统一代码命名风格 (#2910)

openRtpServer时，如果开启tcp模式，从tcp端口池获取

.github/workflows/c-cpp.yml

+name: Linux
+
+on: [push, pull_request]
+
+jobs:
+  build:
+
+    runs-on: ubuntu-20.04
+
+    steps:
+    - uses: actions/checkout@v1
+
+    - name: 下载submodule源码
+      run: mv -f .gitmodules_github .gitmodules && git submodule sync && git submodule update --init
+
+    - name: apt-get安装依赖库(非必选)
+      run: sudo apt-get update && sudo apt-get install -y cmake libssl-dev libsdl-dev libavcodec-dev libavutil-dev libswscale-dev libresample-dev libusrsctp-dev
+
+    - name: 下载 SRTP
+      uses: actions/checkout@v2
+      with:
+        repository: cisco/libsrtp
+        fetch-depth: 1
+        ref: v2.3.0
+        path: 3rdpart/libsrtp
+
+    - name: 编译 SRTP
+      run: cd 3rdpart/libsrtp && ./configure --enable-openssl && make -j4 && sudo make install
+
+    - name: 编译
+      run: mkdir -p linux_build && cd linux_build && cmake .. -DENABLE_WEBRTC=true -DENABLE_FFMPEG=true && make -j $(nproc)
+
+    - name: 运行MediaServer
+      run: pwd && cd release/linux/Debug && sudo ./MediaServer -d &

server/WebHook.cpp

@@ -674,7 +674,7 @@ void installWebHook() {
         });
     });
 
-    NoticeCenter::Instance().addListener(&web_hook_tag, Broadcast::KBroadcastRtpServerTimeout, [](BroadcastRtpServerTimeoutArgs) {
+    NoticeCenter::Instance().addListener(&web_hook_tag, Broadcast::kBroadcastRtpServerTimeout, [](BroadcastRtpServerTimeoutArgs) {
         GET_CONFIG(string, rtp_server_timeout, Hook::kOnRtpServerTimeout);
         if (!hook_enable || rtp_server_timeout.empty()) {
             return;
@@ -703,4 +703,4 @@ void unInstallWebHook() {
 
 void onProcessExited() {
     reportServerExited();
-}
\ No newline at end of file
+}

src/Common/config.cpp

@@ -57,7 +57,7 @@ const string kBroadcastNotFoundStream = "kBroadcastNotFoundStream";
 const string kBroadcastStreamNoneReader = "kBroadcastStreamNoneReader";
 const string kBroadcastHttpBeforeAccess = "kBroadcastHttpBeforeAccess";
 const string kBroadcastSendRtpStopped = "kBroadcastSendRtpStopped";
-const string KBroadcastRtpServerTimeout = "KBroadcastRtpServerTimeout";
+const string kBroadcastRtpServerTimeout = "kBroadcastRtpServerTimeout";
 
 } // namespace Broadcast
 

src/Common/config.h

@@ -106,7 +106,7 @@ extern const std::string kBroadcastReloadConfig;
 #define BroadcastReloadConfigArgs void
 
 // rtp server 超时
-extern const std::string KBroadcastRtpServerTimeout;
+extern const std::string kBroadcastRtpServerTimeout;
 #define BroadcastRtpServerTimeoutArgs uint16_t &local_port, const string &stream_id,int &tcp_mode, bool &re_use_port, uint32_t &ssrc
 
 #define ReloadConfigTag ((void *)(0xFF))

src/Http/HttpSession.cpp

@@ -166,11 +166,11 @@ void HttpSession::onError(const SockException &err) {
     if (_is_live_stream) {
         // flv/ts播放器
         uint64_t duration = _ticker.createdTime() / 1000;
-        WarnP(this) << "FLV/TS/FMP4播放器(" << _mediaInfo.shortUrl() << ")断开:" << err << ",耗时(s):" << duration;
+        WarnP(this) << "FLV/TS/FMP4播放器(" << _media_info.shortUrl() << ")断开:" << err << ",耗时(s):" << duration;
 
         GET_CONFIG(uint32_t, iFlowThreshold, General::kFlowThreshold);
         if (_total_bytes_usage >= iFlowThreshold * 1024) {
-            NOTICE_EMIT(BroadcastFlowReportArgs, Broadcast::kBroadcastFlowReport, _mediaInfo, _total_bytes_usage, duration, true, *this);
+            NOTICE_EMIT(BroadcastFlowReportArgs, Broadcast::kBroadcastFlowReport, _media_info, _total_bytes_usage, duration, true, *this);
         }
         return;
     }
@@ -263,9 +263,9 @@ bool HttpSession::checkLiveStream(const string &schema, const string &url_suffix
     }
 
     // 解析带上协议+参数完整的url
-    _mediaInfo.parse(schema + "://" + _parser["Host"] + url);
+    _media_info.parse(schema + "://" + _parser["Host"] + url);
 
-    if (_mediaInfo.app.empty() || _mediaInfo.stream.empty()) {
+    if (_media_info.app.empty() || _media_info.stream.empty()) {
         // url不合法
         return false;
     }
@@ -288,7 +288,7 @@ bool HttpSession::checkLiveStream(const string &schema, const string &url_suffix
         }
 
         // 异步查找直播流
-        MediaSource::findAsync(strong_self->_mediaInfo, strong_self, [weak_self, close_flag, cb](const MediaSource::Ptr &src) {
+        MediaSource::findAsync(strong_self->_media_info, strong_self, [weak_self, close_flag, cb](const MediaSource::Ptr &src) {
             auto strong_self = weak_self.lock();
             if (!strong_self) {
                 // 本对象已经销毁
@@ -311,7 +311,7 @@ bool HttpSession::checkLiveStream(const string &schema, const string &url_suffix
         }
     };
 
-    auto flag = NOTICE_EMIT(BroadcastMediaPlayedArgs, Broadcast::kBroadcastMediaPlayed, _mediaInfo, invoker, *this);
+    auto flag = NOTICE_EMIT(BroadcastMediaPlayedArgs, Broadcast::kBroadcastMediaPlayed, _media_info, invoker, *this);
     if (!flag) {
         // 该事件无人监听,默认不鉴权
         onRes("");

src/Http/HttpSession.h

@@ -124,7 +124,7 @@ private:
     void setSocketFlags();
 
 protected:
-    MediaInfo _mediaInfo;
+    MediaInfo _media_info;
 
 private:
     bool _is_live_stream = false;

src/Rtp/RtpServer.cpp

@@ -102,7 +102,7 @@ public:
                     process->setOnDetach(std::move(strong_self->_on_detach));
                 }
                 if (!process) { // process 未创建，触发rtp server 超时事件
-                    NOTICE_EMIT(BroadcastRtpServerTimeoutArgs, Broadcast::KBroadcastRtpServerTimeout, strong_self->_local_port, strong_self->_stream_id,
+                    NOTICE_EMIT(BroadcastRtpServerTimeoutArgs, Broadcast::kBroadcastRtpServerTimeout, strong_self->_local_port, strong_self->_stream_id,
                                 (int)strong_self->_tcp_mode, strong_self->_re_use_port, strong_self->_ssrc);
                 }
             }
@@ -161,7 +161,7 @@ void RtpServer::start(uint16_t local_port, const string &stream_id, TcpMode tcp_
     if (local_port == 0) {
         //随机端口，rtp端口采用偶数
         auto pair = std::make_pair(rtp_socket, rtcp_socket);
-        makeSockPair(pair, local_ip, re_use_port);
+        makeSockPair(pair, local_ip, re_use_port, TcpMode::NONE == tcp_mode);
         local_port = rtp_socket->get_local_port();
     } else if (!rtp_socket->bindUdpSock(local_port, local_ip, re_use_port)) {
         //用户指定端口