实现音频转码功能:

- 启用rtc:// schecma
- 增加音频转码码率配置
- aac转码使用原通道数

webrtc/WebRtcTransport.cpp

@@ -1176,10 +1176,10 @@ void push_plugin(Session &sender, const WebRtcArgs &args, const WebRtcPluginMana
             cb(WebRtcException(SockException(Err_other, err)));
             return;
         }
-
+        std::string schema = RTC_SCHEMA;
         RtspMediaSourceImp::Ptr push_src;
         std::shared_ptr<void> push_src_ownership;
-        auto src = MediaSource::find(RTSP_SCHEMA, info.vhost, info.app, info.stream);
+        auto src = MediaSource::find(schema, info.vhost, info.app, info.stream);
         auto push_failed = (bool)src;
 
         while (src) {
@@ -1206,7 +1206,7 @@ void push_plugin(Session &sender, const WebRtcArgs &args, const WebRtcPluginMana
         }
 
         if (!push_src) {
-            push_src = std::make_shared<RtspMediaSourceImp>(info);
+            push_src = std::make_shared<RtspMediaSourceImp>(info, schema);
             push_src_ownership = push_src->getOwnership();
             push_src->setProtocolOption(option);
         }
@@ -1233,9 +1233,8 @@ void play_plugin(Session &sender, const WebRtcArgs &args, const WebRtcPluginMana
             cb(WebRtcException(SockException(Err_other, err)));
             return;
         }
-
         // webrtc播放的是rtsp的源
-        info.schema = RTSP_SCHEMA;
+        info.schema = RTC_SCHEMA;
         MediaSource::findAsync(info, session_ptr, [=](const MediaSource::Ptr &src_in) mutable {
             auto src = dynamic_pointer_cast<RtspMediaSource>(src_in);
             if (!src) {