feat: 增加webrtc代理拉流 (#4389)

- 增加客户端模式，支持主动拉流、推流：
   - addStreamProxy接口新增支持whep主动拉流，拉流地址目前只兼容zlm的whep url。
   - addStreamPusherProxy接口新增支持whip主动推流，推流地址目前只兼容zlm的whip url。
   - 以上推流url格式为webrtc[s]://server_host:server_port/app/stream_id?key=value, 内部会自动转换为http[s]://server_host:server_port/index/api/[whip/whep]?app=app&stream=stream_id&key=value。

- 增加WebRtc p2p 模式：
  - 增加 ICE FULL模式。
  - 增加STUN/TURN 服务器。
  - 增加websocket 信令。
  - 增加P2P代理拉流。

---------

Co-authored-by: xia-chu <771730766@qq.com>
Co-authored-by: mtdxc <mtdxc@126.com>
Co-authored-by: cqm <cqm@97kid.com>

src/Common/macros.cpp

@@ -30,7 +30,7 @@ namespace mediakit {
  * [AUTO-TRANSLATED:f214f734]
  */
 #if !defined(ENABLE_VERSION)
-const char kServerName[] =  "ZLMediaKit-8.0(build in " __DATE__ " " __TIME__ ")";
+const char kServerName[] =  "ZLMediaKit-9.0(build in " __DATE__ " " __TIME__ ")";
 #else
 const char kServerName[] = "ZLMediaKit(git hash:" COMMIT_HASH "/" COMMIT_TIME ",branch:" BRANCH_NAME ",build time:" BUILD_TIME ")";
 #endif

src/Common/macros.h

@@ -62,6 +62,7 @@
     }
 #endif // CLEAR_ARR
 
+#define RTC_SCHEMA "rtc"
 #define RTSP_SCHEMA "rtsp"
 #define RTMP_SCHEMA "rtmp"
 #define TS_SCHEMA "ts"

src/Http/HttpBody.cpp

@@ -163,7 +163,7 @@ static std::shared_ptr<char> getSharedMmap(const string &file_path, int64_t &fil
 
     if (addr_ == nullptr) {
         mmap_close(hfile, hmapping, addr_);
-		WarnL << "MapViewOfFile() " << file_path << " failed:";
+        WarnL << "MapViewOfFile() " << file_path << " failed:";
         return nullptr;
     }
 

src/Player/MediaPlayer.h

@@ -26,6 +26,7 @@ public:
     void play(const std::string &url) override;
     toolkit::EventPoller::Ptr getPoller();
     void setOnCreateSocket(toolkit::Socket::onCreateSocket cb);
+    const PlayerBase::Ptr& getDelegate() const { return _delegate; }
 
 private:
     toolkit::EventPoller::Ptr _poller;

src/Player/PlayerBase.cpp

@@ -18,7 +18,9 @@
 #ifdef ENABLE_SRT
 #include "Srt/SrtPlayerImp.h"
 #endif // ENABLE_SRT
-
+#ifdef ENABLE_WEBRTC
+#include "../webrtc/WebRtcProxyPlayerImp.h"
+#endif // ENABLE_WEBRTC
 using namespace std;
 using namespace toolkit;
 
@@ -84,6 +86,11 @@ PlayerBase::Ptr PlayerBase::createPlayer(const EventPoller::Ptr &in_poller, cons
         return PlayerBase::Ptr(new SrtPlayerImp(poller), release_func);
     }
 #endif//ENABLE_SRT
+#ifdef ENABLE_WEBRTC
+    if ((strcasecmp("webrtc", prefix.data()) == 0 || strcasecmp("webrtcs", prefix.data()) == 0)) {
+        return PlayerBase::Ptr(new WebRtcProxyPlayerImp(poller), release_func);
+    }
+#endif//ENABLE_WEBRTC
 
     throw std::invalid_argument("not supported play schema:" + url_in);
 }

src/Player/PlayerProxy.cpp

@@ -286,6 +286,10 @@ float PlayerProxy::getLossRate(MediaSource &sender, TrackType type) {
     return getPacketLossRate(type);
 }
 
+toolkit::EventPoller::Ptr PlayerProxy::getOwnerPoller(MediaSource &sender) { 
+    return getPoller();
+}
+
 TranslationInfo PlayerProxy::getTranslationInfo() {
     return _transtalion_info;
 }

src/Player/PlayerProxy.h

@@ -151,6 +151,7 @@ private:
     std::string getOriginUrl(MediaSource &sender) const override;
     std::shared_ptr<toolkit::SockInfo> getOriginSock(MediaSource &sender) const override;
     float getLossRate(MediaSource &sender, TrackType type) override;
+    toolkit::EventPoller::Ptr getOwnerPoller(MediaSource &sender) override;
 
     void rePlay(const std::string &strUrl, int iFailedCnt);
     void onPlaySuccess();

src/Pusher/PusherBase.cpp

@@ -15,6 +15,9 @@
 #ifdef ENABLE_SRT
 #include "Srt/SrtPusher.h"
 #endif // ENABLE_SRT
+#ifdef ENABLE_WEBRTC
+#include "../webrtc/WebRtcProxyPusher.h"
+#endif // ENABLE_WEBRTC
 
 using namespace toolkit;
 
@@ -23,7 +26,8 @@ namespace mediakit {
 static bool checkMediaSourceAndUrlMatch(const MediaSource::Ptr &src, const std::string &url) {
     std::string prefix = findSubString(url.data(), NULL, "://");
 
-    if (strcasecmp("rtsps", prefix.data()) == 0 || strcasecmp("rtsp", prefix.data()) == 0) {
+    if (strcasecmp("rtsps", prefix.data()) == 0 || strcasecmp("rtsp", prefix.data()) == 0 ||
+        strcasecmp("webrtcs", prefix.data()) == 0 || strcasecmp("webrtc", prefix.data()) == 0 ) {
         auto rtsp_src = std::dynamic_pointer_cast<RtspMediaSource>(src);
         if (!rtsp_src) {
             return false;
@@ -91,6 +95,11 @@ PusherBase::Ptr PusherBase::createPusher(const EventPoller::Ptr &in_poller,
     }
 #endif//ENABLE_SRT
 
+#ifdef ENABLE_WEBRTC
+    if ((strcasecmp("webrtc", prefix.data()) == 0 || strcasecmp("webrtcs", prefix.data()) == 0)) {
+        return PusherBase::Ptr(new WebRtcProxyPusherImp(poller, std::dynamic_pointer_cast<RtspMediaSource>(src)), release_func);
+    }
+#endif//ENABLE_WEBRTC
 
     throw std::invalid_argument("not supported push schema:" + url);
 }

src/Pusher/PusherBase.h

@@ -69,7 +69,7 @@ public:
 
     virtual size_t getSendSpeed() { return 0; }
     virtual size_t getSendTotalBytes() { return 0; }
-	
+    
 protected:
     virtual void onShutdown(const toolkit::SockException &ex) = 0;
     virtual void onPublishResult(const toolkit::SockException &ex) = 0;
@@ -139,11 +139,11 @@ public:
     size_t getSendSpeed() override {
         return _delegate ?  _delegate->getSendSpeed() : Parent::getSendSpeed();
     }
-	
+    
    size_t getSendTotalBytes() override {
         return _delegate ? _delegate->getSendTotalBytes() : Parent::getSendTotalBytes();
     }
-	
+    
 protected:
     void onShutdown(const toolkit::SockException &ex) override {
         if (_on_shutdown) {

src/Record/HlsMaker.cpp

@@ -17,7 +17,7 @@ using namespace std;
 namespace mediakit {
 
 HlsMaker::HlsMaker(bool is_fmp4, float seg_duration, uint32_t seg_number, bool seg_keep) {
-	_is_fmp4 = is_fmp4;
+    _is_fmp4 = is_fmp4;
     // 最小允许设置为0，0个切片代表点播  [AUTO-TRANSLATED:19235e8e]
     // Minimum allowed setting is 0, 0 slices represent on-demand
     _seg_number = seg_number;

src/Record/Recorder.h

@@ -27,6 +27,11 @@ struct MediaTuple {
     std::string shortUrl() const {
         return vhost + '/' + app + '/' + stream;
     }
+
+    MediaTuple() = default;
+    MediaTuple(std::string vhost, std::string app, std::string stream, std::string params = "")
+        : vhost(std::move(vhost)), app(std::move(app)), stream(std::move(stream)), params(std::move(params)) {
+    }
 };
 
 class RecordInfo: public MediaTuple {

src/Rtmp/RtmpPusher.h

@@ -29,7 +29,7 @@ public:
 
     size_t getSendSpeed() override;
     size_t getSendTotalBytes() override;
-	
+    
 protected:
     //for Tcpclient override
     void onRecv(const toolkit::Buffer::Ptr &buf) override;

src/Srt/SrtCaller.cpp

@@ -1012,9 +1012,9 @@ float SrtCaller::getTimeOutSec() {
     GET_CONFIG(uint32_t, timeout, SRT::kTimeOutSec);
     if (timeout <= 0) {
         WarnL << "config srt " << kTimeOutSec << " not vaild";
-        return 5 * 1000;
+        return 5.0f;
     }
-    return (float)timeout * (float)1000;
+    return (float)timeout;
 };
 
 std::string SrtCaller::generateStreamId() {