Add datachannel c apis and callbacks(#3328)

增加datachannel数据收发的回调通知 #3326，和控制datachannel回显的开关

---------

Co-authored-by: xiongziliang <771730766@qq.com>

api/include/mk_events.h

@@ -177,6 +177,33 @@ typedef struct {
      */
     void(API_CALL *on_mk_media_send_rtp_stop)(const char *vhost, const char *app, const char *stream, const char *ssrc, int err, const char *msg);
 
+    /**
+     * rtc sctp连接中/完成/失败/关闭回调
+     * @param rtc_transport 数据通道对象
+     */
+    void(API_CALL *on_mk_rtc_sctp_connecting)(mk_rtc_transport rtc_transport);
+    void(API_CALL *on_mk_rtc_sctp_connected)(mk_rtc_transport rtc_transport);
+    void(API_CALL *on_mk_rtc_sctp_failed)(mk_rtc_transport rtc_transport);
+    void(API_CALL *on_mk_rtc_sctp_closed)(mk_rtc_transport rtc_transport);
+
+    /**
+     * rtc数据通道发送数据回调
+     * @param rtc_transport 数据通道对象
+     * @param msg 数据
+     * @param len 数据长度
+     */
+    void(API_CALL *on_mk_rtc_sctp_send)(mk_rtc_transport rtc_transport, const uint8_t *msg, size_t len);
+
+    /**
+     * rtc数据通道接收数据回调
+     * @param rtc_transport 数据通道对象
+     * @param streamId 流id
+     * @param ppid 协议id
+     * @param msg 数据
+     * @param len 数据长度
+     */
+    void(API_CALL *on_mk_rtc_sctp_received)(mk_rtc_transport rtc_transport, uint16_t streamId, uint32_t ppid, const uint8_t *msg, size_t len);
+
 } mk_events;
 
 

api/include/mk_events_objects.h

@@ -352,6 +352,20 @@ API_EXPORT mk_auth_invoker API_CALL mk_auth_invoker_clone(const mk_auth_invoker
  */
 API_EXPORT void API_CALL mk_auth_invoker_clone_release(const mk_auth_invoker ctx);
 
+///////////////////////////////////////////WebRtcTransport/////////////////////////////////////////////
+//WebRtcTransport对象的C映射
+typedef struct mk_rtc_transport_t *mk_rtc_transport;
+
+/**
+ * 发送rtc数据通道
+ * @param ctx 数据通道对象
+ * @param streamId 流id
+ * @param ppid 协议id
+ * @param msg 数据
+ * @param len 数据长度
+ */
+API_EXPORT void API_CALL mk_rtc_send_datachannel(const mk_rtc_transport ctx, uint16_t streamId, uint32_t ppid, const char* msg, size_t len);
+
 #ifdef __cplusplus
 }
 #endif

api/source/mk_events.cpp

@@ -14,6 +14,7 @@
 #include "Http/HttpSession.h"
 #include "Rtsp/RtspSession.h"
 #include "Record/MP4Recorder.h"
+#include "webrtc/WebRtcTransport.h"
 
 using namespace toolkit;
 using namespace mediakit;
@@ -167,6 +168,42 @@ API_EXPORT void API_CALL mk_events_listen(const mk_events *events){
                                                    sender.getMediaTuple().stream.c_str(), ssrc.c_str(), ex.getErrCode(), ex.what());
             }
         });
+
+        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);
+            }
+        });
+
+        NoticeCenter::Instance().addListener(&s_tag, Broadcast::kBroadcastRtcSctpConnected,[](BroadcastRtcSctpConnectArgs){
+            if (s_events.on_mk_rtc_sctp_connected) {
+                s_events.on_mk_rtc_sctp_connected((mk_rtc_transport)&sender);
+            }
+        });
+
+        NoticeCenter::Instance().addListener(&s_tag, Broadcast::kBroadcastRtcSctpFailed,[](BroadcastRtcSctpConnectArgs){
+            if (s_events.on_mk_rtc_sctp_failed) {
+                s_events.on_mk_rtc_sctp_failed((mk_rtc_transport)&sender);
+            }
+        });
+
+        NoticeCenter::Instance().addListener(&s_tag, Broadcast::kBroadcastRtcSctpClosed,[](BroadcastRtcSctpConnectArgs){
+            if (s_events.on_mk_rtc_sctp_closed) {
+                s_events.on_mk_rtc_sctp_closed((mk_rtc_transport)&sender);
+            }
+        });
+
+        NoticeCenter::Instance().addListener(&s_tag, Broadcast::kBroadcastRtcSctpSend,[](BroadcastRtcSctpSendArgs){
+            if (s_events.on_mk_rtc_sctp_send) {
+                s_events.on_mk_rtc_sctp_send((mk_rtc_transport)&sender, data, len);
+            }
+        });
+
+        NoticeCenter::Instance().addListener(&s_tag, Broadcast::kBroadcastRtcSctpReceived,[](BroadcastRtcSctpReceivedArgs){
+            if (s_events.on_mk_rtc_sctp_received) {
+                s_events.on_mk_rtc_sctp_received((mk_rtc_transport)&sender, streamId, ppid, msg, len);
+            }
+        });
     });
 
 }

api/source/mk_events_objects.cpp

@@ -17,6 +17,7 @@
 
 #include "Http/HttpClient.h"
 #include "Rtsp/RtspSession.h"
+#include "webrtc/WebRtcTransport.h"
 
 using namespace toolkit;
 using namespace mediakit;
@@ -497,4 +498,22 @@ API_EXPORT void API_CALL mk_auth_invoker_clone_release(const mk_auth_invoker ctx
     assert(ctx);
     Broadcast::AuthInvoker *invoker = (Broadcast::AuthInvoker *)ctx;
     delete invoker;
-}
+}
+
+///////////////////////////////////////////WebRtcTransport/////////////////////////////////////////////
+API_EXPORT void API_CALL mk_rtc_sendDatachannel(const mk_rtc_transport ctx, uint16_t streamId, uint32_t ppid, const char *msg, size_t len) {
+#ifdef ENABLE_WEBRTC
+    assert(ctx && msg);
+    WebRtcTransport *transport = (WebRtcTransport *)ctx;
+    std::string msg_str(msg, len);
+    std::weak_ptr<WebRtcTransport> weak_trans = transport->shared_from_this();
+    transport->getPoller()->async([streamId, ppid, msg_str, weak_trans]() {
+        // 切换线程后再操作
+        if (auto trans = weak_trans.lock()) {
+            trans->sendDatachannel(streamId, ppid, msg_str.c_str(), msg_str.size());
+        }
+    });
+#else
+    WarnL << "未启用webrtc功能, 编译时请开启ENABLE_WEBRTC";
+#endif
+}