Merge pull request #1 from ZLMediaKit/master

同步主线

AUTHORS

@@ -44,7 +44,6 @@ Xinghua Zhao <(holychaossword@hotmail.com>
 [Dw9](https://github.com/Dw9)
 明月惊鹊 <mingyuejingque@gmail.com>
 cgm <2958580318@qq.com>
-hejilin <1724010622@qq.com>
 alexliyu7352 <liyu7352@gmail.com>
 cgm <2958580318@qq.com>
 [haorui wang](https://github.com/HaoruiWang)
@@ -104,3 +103,8 @@ WuPeng <wp@zafu.edu.cn>
 [sandro-qiang](https://github.com/sandro-qiang)
 [Paul Philippov](https://github.com/themactep)
 [张传峰](https://github.com/zhang-chuanfeng)
+[lidaofu-hub](https://github.com/lidaofu-hub)
+[huangcaichun](https://github.com/huangcaichun)
+[jamesZHANG500](https://github.com/jamesZHANG500)
+[weidelong](https://github.com/wdl1697454803)
+[小强先生](https://github.com/linshangqiang)

CMakeLists.txt

@@ -141,8 +141,8 @@ if(GIT_FOUND)
 endif()
 
 configure_file(
-  ${CMAKE_CURRENT_SOURCE_DIR}/version.h.ini
+  ${CMAKE_CURRENT_SOURCE_DIR}/ZLMVersion.h.ini
-  ${CMAKE_CURRENT_BINARY_DIR}/version.h
+  ${CMAKE_CURRENT_BINARY_DIR}/ZLMVersion.h
   @ONLY)
 
 message(STATUS "Git version is ${BRANCH_NAME} ${COMMIT_HASH}/${COMMIT_TIME} ${BUILD_TIME}")

README.md

@@ -358,6 +358,11 @@ bash build_docker_images.sh
 [sandro-qiang](https://github.com/sandro-qiang)
 [Paul Philippov](https://github.com/themactep)
 [张传峰](https://github.com/zhang-chuanfeng)
+[lidaofu-hub](https://github.com/lidaofu-hub)
+[huangcaichun](https://github.com/huangcaichun)
+[jamesZHANG500](https://github.com/jamesZHANG500)
+[weidelong](https://github.com/wdl1697454803)
+[小强先生](https://github.com/linshangqiang)
 
 同时感谢JetBrains对开源项目的支持，本项目使用CLion开发与调试：
 

README_en.md

@@ -516,6 +516,11 @@ Thanks to all those who have supported this project in various ways, including b
 [sandro-qiang](https://github.com/sandro-qiang)
 [Paul Philippov](https://github.com/themactep)
 [张传峰](https://github.com/zhang-chuanfeng)
+[lidaofu-hub](https://github.com/lidaofu-hub)
+[huangcaichun](https://github.com/huangcaichun)
+[jamesZHANG500](https://github.com/jamesZHANG500)
+[weidelong](https://github.com/wdl1697454803)
+[小强先生](https://github.com/linshangqiang)
 
 Also thank to JetBrains for their support for open source project, we developed and debugged zlmediakit with CLion:
 

api/source/mk_common.cpp

@@ -304,10 +304,10 @@ API_EXPORT void API_CALL mk_webrtc_get_answer_sdp2(void *user_data, on_user_data
     std::string offer_str = offer;
     std::shared_ptr<void> ptr(user_data, user_data_free ? user_data_free : [](void *) {});
     auto args = std::make_shared<WebRtcArgsUrl>(url);
-    WebRtcPluginManager::Instance().getAnswerSdp(*session, type, *args,
+    WebRtcPluginManager::Instance().negotiateSdp(*session, type, *args, [offer_str, session, ptr, cb](const WebRtcInterface &exchanger) mutable {
-                                                 [offer_str, session, ptr, cb](const WebRtcInterface &exchanger) mutable {
+        auto &handler = const_cast<WebRtcInterface &>(exchanger);
         try {
-            auto sdp_answer = exchangeSdp(exchanger, offer_str);
+            auto sdp_answer = handler.getAnswerSdp(offer_str);
             cb(ptr.get(), sdp_answer.data(), nullptr);
         } catch (std::exception &ex) {
             cb(ptr.get(), nullptr, ex.what());

api/source/mk_events.cpp

@@ -14,7 +14,10 @@
 #include "Http/HttpSession.h"
 #include "Rtsp/RtspSession.h"
 #include "Record/MP4Recorder.h"
+
+#ifdef ENABLE_WEBRTC
 #include "webrtc/WebRtcTransport.h"
+#endif
 
 using namespace toolkit;
 using namespace mediakit;
@@ -168,7 +171,7 @@ API_EXPORT void API_CALL mk_events_listen(const mk_events *events){
                                                    sender.getMediaTuple().stream.c_str(), ssrc.c_str(), ex.getErrCode(), ex.what());
             }
         });
-
+#ifdef ENABLE_WEBRTC
         NoticeCenter::Instance().addListener(&s_tag, Broadcast::kBroadcastRtcSctpConnecting,[](BroadcastRtcSctpConnectArgs){
             if (s_events.on_mk_rtc_sctp_connecting) {
                 s_events.on_mk_rtc_sctp_connecting((mk_rtc_transport)&sender);
@@ -204,6 +207,7 @@ API_EXPORT void API_CALL mk_events_listen(const mk_events *events){
                 s_events.on_mk_rtc_sctp_received((mk_rtc_transport)&sender, streamId, ppid, msg, len);
             }
         });
+#endif
     });
 
 }

api/source/mk_events_objects.cpp

@@ -17,7 +17,10 @@
 
 #include "Http/HttpClient.h"
 #include "Rtsp/RtspSession.h"
+
+#ifdef ENABLE_WEBRTC
 #include "webrtc/WebRtcTransport.h"
+#endif
 
 using namespace toolkit;
 using namespace mediakit;

conf/config.ini

@@ -357,7 +357,7 @@ tcpPort = 8000
 rembBitRate=0
 #rtc支持的音频codec类型,在前面的优先级更高
 #以下范例为所有支持的音频codec
-preferredCodecA=PCMU,PCMA,opus,mpeg4-generic
+preferredCodecA=PCMA,PCMU,opus,mpeg4-generic
 #rtc支持的视频codec类型,在前面的优先级更高
 #以下范例为所有支持的视频codec
 preferredCodecV=H264,H265,AV1,VP9,VP8

ext-codec/AACRtmp.cpp

@@ -17,7 +17,7 @@ using namespace toolkit;
 namespace mediakit {
 
 void AACRtmpDecoder::inputRtmp(const RtmpPacket::Ptr &pkt) {
-    CHECK(pkt->size() > 2);
+    CHECK_RET(pkt->size() > 2);
     if (pkt->isConfigFrame()) {
         getTrack()->setExtraData((uint8_t *)pkt->data() + 2, pkt->size() - 2);
         return;

ext-codec/H264Rtmp.cpp

@@ -14,14 +14,6 @@
 using namespace std;
 using namespace toolkit;
 
-#define CHECK_RET(...)                                                         \
-    try {                                                                      \
-        CHECK(__VA_ARGS__);                                                    \
-    } catch (AssertFailedException & ex) {                                     \
-        WarnL << ex.what();                                                    \
-        return;                                                                \
-    }
-
 namespace mediakit {
 
 void H264RtmpDecoder::inputRtmp(const RtmpPacket::Ptr &pkt) {

ext-codec/H265Rtmp.cpp

@@ -18,14 +18,6 @@
 using namespace std;
 using namespace toolkit;
 
-#define CHECK_RET(...)                                                         \
-    try {                                                                      \
-        CHECK(__VA_ARGS__);                                                    \
-    } catch (AssertFailedException & ex) {                                     \
-        WarnL << ex.what();                                                    \
-        return;                                                                \
-    }
-
 namespace mediakit {
 
 void H265RtmpDecoder::inputRtmp(const RtmpPacket::Ptr &pkt) {

server/VideoStack.cpp

@@ -62,7 +62,8 @@ void Channel::addParam(const std::weak_ptr<Param>& p)
 void Channel::onFrame(const mediakit::FFmpegFrame::Ptr& frame)
 {
     std::weak_ptr<Channel> weakSelf = shared_from_this();
-    // toolkit::WorkThreadPool::Instance().getFirstPoller()->async([weakSelf, frame]() {
+    _poller = _poller ? _poller : toolkit::WorkThreadPool::Instance().getPoller();
+    _poller->async([weakSelf, frame]() {
         auto self = weakSelf.lock();
         if (!self) {
             return;
@@ -70,7 +71,7 @@ void Channel::onFrame(const mediakit::FFmpegFrame::Ptr& frame)
         self->_tmp = self->_sws->inputFrame(frame);
 
         self->forEachParam([self](const Param::Ptr& p) { self->fillBuffer(p); });
-    // });
+    });
 }
 
 void Channel::forEachParam(const std::function<void(const Param::Ptr&)>& func)
@@ -440,6 +441,7 @@ int VideoStackManager::stopVideoStack(const std::string& id)
     auto it = _stackMap.find(id);
     if (it != _stackMap.end()) {
         _stackMap.erase(it);
+        InfoL << "VideoStack stop: " << id;
         return 0;
     }
     return -1;

server/VideoStack.h

@@ -80,6 +80,7 @@ class Channel : public std::enable_shared_from_this<Channel> {
     std::vector<std::weak_ptr<Param>> _params;
 
     mediakit::FFmpegSws::Ptr _sws;
+    toolkit::EventPoller::Ptr _poller;
 };
 
 class StackPlayer : public std::enable_shared_from_this<StackPlayer> {

server/WebApi.cpp

@@ -59,7 +59,7 @@
 #endif
 
 #if defined(ENABLE_VERSION)
-#include "version.h"
+#include "ZLMVersion.h"
 #endif
 
 #if defined(ENABLE_X264) && defined (ENABLE_FFMPEG)
@@ -119,7 +119,7 @@ static HttpApi toApi(const function<void(API_ARGS_MAP_ASYNC)> &cb) {
 
         //参数解析成map
         auto args = getAllArgs(parser);
-        cb(sender, headerOut, HttpAllArgs<decltype(args)>(parser, args), val, invoker);
+        cb(sender, headerOut, ArgsMap(parser, args), val, invoker);
     };
 }
 
@@ -147,7 +147,7 @@ static HttpApi toApi(const function<void(API_ARGS_JSON_ASYNC)> &cb) {
         Json::Reader reader;
         reader.parse(parser.content(), args);
 
-        cb(sender, headerOut, HttpAllArgs<decltype(args)>(parser, args), val, invoker);
+        cb(sender, headerOut, ArgsJson(parser, args), val, invoker);
     };
 }
 
@@ -167,7 +167,7 @@ static HttpApi toApi(const function<void(API_ARGS_STRING_ASYNC)> &cb) {
         Json::Value val;
         val["code"] = API::Success;
 
-        cb(sender, headerOut, HttpAllArgs<string>(parser, (string &)parser.content()), val, invoker);
+        cb(sender, headerOut, ArgsString(parser, (string &)parser.content()), val, invoker);
     };
 }
 
@@ -584,8 +584,10 @@ void addStreamProxy(const string &vhost, const string &app, const string &stream
     //添加拉流代理
     auto player = s_player_proxy.make(key, vhost, app, stream, option, retry_count);
 
-    // 先透传参数
+    // 先透传拷贝参数
-    player->mINI::operator=(args);
+    for (auto &pr : args) {
+        (*player)[pr.first] = pr.second;
+    }
 
     //指定RTP over TCP(播放rtsp时有效)
     (*player)[Client::kRtpType] = rtp_type;
@@ -660,13 +662,6 @@ void addStreamPusherProxy(const string &schema,
     pusher->publish(url);
 }
 
-template <typename Type>
-static void getArgsValue(const HttpAllArgs<ApiArgsType> &allArgs, const string &key, Type &value) {
-    auto val = allArgs[key];
-    if (!val.empty()) {
-        value = (Type)val;
-    }
-}
 
 /**
  * 安装api接口
@@ -733,7 +728,7 @@ void installWebApi() {
         CHECK_SECRET();
         auto &ini = mINI::Instance();
         int changed = API::Success;
-        for (auto &pr : allArgs.getArgs()) {
+        for (auto &pr : allArgs.args) {
             if (ini.find(pr.first) == ini.end()) {
 #if 1
                 //没有这个key
@@ -1091,7 +1086,7 @@ void installWebApi() {
         CHECK_ARGS("vhost","app","stream","url");
 
         mINI args;
-        for (auto &pr : allArgs.getArgs()) {
+        for (auto &pr : allArgs.args) {
             args.emplace(pr.first, pr.second);
         }
 
@@ -1188,7 +1183,7 @@ void installWebApi() {
     //测试url http://127.0.0.1/index/api/downloadBin
     api_regist("/index/api/downloadBin",[](API_ARGS_MAP_ASYNC){
         CHECK_SECRET();
-        invoker.responseFile(allArgs.getParser().getHeader(),StrCaseMap(),exePath());
+        invoker.responseFile(allArgs.parser.getHeader(), StrCaseMap(), exePath());
     });
 
 #if defined(ENABLE_RTPPROXY)
@@ -1695,7 +1690,7 @@ void installWebApi() {
 
             //截图存在，且未过期，那么返回之
             res_old_snap = true;
-            responseSnap(path, allArgs.getParser().getHeader(), invoker);
+            responseSnap(path, allArgs.parser.getHeader(), invoker);
             //中断遍历
             return false;
         });
@@ -1726,7 +1721,7 @@ void installWebApi() {
                 File::delete_file(new_snap);
                 rename(new_snap_tmp.data(), new_snap.data());
             }
-            responseSnap(new_snap, allArgs.getParser().getHeader(), invoker, err_msg);
+            responseSnap(new_snap, allArgs.parser.getHeader(), invoker, err_msg);
         });
     });
 
@@ -1741,7 +1736,7 @@ void installWebApi() {
 #ifdef ENABLE_WEBRTC
     class WebRtcArgsImp : public WebRtcArgs {
     public:
-        WebRtcArgsImp(const HttpAllArgs<string> &args, std::string session_id)
+        WebRtcArgsImp(const ArgsString &args, std::string session_id)
             : _args(args)
             , _session_id(std::move(session_id)) {}
         ~WebRtcArgsImp() override = default;
@@ -1759,40 +1754,26 @@ void installWebApi() {
             CHECK_ARGS("app", "stream");
 
             return StrPrinter << "rtc://" << _args["Host"] << "/" << _args["app"] << "/"
-                              << _args["stream"] << "?" << _args.getParser().params() + "&session=" + _session_id;
+                              << _args["stream"] << "?" << _args.parser.params() + "&session=" + _session_id;
         }
 
     private:
-        HttpAllArgs<string> _args;
+        ArgsString _args;
         std::string _session_id;
     };
 
     api_regist("/index/api/webrtc",[](API_ARGS_STRING_ASYNC){
         CHECK_ARGS("type");
         auto type = allArgs["type"];
-        auto offer = allArgs.getArgs();
+        auto offer = allArgs.args;
         CHECK(!offer.empty(), "http body(webrtc offer sdp) is empty");
-        std::string host = allArgs.getParser()["Host"];
-        std::string localIp = host.substr(0, host.find(':'));
-
-        auto isVaildIP = [](std::string ip)-> bool {
-            int a,b,c,d;
-            return sscanf(ip.c_str(),"%d.%d.%d.%d", &a, &b, &c, &d) == 4;
-        };
-        if (!isVaildIP(localIp) || localIp=="127.0.0.1") {
-            localIp = "";
-        }
 
+        auto &session = static_cast<Session&>(sender);
         auto args = std::make_shared<WebRtcArgsImp>(allArgs, sender.getIdentifier());
-        WebRtcPluginManager::Instance().getAnswerSdp(static_cast<Session&>(sender), type, *args, [invoker, val, offer, headerOut, localIp](const WebRtcInterface &exchanger) mutable {
+        WebRtcPluginManager::Instance().negotiateSdp(session, type, *args, [invoker, val, offer, headerOut](const WebRtcInterface &exchanger) mutable {
-            //设置返回类型
+            auto &handler = const_cast<WebRtcInterface &>(exchanger);
-            headerOut["Content-Type"] = HttpFileManager::getContentType(".json");
-            //设置跨域
-            headerOut["Access-Control-Allow-Origin"] = "*";
-
             try {
-                setLocalIp(exchanger,localIp);
+                val["sdp"] = handler.getAnswerSdp(offer);
-                val["sdp"] = exchangeSdp(exchanger, offer);
                 val["id"] = exchanger.getIdentifier();
                 val["type"] = "answer";
                 invoker(200, headerOut, val.toStyledString());
@@ -1806,21 +1787,19 @@ void installWebApi() {
 
     static constexpr char delete_webrtc_url [] = "/index/api/delete_webrtc";
     static auto whip_whep_func = [](const char *type, API_ARGS_STRING_ASYNC) {
-        auto offer = allArgs.getArgs();
+        auto offer = allArgs.args;
         CHECK(!offer.empty(), "http body(webrtc offer sdp) is empty");
 
         auto &session = static_cast<Session&>(sender);
-        auto location = std::string("http") + (session.overSsl() ? "s" : "") + "://" + allArgs["host"] + delete_webrtc_url;
+        auto location = std::string(session.overSsl() ? "https://" : "http://") + allArgs["host"] + delete_webrtc_url;
         auto args = std::make_shared<WebRtcArgsImp>(allArgs, sender.getIdentifier());
-        WebRtcPluginManager::Instance().getAnswerSdp(session, type, *args,
+        WebRtcPluginManager::Instance().negotiateSdp(session, type, *args, [invoker, offer, headerOut, location](const WebRtcInterface &exchanger) mutable {
-                                                     [invoker, offer, headerOut, location](const WebRtcInterface &exchanger) mutable {
+            auto &handler = const_cast<WebRtcInterface &>(exchanger);
-                // 设置跨域
-                headerOut["Access-Control-Allow-Origin"] = "*";
             try {
                 // 设置返回类型
                 headerOut["Content-Type"] = "application/sdp";
                 headerOut["Location"] = location + "?id=" + exchanger.getIdentifier() + "&token=" + exchanger.deleteRandStr();
-                    invoker(201, headerOut, exchangeSdp(exchanger, offer));
+                invoker(201, headerOut, handler.getAnswerSdp(offer));
             } catch (std::exception &ex) {
                 headerOut["Content-Type"] = "text/plain";
                 invoker(406, headerOut, ex.what());
@@ -1833,7 +1812,7 @@ void installWebApi() {
 
     api_regist(delete_webrtc_url, [](API_ARGS_MAP_ASYNC) {
         CHECK_ARGS("id", "token");
-        CHECK(allArgs.getParser().method() == "DELETE", "http method is not DELETE: " + allArgs.getParser().method());
+        CHECK(allArgs.parser.method() == "DELETE", "http method is not DELETE: " + allArgs.parser.method());
         auto obj = WebRtcTransportManager::Instance().getItem(allArgs["id"]);
         if (!obj) {
             invoker(404, headerOut, "id not found");
@@ -1919,11 +1898,11 @@ void installWebApi() {
                 if (!save_name.empty()) {
                     res_header.emplace("Content-Disposition", "attachment;filename=\"" + save_name + "\"");
                 }
-                invoker.responseFile(allArgs.getParser().getHeader(), res_header, allArgs["file_path"]);
+                invoker.responseFile(allArgs.parser.getHeader(), res_header, allArgs["file_path"]);
             }
         };
 
-        bool flag = NOTICE_EMIT(BroadcastHttpAccessArgs, Broadcast::kBroadcastHttpAccess, allArgs.getParser(), file_path, false, file_invoker, sender);
+        bool flag = NOTICE_EMIT(BroadcastHttpAccessArgs, Broadcast::kBroadcastHttpAccess, allArgs.parser, file_path, false, file_invoker, sender);
         if (!flag) {
             // 文件下载鉴权事件无人监听，不允许下载
             invoker(401, StrCaseMap {}, "None http access event listener");
@@ -1935,28 +1914,23 @@ void installWebApi() {
     NoticeCenter::Instance().addListener(nullptr, Broadcast::kBroadcastStreamNoneReader, [](BroadcastStreamNoneReaderArgs) {
         auto id = sender.getMediaTuple().stream;
         VideoStackManager::Instance().stopVideoStack(id);
-        InfoL << "VideoStack: " << id <<" stop";
     });
 
     api_regist("/index/api/stack/start", [](API_ARGS_JSON_ASYNC) {
         CHECK_SECRET();
         auto ret = VideoStackManager::Instance().startVideoStack(allArgs.getArgs());
-        if (!ret) {
+        val["code"] = ret;
-            invoker(200, headerOut, "success");
+        val["msg"] = ret ? "failed" : "success";
-        } else {
+        invoker(200, headerOut, val.toStyledString());
-            invoker(200, headerOut, "failed");
-        }
     });
 
     api_regist("/index/api/stack/stop", [](API_ARGS_MAP_ASYNC) {
         CHECK_SECRET();
         CHECK_ARGS("id");
         auto ret = VideoStackManager::Instance().stopVideoStack(allArgs["id"]);
-        if (!ret) {
+        val["code"] = ret;
-            invoker(200, headerOut, "success");
+        val["msg"] = ret ? "failed" : "success";
-        } else {
+        invoker(200, headerOut, val.toStyledString());
-            invoker(200, headerOut, "failed");
-        }
     });
 #endif
 }

server/WebApi.h

@@ -115,72 +115,41 @@ std::string getValue(const mediakit::Parser &parser, Args &args, const First &fi
 
 template<typename Args>
 class HttpAllArgs {
+    mediakit::Parser* _parser = nullptr;
+    Args* _args = nullptr;
 public:
-    HttpAllArgs(const mediakit::Parser &parser, Args &args) {
+    const mediakit::Parser& parser;
-        _get_args = [&args]() {
+    Args& args;
-            return (void *) &args;
-        };
-        _get_parser = [&parser]() -> const mediakit::Parser & {
-            return parser;
-        };
-        _get_value = [](HttpAllArgs &that, const std::string &key) {
-            return getValue(that.getParser(), that.getArgs(), key);
-        };
-        _clone = [&](HttpAllArgs &that) {
-            that._get_args = [args]() {
-                return (void *) &args;
-            };
-            that._get_parser = [parser]() -> const mediakit::Parser & {
-                return parser;
-            };
-            that._get_value = [](HttpAllArgs &that, const std::string &key) {
-                return getValue(that.getParser(), that.getArgs(), key);
-            };
-            that._cache_able = true;
-        };
-    }
 
-    HttpAllArgs(const HttpAllArgs &that) {
+    HttpAllArgs(const mediakit::Parser &p, Args &a): parser(p), args(a) {}
-        if (that._cache_able) {
+
-            _get_args = that._get_args;
+    HttpAllArgs(const HttpAllArgs &that): _parser(new mediakit::Parser(that.parser)),
-            _get_parser = that._get_parser;
+                                          _args(new Args(that.args)),
-            _get_value = that._get_value;
+                                          parser(*_parser), args(*_args) {}
-            _cache_able = true;
+    ~HttpAllArgs() {
-        } else {
+        if (_parser) {
-            that._clone(*this);
+            delete _parser;
+        }
+        if (_args) {
+            delete _args;
         }
     }
 
     template<typename Key>
     toolkit::variant operator[](const Key &key) const {
-        return (toolkit::variant)_get_value(*(HttpAllArgs*)this, key);
+        return (toolkit::variant)getValue(parser, args, key);
     }
-
-    const mediakit::Parser &getParser() const {
-        return _get_parser();
-    }
-
-    Args &getArgs() {
-        return *((Args *) _get_args());
-    }
-
-    const Args &getArgs() const {
-        return *((Args *) _get_args());
-    }
-
-private:
-    bool _cache_able = false;
-    std::function<void *() > _get_args;
-    std::function<const mediakit::Parser &() > _get_parser;
-    std::function<std::string(HttpAllArgs &that, const std::string &key)> _get_value;
-    std::function<void(HttpAllArgs &that) > _clone;
 };
 
-#define API_ARGS_MAP toolkit::SockInfo &sender, mediakit::HttpSession::KeyValue &headerOut, const HttpAllArgs<ApiArgsType> &allArgs, Json::Value &val
+using ArgsMap = HttpAllArgs<ApiArgsType>;
+using ArgsJson = HttpAllArgs<Json::Value>;
+using ArgsString = HttpAllArgs<std::string>;
+
+#define API_ARGS_MAP toolkit::SockInfo &sender, mediakit::HttpSession::KeyValue &headerOut, const ArgsMap &allArgs, Json::Value &val
 #define API_ARGS_MAP_ASYNC API_ARGS_MAP, const mediakit::HttpSession::HttpResponseInvoker &invoker
-#define API_ARGS_JSON toolkit::SockInfo &sender, mediakit::HttpSession::KeyValue &headerOut, const HttpAllArgs<Json::Value> &allArgs, Json::Value &val
+#define API_ARGS_JSON toolkit::SockInfo &sender, mediakit::HttpSession::KeyValue &headerOut, const ArgsJson &allArgs, Json::Value &val
 #define API_ARGS_JSON_ASYNC API_ARGS_JSON, const mediakit::HttpSession::HttpResponseInvoker &invoker
-#define API_ARGS_STRING toolkit::SockInfo &sender, mediakit::HttpSession::KeyValue &headerOut, const HttpAllArgs<std::string> &allArgs, Json::Value &val
+#define API_ARGS_STRING toolkit::SockInfo &sender, mediakit::HttpSession::KeyValue &headerOut, const ArgsString &allArgs, Json::Value &val
 #define API_ARGS_STRING_ASYNC API_ARGS_STRING, const mediakit::HttpSession::HttpResponseInvoker &invoker
 #define API_ARGS_VALUE sender, headerOut, allArgs, val
 

server/main.cpp

@@ -38,7 +38,7 @@
 #endif
 
 #if defined(ENABLE_VERSION)
-#include "version.h"
+#include "ZLMVersion.h"
 #endif
 
 #if !defined(_WIN32)

src/Common/MediaSource.h

@@ -136,6 +136,15 @@ private:
     toolkit::Timer::Ptr _async_close_timer;
 };
 
+
+template <typename MAP, typename KEY, typename TYPE>
+static void getArgsValue(const MAP &allArgs, const KEY &key, TYPE &value) {
+    auto val = ((MAP &)allArgs)[key];
+    if (!val.empty()) {
+        value = (TYPE)val;
+    }
+}
+
 class ProtocolOption {
 public:
     ProtocolOption();
@@ -243,15 +252,6 @@ public:
         GET_OPT_VALUE(stream_replace);
         GET_OPT_VALUE(max_track);
     }
-
-private:
-    template <typename MAP, typename KEY, typename TYPE>
-    static void getArgsValue(const MAP &allArgs, const KEY &key, TYPE &value) {
-        auto val = ((MAP &)allArgs)[key];
-        if (!val.empty()) {
-            value = (TYPE)val;
-        }
-    }
 };
 
 //该对象用于拦截感兴趣的MediaSourceEvent事件

src/Common/Parser.cpp

@@ -294,8 +294,8 @@ void RtspUrl::setup(bool is_ssl, const string &url, const string &user, const st
     splitUrl(ip, ip, port);
 
     _url = std::move(url);
-    _user = strCoding::UrlDecode(std::move(user));
+    _user = strCoding::UrlDecodeComponent(user);
-    _passwd = strCoding::UrlDecode(std::move(passwd));
+    _passwd = strCoding::UrlDecodeComponent(passwd);
     _host = std::move(ip);
     _port = port;
     _is_ssl = is_ssl;

src/Common/Parser.h

@@ -30,7 +30,7 @@ struct StrCaseCompare {
 
 class StrCaseMap : public std::multimap<std::string, std::string, StrCaseCompare> {
 public:
-    using Super = multimap<std::string, std::string, StrCaseCompare>;
+    using Super = std::multimap<std::string, std::string, StrCaseCompare>;
 
     std::string &operator[](const std::string &k) {
         auto it = find(k);

src/Common/macros.cpp

@@ -14,7 +14,7 @@
 using namespace toolkit;
 
 #if defined(ENABLE_VERSION)
-#include "version.h"
+#include "ZLMVersion.h"
 #endif
 
 extern "C" {

src/Common/macros.h

@@ -36,6 +36,16 @@
 #define CHECK(exp, ...) ::mediakit::Assert_ThrowCpp(!(exp), #exp, __FUNCTION__, __FILE__, __LINE__, ##__VA_ARGS__)
 #endif // CHECK
 
+#ifndef CHECK_RET
+#define CHECK_RET(...)                                                         \
+    try {                                                                      \
+        CHECK(__VA_ARGS__);                                                    \
+    } catch (AssertFailedException & ex) {                                     \
+        WarnL << ex.what();                                                    \
+        return;                                                                \
+    }
+#endif
+
 #ifndef MAX
 #define MAX(a, b) ((a) > (b) ? (a) : (b))
 #endif // MAX

src/Common/strCoding.cpp

@@ -53,22 +53,6 @@ char HexStrToBin(const char *str) {
     return (high << 4) | low;
 }
 
-string strCoding::UrlEncode(const string &str) {
-    string out;
-    size_t len = str.size();
-    for (size_t i = 0; i < len; ++i) {
-        char ch = str[i];
-        if (isalnum((uint8_t) ch)) {
-            out.push_back(ch);
-        } else {
-            char buf[4];
-            sprintf(buf, "%%%X%X", (uint8_t) ch >> 4, (uint8_t) ch & 0x0F);
-            out.append(buf);
-        }
-    }
-    return out;
-}
-
 string strCoding::UrlEncodePath(const string &str) {
     const char *dont_escape = "!#&'*+:=?@/._-$,;~()";
     string out;
@@ -103,32 +87,6 @@ string strCoding::UrlEncodeComponent(const string &str) {
     return out;
 }
 
-string strCoding::UrlDecode(const string &str) {
-    string output;
-    size_t i = 0, len = str.length();
-    while (i < len) {
-        if (str[i] == '%') {
-            if (i + 3 > len) {
-                // %后面必须还有两个字节才会反转义
-                output.append(str, i, len - i);
-                break;
-            }
-            char ch = HexStrToBin(&(str[i + 1]));
-            if (ch == -1) {
-                // %后面两个字节不是16进制字符串，转义失败；直接拼接3个原始字符
-                output.append(str, i, 3);
-            } else {
-                output += ch;
-            }
-            i += 3;
-        } else {
-            output += str[i];
-            ++i;
-        }
-    }
-    return output;
-}
-
 string strCoding::UrlDecodePath(const string &str) {
     const char *dont_unescape = "#$&+,/:;=?@";
     string output;
@@ -185,27 +143,6 @@ std::string strCoding::UrlDecodeComponent(const std::string &str) {
     return output;
 }
 
-#if 0
-#include "Util/onceToken.h"
-static toolkit::onceToken token([]() {
-    auto str0 = strCoding::UrlDecode(
-        "rtsp%3A%2F%2Fadmin%3AJm13317934%25jm%40111.47.84.69%3A554%2FStreaming%2FChannels%2F101%3Ftransportmode%3Dunicast%26amp%3Bprofile%3DProfile_1");
-    auto str1 = strCoding::UrlDecode("%j1"); // 测试%后面两个字节不是16进制字符串
-    auto str2 = strCoding::UrlDecode("%a"); // 测试%后面字节数不够
-    auto str3 = strCoding::UrlDecode("%"); // 测试只有%
-    auto str4 = strCoding::UrlDecode("%%%"); // 测试多个%
-    auto str5 = strCoding::UrlDecode("%%%%40"); // 测试多个非法%后恢复正常解析
-    auto str6 = strCoding::UrlDecode("Jm13317934%jm"); // 测试多个非法%后恢复正常解析
-    cout << str0 << endl;
-    cout << str1 << endl;
-    cout << str2 << endl;
-    cout << str3 << endl;
-    cout << str4 << endl;
-    cout << str5 << endl;
-    cout << str6 << endl;
-});
-#endif
-
 ///////////////////////////////windows专用///////////////////////////////////
 #if defined(_WIN32)
 void UnicodeToGB2312(char* pOut, wchar_t uData)

src/Common/strCoding.h

@@ -18,10 +18,8 @@ namespace mediakit {
 
 class strCoding {
 public:
-    [[deprecated]] static std::string UrlEncode(const std::string &str); //url utf8编码, deprecated
     static std::string UrlEncodePath(const std::string &str); //url路径 utf8编码
     static std::string UrlEncodeComponent(const std::string &str); // url参数 utf8编码
-    [[deprecated]] static std::string UrlDecode(const std::string &str); //url utf8解码, deprecated
     static std::string UrlDecodePath(const std::string &str); //url路径 utf8解码
     static std::string UrlDecodeComponent(const std::string &str); // url参数 utf8解码
 #if defined(_WIN32)

src/Http/HttpRequestSplitter.cpp

@@ -65,18 +65,18 @@ void HttpRequestSplitter::input(const char *data,size_t len) {
         _content_len = onRecvHeader(header_ptr, header_size);
     }
 
-    if(_remain_data_size <= 0){
-        //没有剩余数据，清空缓存
-        _remain_data.clear();
-        return;
-    }
-
     /*
      * 恢复末尾字节
      * 移动到这来，目的是防止HttpRequestSplitter::reset()导致内存失效
      */
     tail_ref = tail_tmp;
 
+    if(_remain_data_size <= 0){
+        //没有剩余数据，清空缓存
+        _remain_data.clear();
+        return;
+    }
+
     if(_content_len == 0){
         //尚未找到http头，缓存定位到剩余数据部分
         _remain_data.assign(ptr,_remain_data_size);

src/Http/HttpSession.cpp

@@ -683,18 +683,6 @@ void HttpSession::sendResponse(int code,
     AsyncSender::onSocketFlushed(data);
 }
 
-string HttpSession::urlDecode(const string &str) {
-    auto ret = strCoding::UrlDecode(str);
-#ifdef _WIN32
-    GET_CONFIG(string, charSet, Http::kCharSet);
-    bool isGb2312 = !strcasecmp(charSet.data(), "gb2312");
-    if (isGb2312) {
-        ret = strCoding::UTF8ToGB2312(ret);
-    }
-#endif // _WIN32
-    return ret;
-}
-
 string HttpSession::urlDecodePath(const string &str) {
     auto ret = strCoding::UrlDecodePath(str);
 #ifdef _WIN32

src/Http/HttpSession.h

@@ -44,7 +44,6 @@ public:
     void onRecv(const toolkit::Buffer::Ptr &) override;
     void onError(const toolkit::SockException &err) override;
     void onManager() override;
-    [[deprecated]] static std::string urlDecode(const std::string &str);
     static std::string urlDecodePath(const std::string &str);
     static std::string urlDecodeComponent(const std::string &str);
     void setTimeoutSec(size_t second);

src/Rtmp/RtmpMediaSourceMuxer.h

@@ -29,7 +29,13 @@ public:
         getRtmpRing()->setDelegate(_media_src);
     }
 
-    ~RtmpMediaSourceMuxer() override { RtmpMuxer::flush(); }
+    ~RtmpMediaSourceMuxer() override {
+        try {
+            RtmpMuxer::flush();
+        } catch (std::exception &ex) {
+            WarnL << ex.what();
+        }
+    }
 
     void setListener(const std::weak_ptr<MediaSourceEvent> &listener){
         setDelegate(listener);

src/Rtmp/RtmpProtocol.cpp

@@ -165,14 +165,7 @@ void RtmpProtocol::sendResponse(int type, const string &str) {
 
 void RtmpProtocol::sendInvoke(const string &cmd, const AMFValue &val) {
     AMFEncoder enc;
-    if (val.type() == AMFType::AMF_OBJECT || val.type() == AMFType::AMF_NULL)
-    {
     enc << cmd << ++_send_req_id << val;
-    }
-    else
-    {
-        enc << cmd << ++_send_req_id << AMFValue() << val;
-    }
     sendRequest(MSG_CMD, enc.data());
 }
 
@@ -626,22 +619,12 @@ const char* RtmpProtocol::handle_rtmp(const char *data, size_t len) {
             case 12:
                 chunk_data.is_abs_stamp = true;
                 chunk_data.stream_index = load_le32(header->stream_index);
-                _last_stream_index = chunk_data.stream_index;
             case 8:
                 chunk_data.body_size = load_be24(header->body_size);
                 chunk_data.type_id = header->type_id;
-                _last_body_size = chunk_data.body_size;
-                _last_type_id = chunk_data.type_id;
             case 4:
                 chunk_data.ts_field = load_be24(header->time_stamp);
         }
-        switch (header->fmt) {
-            case 2:
-                chunk_data.type_id = _last_type_id;
-                chunk_data.body_size = _last_body_size;
-            case 1:
-                chunk_data.stream_index = _last_stream_index;
-        }
 
         auto time_stamp = chunk_data.ts_field;
         if (chunk_data.ts_field == 0xFFFFFF) {

src/Rtmp/RtmpProtocol.h

@@ -11,7 +11,6 @@
 #ifndef SRC_RTMP_RTMPPROTOCOL_H_
 #define SRC_RTMP_RTMPPROTOCOL_H_
 
-#include <cstdint>
 #include <memory>
 #include <string>
 #include <functional>
@@ -88,9 +87,6 @@ protected:
 private:
     bool _data_started = false;
     int _now_chunk_id = 0;
-    uint32_t _last_stream_index = 0;
-    size_t _last_body_size = 0;
-    uint8_t _last_type_id = 0;
     ////////////ChunkSize////////////
     size_t _chunk_size_in = DEFAULT_CHUNK_LEN;
     size_t _chunk_size_out = DEFAULT_CHUNK_LEN;

src/Rtmp/RtmpPusher.cpp

@@ -163,18 +163,6 @@ void RtmpPusher::send_connect() {
 }
 
 void RtmpPusher::send_createStream() {
-    // Workaround : 兼容较旧的 FMS3.0
-    {
-        {
-            AMFValue obj(_stream_id);
-            sendInvoke("releaseStream", obj);
-        }
-        {
-            AMFValue obj(_stream_id);
-            sendInvoke("FCPublish", obj);
-        }
-    }
-    {
     AMFValue obj(AMF_NULL);
     sendInvoke("createStream", obj);
     addOnResultCB([this](AMFDecoder &dec) {
@@ -185,8 +173,6 @@ void RtmpPusher::send_createStream() {
     });
 }
 
-}
-
 #define RTMP_STREAM_LIVE    "live"
 void RtmpPusher::send_publish() {
     AMFEncoder enc;

webrtc/Sdp.cpp

@@ -23,10 +23,10 @@ namespace Rtc {
 const string kPreferredCodecA = RTC_FIELD "preferredCodecA";
 const string kPreferredCodecV = RTC_FIELD "preferredCodecV";
 static onceToken token([]() {
-    mINI::Instance()[kPreferredCodecA] = "PCMU,PCMA,opus,mpeg4-generic";
+    mINI::Instance()[kPreferredCodecA] = "PCMA,PCMU,opus,mpeg4-generic";
     mINI::Instance()[kPreferredCodecV] = "H264,H265,AV1,VP9,VP8";
 });
-}
+} // namespace Rtc
 
 using onCreateSdpItem = function<SdpItem::Ptr(const string &key, const string &value)>;
 static map<string, onCreateSdpItem, StrCaseCompare> sdpItemCreator;
@@ -73,9 +73,7 @@ public:
 
 class DirectionInterfaceImp : public SdpItem, public DirectionInterface {
 public:
-    DirectionInterfaceImp(RtpDirection direct){
+    DirectionInterfaceImp(RtpDirection direct) { direction = direct; }
-        direction = direct;
-    }
     const char *getKey() const override { return getRtpDirectionString(getDirection()); }
     RtpDirection getDirection() const override { return direction; }
 
@@ -717,8 +715,7 @@ void SdpAttrCandidate::parse(const string &str)  {
 
 string SdpAttrCandidate::toString() const {
     if (value.empty()) {
-        value = foundation + " " + to_string(component) + " " + transport + " " + to_string(priority) +
+        value = foundation + " " + to_string(component) + " " + transport + " " + to_string(priority) + " " + address + " " + to_string(port) + " typ " + type;
-                " " + address + " " + to_string(port) + " typ " + type;
         for (auto &pr : arr) {
             value += ' ';
             value += pr.first;
@@ -952,8 +949,7 @@ void RtcSession::loadFrom(const string &str) {
             if (fmtp_it != fmtp_map.end()) {
                 plan.fmtp = fmtp_it->second.fmtp;
             }
-            for (auto rtpfb_it = rtcpfb_map.find(pt);
+            for (auto rtpfb_it = rtcpfb_map.find(pt); rtpfb_it != rtcpfb_map.end() && rtpfb_it->second.pt == pt; ++rtpfb_it) {
-                 rtpfb_it != rtcpfb_map.end() && rtpfb_it->second.pt == pt; ++rtpfb_it) {
                 plan.rtcp_fb.emplace(rtpfb_it->second.rtcp_type);
             }
         }
@@ -986,8 +982,7 @@ void RtcSdpBase::toRtsp() {
             case 'a': {
                 auto attr = dynamic_pointer_cast<SdpAttr>(*it);
                 CHECK(attr);
-                if (!strcasecmp(attr->detail->getKey(), "rtpmap")
+                if (!strcasecmp(attr->detail->getKey(), "rtpmap") || !strcasecmp(attr->detail->getKey(), "fmtp")) {
-                    || !strcasecmp(attr->detail->getKey(), "fmtp")) {
                     ++it;
                     break;
                 }
@@ -1221,9 +1216,9 @@ RtcSessionSdp::Ptr RtcSession::toRtcSessionSdp() const{
             }
         }
     }
-    if(ice_lite)
+    if (ice_lite) {
         sdp.addAttr(std::make_shared<SdpCommon>("ice-lite"));
-    
+    }
     return ret;
 }
 
@@ -1427,16 +1422,14 @@ void RtcConfigure::RtcTrackConfigure::setDefaultSetting(TrackType type){
             preferred_codec = s_preferred_codec;
 
             rtcp_fb = { SdpConst::kTWCCRtcpFb, SdpConst::kRembRtcpFb };
-            extmap = {
+            extmap = { { RtpExtType::ssrc_audio_level, RtpDirection::sendrecv },
-                    {RtpExtType::ssrc_audio_level,            RtpDirection::sendrecv},
                        { RtpExtType::csrc_audio_level, RtpDirection::sendrecv },
                        { RtpExtType::abs_send_time, RtpDirection::sendrecv },
                        { RtpExtType::transport_cc, RtpDirection::sendrecv },
                        // rtx重传rtp时，忽略sdes_mid类型的rtp ext,实测发现Firefox在接收rtx时，如果存在sdes_mid的ext,将导致无法播放
                        //{RtpExtType::sdes_mid,RtpDirection::sendrecv},
                        { RtpExtType::sdes_rtp_stream_id, RtpDirection::sendrecv },
-                    {RtpExtType::sdes_repaired_rtp_stream_id, RtpDirection::sendrecv}
+                       { RtpExtType::sdes_repaired_rtp_stream_id, RtpDirection::sendrecv } };
-            };
             break;
         }
         case TrackVideo: {
@@ -1446,8 +1439,7 @@ void RtcConfigure::RtcTrackConfigure::setDefaultSetting(TrackType type){
             preferred_codec = s_preferred_codec;
 
             rtcp_fb = { SdpConst::kTWCCRtcpFb, SdpConst::kRembRtcpFb, "nack", "ccm fir", "nack pli" };
-            extmap = {
+            extmap = { { RtpExtType::abs_send_time, RtpDirection::sendrecv },
-                    {RtpExtType::abs_send_time,               RtpDirection::sendrecv},
                        { RtpExtType::transport_cc, RtpDirection::sendrecv },
                        // rtx重传rtp时，忽略sdes_mid类型的rtp ext,实测发现Firefox在接收rtx时，如果存在sdes_mid的ext,将导致无法播放
                        //{RtpExtType::sdes_mid,RtpDirection::sendrecv},
@@ -1460,8 +1452,7 @@ void RtcConfigure::RtcTrackConfigure::setDefaultSetting(TrackType type){
                        // 手机端推webrtc 会带有旋转角度，rtc协议能正常播放 其他协议拉流画面旋转
                        //{RtpExtType::video_orientation,           RtpDirection::sendrecv},
                        { RtpExtType::toffset, RtpDirection::sendrecv },
-                    {RtpExtType::framemarking,                RtpDirection::sendrecv}
+                       { RtpExtType::framemarking, RtpDirection::sendrecv } };
-            };
             break;
         }
         case TrackApplication: {
@@ -1471,8 +1462,7 @@ void RtcConfigure::RtcTrackConfigure::setDefaultSetting(TrackType type){
     }
 }
 
-void RtcConfigure::setDefaultSetting(string ice_ufrag, string ice_pwd, RtpDirection direction,
+void RtcConfigure::setDefaultSetting(string ice_ufrag, string ice_pwd, RtpDirection direction, const SdpAttrFingerprint &fingerprint) {
-                                     const SdpAttrFingerprint &fingerprint) {
     video.setDefaultSetting(TrackVideo);
     audio.setDefaultSetting(TrackAudio);
     application.setDefaultSetting(TrackApplication);
@@ -1683,8 +1673,7 @@ RETRY:
         answer_media.role = mathDtlsRole(offer_media.role);
 
         // 如果codec匹配失败，那么禁用该track
-        answer_media.direction = check_codec ? matchDirection(offer_media.direction, configure.direction)
+        answer_media.direction = check_codec ? matchDirection(offer_media.direction, configure.direction) : RtpDirection::inactive;
-                                             : RtpDirection::inactive;
         if (answer_media.direction == RtpDirection::invalid) {
             continue;
         }

webrtc/Sdp.h

@@ -77,11 +77,7 @@ enum class DtlsRole {
     actpass,
 };
 
-enum class SdpType {
+enum class SdpType { invalid = -1, offer, answer };
-    invalid = -1,
-    offer,
-    answer
-};
 
 DtlsRole getDtlsRole(const std::string &str);
 const char *getDtlsRoleString(DtlsRole role);
@@ -92,17 +88,11 @@ class SdpItem {
 public:
     using Ptr = std::shared_ptr<SdpItem>;
     virtual ~SdpItem() = default;
-    virtual void parse(const std::string &str) {
+    virtual void parse(const std::string &str) { value = str; }
-        value  = str;
+    virtual std::string toString() const { return value; }
-    }
-    virtual std::string toString() const {
-        return value;
-    }
     virtual const char *getKey() const = 0;
 
-    void reset() {
+    void reset() { value.clear(); }
-        value.clear();
-    }
 
 protected:
     mutable std::string value;
@@ -259,9 +249,7 @@ public:
     void parse(const std::string &str) override;
     std::string toString() const override;
     const char *getKey() const override { return "msid-semantic"; }
-    bool empty() const {
+    bool empty() const { return msid.empty(); }
-        return msid.empty();
-    }
 };
 
 class SdpAttrRtcp : public SdpItem {
@@ -271,12 +259,11 @@ public:
     std::string nettype { "IN" };
     std::string addrtype { "IP4" };
     std::string address { "0.0.0.0" };
-    void parse(const std::string &str) override;;
+    void parse(const std::string &str) override;
+    ;
     std::string toString() const override;
     const char *getKey() const override { return "rtcp"; }
-    bool empty() const {
+    bool empty() const { return address.empty() || !port; }
-        return address.empty() || !port;
-    }
 };
 
 class SdpAttrIceUfrag : public SdpItem {
@@ -364,9 +351,9 @@ public:
     // a=rtcp-fb:98 nack pli
     // a=rtcp-fb:120 nack 支持 nack 重传，nack (Negative-Acknowledgment) 。
     // a=rtcp-fb:120 nack pli 支持 nack 关键帧重传，PLI (Picture Loss Indication) 。
-    //a=rtcp-fb:120 ccm fir 支持编码层关键帧请求，CCM (Codec Control Message)，FIR (Full Intra Request )，通常与 nack pli 有同样的效果，但是 nack pli 是用于重传时的关键帧请求。
+    // a=rtcp-fb:120 ccm fir 支持编码层关键帧请求，CCM (Codec Control Message)，FIR (Full Intra Request )，通常与 nack pli 有同样的效果，但是 nack pli
-    //a=rtcp-fb:120 goog-remb 支持 REMB (Receiver Estimated Maximum Bitrate) 。
+    // 是用于重传时的关键帧请求。 a=rtcp-fb:120 goog-remb 支持 REMB (Receiver Estimated Maximum Bitrate) 。 a=rtcp-fb:120 transport-cc 支持 TCC (Transport
-    //a=rtcp-fb:120 transport-cc 支持 TCC (Transport Congest Control) 。
+    // Congest Control) 。
     uint8_t pt;
     std::string rtcp_type;
     void parse(const std::string &str) override;

webrtc/WebRtcEchoTest.h

@@ -27,7 +27,6 @@ protected:
     void onRtp(const char *buf, size_t len, uint64_t stamp_ms) override;
     void onRtcp(const char *buf, size_t len) override;
 
-    void onRecvRtp(MediaTrack &track, const std::string &rid, RtpPacket::Ptr rtp) override {};
     void onBeforeEncryptRtp(const char *buf, int &len, void *ctx) override {};
     void onBeforeEncryptRtcp(const char *buf, int &len, void *ctx) override {};
 

webrtc/WebRtcPlayer.cpp

@@ -17,9 +17,8 @@ namespace mediakit {
 
 WebRtcPlayer::Ptr WebRtcPlayer::create(const EventPoller::Ptr &poller,
                                        const RtspMediaSource::Ptr &src,
-                                       const MediaInfo &info,
+                                       const MediaInfo &info) {
-                                       bool preferred_tcp) {
+    WebRtcPlayer::Ptr ret(new WebRtcPlayer(poller, src, info), [](WebRtcPlayer *ptr) {
-    WebRtcPlayer::Ptr ret(new WebRtcPlayer(poller, src, info, preferred_tcp), [](WebRtcPlayer *ptr) {
         ptr->onDestory();
         delete ptr;
     });
@@ -29,8 +28,7 @@ WebRtcPlayer::Ptr WebRtcPlayer::create(const EventPoller::Ptr &poller,
 
 WebRtcPlayer::WebRtcPlayer(const EventPoller::Ptr &poller,
                            const RtspMediaSource::Ptr &src,
-                           const MediaInfo &info,
+                           const MediaInfo &info) : WebRtcTransportImp(poller) {
-                           bool preferred_tcp) : WebRtcTransportImp(poller,preferred_tcp) {
     _media_info = info;
     _play_src = src;
     CHECK(src);

webrtc/WebRtcPlayer.h

@@ -19,7 +19,7 @@ namespace mediakit {
 class WebRtcPlayer : public WebRtcTransportImp {
 public:
     using Ptr = std::shared_ptr<WebRtcPlayer>;
-    static Ptr create(const EventPoller::Ptr &poller, const RtspMediaSource::Ptr &src, const MediaInfo &info, bool preferred_tcp = false);
+    static Ptr create(const EventPoller::Ptr &poller, const RtspMediaSource::Ptr &src, const MediaInfo &info);
     MediaInfo getMediaInfo() { return _media_info; }
 
 protected:
@@ -27,10 +27,9 @@ protected:
     void onStartWebRTC() override;
     void onDestory() override;
     void onRtcConfigure(RtcConfigure &configure) const override;
-    void onRecvRtp(MediaTrack &track, const std::string &rid, RtpPacket::Ptr rtp) override {};
 
 private:
-    WebRtcPlayer(const EventPoller::Ptr &poller, const RtspMediaSource::Ptr &src, const MediaInfo &info, bool preferred_tcp);
+    WebRtcPlayer(const EventPoller::Ptr &poller, const RtspMediaSource::Ptr &src, const MediaInfo &info);
 
 private:
     //媒体相关元数据

webrtc/WebRtcPusher.cpp

@@ -20,9 +20,8 @@ WebRtcPusher::Ptr WebRtcPusher::create(const EventPoller::Ptr &poller,
                                        const RtspMediaSource::Ptr &src,
                                        const std::shared_ptr<void> &ownership,
                                        const MediaInfo &info,
-                                       const ProtocolOption &option,
+                                       const ProtocolOption &option) {
-                                       bool preferred_tcp) {
+    WebRtcPusher::Ptr ret(new WebRtcPusher(poller, src, ownership, info, option), [](WebRtcPusher *ptr) {
-    WebRtcPusher::Ptr ret(new WebRtcPusher(poller, src, ownership, info, option,preferred_tcp), [](WebRtcPusher *ptr) {
         ptr->onDestory();
         delete ptr;
     });
@@ -34,8 +33,7 @@ WebRtcPusher::WebRtcPusher(const EventPoller::Ptr &poller,
                            const RtspMediaSource::Ptr &src,
                            const std::shared_ptr<void> &ownership,
                            const MediaInfo &info,
-                           const ProtocolOption &option,
+                           const ProtocolOption &option) : WebRtcTransportImp(poller) {
-                           bool preferred_tcp) : WebRtcTransportImp(poller,preferred_tcp) {
     _media_info = info;
     _push_src = src;
     _push_src_ownership = ownership;

webrtc/WebRtcPusher.h

@@ -20,8 +20,7 @@ class WebRtcPusher : public WebRtcTransportImp, public MediaSourceEvent {
 public:
     using Ptr = std::shared_ptr<WebRtcPusher>;
     static Ptr create(const EventPoller::Ptr &poller, const RtspMediaSource::Ptr &src,
-                      const std::shared_ptr<void> &ownership, const MediaInfo &info, const ProtocolOption &option, bool preferred_tcp = false);
+                      const std::shared_ptr<void> &ownership, const MediaInfo &info, const ProtocolOption &option);
-
 
 protected:
     ///////WebRtcTransportImp override///////
@@ -53,7 +52,7 @@ protected:
 
 private:
     WebRtcPusher(const EventPoller::Ptr &poller, const RtspMediaSource::Ptr &src,
-                 const std::shared_ptr<void> &ownership, const MediaInfo &info, const ProtocolOption &option, bool preferred_tcp);
+                 const std::shared_ptr<void> &ownership, const MediaInfo &info, const ProtocolOption &option);
 
 private:
     bool _simulcast = false;

webrtc/WebRtcTransport.cpp

@@ -378,6 +378,12 @@ void WebRtcTransport::setRemoteDtlsFingerprint(const RtcSession &remote) {
 }
 
 void WebRtcTransport::onRtcConfigure(RtcConfigure &configure) const {
+    SdpAttrFingerprint fingerprint;
+    fingerprint.algorithm = _offer_sdp->media[0].fingerprint.algorithm;
+    fingerprint.hash = getFingerprint(fingerprint.algorithm, _dtls_transport);
+    configure.setDefaultSetting(
+            _ice_server->GetUsernameFragment(), _ice_server->GetPassword(), RtpDirection::sendrecv, fingerprint);
+
     // 开启remb后关闭twcc，因为开启twcc后remb无效
     GET_CONFIG(size_t, remb_bit_rate, Rtc::kRembBitRate);
     configure.enableTWCC(!remb_bit_rate);
@@ -407,12 +413,7 @@ std::string WebRtcTransport::getAnswerSdp(const string &offer) {
         setRemoteDtlsFingerprint(*_offer_sdp);
 
         //// sdp 配置 ////
-        SdpAttrFingerprint fingerprint;
-        fingerprint.algorithm = _offer_sdp->media[0].fingerprint.algorithm;
-        fingerprint.hash = getFingerprint(fingerprint.algorithm, _dtls_transport);
         RtcConfigure configure;
-        configure.setDefaultSetting(
-            _ice_server->GetUsernameFragment(), _ice_server->GetPassword(), RtpDirection::sendrecv, fingerprint);
         onRtcConfigure(configure);
 
         //// 生成answer sdp ////
@@ -431,10 +432,6 @@ static bool isDtls(char *buf) {
     return ((*buf > 19) && (*buf < 64));
 }
 
-static string getPeerAddress(RTC::TransportTuple *tuple) {
-    return tuple->get_peer_ip();
-}
-
 void WebRtcTransport::inputSockData(char *buf, int len, RTC::TransportTuple *tuple) {
     if (RTC::StunPacket::IsStun((const uint8_t *)buf, len)) {
         std::unique_ptr<RTC::StunPacket> packet(RTC::StunPacket::Parse((const uint8_t *)buf, len));
@@ -451,7 +448,7 @@ void WebRtcTransport::inputSockData(char *buf, int len, RTC::TransportTuple *tup
     }
     if (isRtp(buf, len)) {
         if (!_srtp_session_recv) {
-            WarnL << "received rtp packet when dtls not completed from:" << getPeerAddress(tuple);
+            WarnL << "received rtp packet when dtls not completed from:" << tuple->get_peer_ip();
             return;
         }
         if (_srtp_session_recv->DecryptSrtp((uint8_t *)buf, &len)) {
@@ -461,7 +458,7 @@ void WebRtcTransport::inputSockData(char *buf, int len, RTC::TransportTuple *tup
     }
     if (isRtcp(buf, len)) {
         if (!_srtp_session_recv) {
-            WarnL << "received rtcp packet when dtls not completed from:" << getPeerAddress(tuple);
+            WarnL << "received rtcp packet when dtls not completed from:" << tuple->get_peer_ip();
             return;
         }
         if (_srtp_session_recv->DecryptSrtcp((uint8_t *)buf, &len)) {
@@ -533,8 +530,7 @@ void WebRtcTransportImp::OnDtlsTransportApplicationDataReceived(const RTC::DtlsT
 #endif
 }
 
-WebRtcTransportImp::WebRtcTransportImp(const EventPoller::Ptr &poller,bool preferred_tcp)
+WebRtcTransportImp::WebRtcTransportImp(const EventPoller::Ptr &poller) : WebRtcTransport(poller) {
-    : WebRtcTransport(poller), _preferred_tcp(preferred_tcp) {
     InfoL << getIdentifier();
 }
 
@@ -674,7 +670,7 @@ void WebRtcTransportImp::onCheckAnswer(RtcSession &sdp) {
     });
     for (auto &m : sdp.media) {
         m.addr.reset();
-        m.addr.address = extern_ips.empty() ? _localIp.empty() ? SockUtil::get_local_ip() : _localIp : extern_ips[0];
+        m.addr.address = extern_ips.empty() ? _local_ip.empty() ? SockUtil::get_local_ip() : _local_ip : extern_ips[0];
         m.rtcp_addr.reset();
         m.rtcp_addr.address = m.addr.address;
 
@@ -769,7 +765,7 @@ void WebRtcTransportImp::onRtcConfigure(RtcConfigure &configure) const {
         return ret;
     });
     if (extern_ips.empty()) {
-        std::string local_ip = _localIp.empty() ? SockUtil::get_local_ip() : _localIp;
+        std::string local_ip = _local_ip.empty() ? SockUtil::get_local_ip() : _local_ip;
         if (local_udp_port) { configure.addCandidate(*makeIceCandidate(local_ip, local_udp_port, 120, "udp")); }
         if (local_tcp_port) { configure.addCandidate(*makeIceCandidate(local_ip, local_tcp_port, _preferred_tcp ? 125 : 115, "tcp")); }
     } else {
@@ -783,12 +779,16 @@ void WebRtcTransportImp::onRtcConfigure(RtcConfigure &configure) const {
     }
 }
 
-void WebRtcTransportImp::setIceCandidate(vector<SdpAttrCandidate> cands) {
+void WebRtcTransportImp::setPreferredTcp(bool flag) {
-    _cands = std::move(cands);
+    _preferred_tcp = flag;
 }
 
-void WebRtcTransportImp::setLocalIp(const std::string &localIp) {
+void WebRtcTransportImp::setLocalIp(std::string local_ip) {
-    _localIp = localIp;
+    _local_ip = std::move(local_ip);
+}
+
+void WebRtcTransportImp::setIceCandidate(vector<SdpAttrCandidate> cands) {
+    _cands = std::move(cands);
 }
 
 ///////////////////////////////////////////////////////////////////
@@ -1278,21 +1278,14 @@ 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 setLocalIp(const WebRtcInterface& exchanger, const std::string& localIp) {
-    return const_cast<WebRtcInterface &>(exchanger).setLocalIp(localIp);
-}
 
 void WebRtcPluginManager::setListener(Listener cb) {
     lock_guard<mutex> lck(_mtx_creator);
     _listener = std::move(cb);
 }
 
-void WebRtcPluginManager::getAnswerSdp(Session &sender, const string &type, const WebRtcArgs &args, const onCreateRtc &cb_in) {
+void WebRtcPluginManager::negotiateSdp(Session &sender, const string &type, const WebRtcArgs &args, const onCreateWebRtc &cb_in) {
-    onCreateRtc cb;
+    onCreateWebRtc cb;
     lock_guard<mutex> lck(_mtx_creator);
     if (_listener) {
         auto listener = _listener;
@@ -1308,21 +1301,19 @@ void WebRtcPluginManager::getAnswerSdp(Session &sender, const string &type, cons
 
     auto it = _map_creator.find(type);
     if (it == _map_creator.end()) {
-        cb(WebRtcException(SockException(Err_other, "the type can not supported")));
+        cb_in(WebRtcException(SockException(Err_other, "the type can not supported")));
         return;
     }
     it->second(sender, args, cb);
 }
 
-void echo_plugin(Session &sender, const WebRtcArgs &args, const WebRtcPluginManager::onCreateRtc &cb) {
+void echo_plugin(Session &sender, const WebRtcArgs &args, const onCreateWebRtc &cb) {
     cb(*WebRtcEchoTest::create(EventPollerPool::Instance().getPoller()));
 }
 
-void push_plugin(Session &sender, const WebRtcArgs &args, const WebRtcPluginManager::onCreateRtc &cb) {
+void push_plugin(Session &sender, const WebRtcArgs &args, const onCreateWebRtc &cb) {
     MediaInfo info(args["url"]);
-    bool preferred_tcp = args["preferred_tcp"];
+    Broadcast::PublishAuthInvoker invoker = [cb, info](const string &err, const ProtocolOption &option) mutable {
-
-    Broadcast::PublishAuthInvoker invoker = [cb, info, preferred_tcp](const string &err, const ProtocolOption &option) mutable {
         if (!err.empty()) {
             cb(WebRtcException(SockException(Err_other, err)));
             return;
@@ -1361,7 +1352,7 @@ void push_plugin(Session &sender, const WebRtcArgs &args, const WebRtcPluginMana
             push_src_ownership = push_src->getOwnership();
             push_src->setProtocolOption(option);
         }
-        auto rtc = WebRtcPusher::create(EventPollerPool::Instance().getPoller(), push_src, push_src_ownership, info, option, preferred_tcp);
+        auto rtc = WebRtcPusher::create(EventPollerPool::Instance().getPoller(), push_src, push_src_ownership, info, option);
         push_src->setListener(rtc);
         cb(*rtc);
     };
@@ -1374,12 +1365,10 @@ void push_plugin(Session &sender, const WebRtcArgs &args, const WebRtcPluginMana
     }
 }
 
-void play_plugin(Session &sender, const WebRtcArgs &args, const WebRtcPluginManager::onCreateRtc &cb) {
+void play_plugin(Session &sender, const WebRtcArgs &args, const onCreateWebRtc &cb) {
     MediaInfo info(args["url"]);
-    bool preferred_tcp = args["preferred_tcp"];
-
     auto session_ptr = static_pointer_cast<Session>(sender.shared_from_this());
-    Broadcast::AuthInvoker invoker = [cb, info, session_ptr, preferred_tcp](const string &err) mutable {
+    Broadcast::AuthInvoker invoker = [cb, info, session_ptr](const string &err) mutable {
         if (!err.empty()) {
             cb(WebRtcException(SockException(Err_other, err)));
             return;
@@ -1395,7 +1384,7 @@ void play_plugin(Session &sender, const WebRtcArgs &args, const WebRtcPluginMana
             }
             // 还原成rtc，目的是为了hook时识别哪种播放协议
             info.schema = "rtc";
-            auto rtc = WebRtcPlayer::create(EventPollerPool::Instance().getPoller(), src, info, preferred_tcp);
+            auto rtc = WebRtcPlayer::create(EventPollerPool::Instance().getPoller(), src, info);
             cb(*rtc);
         });
     };
@@ -1408,14 +1397,35 @@ void play_plugin(Session &sender, const WebRtcArgs &args, const WebRtcPluginMana
     }
 }
 
-static void set_webrtc_cands(const WebRtcArgs &args, const WebRtcInterface &rtc) {
+static void setWebRtcArgs(const WebRtcArgs &args, WebRtcInterface &rtc) {
+    {
+        static auto is_vaild_ip = [](const std::string &ip) -> bool {
+            int a, b, c, d;
+            return sscanf(ip.c_str(), "%d.%d.%d.%d", &a, &b, &c, &d) == 4;
+        };
+        std::string host = args["Host"];
+        if (!host.empty()) {
+            auto local_ip = host.substr(0, host.find(':'));
+            if (!is_vaild_ip(local_ip) || local_ip == "127.0.0.1") {
+                local_ip = "";
+            }
+            rtc.setLocalIp(std::move(local_ip));
+        }
+    }
+
+    bool preferred_tcp = args["preferred_tcp"];
+    {
+        rtc.setPreferredTcp(preferred_tcp);
+    }
+
+    {
         vector<SdpAttrCandidate> cands;
         {
             auto cand_str = trim(args["cand_udp"]);
             auto ip_port = toolkit::split(cand_str, ":");
             if (ip_port.size() == 2) {
                 // udp优先
-            auto ice_cand = makeIceCandidate(ip_port[0], atoi(ip_port[1].data()), 120, "udp");
+                auto ice_cand = makeIceCandidate(ip_port[0], atoi(ip_port[1].data()), preferred_tcp ? 100 : 120, "udp");
                 cands.emplace_back(std::move(*ice_cand));
             }
         }
@@ -1424,23 +1434,26 @@ static void set_webrtc_cands(const WebRtcArgs &args, const WebRtcInterface &rtc)
             auto ip_port = toolkit::split(cand_str, ":");
             if (ip_port.size() == 2) {
                 // tcp模式
-            auto ice_cand = makeIceCandidate(ip_port[0], atoi(ip_port[1].data()), 100, "tcp");
+                auto ice_cand = makeIceCandidate(ip_port[0], atoi(ip_port[1].data()), preferred_tcp ? 120 : 100, "tcp");
                 cands.emplace_back(std::move(*ice_cand));
             }
         }
         if (!cands.empty()) {
             // udp优先
-        const_cast<WebRtcInterface &>(rtc).setIceCandidate(std::move(cands));
+            rtc.setIceCandidate(std::move(cands));
+        }
     }
 }
 
 static onceToken s_rtc_auto_register([]() {
+#if !defined (NDEBUG)
+    // debug模式才开启echo插件
     WebRtcPluginManager::Instance().registerPlugin("echo", echo_plugin);
+#endif
     WebRtcPluginManager::Instance().registerPlugin("push", push_plugin);
     WebRtcPluginManager::Instance().registerPlugin("play", play_plugin);
-
     WebRtcPluginManager::Instance().setListener([](Session &sender, const std::string &type, const WebRtcArgs &args, const WebRtcInterface &rtc) {
-        set_webrtc_cands(args, rtc);
+        setWebRtcArgs(args, const_cast<WebRtcInterface&>(rtc));
     });
 });
 

webrtc/WebRtcTransport.h

@@ -42,13 +42,10 @@ public:
     virtual const std::string& getIdentifier() const = 0;
     virtual const std::string& deleteRandStr() const { static std::string s_null; return s_null; }
     virtual void setIceCandidate(std::vector<SdpAttrCandidate> cands) {}
-    virtual void setLocalIp(const std::string &localIp) {}
+    virtual void setLocalIp(std::string localIp) {}
+    virtual void setPreferredTcp(bool flag) {}
 };
 
-std::string exchangeSdp(const WebRtcInterface &exchanger, const std::string& offer);
-
-void setLocalIp(const WebRtcInterface &exchanger, const std::string &localIp);
-
 class WebRtcException : public WebRtcInterface {
 public:
     WebRtcException(const SockException &ex) : _ex(ex) {};
@@ -88,7 +85,7 @@ public:
      * @param offer offer sdp
      * @return answer sdp
      */
-    std::string getAnswerSdp(const std::string &offer) override;
+    std::string getAnswerSdp(const std::string &offer) override final;
 
     /**
      * 获取对象唯一id
@@ -252,14 +249,16 @@ public:
     void onSendRtp(const RtpPacket::Ptr &rtp, bool flush, bool rtx = false);
 
     void createRtpChannel(const std::string &rid, uint32_t ssrc, MediaTrack &track);
-    void setIceCandidate(std::vector<SdpAttrCandidate> cands) override;
     void removeTuple(RTC::TransportTuple* tuple);
     void safeShutdown(const SockException &ex);
 
-    void setLocalIp(const std::string &localIp) override;
+    void setPreferredTcp(bool flag) override;
+    void setLocalIp(std::string local_ip) override;
+    void setIceCandidate(std::vector<SdpAttrCandidate> cands) override;
+
 protected:
     void OnIceServerSelectedTuple(const RTC::IceServer *iceServer, RTC::TransportTuple *tuple) override;
-    WebRtcTransportImp(const EventPoller::Ptr &poller,bool preferred_tcp = false);
+    WebRtcTransportImp(const EventPoller::Ptr &poller);
     void OnDtlsTransportApplicationDataReceived(const RTC::DtlsTransport *dtlsTransport, const uint8_t *data, size_t len) override;
     void onStartWebRTC() override;
     void onSendSockData(Buffer::Ptr buf, bool flush = true, RTC::TransportTuple *tuple = nullptr) override;
@@ -273,7 +272,7 @@ protected:
     void onCreate() override;
     void onDestory() override;
     void onShutdown(const SockException &ex) override;
-    virtual void onRecvRtp(MediaTrack &track, const std::string &rid, RtpPacket::Ptr rtp) = 0;
+    virtual void onRecvRtp(MediaTrack &track, const std::string &rid, RtpPacket::Ptr rtp) {}
     void updateTicker();
     float getLossRate(TrackType type);
     void onRtcpBye() override;
@@ -289,7 +288,7 @@ private:
     void onCheckAnswer(RtcSession &sdp);
 
 private:
-    bool _preferred_tcp;
+    bool _preferred_tcp = false;
     uint16_t _rtx_seq[2] = {0, 0};
     //用掉的总流量
     uint64_t _bytes_usage = 0;
@@ -310,8 +309,8 @@ private:
     //根据接收rtp的pt获取相关信息
     std::unordered_map<uint8_t/*pt*/, std::unique_ptr<WrappedMediaTrack>> _pt_to_track;
     std::vector<SdpAttrCandidate> _cands;
-    //源访问的hostip
+    //http访问时的host ip
-    std::string _localIp;
+    std::string _local_ip;
 };
 
 class WebRtcTransportManager {
@@ -333,21 +332,20 @@ private:
 class WebRtcArgs : public std::enable_shared_from_this<WebRtcArgs> {
 public:
     virtual ~WebRtcArgs() = default;
-
     virtual variant operator[](const std::string &key) const = 0;
 };
 
+using onCreateWebRtc = std::function<void(const WebRtcInterface &rtc)>;
 class WebRtcPluginManager {
 public:
-    using onCreateRtc = std::function<void(const WebRtcInterface &rtc)>;
+    using Plugin = std::function<void(Session &sender, const WebRtcArgs &args, const onCreateWebRtc &cb)>;
-    using Plugin = std::function<void(Session &sender, const WebRtcArgs &args, const onCreateRtc &cb)>;
     using Listener = std::function<void(Session &sender, const std::string &type, const WebRtcArgs &args, const WebRtcInterface &rtc)>;
 
     static WebRtcPluginManager &Instance();
 
     void registerPlugin(const std::string &type, Plugin cb);
-    void getAnswerSdp(Session &sender, const std::string &type, const WebRtcArgs &args, const onCreateRtc &cb);
     void setListener(Listener cb);
+    void negotiateSdp(Session &sender, const std::string &type, const WebRtcArgs &args, const onCreateWebRtc &cb);
 
 private:
     WebRtcPluginManager() = default;