注释，说明

注释，说明

src/Common/MediaSink.cpp

@@ -70,7 +70,7 @@ void MediaSink::inputFrame(const Frame::Ptr &frame) {
     it->second->inputFrame(frame);
 
     if(!_allTrackReady && !_trackReadyCallback.empty() && it->second->ready()){
-        //Track由未就绪状态装换成就绪状态，我们就触发onTrackReady回调
+        //Track由未就绪状态转换成就绪状态，我们就触发onTrackReady回调
         auto it_callback = _trackReadyCallback.find(codec_id);
         if(it_callback != _trackReadyCallback.end()){
             it_callback->second();

src/Common/MediaSource.cpp

@@ -123,6 +123,7 @@ MediaSource::Ptr MediaSource::find(
 
     lock_guard<recursive_mutex> lock(g_mtxMediaSrc);
     MediaSource::Ptr ret;
+	//查找某一媒体源，找到后返回
     searchMedia(schema, vhost, app, id,
                 [&](SchemaVhostAppStreamMap::iterator &it0 ,
                     VhostAppStreamMap::iterator &it1,
@@ -138,7 +139,7 @@ MediaSource::Ptr MediaSource::find(
                     return true;
                 });
     if(!ret && bMake){
-        //查找某一媒体源，找到后返回
+        //未查找媒体源，则创建一个
         ret = MediaReader::onMakeMediaSource(schema, vhost,app,id);
     }
     return ret;

src/Extension/AAC.h

@@ -48,7 +48,7 @@ string 	makeAdtsConfig(const uint8_t *pcAdts);
 void 	getAACInfo(const AACFrame &adts,int &iSampleRate,int &iChannel);
 
 
-    /**
+/**
  * aac帧，包含adts头
  */
 class AACFrame : public Frame {

src/Extension/H264Rtmp.h

@@ -36,6 +36,7 @@ using namespace toolkit;
 namespace mediakit{
 /**
  * h264 Rtmp解码类
+ * 将 h264 over rtmp 解复用出 h264-Frame
  */
 class H264RtmpDecoder : public RtmpCodec ,public ResourcePoolHelper<H264Frame> {
 public:

src/Extension/H264Rtp.cpp

@@ -90,7 +90,24 @@ bool H264RtpDecoder::decodeRtp(const RtpPacket::Ptr &rtppack) {
      * Type==7:SPS frame
      * Type==8:PPS frame
      */
+	/*
+	RTF3984 5.2节  Common Structure of the RTP Payload Format
+    Table 1.  Summary of NAL unit types and their payload structures
 
+	   Type   Packet    Type name                        Section
+	   ---------------------------------------------------------
+	   0      undefined                                    -
+	   1-23   NAL unit  Single NAL unit packet per H.264   5.6
+	   24     STAP-A    Single-time aggregation packet     5.7.1
+	   25     STAP-B    Single-time aggregation packet     5.7.1
+	   26     MTAP16    Multi-time aggregation packet      5.7.2
+	   27     MTAP24    Multi-time aggregation packet      5.7.2
+	   28     FU-A      Fragmentation unit                 5.8
+	   29     FU-B      Fragmentation unit                 5.8
+	   30-31  undefined                                    -
+
+
+	*/
     const uint8_t *frame = (uint8_t *) rtppack->data() + rtppack->offset;
     int length = rtppack->size() - rtppack->offset;
     NALU nal;
@@ -145,7 +162,7 @@ bool H264RtpDecoder::decodeRtp(const RtpPacket::Ptr &rtppack) {
             FU fu;
             MakeFU(frame[1], fu);
             if (fu.S) {
-                //该帧的第一个rtp包
+                //该帧的第一个rtp包  FU-A start
                 char tmp = (nal.forbidden_zero_bit << 7 | nal.nal_ref_idc << 5 | fu.type);
                 _h264frame->buffer.assign("\x0\x0\x0\x1", 4);
                 _h264frame->buffer.push_back(tmp);
@@ -164,14 +181,14 @@ bool H264RtpDecoder::decodeRtp(const RtpPacket::Ptr &rtppack) {
             }
 
             if (!fu.E) {
-                //该帧的中间rtp包
+                //该帧的中间rtp包  FU-A mid
                 _h264frame->buffer.append((char *)frame + 2, length - 2);
                 //该函数return时，保存下当前sequence,以便下次对比seq是否连续
                 _lastSeq = rtppack->sequence;
                 return false;
             }
 
-            //该帧最后一个rtp包
+            //该帧最后一个rtp包  FU-A end
             _h264frame->buffer.append((char *)frame + 2, length - 2);
             _h264frame->timeStamp = rtppack->timeStamp;
             auto key = _h264frame->keyFrame();

src/Extension/H264Rtp.h

@@ -36,6 +36,8 @@ namespace mediakit{
 
 /**
  * h264 rtp解码类
+ * 将 h264 over rtsp-rtp 解复用出 h264-Frame
+ * rfc3984
  */
 class H264RtpDecoder : public RtpCodec , public ResourcePoolHelper<H264Frame> {
 public:

src/Extension/H265Rtp.h

@@ -37,6 +37,8 @@ namespace mediakit{
 
 /**
  * h265 rtp解码类
+ * 将 h265 over rtsp-rtp 解复用出 h265-Frame
+ * 《草案（H265-over-RTP）draft-ietf-payload-rtp-h265-07.pdf》
  */
 class H265RtpDecoder : public RtpCodec , public ResourcePoolHelper<H265Frame> {
 public:

src/Http/HttpSession.cpp

@@ -490,6 +490,7 @@ inline void HttpSession::Handle_Req_GET(int64_t &content_len) {
 
     //再看看是否为http-flv直播请求
 	if(checkLiveFlvStream()){
+		//若是，return！
 		return;
 	}
 

src/Rtmp/RtmpPlayer.h

@@ -44,7 +44,7 @@ using namespace toolkit;
 using namespace mediakit::Client;
 
 namespace mediakit {
-
+//实现了rtmp播放器协议部分的功能，及数据接收功能
 class RtmpPlayer:public PlayerBase, public TcpClient,  public RtmpProtocol{
 public:
 	typedef std::shared_ptr<RtmpPlayer> Ptr;
@@ -63,11 +63,11 @@ protected:
 	void onMediaData_l(const RtmpPacket::Ptr &chunkData);
 	void onPlayResult_l(const SockException &ex);
 
-	//for Tcpclient
+	//form Tcpclient
 	void onRecv(const Buffer::Ptr &pBuf) override;
 	void onConnect(const SockException &err) override;
 	void onErr(const SockException &ex) override;
-	//fro RtmpProtocol
+	//from RtmpProtocol
 	void onRtmpChunk(RtmpPacket &chunkData) override;
 	void onStreamDry(uint32_t ui32StreamId) override;
     void onSendRawData(const Buffer::Ptr &buffer) override{

src/Rtsp/RtspMuxer.cpp

@@ -39,7 +39,7 @@ RtspMuxer::RtspMuxer(const TitleSdp::Ptr &title){
 }
 
 void RtspMuxer::onTrackReady(const Track::Ptr &track) {
-    //根据track生产sdp
+    //根据track生成sdp
     Sdp::Ptr sdp = track->getSdp();
     if (!sdp) {
         return;

src/Rtsp/RtspPlayer.h

@@ -46,7 +46,7 @@ using namespace toolkit;
 
 namespace mediakit {
 
-//实现了rtsp播放器协议部分的功能
+//实现了rtsp播放器协议部分的功能，及数据接收功能
 class RtspPlayer: public PlayerBase,public TcpClient, public RtspSplitter, public RtpReceiver {
 public:
 	typedef std::shared_ptr<RtspPlayer> Ptr;

src/Rtsp/RtspSession.cpp

@@ -361,10 +361,11 @@ void RtspSession::onAuthSuccess() {
             strongSelf->shutdown(SockException(Err_shutdown,err));
             return;
         }
-        //找到了响应的rtsp流
+        //找到了相应的rtsp流
         strongSelf->_aTrackInfo = SdpParser(rtsp_src->getSdp()).getAvailableTrack();
         if (strongSelf->_aTrackInfo.empty()) {
             //该流无效
+			DebugL << "无trackInfo，该流无效";
             strongSelf->send_StreamNotFound();
             strongSelf->shutdown(SockException(Err_shutdown,"can not find any availabe track in sdp"));
             return;