新增webrtc播放器datachannel消息广播机制和接口

ZLToolKit @ 5d74e09b

-Subproject commit 97f9b9a2ac58353f72f085830690d27833b8ad88
+Subproject commit 5d74e09b8c84cccc46036ed2ef1a62f670c423d4

server/WebApi.cpp

@@ -816,6 +816,19 @@ void installWebApi() {
             });
     });
 
+    api_regist("/index/api/broadcastMessage", [](API_ARGS_MAP) {
+        CHECK_SECRET();
+        CHECK_ARGS("schema", "vhost", "app", "stream", "msg");
+        auto src = MediaSource::find(allArgs["schema"], allArgs["vhost"], allArgs["app"], allArgs["stream"]);
+        if (!src) {
+            throw ApiRetException("can not find the stream", API::NotFound);
+        }
+        Any any;
+        Buffer::Ptr buffer = std::make_shared<BufferLikeString>(allArgs["msg"]);
+        any.set(std::move(buffer));
+        src->broadcastMessage(any);
+    });
+
     //测试url http://127.0.0.1/index/api/getMediaInfo?schema=rtsp&vhost=__defaultVhost__&app=live&stream=obs
     api_regist("/index/api/getMediaInfo",[](API_ARGS_MAP_ASYNC){
         CHECK_SECRET();

src/Common/MediaSource.h

@@ -353,6 +353,8 @@ public:
         cb(std::list<toolkit::Any>());
     }
 
+    virtual bool broadcastMessage(const toolkit::Any &data) { return false; }
+
     // 获取媒体源类型
     MediaOriginType getOriginType() const;
     // 获取媒体源url或者文件路径

src/Rtsp/RtspMediaSource.h

@@ -55,9 +55,16 @@ public:
 
     void getPlayerList(const std::function<void(const std::list<toolkit::Any> &info_list)> &cb,
                        const std::function<toolkit::Any(toolkit::Any &&info)> &on_change) override {
+        assert(_ring);
         _ring->getInfoList(cb, on_change);
     }
 
+    bool broadcastMessage(const toolkit::Any &data) override {
+        assert(_ring);
+        _ring->sendMessage(data);
+        return true;
+    }
+
     /**
      * 获取播放器个数
      */

webrtc/WebRtcPlayer.cpp

@@ -71,6 +71,21 @@ void WebRtcPlayer::onStartWebRTC() {
             }
             strong_self->onShutdown(SockException(Err_shutdown, "rtsp ring buffer detached"));
         });
+
+        _reader->setMessageCB([weak_self] (const toolkit::Any &data) {
+            auto strong_self = weak_self.lock();
+            if (!strong_self) {
+                return;
+            }
+            if (data.is<Buffer>()) {
+                auto &buffer = data.get<Buffer>();
+                // PPID 51: 文本string
+                // PPID 53: 二进制
+                strong_self->sendDatachannel(0, 51, buffer.data(), buffer.size());
+            } else {
+                WarnL << "Send unknown message type to webrtc player: " << data.type_name();
+            }
+        });
     }
 }
 void WebRtcPlayer::onDestory() {

webrtc/WebRtcTransport.cpp

@@ -229,6 +229,19 @@ void WebRtcTransport::OnSctpAssociationMessageReceived(
     _sctp->SendSctpMessage(params, ppid, msg, len);
 }
 #endif
+
+void WebRtcTransport::sendDatachannel(uint16_t streamId, uint32_t ppid, const char *msg, size_t len) {
+#ifdef ENABLE_SCTP
+    if (_sctp) {
+        RTC::SctpStreamParameters params;
+        params.streamId = streamId;
+        _sctp->SendSctpMessage(params, ppid, (uint8_t *)msg, len);
+    }
+#else
+    WarnL << "WebRTC datachannel disabled!";
+#endif
+}
+
 //////////////////////////////////////////////////////////////////////////////////////////////////////////////////////
 
 void WebRtcTransport::sendSockData(const char *buf, size_t len, RTC::TransportTuple *tuple) {

webrtc/WebRtcTransport.h

@@ -112,6 +112,7 @@ public:
      */
     void sendRtpPacket(const char *buf, int len, bool flush, void *ctx = nullptr);
     void sendRtcpPacket(const char *buf, int len, bool flush, void *ctx = nullptr);
+    void sendDatachannel(uint16_t streamId, uint32_t ppid, const char *msg, size_t len);
 
     const EventPoller::Ptr& getPoller() const;
     Session::Ptr getSession() const;