diff --git a/rtmp/rtmp_session.go b/rtmp/rtmp_session.go
index 5b2d469..202ee28 100644
--- a/rtmp/rtmp_session.go
+++ b/rtmp/rtmp_session.go
@@ -38,11 +38,11 @@ func (s *Session) OnPublish(app, stream_ string, response chan utils.HookState)
 	//设置推流的音视频回调
 	s.stack.SetOnPublishHandler(source)
 
-	//初始化放在add source前面, 以防add后再init,空窗期拉流队列空指针.
+	//初始化放在add source前面, 以防add后再init, 空窗期拉流队列空指针.
 	source.Init(source.Input, source.Close, stream.ReceiveBufferTCPBlockCount)
 	source.SetUrlValues(values)
 
-	//推流事件Source统一处理, 是否已经存在, Hook回调....
+	//统一处理source推流事件, source是否已经存在, hook回调....
 	_, state := stream.PreparePublishSource(source, true)
 	if utils.HookStateOK != state {
 		log.Sugar.Errorf("rtmp推流失败 source:%s", sourceId)
@@ -87,7 +87,7 @@ func (s *Session) Input(conn net.Conn, data []byte) error {
 }
 
 func (s *Session) Close() {
-	//session/conn/stack相关引用, go释放不了...手动赋值为nil
+	//session/conn/stack相互引用, go释放不了...手动赋值为nil
 	s.conn = nil
 
 	defer func() {
diff --git a/stream/source.go b/stream/source.go
index 9bb4e11..11f950b 100644
--- a/stream/source.go
+++ b/stream/source.go
@@ -143,7 +143,7 @@ type PublishSource struct {
 
 	TransDeMuxer     stream.DeMuxer            //负责从推流协议中解析出AVStream和AVPacket
 	recordSink       Sink                      //每个Source的录制流
-	hlsStream        TransStream               //如果开开启HLS传输流, 不等拉流时, 创建直接生成
+	hlsStream        TransStream               //HLS传输流, 如果开启, 在@seee writeHeader 直接创建, 如果等拉流时再创建, 会进一步加大HLS延迟.
 	audioTranscoders []transcode.Transcoder    //音频解码器
 	videoTranscoders []transcode.Transcoder    //视频解码器
 	originStreams    StreamManager             //推流的音视频Streams