtc offer sdp) is empty");
 
-        WebRtcPluginManager::Instance().getAnswerSdp(*(static_cast<Session *>(&sender)), type,
+        WebRtcPluginManager::Instance().getAnswerSdp(static_cast<Session&>(sender), type,
                                                      WebRtcArgsImp(allArgs, sender.getIdentifier()),
                                                      [invoker, val, offer, headerOut](const WebRtcInterface &exchanger) mutable {
             //设置返回类型
@@ -1604,7 +1616,7 @@ void installWebApi() {
             headerOut["Access-Control-Allow-Origin"] = "*";
 
             try {
-                val["sdp"] = const_cast<WebRtcInterface &>(exchanger).getAnswerSdp(offer);
+                val["sdp"] = exchangeSdp(exchanger, offer);
                 val["id"] = exchanger.getIdentifier();
                 val["type"] = "answer";
                 invoker(200, headerOut, val.toStyledString());
@@ -1620,7 +1632,7 @@ void installWebApi() {
         auto offer = allArgs.getArgs();
         CHECK(!offer.empty(), "http body(webrtc offer sdp) is empty");
 
-        WebRtcPluginManager::Instance().getAnswerSdp(*(static_cast<Session *>(&sender)), type,
+        WebRtcPluginManager::Instance().getAnswerSdp(static_cast<Session&>(sender), type,
                                                      WebRtcArgsImp(allArgs, sender.getIdentifier()),
                                                      [invoker, offer, headerOut](const WebRtcInterface &exchanger) mutable {
                 // 设置跨域
@@ -1628,7 +1640,7 @@ void installWebApi() {
                 try {
                     // 设置返回类型
                     headerOut["Content-Type"] = "application/sdp";
-                    invoker(201, headerOut, const_cast<WebRtcInterface &>(exchanger).getAnswerSdp(offer));
+                    invoker(201, headerOut, exchangeSdp(exchanger, offer));
                 } catch (std::exception &ex) {
                     headerOut["Content-Type"] = "text/plain";
                     invoker(406, headerOut, ex.what());

server/WebHook.cpp

@@ -14,10 +14,10 @@
 #include "Util/NoticeCenter.h"
 #include "Common/config.h"
 #include "Common/MediaSource.h"
+#include "Http/HttpSession.h"
 #include "Http/HttpRequester.h"
 #include "Network/Session.h"
 #include "Rtsp/RtspSession.h"
-#include "Http/HttpSession.h"
 #include "WebHook.h"
 #include "WebApi.h"
 
@@ -52,7 +52,7 @@ const string kAliveInterval = HOOK_FIELD"alive_interval";
 const string kRetry = HOOK_FIELD "retry";
 const string kRetryDelay = HOOK_FIELD "retry_delay";
 
-onceToken token([](){
+static onceToken token([]() {
     mINI::Instance()[kEnable] = false;
     mINI::Instance()[kTimeoutSec] = 10;
     // 默认hook地址设置为空，采用默认行为(例如不鉴权)
@@ -76,7 +76,7 @@ onceToken token([](){
     mINI::Instance()[kAliveInterval] = 30.0;
     mINI::Instance()[kRetry] = 1;
     mINI::Instance()[kRetryDelay] = 3.0;
-},nullptr);
+});
 } // namespace Hook
 
 namespace Cluster {
@@ -93,11 +93,10 @@ static onceToken token([]() {
 
 } // namespace Cluster
 
-static void parse_http_response(const SockException &ex, const Parser &res,
+static void parse_http_response(const SockException &ex, const Parser &res, const function<void(const Value &, const string &, bool)> &fun) {
-                                const function<void(const Value &,const string &,const bool &)> &fun){
     bool should_retry = true;
     if (ex) {
-        auto errStr = StrPrinter << "[network err]:" << ex.what() << endl;
+        auto errStr = StrPrinter << "[network err]:" << ex << endl;
         fun(Json::nullValue, errStr, should_retry);
         return;
     }
@@ -183,7 +182,7 @@ void do_http_hook(const string &url, const ArgsType &body, const function<void(c
     Ticker ticker;
     requester->startRequester(url, [url, func, bodyStr, body, requester, ticker, retry](const SockException &ex, const Parser &res) mutable {
         onceToken token(nullptr, [&]() mutable { requester.reset(); });
-            parse_http_response(ex, res, [&](const Value &obj, const string &err,const bool &should_retry) {
+        parse_http_response(ex, res, [&](const Value &obj, const string &err, bool should_retry) {
             if (!err.empty()) {
                 // hook失败
                 WarnL << "hook " << url << " " << ticker.elapsedTime() << "ms,failed" << err << ":" << bodyStr;
@@ -272,9 +271,7 @@ static string getPullUrl(const string &origin_fmt, const MediaInfo &info) {
     return string(url) + '?' + kEdgeServerParam + '&' + VHOST_KEY + '=' + info._vhost + '&' + info._param_strs;
 }
 
-static void pullStreamFromOrigin(const vector<string>& urls, size_t index, size_t failed_cnt, const MediaInfo &args,
+static void pullStreamFromOrigin(const vector<string> &urls, size_t index, size_t failed_cnt, const MediaInfo &args, const function<void()> &closePlayer) {
-                                 const function<void()> &closePlayer) {
-
     GET_CONFIG(float, cluster_timeout_sec, Cluster::kTimeoutSec);
     GET_CONFIG(int, retry_count, Cluster::kRetryCount);
 
@@ -286,8 +283,7 @@ static void pullStreamFromOrigin(const vector<string>& urls, size_t index, size_
     option.enable_hls = option.enable_hls || (args._schema == HLS_SCHEMA);
     option.enable_mp4 = false;
 
-    addStreamProxy(args._vhost, args._app, args._streamid, url, retry_count, option, Rtsp::RTP_TCP, timeout_sec,
+    addStreamProxy(args._vhost, args._app, args._streamid, url, retry_count, option, Rtsp::RTP_TCP, timeout_sec, [=](const SockException &ex, const string &key) mutable {
-                  [=](const SockException &ex, const string &key) mutable {
         if (!ex) {
             return;
         }
@@ -359,9 +355,7 @@ void installWebHook(){
         body["port"] = sender.get_peer_port();
         body["id"] = sender.getIdentifier();
         // 执行hook
-        do_http_hook(hook_play,body,[invoker](const Value &obj,const string &err){
+        do_http_hook(hook_play, body, [invoker](const Value &obj, const string &err) { invoker(err); });
-            invoker(err);
-        });
     });
 
     NoticeCenter::Instance().addListener(&web_hook_tag, Broadcast::kBroadcastFlowReport, [](BroadcastFlowReportArgs) {
@@ -380,7 +374,6 @@ void installWebHook(){
         do_http_hook(hook_flowreport, body, nullptr);
     });
 
-
     static const string unAuthedRealm = "unAuthedRealm";
 
     // 监听kBroadcastOnGetRtspRealm事件决定rtsp链接是否需要鉴权(传统的rtsp鉴权方案)才能访问
@@ -432,7 +425,6 @@ void installWebHook(){
         });
     });
 
-
     // 监听rtsp、rtmp源注册或注销事件
     NoticeCenter::Instance().addListener(&web_hook_tag, Broadcast::kBroadcastMediaChanged, [](BroadcastMediaChangedArgs) {
         GET_CONFIG(string, hook_stream_chaned, Hook::kOnStreamChanged);
@@ -552,9 +544,7 @@ void installWebHook(){
         body["passwd"] = passwd;
 
         // 执行hook
-        do_http_hook(hook_shell_login,body, [invoker](const Value &,const string &err){
+        do_http_hook(hook_shell_login, body, [invoker](const Value &, const string &err) { invoker(err); });
-            invoker(err);
-        });
     });
 
     NoticeCenter::Instance().addListener(&web_hook_tag, Broadcast::kBroadcastStreamNoneReader, [](BroadcastStreamNoneReaderArgs) {

server/main.cpp

@@ -123,7 +123,7 @@ public:
         (*_parser) << Option('l',/*该选项简称，如果是\x00则说明无简称*/
                              "level",/*该选项全称,每个选项必须有全称；不得为null或空字符串*/
                              Option::ArgRequired,/*该选项后面必须跟值*/
-                             to_string(LTrace).data(),/*该选项默认值*/
+                             to_string(LDebug).data(),/*该选项默认值*/
                              false,/*该选项是否必须赋值，如果没有默认值且为ArgRequired时用户必须提供该参数否则将抛异常*/
                              "日志等级,LTrace~LError(0~4)",/*该选项说明文字*/
                              nullptr);
@@ -160,6 +160,14 @@ public:
                              "启动事件触发线程数",/*该选项说明文字*/
                              nullptr);
 
+        (*_parser) << Option(0,/*该选项简称，如果是\x00则说明无简称*/
+                             "affinity",/*该选项全称,每个选项必须有全称；不得为null或空字符串*/
+                             Option::ArgRequired,/*该选项后面必须跟值*/
+                             to_string(1).data(),/*该选项默认值*/
+                             false,/*该选项是否必须赋值，如果没有默认值且为ArgRequired时用户必须提供该参数否则将抛异常*/
+                             "是否启动cpu亲和性设置",/*该选项说明文字*/
+                             nullptr);
+
 #if defined(ENABLE_VERSION)
         (*_parser) << Option('v', "version", Option::ArgNone, nullptr, false, "显示版本号",
                              [](const std::shared_ptr<ostream> &stream, const string &arg) -> bool {
@@ -200,15 +208,16 @@ int start_main(int argc,char *argv[]) {
         g_ini_file = cmd_main["config"];
         string ssl_file = cmd_main["ssl"];
         int threads = cmd_main["threads"];
+        bool affinity = cmd_main["affinity"];
 
         //设置日志
         Logger::Instance().add(std::make_shared<ConsoleChannel>("ConsoleChannel", logLevel));
-#ifndef ANDROID
+#if !defined(ANDROID)
         auto fileChannel = std::make_shared<FileChannel>("FileChannel", exeDir() + "log/", logLevel);
         // 日志最多保存天数
         fileChannel->setMaxDay(cmd_main["max_day"]);
         Logger::Instance().add(fileChannel);
-#endif//
+#endif // !defined(ANDROID)
 
 #if !defined(_WIN32)
         pid_t pid = getpid();
@@ -252,24 +261,27 @@ int start_main(int argc,char *argv[]) {
         uint16_t httpsPort = mINI::Instance()[Http::kSSLPort];
         uint16_t rtpPort = mINI::Instance()[RtpProxy::kPort];
 
-        //设置poller线程数,该函数必须在使用ZLToolKit网络相关对象之前调用才能生效
+        //设置poller线程数和cpu亲和性,该函数必须在使用ZLToolKit网络相关对象之前调用才能生效
+        //如果需要调用getSnap和addFFmpegSource接口，可以关闭cpu亲和性
+
         EventPollerPool::setPoolSize(threads);
+        EventPollerPool::enableCpuAffinity(affinity);
 
         //简单的telnet服务器，可用于服务器调试，但是不能使用23端口，否则telnet上了莫名其妙的现象
         //测试方法:telnet 127.0.0.1 9000
         auto shellSrv = std::make_shared<TcpServer>();
 
         //rtsp[s]服务器, 可用于诸如亚马逊echo show这样的设备访问
-        auto rtspSrv = std::make_shared<TcpServer>();;
+        auto rtspSrv = std::make_shared<TcpServer>();
-        auto rtspSSLSrv = std::make_shared<TcpServer>();;
+        auto rtspSSLSrv = std::make_shared<TcpServer>();
 
         //rtmp[s]服务器
-        auto rtmpSrv = std::make_shared<TcpServer>();;
+        auto rtmpSrv = std::make_shared<TcpServer>();
-        auto rtmpsSrv = std::make_shared<TcpServer>();;
+        auto rtmpsSrv = std::make_shared<TcpServer>();
 
         //http[s]服务器
-        auto httpSrv = std::make_shared<TcpServer>();;
+        auto httpSrv = std::make_shared<TcpServer>();
-        auto httpsSrv = std::make_shared<TcpServer>();;
+        auto httpsSrv = std::make_shared<TcpServer>();
 
 #if defined(ENABLE_RTPPROXY)
         //GB28181 rtp推流端口，支持UDP/TCP

src/Common/MultiMediaSourceMuxer.cpp

@@ -270,7 +270,7 @@ void MultiMediaSourceMuxer::startSendRtp(MediaSource &sender, const MediaSourceE
             if (auto strong_self = weak_self.lock()) {
                 // 可能归属线程发生变更
                 strong_self->getOwnerPoller(MediaSource::NullMediaSource())->async([=]() {
-                    WarnL << "stream:" << strong_self->shortUrl() << " stop send rtp:" << ssrc << ", reason:" << ex.what();
+                    WarnL << "stream:" << strong_self->shortUrl() << " stop send rtp:" << ssrc << ", reason:" << ex;
                     strong_self->_rtp_sender.erase(ssrc);
                     NoticeCenter::Instance().emitEvent(Broadcast::kBroadcastSendRtpStopped, *strong_self, ssrc, ex);
                 });
@@ -393,12 +393,11 @@ void MultiMediaSourceMuxer::createGopCacheIfNeed() {
         return;
     }
     weak_ptr<MultiMediaSourceMuxer> weak_self = shared_from_this();
-    _ring = std::make_shared<RingType>(1024, [weak_self](int size) {
+    auto src = std::make_shared<MediaSourceForMuxer>(weak_self.lock());
-        auto strong_self = weak_self.lock();
+    _ring = std::make_shared<RingType>(1024, [weak_self, src](int size) {
-        if (strong_self) {
+        if (auto strong_self = weak_self.lock()) {
             // 切换到归属线程
             strong_self->getOwnerPoller(MediaSource::NullMediaSource())->async([=]() {
-                auto src = std::make_shared<MediaSourceForMuxer>(strong_self);
                 strong_self->onReaderChanged(*src, strong_self->totalReaderCount());
             });
         }

src/Extension/Track.h

@@ -24,9 +24,8 @@ namespace mediakit{
 class Track : public FrameDispatcher , public CodecInfo{
 public:
     using Ptr = std::shared_ptr<Track>;
-    Track(){}
+    Track() = default;
-
+    virtual ~Track() = default;
-    virtual ~Track(){}
 
     /**
      * 是否准备好，准备好才能获取譬如sps pps等信息
@@ -177,8 +176,8 @@ private:
 
 class TrackSource{
 public:
-    TrackSource(){}
+    TrackSource() = default;
-    virtual ~TrackSource(){}
+    virtual ~TrackSource() = default;
 
     /**
      * 获取全部的Track

src/Http/HlsPlayer.cpp

@@ -118,7 +118,7 @@ void HlsPlayer::fetchSegment() {
             return;
         }
         if (err) {
-            WarnL << "Download ts segment " << url << " failed:" << err.what();
+            WarnL << "Download ts segment " << url << " failed:" << err;
             if (err.getErrCode() == Err_timeout) {
                 strong_self->_timeout_multiple = MAX(strong_self->_timeout_multiple + 1, MAX_TIMEOUT_MULTIPLE);
             }else{

src/Http/HttpBody.cpp

@@ -189,7 +189,7 @@ public:
         _data = map_addr.get() + offset;
         _size = size;
     }
-    ~BufferMmap() override {};
+    ~BufferMmap() override = default;
     //返回数据长度
     char *data() const override { return _data; }
     size_t size() const override { return _size; }

src/Http/HttpBody.h

@@ -30,9 +30,9 @@ namespace mediakit {
 class HttpBody : public std::enable_shared_from_this<HttpBody>{
 public:
     using Ptr = std::shared_ptr<HttpBody>;
-    HttpBody(){}
+    HttpBody() = default;
 
-    virtual ~HttpBody(){}
+    virtual ~HttpBody() = default;
 
     /**
      * 剩余数据大小，如果返回-1, 那么就不设置content-length
@@ -151,7 +151,7 @@ public:
      * @param boundary boundary字符串
      */
     HttpMultiFormBody(const HttpArgs &args,const std::string &filePath,const std::string &boundary = "0xKhTmLbOuNdArY");
-    virtual ~HttpMultiFormBody(){}
+    virtual ~HttpMultiFormBody() = default;
     int64_t remainSize() override ;
     toolkit::Buffer::Ptr readData(size_t size) override;
 

src/Http/HttpCookie.h

@@ -27,8 +27,8 @@ class HttpCookie {
 public:
     using Ptr = std::shared_ptr<HttpCookie>;
     friend class HttpCookieStorage;
-    HttpCookie(){}
+    HttpCookie() = default;
-    ~HttpCookie(){}
+    ~HttpCookie() = default;
 
     void setPath(const std::string &path);
     void setHost(const std::string &host);
@@ -52,12 +52,14 @@ private:
  */
 class HttpCookieStorage{
 public:
-    ~HttpCookieStorage(){}
+    ~HttpCookieStorage() = default;
     static HttpCookieStorage &Instance();
     void set(const HttpCookie::Ptr &cookie);
     std::vector<HttpCookie::Ptr> get(const std::string &host,const std::string &path);
+
 private:
-    HttpCookieStorage(){};
+    HttpCookieStorage() = default;
+
 private:
     std::unordered_map<std::string/*host*/, std::map<std::string/*cookie path*/,std::map<std::string/*cookie_key*/, HttpCookie::Ptr> > > _all_cookie;
     std::mutex _mtx_cookie;

src/Http/HttpFileManager.h

@@ -24,8 +24,8 @@ public:
     typedef std::function<void(int code, const StrCaseMap &headerOut, const HttpBody::Ptr &body)> HttpResponseInvokerLambda0;
     typedef std::function<void(int code, const StrCaseMap &headerOut, const std::string &body)> HttpResponseInvokerLambda1;
 
-    HttpResponseInvokerImp(){}
+    HttpResponseInvokerImp() = default;
-    ~HttpResponseInvokerImp(){}
+    ~HttpResponseInvokerImp() = default;
     template<typename C>
     HttpResponseInvokerImp(const C &c):HttpResponseInvokerImp(typename toolkit::function_traits<C>::stl_function_type(c)) {}
     HttpResponseInvokerImp(const HttpResponseInvokerLambda0 &lambda);

src/Http/HttpRequestSplitter.h

@@ -18,8 +18,8 @@ namespace mediakit {
 
 class HttpRequestSplitter {
 public:
-    HttpRequestSplitter(){};
+    HttpRequestSplitter() = default;
-    virtual ~HttpRequestSplitter(){};
+    virtual ~HttpRequestSplitter() = default;
 
     /**
      * 添加数据

src/Http/HttpSession.cpp

@@ -24,14 +24,11 @@ using namespace toolkit;
 namespace mediakit {
 
 HttpSession::HttpSession(const Socket::Ptr &pSock) : Session(pSock) {
-    TraceP(this);
     GET_CONFIG(uint32_t,keep_alive_sec,Http::kKeepAliveSecond);
     pSock->setSendTimeOutSecond(keep_alive_sec);
 }
 
-HttpSession::~HttpSession() {
+HttpSession::~HttpSession() = default;
-    TraceP(this);
-}
 
 void HttpSession::Handle_Req_HEAD(ssize_t &content_len){
     //暂时全部返回200 OK，因为HTTP GET存在按需生成流的操作，所以不能按照HTTP GET的流程返回
@@ -104,7 +101,7 @@ void HttpSession::onError(const SockException& err) {
         uint64_t duration = _ticker.createdTime() / 1000;
         WarnP(this) << "FLV/TS/FMP4播放器("
                     << _mediaInfo.shortUrl()
-                    << ")断开:" << err.what()
+                    << ")断开:" << err
                     << ",耗时(s):" << duration;
 
         GET_CONFIG(uint32_t, iFlowThreshold, General::kFlowThreshold);
@@ -114,9 +111,6 @@ void HttpSession::onError(const SockException& err) {
         }
         return;
     }
-
-    //http客户端
-    TraceP(this) << err.what();
 }
 
 void HttpSession::onManager() {

src/Http/WebSocketClient.h

@@ -37,7 +37,7 @@ public:
 
     template <typename... ArgsType>
     ClientTypeImp(ArgsType &&...args) : ClientType(std::forward<ArgsType>(args)...) {}
-    ~ClientTypeImp() override {};
+    ~ClientTypeImp() override  = default;
 
 protected:
     /**

src/Http/WebSocketSession.h

@@ -82,7 +82,7 @@ template<typename Creator, typename HttpSessionType = mediakit::HttpSession, med
 class WebSocketSessionBase : public HttpSessionType {
 public:
     WebSocketSessionBase(const toolkit::Socket::Ptr &pSock) : HttpSessionType(pSock){}
-    virtual ~WebSocketSessionBase(){}
+    virtual ~WebSocketSessionBase() = default;
 
     //收到eof或其他导致脱离TcpServer事件的回调
     void onError(const toolkit::SockException &err) override{
@@ -248,7 +248,7 @@ template<typename SessionType,typename HttpSessionType = mediakit::HttpSession,
 class WebSocketSession : public WebSocketSessionBase<SessionCreator<SessionType>,HttpSessionType,DataType>{
 public:
     WebSocketSession(const toolkit::Socket::Ptr &pSock) : WebSocketSessionBase<SessionCreator<SessionType>,HttpSessionType,DataType>(pSock){}
-    virtual ~WebSocketSession(){}
+    virtual ~WebSocketSession() = default;
 };
 
 #endif //ZLMEDIAKIT_WEBSOCKETSESSION_H

src/Http/WebSocketSplitter.h

@@ -51,7 +51,7 @@ public:
         //根据内存地址设置掩码随机数
         _mask.assign((uint8_t*)(&ptr), (uint8_t*)(&ptr) + 4);
     }
-    virtual ~WebSocketHeader(){}
+    virtual ~WebSocketHeader() = default;
 
 public:
     bool _fin;
@@ -71,7 +71,7 @@ public:
     WebSocketBuffer(WebSocketHeader::Type headType, bool fin, ARGS &&...args)
             :  toolkit::BufferString(std::forward<ARGS>(args)...), _fin(fin), _head_type(headType){}
 
-    ~WebSocketBuffer() override {}
+    ~WebSocketBuffer() override = default;
 
     WebSocketHeader::Type headType() const { return _head_type; }
 
@@ -84,8 +84,8 @@ private:
 
 class WebSocketSplitter : public WebSocketHeader{
 public:
-    WebSocketSplitter(){}
+    WebSocketSplitter() = default;
-    virtual ~WebSocketSplitter(){}
+    virtual ~WebSocketSplitter() = default;
 
     /**
      * 输入数据以便解包webSocket数据以及处理粘包问题

src/Player/PlayerProxy.cpp

@@ -100,7 +100,13 @@ void PlayerProxy::play(const string &strUrlTmp) {
             strongSelf->_on_close(err);
         }
     });
+    try {
         MediaPlayer::play(strUrlTmp);
+    } catch (std::exception &ex) {
+        ErrorL << ex.what();
+        _on_play_result(SockException(Err_other, ex.what()));
+        return;
+    }
     _pull_url = strUrlTmp;
     setDirectProxy();
 }

src/Record/MP4Muxer.cpp

@@ -121,6 +121,18 @@ bool MP4MuxerInterface::inputFrame(const Frame::Ptr &frame) {
             });
             break;
         }
+        case CodecJPEG:{
+            int64_t dts_out, pts_out;
+            track_info.stamp.revise(frame->dts(), frame->pts(), dts_out, pts_out);
+            mp4_writer_write(_mov_writter.get(),
+                             track_info.track_id,
+                             frame->data(),
+                             frame->size(),
+                             pts_out,
+                             dts_out,
+                             frame->keyFrame() ? MOV_AV_FLAG_KEYFREAME : 0);
+            break;
+        }
 
         default: {
             int64_t dts_out, pts_out;

src/Record/MPEG.h

@@ -86,7 +86,7 @@ namespace mediakit {
 
 class MpegMuxer : public MediaSinkInterface {
 public:
-    MpegMuxer(bool is_ps) {};
+    MpegMuxer(bool is_ps) = default;
     ~MpegMuxer() override = default;
     bool addTrack(const Track::Ptr &track) override { return false; }
     void resetTracks() override {}

src/Rtcp/Rtcp.cpp

@@ -242,11 +242,9 @@ vector<RtcpHeader *> RtcpHeader::loadFromBytes(char *data, size_t len) {
 class BufferRtcp : public Buffer {
 public:
     BufferRtcp(std::shared_ptr<RtcpHeader> rtcp) { _rtcp = std::move(rtcp); }
-
+    ~BufferRtcp() override  = default;
-    ~BufferRtcp() override {}
 
     char *data() const override { return (char *)_rtcp.get(); }
-
     size_t size() const override { return _rtcp->getSize(); }
 
 private:
@@ -551,7 +549,7 @@ const void *RtcpFB::getFciPtr() const {
 
 size_t RtcpFB::getFciSize() const {
     auto fci_len = (ssize_t)getSize() - getPaddingSize() - sizeof(RtcpFB);
-    CHECK(fci_len >= 0);
+    CHECK(getSize() >= getPaddingSize() + sizeof(RtcpFB));
     return fci_len;
 }
 

src/Rtmp/Rtmp.cpp

@@ -16,7 +16,7 @@ TitleMeta::TitleMeta(float dur_sec, size_t fileSize, const std::map<std::string,
 {
     _metadata.set("duration", dur_sec);
     _metadata.set("fileSize", (int)fileSize);
-    _metadata.set("server", kServerName);
+    _metadata.set("title", std::string("Streamed by ") + kServerName);
     for (auto &pr : header) {
         _metadata.set(pr.first, pr.second);
     }

src/Rtmp/Rtmp.h

@@ -208,7 +208,7 @@ public:
     using Ptr = std::shared_ptr<Metadata>;
 
     Metadata(): _metadata(AMF_OBJECT) {}
-    virtual ~Metadata(){}
+    virtual ~Metadata() = default;
     const AMFValue &getMetadata() const{
         return _metadata;
     }
@@ -239,7 +239,7 @@ public:
     using Ptr = std::shared_ptr<VideoMeta>;
 
     VideoMeta(const VideoTrack::Ptr &video);
-    virtual ~VideoMeta(){}
+    virtual ~VideoMeta() = default;
 
     CodecId getCodecId() const override{
         return _codecId;
@@ -253,8 +253,7 @@ public:
     using Ptr = std::shared_ptr<AudioMeta>;
 
     AudioMeta(const AudioTrack::Ptr &audio);
-
+    virtual ~AudioMeta() = default;
-    virtual ~AudioMeta(){}
 
     CodecId getCodecId() const override{
         return _codecId;

src/Rtmp/RtmpMediaSource.h

@@ -99,7 +99,7 @@ public:
      */
     virtual void setMetaData(const AMFValue &metadata) {
         _metadata = metadata;
-        _metadata.set("server", kServerName);
+        _metadata.set("title", std::string("Streamed by ") + kServerName);
         _have_video = _metadata["videocodecid"];
         _have_audio = _metadata["audiocodecid"];
         if (_ring) {

src/Rtmp/RtmpPlayer.cpp

@@ -101,7 +101,7 @@ void RtmpPlayer::onPlayResult_l(const SockException &ex, bool handshake_done) {
         return;
     }
 
-    WarnL << ex.getErrCode() << " " << ex.what();
+    WarnL << ex.getErrCode() << " " << ex;
     if (!handshake_done) {
         //开始播放阶段
         _play_timer.reset();

src/Rtmp/RtmpProtocol.cpp

@@ -187,7 +187,7 @@ public:
         _size = size;
     }
 
-    ~BufferPartial() override{}
+    ~BufferPartial() override = default;
 
     char *data() const override {
         return _data;

src/Rtmp/RtmpSession.cpp

@@ -19,14 +19,11 @@ using namespace toolkit;
 namespace mediakit {
 
 RtmpSession::RtmpSession(const Socket::Ptr &sock) : Session(sock) {
-    DebugP(this);
     GET_CONFIG(uint32_t,keep_alive_sec,Rtmp::kKeepAliveSecond);
     sock->setSendTimeOutSecond(keep_alive_sec);
 }
 
-RtmpSession::~RtmpSession() {
+RtmpSession::~RtmpSession() = default;
-    DebugP(this);
-}
 
 void RtmpSession::onError(const SockException& err) {
     bool is_player = !_push_src_ownership;

src/Rtp/RtpSender.cpp

@@ -336,7 +336,7 @@ void RtpSender::onFlushRtpList(shared_ptr<List<Buffer::Ptr> > rtp_list) {
 
 void RtpSender::onErr(const SockException &ex) {
     _is_connect = false;
-    WarnL << "send rtp connection lost: " << ex.what();
+    WarnL << "send rtp connection lost: " << ex;
     onClose(ex);
 }
 

src/Rtp/RtpServer.cpp

@@ -102,7 +102,8 @@ public:
                     process->setOnDetach(std::move(strong_self->_on_detach));
                 }
                 if (!process) { // process 未创建，触发rtp server 超时事件
-                    NoticeCenter::Instance().emitEvent(Broadcast::KBroadcastRtpServerTimeout,strong_self->_local_port,strong_self->_stream_id,(int)strong_self->_tcp_mode,strong_self->_re_use_port,strong_self->_ssrc);
+                    NoticeCenter::Instance().emitEvent(Broadcast::KBroadcastRtpServerTimeout, strong_self->_local_port, strong_self->_stream_id,
+                                                       (int)strong_self->_tcp_mode, strong_self->_re_use_port, strong_self->_ssrc);
                 }
             }
             return 0;
@@ -198,11 +199,14 @@ void RtpServer::start(uint16_t local_port, const string &stream_id, TcpMode tcp_
         helper->startRtcp();
         helper->setRtpServerInfo(local_port, tcp_mode, re_use_port, ssrc, only_audio);
         bool bind_peer_addr = false;
-        rtp_socket->setOnRead([rtp_socket, helper, ssrc, bind_peer_addr](const Buffer::Ptr &buf, struct sockaddr *addr, int addr_len) mutable {
+        auto ssrc_ptr = std::make_shared<uint32_t>(ssrc);
+        _ssrc = ssrc_ptr;
+        rtp_socket->setOnRead([rtp_socket, helper, ssrc_ptr, bind_peer_addr](const Buffer::Ptr &buf, struct sockaddr *addr, int addr_len) mutable {
             RtpHeader *header = (RtpHeader *)buf->data();
             auto rtp_ssrc = ntohl(header->ssrc);
+            auto ssrc = *ssrc_ptr;
             if (ssrc && rtp_ssrc != ssrc) {
-                WarnL << "ssrc不匹配,rtp已丢弃:" << rtp_ssrc << " != " << ssrc;
+                WarnL << "ssrc mismatched, rtp dropped: " << rtp_ssrc << " != " << ssrc;
             } else {
                 if (!bind_peer_addr) {
                     //绑定对方ip+端口，防止多个设备或一个设备多次推流从而日志报ssrc不匹配问题
@@ -213,19 +217,11 @@ void RtpServer::start(uint16_t local_port, const string &stream_id, TcpMode tcp_
             }
         });
     } else {
-#if 1
         //单端口多线程接收多个流，根据ssrc区分流
         udp_server = std::make_shared<UdpServer>(rtp_socket->getPoller());
         (*udp_server)[RtpSession::kOnlyAudio] = only_audio;
         udp_server->start<RtpSession>(local_port, local_ip);
         rtp_socket = nullptr;
-#else
-        //单端口单线程接收多个流
-        auto &ref = RtpSelector::Instance();
-        rtp_socket->setOnRead([&ref, rtp_socket](const Buffer::Ptr &buf, struct sockaddr *addr, int) {
-            ref.inputRtp(rtp_socket, buf->data(), buf->size(), addr);
-        });
-#endif
     }
 
     _on_cleanup = [rtp_socket, stream_id]() {
@@ -264,7 +260,7 @@ void RtpServer::connectToServer(const std::string &url, uint16_t port, const fun
             return;
         }
         if (err) {
-            WarnL << "连接到服务器 " << url << ":" << port << " 失败 " << err.what();
+            WarnL << "连接到服务器 " << url << ":" << port << " 失败 " << err;
         } else {
             InfoL << "连接到服务器 " << url << ":" << port << " 成功";
             strong_self->onConnect();
@@ -288,5 +284,15 @@ void RtpServer::onConnect() {
     });
 }
 
+void RtpServer::updateSSRC(uint32_t ssrc) {
+    if (_ssrc) {
+        *_ssrc = ssrc;
+    }
+
+    if (_tcp_server) {
+        (*_tcp_server)[RtpSession::kSSRC] = ssrc;
+    }
+}
+
 }//namespace mediakit
 #endif//defined(ENABLE_RTPPROXY)

src/Rtp/RtpServer.h

@@ -64,6 +64,11 @@ public:
      */
     void setOnDetach(std::function<void()> cb);
 
+    /**
+     * 更新ssrc
+     */
+    void updateSSRC(uint32_t ssrc);
+
 private:
     // tcp主动模式连接服务器成功回调
     void onConnect();
@@ -72,6 +77,7 @@ protected:
     toolkit::Socket::Ptr _rtp_socket;
     toolkit::UdpServer::Ptr _udp_server;
     toolkit::TcpServer::Ptr _tcp_server;
+    std::shared_ptr<uint32_t> _ssrc;
     std::shared_ptr<RtcpHelper> _rtcp_helper;
     std::function<void()> _on_cleanup;
 

src/Rtp/RtpSession.cpp

@@ -36,7 +36,6 @@ void RtpSession::setParams(mINI &ini) {
 }
 
 RtpSession::RtpSession(const Socket::Ptr &sock) : Session(sock) {
-    DebugP(this);
     socklen_t addr_len = sizeof(_addr);
     getpeername(sock->rawFD(), (struct sockaddr *)&_addr, &addr_len);
     _is_udp = sock->sockType() == SockNum::Sock_UDP;
@@ -47,7 +46,6 @@ RtpSession::RtpSession(const Socket::Ptr &sock) : Session(sock) {
 }
 
 RtpSession::~RtpSession() {
-    DebugP(this);
     if(_process){
         RtpSelector::Instance().delProcess(_stream_id,_process.get());
     }
@@ -62,7 +60,7 @@ void RtpSession::onRecv(const Buffer::Ptr &data) {
 }
 
 void RtpSession::onError(const SockException &err) {
-    WarnP(this) << _stream_id << " " << err.what();
+    WarnP(this) << _stream_id << " " << err;
 }
 
 void RtpSession::onManager() {
@@ -130,7 +128,7 @@ void RtpSession::onRtpPacket(const char *data, size_t len) {
         uint32_t rtp_ssrc = 0;
         RtpSelector::getSSRC(data, len, rtp_ssrc);
         if (rtp_ssrc != _ssrc) {
-            WarnP(this) << "ssrc不匹配,rtp已丢弃:" << rtp_ssrc << " != " << _ssrc;
+            WarnP(this) << "ssrc mismatched, rtp dropped: " << rtp_ssrc << " != " << _ssrc;
             return;
         }
         _process->inputRtp(false, getSock(), data, len, (struct sockaddr *)&_addr);

src/Rtp/RtpSplitter.cpp

@@ -13,12 +13,8 @@
 #include "RtpSplitter.h"
 namespace mediakit{
 
-static const char kEHOME_MAGIC[] = "\x01\x00\x01\x00";
 static const int  kEHOME_OFFSET = 256;
 
-RtpSplitter::RtpSplitter() {}
-RtpSplitter::~RtpSplitter() {}
-
 ssize_t RtpSplitter::onRecvHeader(const char *data,size_t len){
     //忽略偏移量
     data += _offset;
@@ -42,7 +38,6 @@ static bool isEhome(const char *data, size_t len){
         return true;
     }
     return false;
-    //return memcmp(data, kEHOME_MAGIC, sizeof(kEHOME_MAGIC) - 1) == 0;
 }
 
 const char *RtpSplitter::onSearchPacketTail(const char *data, size_t len) {

src/Rtp/RtpSplitter.h

@@ -18,8 +18,8 @@ namespace mediakit{
 
 class RtpSplitter : public HttpRequestSplitter{
 public:
-    RtpSplitter();
+    RtpSplitter() = default;
-    ~RtpSplitter() override;
+    ~RtpSplitter() override = default;
 
 protected:
     /**

src/Rtp/TSDecoder.h

@@ -25,7 +25,7 @@ class TSSegment : public HttpRequestSplitter {
 public:
     typedef std::function<void(const char *data,size_t len)> onSegment;
     TSSegment(size_t size = TS_PACKET_SIZE) : _size(size){}
-    ~TSSegment(){}
+    ~TSSegment() = default;
     void setOnSegment(onSegment cb);
     static bool isTSPacket(const char *data, size_t len);
 

src/Rtsp/RtpCodec.h

@@ -74,7 +74,7 @@ public:
         _interleaved = interleaved;
     }
 
-    virtual ~RtpInfo() {}
+    virtual ~RtpInfo() = default;
 
     //返回rtp负载最大长度
     size_t getMaxSize() const {

src/Rtsp/Rtsp.h

@@ -237,9 +237,9 @@ class SdpParser {
 public:
     using Ptr = std::shared_ptr<SdpParser>;
 
-    SdpParser() {}
+    SdpParser() = default;
     SdpParser(const std::string &sdp) { load(sdp); }
-    ~SdpParser() {}
+    ~SdpParser() = default;
 
     void load(const std::string &sdp);
     bool available() const;
@@ -268,7 +268,7 @@ public:
         _payload_type = payload_type;
     }
 
-    virtual ~Sdp(){}
+    virtual ~Sdp() = default;
 
     /**
      * 获取sdp字符串

src/Rtsp/RtspSession.cpp

@@ -53,14 +53,11 @@ static unordered_map<string, weak_ptr<RtspSession> > g_mapGetter;
 static recursive_mutex g_mtxGetter;
 
 RtspSession::RtspSession(const Socket::Ptr &sock) : Session(sock) {
-    DebugP(this);
     GET_CONFIG(uint32_t,keep_alive_sec,Rtsp::kKeepAliveSecond);
     sock->setSendTimeOutSecond(keep_alive_sec);
 }
 
-RtspSession::~RtspSession() {
+RtspSession::~RtspSession() = default;
-    DebugP(this);
-}
 
 void RtspSession::onError(const SockException &err) {
     bool is_player = !_push_src_ownership;
@@ -454,7 +451,6 @@ void RtspSession::handleReq_Describe(const Parser &parser) {
 }
 
 void RtspSession::onAuthSuccess() {
-    TraceP(this);
     weak_ptr<RtspSession> weak_self = dynamic_pointer_cast<RtspSession>(shared_from_this());
     MediaSource::findAsync(_media_info, weak_self.lock(), [weak_self](const MediaSource::Ptr &src){
         auto strong_self = weak_self.lock();

src/Rtsp/RtspSplitter.h

@@ -18,8 +18,8 @@ namespace mediakit{
 
 class RtspSplitter : public HttpRequestSplitter{
 public:
-    RtspSplitter(){}
+    RtspSplitter() = default;
-    virtual ~RtspSplitter(){}
+    virtual ~RtspSplitter() = default;
 
     /**
     * 是否允许接收rtp包

src/Shell/ShellSession.cpp

@@ -25,13 +25,10 @@ static onceToken s_token([]() {
 }, nullptr);
 
 ShellSession::ShellSession(const Socket::Ptr &_sock) : Session(_sock) {
-    DebugP(this);
     pleaseInputUser();
 }
 
-ShellSession::~ShellSession() {
+ShellSession::~ShellSession() = default;
-    DebugP(this);
-}
 
 void ShellSession::onRecv(const Buffer::Ptr&buf) {
     //DebugL << hexdump(buf->data(), buf->size());
@@ -60,7 +57,7 @@ void ShellSession::onRecv(const Buffer::Ptr&buf) {
 }
 
 void ShellSession::onError(const SockException &err){
-    WarnP(this) << err.what();
+    WarnP(this) << err;
 }
 
 void ShellSession::onManager() {

srt/SrtSession.cpp

@@ -16,9 +16,7 @@ SrtSession::SrtSession(const Socket::Ptr &sock)
     // TraceL<<"after addr len "<<addr_len<<" family "<<_peer_addr.ss_family;
 }
 
-SrtSession::~SrtSession() {
+SrtSession::~SrtSession() = default;
-    InfoP(this);
-}
 
 EventPoller::Ptr SrtSession::queryPoller(const Buffer::Ptr &buffer) {
     uint8_t *data = (uint8_t *)buffer->data();
@@ -119,7 +117,7 @@ void SrtSession::onError(const SockException &err) {
     // udp链接超时，但是srt链接不一定超时，因为可能存在udp链接迁移的情况
     //在udp链接迁移时，新的SrtSession对象将接管SrtSession对象的生命周期
     //本SrtSession对象将在超时后自动销毁
-    WarnP(this) << err.what();
+    WarnP(this) << err;
 
     if (!_transport) {
         return;

srt/SrtTransportImp.cpp

@@ -6,10 +6,10 @@
 
 namespace SRT {
 SrtTransportImp::SrtTransportImp(const EventPoller::Ptr &poller)
-    : SrtTransport(poller) {}
+    : SrtTransport(poller) {
+}
 
 SrtTransportImp::~SrtTransportImp() {
-    InfoP(this);
     uint64_t duration = _alive_ticker.createdTime() / 1000;
     WarnP(this) << (_is_pusher ? "srt 推流器(" : "srt 播放器(") << _media_info.shortUrl() << ")断开,耗时(s):" << duration;
 

tests/test_wsClient.cpp

@@ -33,7 +33,7 @@ protected:
     }
     //被动断开连接回调
     void onErr(const SockException &ex) override {
-        WarnL << ex.what();
+        WarnL << ex;
     }
     //tcp连接成功后每2秒触发一次该事件
     void onManager() override {
@@ -42,7 +42,7 @@ protected:
     }
     //连接服务器结果回调
     void onConnect(const SockException &ex) override{
-        DebugL << ex.what();
+        DebugL << ex;
     }
 
     //数据全部发送完毕后回调

webrtc/WebRtcSession.cpp

@@ -51,9 +51,7 @@ WebRtcSession::WebRtcSession(const Socket::Ptr &sock) : Session(sock) {
     _over_tcp = sock->sockType() == SockNum::Sock_TCP;
 }
 
-WebRtcSession::~WebRtcSession() {
+WebRtcSession::~WebRtcSession() = default;
-    InfoP(this);
-}
 
 void WebRtcSession::attachServer(const Server &server) {
     _server = std::dynamic_pointer_cast<toolkit::TcpServer>(const_cast<Server &>(server).shared_from_this());
@@ -71,7 +69,7 @@ void WebRtcSession::onRecv_l(const char *data, size_t len) {
         if (!transport->getPoller()->isCurrentThread()) {
             auto sock = Socket::createSocket(transport->getPoller(), false);
             //1、克隆socket(fd不变)，切换poller线程到WebRtcTransport所在线程
-            sock->cloneFromPeerSocket(*(getSock()));
+            sock->cloneSocket(*(getSock()));
             auto server = _server;
             std::string str(data, len);
             sock->getPoller()->async([sock, server, str](){
@@ -105,7 +103,7 @@ void WebRtcSession::onError(const SockException &err) {
     //udp链接超时，但是rtc链接不一定超时，因为可能存在链接迁移的情况
     //在udp链接迁移时，新的WebRtcSession对象将接管WebRtcTransport对象的生命周期
     //本WebRtcSession对象将在超时后自动销毁
-    WarnP(this) << err.what();
+    WarnP(this) << err;
 
     if (!_transport) {
         return;

webrtc/WebRtcTransport.cpp

@@ -1054,7 +1054,7 @@ void WebRtcTransportImp::onBeforeEncryptRtp(const char *buf, int &len, void *ctx
 }
 
 void WebRtcTransportImp::onShutdown(const SockException &ex) {
-    WarnL << ex.what();
+    WarnL << ex;
     unrefSelf();
     for (auto &tuple : _ice_server->GetTuples()) {
         tuple->shutdown(ex);
@@ -1131,6 +1131,10 @@ void WebRtcPluginManager::registerPlugin(const string &type, Plugin cb) {
     _map_creator[type] = std::move(cb);
 }
 
+std::string exchangeSdp(const WebRtcInterface &exchanger, const std::string& offer) {
+    return const_cast<WebRtcInterface &>(exchanger).getAnswerSdp(offer);
+}
+
 void WebRtcPluginManager::getAnswerSdp(Session &sender, const string &type, const WebRtcArgs &args, const onCreateRtc &cb) {
     lock_guard<mutex> lck(_mtx_creator);
     auto it = _map_creator.find(type);

webrtc/WebRtcTransport.h

@@ -43,6 +43,8 @@ public:
     virtual const std::string &getIdentifier() const = 0;
 };
 
+std::string exchangeSdp(const WebRtcInterface &exchanger, const std::string& offer);
+
 class WebRtcException : public WebRtcInterface {
 public:
     WebRtcException(const SockException &ex) : _ex(ex) {};