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119 lines
3.0 KiB
Go
119 lines
3.0 KiB
Go
package rtmp
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import (
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"github.com/yangjiechina/avformat/librtmp"
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"github.com/yangjiechina/avformat/utils"
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"github.com/yangjiechina/lkm/log"
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"github.com/yangjiechina/lkm/stream"
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"net"
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)
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// Session 负责除连接和断开以外的所有RTMP生命周期处理
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type Session struct {
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//解析rtmp协议栈
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stack *librtmp.Stack
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//Publisher/sink, 在publish或play成功后赋值
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handle interface{}
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isPublisher bool
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conn net.Conn
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receiveBuffer *stream.ReceiveBuffer
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}
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func (s *Session) generateSourceId(app, stream_ string) string {
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if len(app) == 0 {
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return stream_
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} else if len(stream_) == 0 {
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return app
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} else {
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return app + "/" + stream_
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}
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}
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func (s *Session) OnPublish(app, stream_ string, response chan utils.HookState) {
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log.Sugar.Infof("rtmp onpublish app:%s stream:%s conn:%s", app, stream_, s.conn.RemoteAddr().String())
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sourceId := s.generateSourceId(app, stream_)
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source := NewPublisher(sourceId, s.stack, s.conn)
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//设置推流的音视频回调
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s.stack.SetOnPublishHandler(source)
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//初始化放在add source前面, 以防add-init空窗期, 拉流队列空指针.
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source.Init(source.Input, source.Close, stream.ReceiveBufferTCPBlockCount)
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//推流事件Source统一处理, 是否已经存在, Hook回调....
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_, state := stream.PreparePublishSource(source, true)
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if utils.HookStateOK != state {
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log.Sugar.Errorf("rtmp推流失败 source:%s", sourceId)
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} else {
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s.handle = source
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s.isPublisher = true
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s.receiveBuffer = stream.NewTCPReceiveBuffer()
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go source.LoopEvent()
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}
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response <- state
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}
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func (s *Session) OnPlay(app, stream_ string, response chan utils.HookState) {
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sourceId := s.generateSourceId(app, stream_)
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//拉流事件Sink统一处理
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sink := NewSink(stream.GenerateSinkId(s.conn.RemoteAddr()), sourceId, s.conn)
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log.Sugar.Infof("rtmp onplay app:%s stream:%s sink:%v conn:%s", app, stream_, sink.Id(), s.conn.RemoteAddr().String())
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_, state := stream.PreparePlaySink(sink)
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if utils.HookStateOK != state {
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log.Sugar.Errorf("rtmp拉流失败 source:%s sink:%s", sourceId, sink.Id())
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} else {
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s.handle = sink
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}
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response <- state
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}
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func (s *Session) Input(conn net.Conn, data []byte) error {
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//如果是推流，并且握手成功，后续收到的包，都将发送给LoopEvent处理
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if s.isPublisher {
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s.handle.(*Publisher).PublishSource.Input(data)
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return nil
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} else {
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return s.stack.Input(conn, data)
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}
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}
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func (s *Session) Close() {
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//释放协议栈
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if s.stack != nil {
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s.stack.Close()
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}
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//还没到publish/play
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if s.handle == nil {
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return
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}
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publisher, ok := s.handle.(*Publisher)
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if ok {
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log.Sugar.Infof("rtmp推流结束 %s", publisher.PrintInfo())
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if s.isPublisher {
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s.handle.(*Publisher).Close()
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}
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} else {
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sink := s.handle.(stream.Sink)
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log.Sugar.Infof("rtmp拉流结束 %s", sink.PrintInfo())
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sink.Close()
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}
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}
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func NewSession(conn net.Conn) *Session {
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session := &Session{}
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stack := librtmp.NewStack(session)
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session.stack = stack
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session.conn = conn
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return session
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}
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